diff --git a/src/external/miniaudio.h b/src/external/miniaudio.h
index 5793f20d6..814b7541e 100644
--- a/src/external/miniaudio.h
+++ b/src/external/miniaudio.h
@@ -1,6 +1,6 @@
/*
Audio playback and capture library. Choice of public domain or MIT-0. See license statements at the end of this file.
-miniaudio - v0.10.42 - 2021-08-22
+miniaudio - v0.11.8 - 2022-02-12
David Reid - mackron@gmail.com
@@ -12,7 +12,8 @@ GitHub: https://github.com/mackron/miniaudio
/*
1. Introduction
===============
-miniaudio is a single file library for audio playback and capture. To use it, do the following in one .c file:
+miniaudio is a single file library for audio playback and capture. To use it, do the following in
+one .c file:
```c
#define MINIAUDIO_IMPLEMENTATION
@@ -21,16 +22,44 @@ miniaudio is a single file library for audio playback and capture. To use it, do
You can do `#include "miniaudio.h"` in other parts of the program just like any other header.
-miniaudio uses the concept of a "device" as the abstraction for physical devices. The idea is that you choose a physical device to emit or capture audio from,
-and then move data to/from the device when miniaudio tells you to. Data is delivered to and from devices asynchronously via a callback which you specify when
-initializing the device.
+miniaudio includes both low level and high level APIs. The low level API is good for those who want
+to do all of their mixing themselves and only require a light weight interface to the underlying
+audio device. The high level API is good for those who have complex mixing and effect requirements.
-When initializing the device you first need to configure it. The device configuration allows you to specify things like the format of the data delivered via
-the callback, the size of the internal buffer and the ID of the device you want to emit or capture audio from.
+In miniaudio, objects are transparent structures. Unlike many other libraries, there are no handles
+to opaque objects which means you need to allocate memory for objects yourself. In the examples
+presented in this documentation you will often see objects declared on the stack. You need to be
+careful when translating these examples to your own code so that you don't accidentally declare
+your objects on the stack and then cause them to become invalid once the function returns. In
+addition, you must ensure the memory address of your objects remain the same throughout their
+lifetime. You therefore cannot be making copies of your objects.
-Once you have the device configuration set up you can initialize the device. When initializing a device you need to allocate memory for the device object
-beforehand. This gives the application complete control over how the memory is allocated. In the example below we initialize a playback device on the stack,
-but you could allocate it on the heap if that suits your situation better.
+A config/init pattern is used throughout the entire library. The idea is that you set up a config
+object and pass that into the initialization routine. The advantage to this system is that the
+config object can be initialized with logical defaults and new properties added to it without
+breaking the API. The config object can be allocated on the stack and does not need to be
+maintained after initialization of the corresponding object.
+
+
+1.1. Low Level API
+------------------
+The low level API gives you access to the raw audio data of an audio device. It supports playback,
+capture, full-duplex and loopback (WASAPI only). You can enumerate over devices to determine which
+physical device(s) you want to connect to.
+
+The low level API uses the concept of a "device" as the abstraction for physical devices. The idea
+is that you choose a physical device to emit or capture audio from, and then move data to/from the
+device when miniaudio tells you to. Data is delivered to and from devices asynchronously via a
+callback which you specify when initializing the device.
+
+When initializing the device you first need to configure it. The device configuration allows you to
+specify things like the format of the data delivered via the callback, the size of the internal
+buffer and the ID of the device you want to emit or capture audio from.
+
+Once you have the device configuration set up you can initialize the device. When initializing a
+device you need to allocate memory for the device object beforehand. This gives the application
+complete control over how the memory is allocated. In the example below we initialize a playback
+device on the stack, but you could allocate it on the heap if that suits your situation better.
```c
void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
@@ -63,20 +92,27 @@ but you could allocate it on the heap if that suits your situation better.
}
```
-In the example above, `data_callback()` is where audio data is written and read from the device. The idea is in playback mode you cause sound to be emitted
-from the speakers by writing audio data to the output buffer (`pOutput` in the example). In capture mode you read data from the input buffer (`pInput`) to
-extract sound captured by the microphone. The `frameCount` parameter tells you how many frames can be written to the output buffer and read from the input
-buffer. A "frame" is one sample for each channel. For example, in a stereo stream (2 channels), one frame is 2 samples: one for the left, one for the right.
-The channel count is defined by the device config. The size in bytes of an individual sample is defined by the sample format which is also specified in the
-device config. Multi-channel audio data is always interleaved, which means the samples for each frame are stored next to each other in memory. For example, in
-a stereo stream the first pair of samples will be the left and right samples for the first frame, the second pair of samples will be the left and right samples
-for the second frame, etc.
-
-The configuration of the device is defined by the `ma_device_config` structure. The config object is always initialized with `ma_device_config_init()`. It's
-important to always initialize the config with this function as it initializes it with logical defaults and ensures your program doesn't break when new members
-are added to the `ma_device_config` structure. The example above uses a fairly simple and standard device configuration. The call to `ma_device_config_init()`
-takes a single parameter, which is whether or not the device is a playback, capture, duplex or loopback device (loopback devices are not supported on all
-backends). The `config.playback.format` member sets the sample format which can be one of the following (all formats are native-endian):
+In the example above, `data_callback()` is where audio data is written and read from the device.
+The idea is in playback mode you cause sound to be emitted from the speakers by writing audio data
+to the output buffer (`pOutput` in the example). In capture mode you read data from the input
+buffer (`pInput`) to extract sound captured by the microphone. The `frameCount` parameter tells you
+how many frames can be written to the output buffer and read from the input buffer. A "frame" is
+one sample for each channel. For example, in a stereo stream (2 channels), one frame is 2
+samples: one for the left, one for the right. The channel count is defined by the device config.
+The size in bytes of an individual sample is defined by the sample format which is also specified
+in the device config. Multi-channel audio data is always interleaved, which means the samples for
+each frame are stored next to each other in memory. For example, in a stereo stream the first pair
+of samples will be the left and right samples for the first frame, the second pair of samples will
+be the left and right samples for the second frame, etc.
+
+The configuration of the device is defined by the `ma_device_config` structure. The config object
+is always initialized with `ma_device_config_init()`. It's important to always initialize the
+config with this function as it initializes it with logical defaults and ensures your program
+doesn't break when new members are added to the `ma_device_config` structure. The example above
+uses a fairly simple and standard device configuration. The call to `ma_device_config_init()` takes
+a single parameter, which is whether or not the device is a playback, capture, duplex or loopback
+device (loopback devices are not supported on all backends). The `config.playback.format` member
+sets the sample format which can be one of the following (all formats are native-endian):
+---------------+----------------------------------------+---------------------------+
| Symbol | Description | Range |
@@ -88,22 +124,30 @@ backends). The `config.playback.format` member sets the sample format which can
| ma_format_u8 | 8-bit unsigned integer | [0, 255] |
+---------------+----------------------------------------+---------------------------+
-The `config.playback.channels` member sets the number of channels to use with the device. The channel count cannot exceed MA_MAX_CHANNELS. The
-`config.sampleRate` member sets the sample rate (which must be the same for both playback and capture in full-duplex configurations). This is usually set to
-44100 or 48000, but can be set to anything. It's recommended to keep this between 8000 and 384000, however.
-
-Note that leaving the format, channel count and/or sample rate at their default values will result in the internal device's native configuration being used
-which is useful if you want to avoid the overhead of miniaudio's automatic data conversion.
-
-In addition to the sample format, channel count and sample rate, the data callback and user data pointer are also set via the config. The user data pointer is
-not passed into the callback as a parameter, but is instead set to the `pUserData` member of `ma_device` which you can access directly since all miniaudio
-structures are transparent.
-
-Initializing the device is done with `ma_device_init()`. This will return a result code telling you what went wrong, if anything. On success it will return
-`MA_SUCCESS`. After initialization is complete the device will be in a stopped state. To start it, use `ma_device_start()`. Uninitializing the device will stop
-it, which is what the example above does, but you can also stop the device with `ma_device_stop()`. To resume the device simply call `ma_device_start()` again.
-Note that it's important to never stop or start the device from inside the callback. This will result in a deadlock. Instead you set a variable or signal an
-event indicating that the device needs to stop and handle it in a different thread. The following APIs must never be called inside the callback:
+The `config.playback.channels` member sets the number of channels to use with the device. The
+channel count cannot exceed MA_MAX_CHANNELS. The `config.sampleRate` member sets the sample rate
+(which must be the same for both playback and capture in full-duplex configurations). This is
+usually set to 44100 or 48000, but can be set to anything. It's recommended to keep this between
+8000 and 384000, however.
+
+Note that leaving the format, channel count and/or sample rate at their default values will result
+in the internal device's native configuration being used which is useful if you want to avoid the
+overhead of miniaudio's automatic data conversion.
+
+In addition to the sample format, channel count and sample rate, the data callback and user data
+pointer are also set via the config. The user data pointer is not passed into the callback as a
+parameter, but is instead set to the `pUserData` member of `ma_device` which you can access
+directly since all miniaudio structures are transparent.
+
+Initializing the device is done with `ma_device_init()`. This will return a result code telling you
+what went wrong, if anything. On success it will return `MA_SUCCESS`. After initialization is
+complete the device will be in a stopped state. To start it, use `ma_device_start()`.
+Uninitializing the device will stop it, which is what the example above does, but you can also stop
+the device with `ma_device_stop()`. To resume the device simply call `ma_device_start()` again.
+Note that it's important to never stop or start the device from inside the callback. This will
+result in a deadlock. Instead you set a variable or signal an event indicating that the device
+needs to stop and handle it in a different thread. The following APIs must never be called inside
+the callback:
```c
ma_device_init()
@@ -113,12 +157,14 @@ event indicating that the device needs to stop and handle it in a different thre
ma_device_stop()
```
-You must never try uninitializing and reinitializing a device inside the callback. You must also never try to stop and start it from inside the callback. There
-are a few other things you shouldn't do in the callback depending on your requirements, however this isn't so much a thread-safety thing, but rather a
-real-time processing thing which is beyond the scope of this introduction.
+You must never try uninitializing and reinitializing a device inside the callback. You must also
+never try to stop and start it from inside the callback. There are a few other things you shouldn't
+do in the callback depending on your requirements, however this isn't so much a thread-safety
+thing, but rather a real-time processing thing which is beyond the scope of this introduction.
-The example above demonstrates the initialization of a playback device, but it works exactly the same for capture. All you need to do is change the device type
-from `ma_device_type_playback` to `ma_device_type_capture` when setting up the config, like so:
+The example above demonstrates the initialization of a playback device, but it works exactly the
+same for capture. All you need to do is change the device type from `ma_device_type_playback` to
+`ma_device_type_capture` when setting up the config, like so:
```c
ma_device_config config = ma_device_config_init(ma_device_type_capture);
@@ -126,8 +172,9 @@ from `ma_device_type_playback` to `ma_device_type_capture` when setting up the c
config.capture.channels = MY_CHANNEL_COUNT;
```
-In the data callback you just read from the input buffer (`pInput` in the example above) and leave the output buffer alone (it will be set to NULL when the
-device type is set to `ma_device_type_capture`).
+In the data callback you just read from the input buffer (`pInput` in the example above) and leave
+the output buffer alone (it will be set to NULL when the device type is set to
+`ma_device_type_capture`).
These are the available device types and how you should handle the buffers in the callback:
@@ -140,23 +187,29 @@ These are the available device types and how you should handle the buffers in th
| ma_device_type_loopback | Read from input buffer, leave output buffer untouched. |
+-------------------------+--------------------------------------------------------+
-You will notice in the example above that the sample format and channel count is specified separately for playback and capture. This is to support different
-data formats between the playback and capture devices in a full-duplex system. An example may be that you want to capture audio data as a monaural stream (one
-channel), but output sound to a stereo speaker system. Note that if you use different formats between playback and capture in a full-duplex configuration you
-will need to convert the data yourself. There are functions available to help you do this which will be explained later.
+You will notice in the example above that the sample format and channel count is specified
+separately for playback and capture. This is to support different data formats between the playback
+and capture devices in a full-duplex system. An example may be that you want to capture audio data
+as a monaural stream (one channel), but output sound to a stereo speaker system. Note that if you
+use different formats between playback and capture in a full-duplex configuration you will need to
+convert the data yourself. There are functions available to help you do this which will be
+explained later.
-The example above did not specify a physical device to connect to which means it will use the operating system's default device. If you have multiple physical
-devices connected and you want to use a specific one you will need to specify the device ID in the configuration, like so:
+The example above did not specify a physical device to connect to which means it will use the
+operating system's default device. If you have multiple physical devices connected and you want to
+use a specific one you will need to specify the device ID in the configuration, like so:
```c
config.playback.pDeviceID = pMyPlaybackDeviceID; // Only if requesting a playback or duplex device.
config.capture.pDeviceID = pMyCaptureDeviceID; // Only if requesting a capture, duplex or loopback device.
```
-To retrieve the device ID you will need to perform device enumeration, however this requires the use of a new concept called the "context". Conceptually
-speaking the context sits above the device. There is one context to many devices. The purpose of the context is to represent the backend at a more global level
-and to perform operations outside the scope of an individual device. Mainly it is used for performing run-time linking against backend libraries, initializing
-backends and enumerating devices. The example below shows how to enumerate devices.
+To retrieve the device ID you will need to perform device enumeration, however this requires the
+use of a new concept called the "context". Conceptually speaking the context sits above the device.
+There is one context to many devices. The purpose of the context is to represent the backend at a
+more global level and to perform operations outside the scope of an individual device. Mainly it is
+used for performing run-time linking against backend libraries, initializing backends and
+enumerating devices. The example below shows how to enumerate devices.
```c
ma_context context;
@@ -197,44 +250,230 @@ backends and enumerating devices. The example below shows how to enumerate devic
ma_context_uninit(&context);
```
-The first thing we do in this example is initialize a `ma_context` object with `ma_context_init()`. The first parameter is a pointer to a list of `ma_backend`
-values which are used to override the default backend priorities. When this is NULL, as in this example, miniaudio's default priorities are used. The second
-parameter is the number of backends listed in the array pointed to by the first parameter. The third parameter is a pointer to a `ma_context_config` object
-which can be NULL, in which case defaults are used. The context configuration is used for setting the logging callback, custom memory allocation callbacks,
-user-defined data and some backend-specific configurations.
+The first thing we do in this example is initialize a `ma_context` object with `ma_context_init()`.
+The first parameter is a pointer to a list of `ma_backend` values which are used to override the
+default backend priorities. When this is NULL, as in this example, miniaudio's default priorities
+are used. The second parameter is the number of backends listed in the array pointed to by the
+first parameter. The third parameter is a pointer to a `ma_context_config` object which can be
+NULL, in which case defaults are used. The context configuration is used for setting the logging
+callback, custom memory allocation callbacks, user-defined data and some backend-specific
+configurations.
+
+Once the context has been initialized you can enumerate devices. In the example above we use the
+simpler `ma_context_get_devices()`, however you can also use a callback for handling devices by
+using `ma_context_enumerate_devices()`. When using `ma_context_get_devices()` you provide a pointer
+to a pointer that will, upon output, be set to a pointer to a buffer containing a list of
+`ma_device_info` structures. You also provide a pointer to an unsigned integer that will receive
+the number of items in the returned buffer. Do not free the returned buffers as their memory is
+managed internally by miniaudio.
+
+The `ma_device_info` structure contains an `id` member which is the ID you pass to the device
+config. It also contains the name of the device which is useful for presenting a list of devices
+to the user via the UI.
+
+When creating your own context you will want to pass it to `ma_device_init()` when initializing the
+device. Passing in NULL, like we do in the first example, will result in miniaudio creating the
+context for you, which you don't want to do since you've already created a context. Note that
+internally the context is only tracked by it's pointer which means you must not change the location
+of the `ma_context` object. If this is an issue, consider using `malloc()` to allocate memory for
+the context.
+
+
+1.2. High Level API
+-------------------
+The high level API consists of three main parts:
+
+ * Resource management for loading and streaming sounds.
+ * A node graph for advanced mixing and effect processing.
+ * A high level "engine" that wraps around the resource manager and node graph.
+
+The resource manager (`ma_resource_manager`) is used for loading sounds. It supports loading sounds
+fully into memory and also streaming. It will also deal with reference counting for you which
+avoids the same sound being loaded multiple times.
+
+The node graph is used for mixing and effect processing. The idea is that you connect a number of
+nodes into the graph by connecting each node's outputs to another node's inputs. Each node can
+implement it's own effect. By chaining nodes together, advanced mixing and effect processing can
+be achieved.
+
+The engine encapsulates both the resource manager and the node graph to create a simple, easy to
+use high level API. The resource manager and node graph APIs are covered in more later sections of
+this manual.
+
+The code below shows how you can initialize an engine using it's default configuration.
+
+ ```c
+ ma_result result;
+ ma_engine engine;
+
+ result = ma_engine_init(NULL, &engine);
+ if (result != MA_SUCCESS) {
+ return result; // Failed to initialize the engine.
+ }
+ ```
+
+This creates an engine instance which will initialize a device internally which you can access with
+`ma_engine_get_device()`. It will also initialize a resource manager for you which can be accessed
+with `ma_engine_get_resource_manager()`. The engine itself is a node graph (`ma_node_graph`) which
+means you can pass a pointer to the engine object into any of the `ma_node_graph` APIs (with a
+cast). Alternatively, you can use `ma_engine_get_node_graph()` instead of a cast.
+
+Note that all objects in miniaudio, including the `ma_engine` object in the example above, are
+transparent structures. There are no handles to opaque structures in miniaudio which means you need
+to be mindful of how you declare them. In the example above we are declaring it on the stack, but
+this will result in the struct being invalidated once the function encapsulating it returns. If
+allocating the engine on the heap is more appropriate, you can easily do so with a standard call
+to `malloc()` or whatever heap allocation routine you like:
+
+ ```c
+ ma_engine* pEngine = malloc(sizeof(*pEngine));
+ ```
+
+The `ma_engine` API uses the same config/init pattern used all throughout miniaudio. To configure
+an engine, you can fill out a `ma_engine_config` object and pass it into the first parameter of
+`ma_engine_init()`:
+
+ ```c
+ ma_result result;
+ ma_engine engine;
+ ma_engine_config engineConfig;
+
+ engineConfig = ma_engine_config_init();
+ engineConfig.pResourceManager = &myCustomResourceManager; // <-- Initialized as some earlier stage.
+
+ result = ma_engine_init(&engineConfig, &engine);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ ```
+
+This creates an engine instance using a custom config. In this particular example it's showing how
+you can specify a custom resource manager rather than having the engine initialize one internally.
+This is particularly useful if you want to have multiple engine's share the same resource manager.
+
+The engine must be uninitialized with `ma_engine_uninit()` when it's no longer needed.
+
+By default the engine will be started, but nothing will be playing because no sounds have been
+initialized. The easiest but least flexible way of playing a sound is like so:
+
+ ```c
+ ma_engine_play_sound(&engine, "my_sound.wav", NULL);
+ ```
+
+This plays what miniaudio calls an "inline" sound. It plays the sound once, and then puts the
+internal sound up for recycling. The last parameter is used to specify which sound group the sound
+should be associated with which will be explained later. This particular way of playing a sound is
+simple, but lacks flexibility and features. A more flexible way of playing a sound is to first
+initialize a sound:
+
+ ```c
+ ma_result result;
+ ma_sound sound;
+
+ result = ma_sound_init_from_file(&engine, "my_sound.wav", 0, NULL, NULL, &sound);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ ma_sound_start(&sound);
+ ```
+
+This returns a `ma_sound` object which represents a single instance of the specified sound file. If
+you want to play the same file multiple times simultaneously, you need to create one sound for each
+instance.
+
+Sounds should be uninitialized with `ma_sound_uninit()`.
+
+Sounds are not started by default. Start a sound with `ma_sound_start()` and stop it with
+`ma_sound_stop()`. When a sound is stopped, it is not rewound to the start. Use
+`ma_sound_seek_to_pcm_frames(&sound, 0)` to seek back to the start of a sound. By default, starting
+and stopping sounds happens immediately, but sometimes it might be convenient to schedule the sound
+the be started and/or stopped at a specific time. This can be done with the following functions:
+
+ ```c
+ ma_sound_set_start_time_in_pcm_frames()
+ ma_sound_set_start_time_in_milliseconds()
+ ma_sound_set_stop_time_in_pcm_frames()
+ ma_sound_set_stop_time_in_milliseconds()
+ ```
+
+The start/stop time needs to be specified based on the absolute timer which is controlled by the
+engine. The current global time time in PCM frames can be retrieved with `ma_engine_get_time()`.
+The engine's global time can be changed with `ma_engine_set_time()` for synchronization purposes if
+required.
+
+The third parameter of `ma_sound_init_from_file()` is a set of flags that control how the sound be
+loaded and a few options on which features should be enabled for that sound. By default, the sound
+is synchronously loaded fully into memory straight from the file system without any kind of
+decoding. If you want to decode the sound before storing it in memory, you need to specify the
+`MA_SOUND_FLAG_DECODE` flag. This is useful if you want to incur the cost of decoding at an earlier
+stage, such as a loading stage. Without this option, decoding will happen dynamically at mixing
+time which might be too expensive on the audio thread.
+
+If you want to load the sound asynchronously, you can specify the `MA_SOUND_FLAG_ASYNC` flag. This
+will result in `ma_sound_init_from_file()` returning quickly, but the sound will not start playing
+until the sound has had some audio decoded.
+
+The fourth parameter is a pointer to sound group. A sound group is used as a mechanism to organise
+sounds into groups which have their own effect processing and volume control. An example is a game
+which might have separate groups for sfx, voice and music. Each of these groups have their own
+independent volume control. Use `ma_sound_group_init()` or `ma_sound_group_init_ex()` to initialize
+a sound group.
+
+Sounds and sound groups are nodes in the engine's node graph and can be plugged into any `ma_node`
+API. This makes it possible to connect sounds and sound groups to effect nodes to produce complex
+effect chains.
-Once the context has been initialized you can enumerate devices. In the example above we use the simpler `ma_context_get_devices()`, however you can also use a
-callback for handling devices by using `ma_context_enumerate_devices()`. When using `ma_context_get_devices()` you provide a pointer to a pointer that will,
-upon output, be set to a pointer to a buffer containing a list of `ma_device_info` structures. You also provide a pointer to an unsigned integer that will
-receive the number of items in the returned buffer. Do not free the returned buffers as their memory is managed internally by miniaudio.
+A sound can have it's volume changed with `ma_sound_set_volume()`. If you prefer decibel volume
+control you can use `ma_volume_db_to_linear()` to convert from decibel representation to linear.
-The `ma_device_info` structure contains an `id` member which is the ID you pass to the device config. It also contains the name of the device which is useful
-for presenting a list of devices to the user via the UI.
+Panning and pitching is supported with `ma_sound_set_pan()` and `ma_sound_set_pitch()`. If you know
+a sound will never have it's pitch changed with `ma_sound_set_pitch()` or via the doppler effect,
+you can specify the `MA_SOUND_FLAG_NO_PITCH` flag when initializing the sound for an optimization.
-When creating your own context you will want to pass it to `ma_device_init()` when initializing the device. Passing in NULL, like we do in the first example,
-will result in miniaudio creating the context for you, which you don't want to do since you've already created a context. Note that internally the context is
-only tracked by it's pointer which means you must not change the location of the `ma_context` object. If this is an issue, consider using `malloc()` to
-allocate memory for the context.
+By default, sounds and sound groups have spatialization enabled. If you don't ever want to
+spatialize your sounds, initialize the sound with the `MA_SOUND_FLAG_NO_SPATIALIZATION` flag. The
+spatialization model is fairly simple and is roughly on feature parity with OpenAL. HRTF and
+environmental occlusion are not currently supported, but planned for the future. The supported
+features include:
+
+ * Sound and listener positioning and orientation with cones
+ * Attenuation models: none, inverse, linear and exponential
+ * Doppler effect
+
+Sounds can be faded in and out with `ma_sound_set_fade_in_pcm_frames()`.
+
+To check if a sound is currently playing, you can use `ma_sound_is_playing()`. To check if a sound
+is at the end, use `ma_sound_at_end()`. Looping of a sound can be controlled with
+`ma_sound_set_looping()`. Use `ma_sound_is_looping()` to check whether or not the sound is looping.
2. Building
===========
-miniaudio should work cleanly out of the box without the need to download or install any dependencies. See below for platform-specific details.
+miniaudio should work cleanly out of the box without the need to download or install any
+dependencies. See below for platform-specific details.
2.1. Windows
------------
-The Windows build should compile cleanly on all popular compilers without the need to configure any include paths nor link to any libraries.
+The Windows build should compile cleanly on all popular compilers without the need to configure any
+include paths nor link to any libraries.
+
+The UWP build may require linking to mmdevapi.lib if you get errors about an unresolved external
+symbol for `ActivateAudioInterfaceAsync()`.
+
2.2. macOS and iOS
------------------
-The macOS build should compile cleanly without the need to download any dependencies nor link to any libraries or frameworks. The iOS build needs to be
-compiled as Objective-C and will need to link the relevant frameworks but should compile cleanly out of the box with Xcode. Compiling through the command line
-requires linking to `-lpthread` and `-lm`.
+The macOS build should compile cleanly without the need to download any dependencies nor link to
+any libraries or frameworks. The iOS build needs to be compiled as Objective-C and will need to
+link the relevant frameworks but should compile cleanly out of the box with Xcode. Compiling
+through the command line requires linking to `-lpthread` and `-lm`.
-Due to the way miniaudio links to frameworks at runtime, your application may not pass Apple's notarization process. To fix this there are two options. The
-first is to use the `MA_NO_RUNTIME_LINKING` option, like so:
+Due to the way miniaudio links to frameworks at runtime, your application may not pass Apple's
+notarization process. To fix this there are two options. The first is to use the
+`MA_NO_RUNTIME_LINKING` option, like so:
```c
#ifdef __APPLE__
@@ -244,8 +483,9 @@ first is to use the `MA_NO_RUNTIME_LINKING` option, like so:
#include "miniaudio.h"
```
-This will require linking with `-framework CoreFoundation -framework CoreAudio -framework AudioUnit`. Alternatively, if you would rather keep using runtime
-linking you can add the following to your entitlements.xcent file:
+This will require linking with `-framework CoreFoundation -framework CoreAudio -framework AudioUnit`.
+Alternatively, if you would rather keep using runtime linking you can add the following to your
+entitlements.xcent file:
```
com.apple.security.cs.allow-dyld-environment-variables
@@ -254,26 +494,37 @@ linking you can add the following to your entitlements.xcent file:
```
+See this discussion for more info: https://github.com/mackron/miniaudio/issues/203.
+
2.3. Linux
----------
-The Linux build only requires linking to `-ldl`, `-lpthread` and `-lm`. You do not need any development packages.
+The Linux build only requires linking to `-ldl`, `-lpthread` and `-lm`. You do not need any
+development packages. You may need to link with `-latomic` if you're compiling for 32-bit ARM.
+
2.4. BSD
--------
-The BSD build only requires linking to `-lpthread` and `-lm`. NetBSD uses audio(4), OpenBSD uses sndio and FreeBSD uses OSS.
+The BSD build only requires linking to `-lpthread` and `-lm`. NetBSD uses audio(4), OpenBSD uses
+sndio and FreeBSD uses OSS. You may need to link with `-latomic` if you're compiling for 32-bit
+ARM.
+
2.5. Android
------------
-AAudio is the highest priority backend on Android. This should work out of the box without needing any kind of compiler configuration. Support for AAudio
-starts with Android 8 which means older versions will fall back to OpenSL|ES which requires API level 16+.
+AAudio is the highest priority backend on Android. This should work out of the box without needing
+any kind of compiler configuration. Support for AAudio starts with Android 8 which means older
+versions will fall back to OpenSL|ES which requires API level 16+.
+
+There have been reports that the OpenSL|ES backend fails to initialize on some Android based
+devices due to `dlopen()` failing to open "libOpenSLES.so". If this happens on your platform
+you'll need to disable run-time linking with `MA_NO_RUNTIME_LINKING` and link with -lOpenSLES.
-There have been reports that the OpenSL|ES backend fails to initialize on some Android based devices due to `dlopen()` failing to open "libOpenSLES.so". If
-this happens on your platform you'll need to disable run-time linking with `MA_NO_RUNTIME_LINKING` and link with -lOpenSLES.
2.6. Emscripten
---------------
-The Emscripten build emits Web Audio JavaScript directly and should compile cleanly out of the box. You cannot use -std=c* compiler flags, nor -ansi.
+The Emscripten build emits Web Audio JavaScript directly and should compile cleanly out of the box.
+You cannot use `-std=c*` compiler flags, nor `-ansi`.
2.7. Build Options
@@ -366,28 +617,26 @@ The Emscripten build emits Web Audio JavaScript directly and should compile clea
+----------------------------------+--------------------------------------------------------------------+
| MA_NO_MP3 | Disables the built-in MP3 decoder. |
+----------------------------------+--------------------------------------------------------------------+
- | MA_NO_DEVICE_IO | Disables playback and recording. This will disable ma_context and |
- | | ma_device APIs. This is useful if you only want to use miniaudio's |
- | | data conversion and/or decoding APIs. |
+ | MA_NO_DEVICE_IO | Disables playback and recording. This will disable `ma_context` |
+ | | and `ma_device` APIs. This is useful if you only want to use |
+ | | miniaudio's data conversion and/or decoding APIs. |
+----------------------------------+--------------------------------------------------------------------+
- | MA_NO_THREADING | Disables the ma_thread, ma_mutex, ma_semaphore and ma_event APIs. |
- | | This option is useful if you only need to use miniaudio for data |
- | | conversion, decoding and/or encoding. Some families of APIs |
- | | require threading which means the following options must also be |
- | | set: |
+ | MA_NO_THREADING | Disables the `ma_thread`, `ma_mutex`, `ma_semaphore` and |
+ | | `ma_event` APIs. This option is useful if you only need to use |
+ | | miniaudio for data conversion, decoding and/or encoding. Some |
+ | | families of APIsrequire threading which means the following |
+ | | options must also be set: |
| | |
| | ``` |
| | MA_NO_DEVICE_IO |
| | ``` |
+----------------------------------+--------------------------------------------------------------------+
- | MA_NO_GENERATION | Disables generation APIs such a ma_waveform and ma_noise. |
+ | MA_NO_GENERATION | Disables generation APIs such a `ma_waveform` and `ma_noise`. |
+----------------------------------+--------------------------------------------------------------------+
| MA_NO_SSE2 | Disables SSE2 optimizations. |
+----------------------------------+--------------------------------------------------------------------+
| MA_NO_AVX2 | Disables AVX2 optimizations. |
+----------------------------------+--------------------------------------------------------------------+
- | MA_NO_AVX512 | Disables AVX-512 optimizations. |
- +----------------------------------+--------------------------------------------------------------------+
| MA_NO_NEON | Disables NEON optimizations. |
+----------------------------------+--------------------------------------------------------------------+
| MA_NO_RUNTIME_LINKING | Disables runtime linking. This is useful for passing Apple's |
@@ -399,47 +648,47 @@ The Emscripten build emits Web Audio JavaScript directly and should compile clea
| | You may need to enable this if your target platform does not allow |
| | runtime linking via `dlopen()`. |
+----------------------------------+--------------------------------------------------------------------+
- | MA_DEBUG_OUTPUT | Enable processing of MA_LOG_LEVEL_DEBUG messages and `printf()` |
- | | output. |
+ | MA_DEBUG_OUTPUT | Enable `printf()` output of debug logs (`MA_LOG_LEVEL_DEBUG`). |
+----------------------------------+--------------------------------------------------------------------+
| MA_COINIT_VALUE | Windows only. The value to pass to internal calls to |
| | `CoInitializeEx()`. Defaults to `COINIT_MULTITHREADED`. |
+----------------------------------+--------------------------------------------------------------------+
| MA_API | Controls how public APIs should be decorated. Default is `extern`. |
+----------------------------------+--------------------------------------------------------------------+
- | MA_DLL | If set, configures MA_API to either import or export APIs |
- | | depending on whether or not the implementation is being defined. |
- | | If defining the implementation, MA_API will be configured to |
- | | export. Otherwise it will be configured to import. This has no |
- | | effect if MA_API is defined externally. |
- +----------------------------------+--------------------------------------------------------------------+
3. Definitions
==============
-This section defines common terms used throughout miniaudio. Unfortunately there is often ambiguity in the use of terms throughout the audio space, so this
-section is intended to clarify how miniaudio uses each term.
+This section defines common terms used throughout miniaudio. Unfortunately there is often ambiguity
+in the use of terms throughout the audio space, so this section is intended to clarify how miniaudio
+uses each term.
3.1. Sample
-----------
-A sample is a single unit of audio data. If the sample format is f32, then one sample is one 32-bit floating point number.
+A sample is a single unit of audio data. If the sample format is f32, then one sample is one 32-bit
+floating point number.
3.2. Frame / PCM Frame
----------------------
-A frame is a group of samples equal to the number of channels. For a stereo stream a frame is 2 samples, a mono frame is 1 sample, a 5.1 surround sound frame
-is 6 samples, etc. The terms "frame" and "PCM frame" are the same thing in miniaudio. Note that this is different to a compressed frame. If ever miniaudio
-needs to refer to a compressed frame, such as a FLAC frame, it will always clarify what it's referring to with something like "FLAC frame".
+A frame is a group of samples equal to the number of channels. For a stereo stream a frame is 2
+samples, a mono frame is 1 sample, a 5.1 surround sound frame is 6 samples, etc. The terms "frame"
+and "PCM frame" are the same thing in miniaudio. Note that this is different to a compressed frame.
+If ever miniaudio needs to refer to a compressed frame, such as a FLAC frame, it will always
+clarify what it's referring to with something like "FLAC frame".
3.3. Channel
------------
-A stream of monaural audio that is emitted from an individual speaker in a speaker system, or received from an individual microphone in a microphone system. A
-stereo stream has two channels (a left channel, and a right channel), a 5.1 surround sound system has 6 channels, etc. Some audio systems refer to a channel as
-a complex audio stream that's mixed with other channels to produce the final mix - this is completely different to miniaudio's use of the term "channel" and
-should not be confused.
+A stream of monaural audio that is emitted from an individual speaker in a speaker system, or
+received from an individual microphone in a microphone system. A stereo stream has two channels (a
+left channel, and a right channel), a 5.1 surround sound system has 6 channels, etc. Some audio
+systems refer to a channel as a complex audio stream that's mixed with other channels to produce
+the final mix - this is completely different to miniaudio's use of the term "channel" and should
+not be confused.
3.4. Sample Rate
----------------
-The sample rate in miniaudio is always expressed in Hz, such as 44100, 48000, etc. It's the number of PCM frames that are processed per second.
+The sample rate in miniaudio is always expressed in Hz, such as 44100, 48000, etc. It's the number
+of PCM frames that are processed per second.
3.5. Formats
------------
@@ -459,52347 +708,72857 @@ All formats are native-endian.
-4. Decoding
-===========
-The `ma_decoder` API is used for reading audio files. Decoders are completely decoupled from devices and can be used independently. The following formats are
-supported:
+4. Data Sources
+===============
+The data source abstraction in miniaudio is used for retrieving audio data from some source. A few
+examples include `ma_decoder`, `ma_noise` and `ma_waveform`. You will need to be familiar with data
+sources in order to make sense of some of the higher level concepts in miniaudio.
- +---------+------------------+----------+
- | Format | Decoding Backend | Built-In |
- +---------+------------------+----------+
- | WAV | dr_wav | Yes |
- | MP3 | dr_mp3 | Yes |
- | FLAC | dr_flac | Yes |
- | Vorbis | stb_vorbis | No |
- +---------+------------------+----------+
+The `ma_data_source` API is a generic interface for reading from a data source. Any object that
+implements the data source interface can be plugged into any `ma_data_source` function.
-Vorbis is supported via stb_vorbis which can be enabled by including the header section before the implementation of miniaudio, like the following:
+To read data from a data source:
```c
- #define STB_VORBIS_HEADER_ONLY
- #include "extras/stb_vorbis.c" // Enables Vorbis decoding.
-
- #define MINIAUDIO_IMPLEMENTATION
- #include "miniaudio.h"
+ ma_result result;
+ ma_uint64 framesRead;
- // The stb_vorbis implementation must come after the implementation of miniaudio.
- #undef STB_VORBIS_HEADER_ONLY
- #include "extras/stb_vorbis.c"
+ result = ma_data_source_read_pcm_frames(pDataSource, pFramesOut, frameCount, &framesRead, loop);
+ if (result != MA_SUCCESS) {
+ return result; // Failed to read data from the data source.
+ }
```
-A copy of stb_vorbis is included in the "extras" folder in the miniaudio repository (https://github.com/mackron/miniaudio).
+If you don't need the number of frames that were successfully read you can pass in `NULL` to the
+`pFramesRead` parameter. If this returns a value less than the number of frames requested it means
+the end of the file has been reached. `MA_AT_END` will be returned only when the number of frames
+read is 0.
-Built-in decoders are amalgamated into the implementation section of miniaudio. You can disable the built-in decoders by specifying one or more of the
-following options before the miniaudio implementation:
+When calling any data source function, with the exception of `ma_data_source_init()` and
+`ma_data_source_uninit()`, you can pass in any object that implements a data source. For example,
+you could plug in a decoder like so:
```c
- #define MA_NO_WAV
- #define MA_NO_MP3
- #define MA_NO_FLAC
+ ma_result result;
+ ma_uint64 framesRead;
+ ma_decoder decoder; // <-- This would be initialized with `ma_decoder_init_*()`.
+
+ result = ma_data_source_read_pcm_frames(&decoder, pFramesOut, frameCount, &framesRead, loop);
+ if (result != MA_SUCCESS) {
+ return result; // Failed to read data from the decoder.
+ }
```
-Disabling built-in decoding libraries is useful if you use these libraries independantly of the `ma_decoder` API.
+If you want to seek forward you can pass in `NULL` to the `pFramesOut` parameter. Alternatively you
+can use `ma_data_source_seek_pcm_frames()`.
-A decoder can be initialized from a file with `ma_decoder_init_file()`, a block of memory with `ma_decoder_init_memory()`, or from data delivered via callbacks
-with `ma_decoder_init()`. Here is an example for loading a decoder from a file:
+To seek to a specific PCM frame:
```c
- ma_decoder decoder;
- ma_result result = ma_decoder_init_file("MySong.mp3", NULL, &decoder);
+ result = ma_data_source_seek_to_pcm_frame(pDataSource, frameIndex);
if (result != MA_SUCCESS) {
- return false; // An error occurred.
+ return result; // Failed to seek to PCM frame.
}
-
- ...
-
- ma_decoder_uninit(&decoder);
```
-When initializing a decoder, you can optionally pass in a pointer to a `ma_decoder_config` object (the `NULL` argument in the example above) which allows you
-to configure the output format, channel count, sample rate and channel map:
+You can retrieve the total length of a data source in PCM frames, but note that some data sources
+may not have the notion of a length, such as noise and waveforms, and others may just not have a
+way of determining the length such as some decoders. To retrieve the length:
```c
- ma_decoder_config config = ma_decoder_config_init(ma_format_f32, 2, 48000);
+ ma_uint64 length;
+
+ result = ma_data_source_get_length_in_pcm_frames(pDataSource, &length);
+ if (result != MA_SUCCESS) {
+ return result; // Failed to retrieve the length.
+ }
```
-When passing in `NULL` for decoder config in `ma_decoder_init*()`, the output format will be the same as that defined by the decoding backend.
+Care should be taken when retrieving the length of a data source where the underlying decoder is
+pulling data from a data stream with an undefined length, such as internet radio or some kind of
+broadcast. If you do this, `ma_data_source_get_length_in_pcm_frames()` may never return.
-Data is read from the decoder as PCM frames. This will return the number of PCM frames actually read. If the return value is less than the requested number of
-PCM frames it means you've reached the end:
+The current position of the cursor in PCM frames can also be retrieved:
```c
- ma_uint64 framesRead = ma_decoder_read_pcm_frames(pDecoder, pFrames, framesToRead);
- if (framesRead < framesToRead) {
- // Reached the end.
+ ma_uint64 cursor;
+
+ result = ma_data_source_get_cursor_in_pcm_frames(pDataSource, &cursor);
+ if (result != MA_SUCCESS) {
+ return result; // Failed to retrieve the cursor.
}
```
-You can also seek to a specific frame like so:
+You will often need to know the data format that will be returned after reading. This can be
+retrieved like so:
```c
- ma_result result = ma_decoder_seek_to_pcm_frame(pDecoder, targetFrame);
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ ma_channel channelMap[MA_MAX_CHANNELS];
+
+ result = ma_data_source_get_data_format(pDataSource, &format, &channels, &sampleRate, channelMap, MA_MAX_CHANNELS);
if (result != MA_SUCCESS) {
- return false; // An error occurred.
+ return result; // Failed to retrieve data format.
}
```
-If you want to loop back to the start, you can simply seek back to the first PCM frame:
-
- ```c
- ma_decoder_seek_to_pcm_frame(pDecoder, 0);
- ```
+If you do not need a specific data format property, just pass in NULL to the respective parameter.
-When loading a decoder, miniaudio uses a trial and error technique to find the appropriate decoding backend. This can be unnecessarily inefficient if the type
-is already known. In this case you can use `encodingFormat` variable in the device config to specify a specific encoding format you want to decode:
+There may be cases where you want to implement something like a sound bank where you only want to
+read data within a certain range of the underlying data. To do this you can use a range:
```c
- decoderConfig.encodingFormat = ma_encoding_format_wav;
+ result = ma_data_source_set_range_in_pcm_frames(pDataSource, rangeBegInFrames, rangeEndInFrames);
+ if (result != MA_SUCCESS) {
+ return result; // Failed to set the range.
+ }
```
-See the `ma_encoding_format` enum for possible encoding formats.
+This is useful if you have a sound bank where many sounds are stored in the same file and you want
+the data source to only play one of those sub-sounds.
-The `ma_decoder_init_file()` API will try using the file extension to determine which decoding backend to prefer.
+Custom loop points can also be used with data sources. By default, data sources will loop after
+they reach the end of the data source, but if you need to loop at a specific location, you can do
+the following:
+ ```c
+ result = ma_data_set_loop_point_in_pcm_frames(pDataSource, loopBegInFrames, loopEndInFrames);
+ if (result != MA_SUCCESS) {
+ return result; // Failed to set the loop point.
+ }
+ ```
+The loop point is relative to the current range.
-5. Encoding
-===========
-The `ma_encoding` API is used for writing audio files. The only supported output format is WAV which is achieved via dr_wav which is amalgamated into the
-implementation section of miniaudio. This can be disabled by specifying the following option before the implementation of miniaudio:
+It's sometimes useful to chain data sources together so that a seamless transition can be achieved.
+To do this, you can use chaining:
```c
- #define MA_NO_WAV
- ```
+ ma_decoder decoder1;
+ ma_decoder decoder2;
-An encoder can be initialized to write to a file with `ma_encoder_init_file()` or from data delivered via callbacks with `ma_encoder_init()`. Below is an
-example for initializing an encoder to output to a file.
+ // ... initialize decoders with ma_decoder_init_*() ...
- ```c
- ma_encoder_config config = ma_encoder_config_init(ma_resource_format_wav, FORMAT, CHANNELS, SAMPLE_RATE);
- ma_encoder encoder;
- ma_result result = ma_encoder_init_file("my_file.wav", &config, &encoder);
+ result = ma_data_source_set_next(&decoder1, &decoder2);
if (result != MA_SUCCESS) {
- // Error
+ return result; // Failed to set the next data source.
}
- ...
-
- ma_encoder_uninit(&encoder);
+ result = ma_data_source_read_pcm_frames(&decoder1, pFramesOut, frameCount, pFramesRead, MA_FALSE);
+ if (result != MA_SUCCESS) {
+ return result; // Failed to read from the decoder.
+ }
```
-When initializing an encoder you must specify a config which is initialized with `ma_encoder_config_init()`. Here you must specify the file type, the output
-sample format, output channel count and output sample rate. The following file types are supported:
-
- +------------------------+-------------+
- | Enum | Description |
- +------------------------+-------------+
- | ma_resource_format_wav | WAV |
- +------------------------+-------------+
+In the example above we're using decoders. When reading from a chain, you always want to read from
+the top level data source in the chain. In the example above, `decoder1` is the top level data
+source in the chain. When `decoder1` reaches the end, `decoder2` will start seamlessly without any
+gaps.
-If the format, channel count or sample rate is not supported by the output file type an error will be returned. The encoder will not perform data conversion so
-you will need to convert it before outputting any audio data. To output audio data, use `ma_encoder_write_pcm_frames()`, like in the example below:
+Note that the `loop` parameter is set to false in the example above. When this is set to true, only
+the current data source will be looped. You can loop the entire chain by linking in a loop like so:
```c
- framesWritten = ma_encoder_write_pcm_frames(&encoder, pPCMFramesToWrite, framesToWrite);
+ ma_data_source_set_next(&decoder1, &decoder2); // decoder1 -> decoder2
+ ma_data_source_set_next(&decoder2, &decoder1); // decoder2 -> decoder1 (loop back to the start).
```
-Encoders must be uninitialized with `ma_encoder_uninit()`.
-
-
-6. Data Conversion
-==================
-A data conversion API is included with miniaudio which supports the majority of data conversion requirements. This supports conversion between sample formats,
-channel counts (with channel mapping) and sample rates.
-
-
-6.1. Sample Format Conversion
------------------------------
-Conversion between sample formats is achieved with the `ma_pcm_*_to_*()`, `ma_pcm_convert()` and `ma_convert_pcm_frames_format()` APIs. Use `ma_pcm_*_to_*()`
-to convert between two specific formats. Use `ma_pcm_convert()` to convert based on a `ma_format` variable. Use `ma_convert_pcm_frames_format()` to convert
-PCM frames where you want to specify the frame count and channel count as a variable instead of the total sample count.
-
+Note that setting up chaining is not thread safe, so care needs to be taken if you're dynamically
+changing links while the audio thread is in the middle of reading.
-6.1.1. Dithering
-----------------
-Dithering can be set using the ditherMode parameter.
-
-The different dithering modes include the following, in order of efficiency:
+Do not use `ma_decoder_seek_to_pcm_frame()` as a means to reuse a data source to play multiple
+instances of the same sound simultaneously. Instead, initialize multiple data sources for each
+instance. This can be extremely inefficient depending on the data source and can result in
+glitching due to subtle changes to the state of internal filters.
- +-----------+--------------------------+
- | Type | Enum Token |
- +-----------+--------------------------+
- | None | ma_dither_mode_none |
- | Rectangle | ma_dither_mode_rectangle |
- | Triangle | ma_dither_mode_triangle |
- +-----------+--------------------------+
-Note that even if the dither mode is set to something other than `ma_dither_mode_none`, it will be ignored for conversions where dithering is not needed.
-Dithering is available for the following conversions:
+4.1. Custom Data Sources
+------------------------
+You can implement a custom data source by implementing the functions in `ma_data_source_vtable`.
+Your custom object must have `ma_data_source_base` as it's first member:
- ```
- s16 -> u8
- s24 -> u8
- s32 -> u8
- f32 -> u8
- s24 -> s16
- s32 -> s16
- f32 -> s16
+ ```c
+ struct my_data_source
+ {
+ ma_data_source_base base;
+ ...
+ };
```
-Note that it is not an error to pass something other than ma_dither_mode_none for conversions where dither is not used. It will just be ignored.
-
-
-
-6.2. Channel Conversion
------------------------
-Channel conversion is used for channel rearrangement and conversion from one channel count to another. The `ma_channel_converter` API is used for channel
-conversion. Below is an example of initializing a simple channel converter which converts from mono to stereo.
+In your initialization routine, you need to call `ma_data_source_init()` in order to set up the
+base object (`ma_data_source_base`):
```c
- ma_channel_converter_config config = ma_channel_converter_config_init(
- ma_format, // Sample format
- 1, // Input channels
- NULL, // Input channel map
- 2, // Output channels
- NULL, // Output channel map
- ma_channel_mix_mode_default); // The mixing algorithm to use when combining channels.
+ static ma_result my_data_source_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+ {
+ // Read data here. Output in the same format returned by my_data_source_get_data_format().
+ }
- result = ma_channel_converter_init(&config, &converter);
- if (result != MA_SUCCESS) {
- // Error.
+ static ma_result my_data_source_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
+ {
+ // Seek to a specific PCM frame here. Return MA_NOT_IMPLEMENTED if seeking is not supported.
}
- ```
-To perform the conversion simply call `ma_channel_converter_process_pcm_frames()` like so:
+ static ma_result my_data_source_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
+ {
+ // Return the format of the data here.
+ }
- ```c
- ma_result result = ma_channel_converter_process_pcm_frames(&converter, pFramesOut, pFramesIn, frameCount);
- if (result != MA_SUCCESS) {
- // Error.
+ static ma_result my_data_source_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
+ {
+ // Retrieve the current position of the cursor here. Return MA_NOT_IMPLEMENTED and set *pCursor to 0 if there is no notion of a cursor.
}
- ```
-It is up to the caller to ensure the output buffer is large enough to accomodate the new PCM frames.
+ static ma_result my_data_source_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
+ {
+ // Retrieve the length in PCM frames here. Return MA_NOT_IMPLEMENTED and set *pLength to 0 if there is no notion of a length or if the length is unknown.
+ }
-Input and output PCM frames are always interleaved. Deinterleaved layouts are not supported.
+ static g_my_data_source_vtable =
+ {
+ my_data_source_read,
+ my_data_source_seek,
+ my_data_source_get_data_format,
+ my_data_source_get_cursor,
+ my_data_source_get_length
+ };
+ ma_result my_data_source_init(my_data_source* pMyDataSource)
+ {
+ ma_result result;
+ ma_data_source_config baseConfig;
-6.2.1. Channel Mapping
-----------------------
-In addition to converting from one channel count to another, like the example above, the channel converter can also be used to rearrange channels. When
-initializing the channel converter, you can optionally pass in channel maps for both the input and output frames. If the channel counts are the same, and each
-channel map contains the same channel positions with the exception that they're in a different order, a simple shuffling of the channels will be performed. If,
-however, there is not a 1:1 mapping of channel positions, or the channel counts differ, the input channels will be mixed based on a mixing mode which is
-specified when initializing the `ma_channel_converter_config` object.
+ baseConfig = ma_data_source_config_init();
+ baseConfig.vtable = &g_my_data_source_vtable;
-When converting from mono to multi-channel, the mono channel is simply copied to each output channel. When going the other way around, the audio of each output
-channel is simply averaged and copied to the mono channel.
+ result = ma_data_source_init(&baseConfig, &pMyDataSource->base);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
-In more complicated cases blending is used. The `ma_channel_mix_mode_simple` mode will drop excess channels and silence extra channels. For example, converting
-from 4 to 2 channels, the 3rd and 4th channels will be dropped, whereas converting from 2 to 4 channels will put silence into the 3rd and 4th channels.
+ // ... do the initialization of your custom data source here ...
-The `ma_channel_mix_mode_rectangle` mode uses spacial locality based on a rectangle to compute a simple distribution between input and output. Imagine sitting
-in the middle of a room, with speakers on the walls representing channel positions. The MA_CHANNEL_FRONT_LEFT position can be thought of as being in the corner
-of the front and left walls.
+ return MA_SUCCESS;
+ }
-Finally, the `ma_channel_mix_mode_custom_weights` mode can be used to use custom user-defined weights. Custom weights can be passed in as the last parameter of
-`ma_channel_converter_config_init()`.
+ void my_data_source_uninit(my_data_source* pMyDataSource)
+ {
+ // ... do the uninitialization of your custom data source here ...
+
+ // You must uninitialize the base data source.
+ ma_data_source_uninit(&pMyDataSource->base);
+ }
+ ```
-Predefined channel maps can be retrieved with `ma_get_standard_channel_map()`. This takes a `ma_standard_channel_map` enum as it's first parameter, which can
-be one of the following:
+Note that `ma_data_source_init()` and `ma_data_source_uninit()` are never called directly outside
+of the custom data source. It's up to the custom data source itself to call these within their own
+init/uninit functions.
- +-----------------------------------+-----------------------------------------------------------+
- | Name | Description |
- +-----------------------------------+-----------------------------------------------------------+
- | ma_standard_channel_map_default | Default channel map used by miniaudio. See below. |
- | ma_standard_channel_map_microsoft | Channel map used by Microsoft's bitfield channel maps. |
- | ma_standard_channel_map_alsa | Default ALSA channel map. |
- | ma_standard_channel_map_rfc3551 | RFC 3551. Based on AIFF. |
- | ma_standard_channel_map_flac | FLAC channel map. |
- | ma_standard_channel_map_vorbis | Vorbis channel map. |
- | ma_standard_channel_map_sound4 | FreeBSD's sound(4). |
- | ma_standard_channel_map_sndio | sndio channel map. http://www.sndio.org/tips.html. |
- | ma_standard_channel_map_webaudio | https://webaudio.github.io/web-audio-api/#ChannelOrdering |
- +-----------------------------------+-----------------------------------------------------------+
-Below are the channel maps used by default in miniaudio (`ma_standard_channel_map_default`):
- +---------------+---------------------------------+
- | Channel Count | Mapping |
- +---------------+---------------------------------+
- | 1 (Mono) | 0: MA_CHANNEL_MONO |
- +---------------+---------------------------------+
- | 2 (Stereo) | 0: MA_CHANNEL_FRONT_LEFT
|
- | | 1: MA_CHANNEL_FRONT_RIGHT |
- +---------------+---------------------------------+
- | 3 | 0: MA_CHANNEL_FRONT_LEFT
|
- | | 1: MA_CHANNEL_FRONT_RIGHT
|
- | | 2: MA_CHANNEL_FRONT_CENTER |
- +---------------+---------------------------------+
- | 4 (Surround) | 0: MA_CHANNEL_FRONT_LEFT
|
- | | 1: MA_CHANNEL_FRONT_RIGHT
|
- | | 2: MA_CHANNEL_FRONT_CENTER
|
- | | 3: MA_CHANNEL_BACK_CENTER |
- +---------------+---------------------------------+
- | 5 | 0: MA_CHANNEL_FRONT_LEFT
|
- | | 1: MA_CHANNEL_FRONT_RIGHT
|
- | | 2: MA_CHANNEL_FRONT_CENTER
|
- | | 3: MA_CHANNEL_BACK_LEFT
|
- | | 4: MA_CHANNEL_BACK_RIGHT |
- +---------------+---------------------------------+
- | 6 (5.1) | 0: MA_CHANNEL_FRONT_LEFT
|
- | | 1: MA_CHANNEL_FRONT_RIGHT
|
- | | 2: MA_CHANNEL_FRONT_CENTER
|
- | | 3: MA_CHANNEL_LFE
|
- | | 4: MA_CHANNEL_SIDE_LEFT
|
- | | 5: MA_CHANNEL_SIDE_RIGHT |
- +---------------+---------------------------------+
- | 7 | 0: MA_CHANNEL_FRONT_LEFT
|
- | | 1: MA_CHANNEL_FRONT_RIGHT
|
- | | 2: MA_CHANNEL_FRONT_CENTER
|
- | | 3: MA_CHANNEL_LFE
|
- | | 4: MA_CHANNEL_BACK_CENTER
|
- | | 4: MA_CHANNEL_SIDE_LEFT
|
- | | 5: MA_CHANNEL_SIDE_RIGHT |
- +---------------+---------------------------------+
- | 8 (7.1) | 0: MA_CHANNEL_FRONT_LEFT
|
- | | 1: MA_CHANNEL_FRONT_RIGHT
|
- | | 2: MA_CHANNEL_FRONT_CENTER
|
- | | 3: MA_CHANNEL_LFE
|
- | | 4: MA_CHANNEL_BACK_LEFT
|
- | | 5: MA_CHANNEL_BACK_RIGHT
|
- | | 6: MA_CHANNEL_SIDE_LEFT
|
- | | 7: MA_CHANNEL_SIDE_RIGHT |
- +---------------+---------------------------------+
- | Other | All channels set to 0. This |
- | | is equivalent to the same |
- | | mapping as the device. |
- +---------------+---------------------------------+
+5. Engine
+=========
+The `ma_engine` API is a high level API for managing and mixing sounds and effect processing. The
+`ma_engine` object encapsulates a resource manager and a node graph, both of which will be
+explained in more detail later.
+Sounds are called `ma_sound` and are created from an engine. Sounds can be associated with a mixing
+group called `ma_sound_group` which are also created from the engine. Both `ma_sound` and
+`ma_sound_group` objects are nodes within the engine's node graph.
+When the engine is initialized, it will normally create a device internally. If you would rather
+manage the device yourself, you can do so and just pass a pointer to it via the engine config when
+you initialize the engine. You can also just use the engine without a device, which again can be
+configured via the engine config.
-6.3. Resampling
----------------
-Resampling is achieved with the `ma_resampler` object. To create a resampler object, do something like the following:
+The most basic way to initialize the engine is with a default config, like so:
```c
- ma_resampler_config config = ma_resampler_config_init(
- ma_format_s16,
- channels,
- sampleRateIn,
- sampleRateOut,
- ma_resample_algorithm_linear);
+ ma_result result;
+ ma_engine engine;
- ma_resampler resampler;
- ma_result result = ma_resampler_init(&config, &resampler);
+ result = ma_engine_init(NULL, &engine);
if (result != MA_SUCCESS) {
- // An error occurred...
+ return result; // Failed to initialize the engine.
}
```
-Do the following to uninitialize the resampler:
+This will result in the engine initializing a playback device using the operating system's default
+device. This will be sufficient for many use cases, but if you need more flexibility you'll want to
+configure the engine with an engine config:
```c
- ma_resampler_uninit(&resampler);
- ```
+ ma_result result;
+ ma_engine engine;
+ ma_engine_config engineConfig;
-The following example shows how data can be processed
+ engineConfig = ma_engine_config_init();
+ engineConfig.pPlaybackDevice = &myDevice;
- ```c
- ma_uint64 frameCountIn = 1000;
- ma_uint64 frameCountOut = 2000;
- ma_result result = ma_resampler_process_pcm_frames(&resampler, pFramesIn, &frameCountIn, pFramesOut, &frameCountOut);
+ result = ma_engine_init(&engineConfig, &engine);
if (result != MA_SUCCESS) {
- // An error occurred...
+ return result; // Failed to initialize the engine.
}
+ ```
- // At this point, frameCountIn contains the number of input frames that were consumed and frameCountOut contains the
- // number of output frames written.
+In the example above we're passing in a pre-initialized device. Since the caller is the one in
+control of the device's data callback, it's their responsibility to manually call
+`ma_engine_read_pcm_frames()` from inside their data callback:
+
+ ```c
+ void playback_data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
+ {
+ ma_engine_read_pcm_frames(&g_Engine, pOutput, frameCount, NULL);
+ }
```
-To initialize the resampler you first need to set up a config (`ma_resampler_config`) with `ma_resampler_config_init()`. You need to specify the sample format
-you want to use, the number of channels, the input and output sample rate, and the algorithm.
+You can also use the engine independent of a device entirely:
-The sample format can be either `ma_format_s16` or `ma_format_f32`. If you need a different format you will need to perform pre- and post-conversions yourself
-where necessary. Note that the format is the same for both input and output. The format cannot be changed after initialization.
+ ```c
+ ma_result result;
+ ma_engine engine;
+ ma_engine_config engineConfig;
-The resampler supports multiple channels and is always interleaved (both input and output). The channel count cannot be changed after initialization.
+ engineConfig = ma_engine_config_init();
+ engineConfig.noDevice = MA_TRUE;
+ engineConfig.channels = 2; // Must be set when not using a device.
+ engineConfig.sampleRate = 48000; // Must be set when not using a device.
-The sample rates can be anything other than zero, and are always specified in hertz. They should be set to something like 44100, etc. The sample rate is the
-only configuration property that can be changed after initialization.
+ result = ma_engine_init(&engineConfig, &engine);
+ if (result != MA_SUCCESS) {
+ return result; // Failed to initialize the engine.
+ }
+ ```
-The miniaudio resampler supports multiple algorithms:
+Note that when you're not using a device, you must set the channel count and sample rate in the
+config or else miniaudio won't know what to use (miniaudio will use the device to determine this
+normally). When not using a device, you need to use `ma_engine_read_pcm_frames()` to process audio
+data from the engine. This kind of setup is useful if you want to do something like offline
+processing.
- +-----------+------------------------------+
- | Algorithm | Enum Token |
- +-----------+------------------------------+
- | Linear | ma_resample_algorithm_linear |
- | Speex | ma_resample_algorithm_speex |
- +-----------+------------------------------+
+When a sound is loaded it goes through a resource manager. By default the engine will initialize a
+resource manager internally, but you can also specify a pre-initialized resource manager:
-Because Speex is not public domain it is strictly opt-in and the code is stored in separate files. if you opt-in to the Speex backend you will need to consider
-it's license, the text of which can be found in it's source files in "extras/speex_resampler". Details on how to opt-in to the Speex resampler is explained in
-the Speex Resampler section below.
+ ```c
+ ma_result result;
+ ma_engine engine1;
+ ma_engine engine2;
+ ma_engine_config engineConfig;
-The algorithm cannot be changed after initialization.
+ engineConfig = ma_engine_config_init();
+ engineConfig.pResourceManager = &myResourceManager;
-Processing always happens on a per PCM frame basis and always assumes interleaved input and output. De-interleaved processing is not supported. To process
-frames, use `ma_resampler_process_pcm_frames()`. On input, this function takes the number of output frames you can fit in the output buffer and the number of
-input frames contained in the input buffer. On output these variables contain the number of output frames that were written to the output buffer and the
-number of input frames that were consumed in the process. You can pass in NULL for the input buffer in which case it will be treated as an infinitely large
-buffer of zeros. The output buffer can also be NULL, in which case the processing will be treated as seek.
+ ma_engine_init(&engineConfig, &engine1);
+ ma_engine_init(&engineConfig, &engine2);
+ ```
-The sample rate can be changed dynamically on the fly. You can change this with explicit sample rates with `ma_resampler_set_rate()` and also with a decimal
-ratio with `ma_resampler_set_rate_ratio()`. The ratio is in/out.
+In this example we are initializing two engines, both of which are sharing the same resource
+manager. This is especially useful for saving memory when loading the same file across multiple
+engines. If you were not to use a shared resource manager, each engine instance would use their own
+which would result in any sounds that are used between both engine's being loaded twice. By using
+a shared resource manager, it would only be loaded once. Using multiple engine's is useful when you
+need to output to multiple playback devices, such as in a local multiplayer game where each player
+is using their own set of headphones.
-Sometimes it's useful to know exactly how many input frames will be required to output a specific number of frames. You can calculate this with
-`ma_resampler_get_required_input_frame_count()`. Likewise, it's sometimes useful to know exactly how many frames would be output given a certain number of
-input frames. You can do this with `ma_resampler_get_expected_output_frame_count()`.
+By default an engine will be in a started state. To make it so the engine is not automatically
+started you can configure it as such:
-Due to the nature of how resampling works, the resampler introduces some latency. This can be retrieved in terms of both the input rate and the output rate
-with `ma_resampler_get_input_latency()` and `ma_resampler_get_output_latency()`.
+ ```c
+ engineConfig.noAutoStart = MA_TRUE;
+ // The engine will need to be started manually.
+ ma_engine_start(&engine);
-6.3.1. Resampling Algorithms
-----------------------------
-The choice of resampling algorithm depends on your situation and requirements. The linear resampler is the most efficient and has the least amount of latency,
-but at the expense of poorer quality. The Speex resampler is higher quality, but slower with more latency. It also performs several heap allocations internally
-for memory management.
+ // Later on the engine can be stopped with ma_engine_stop().
+ ma_engine_stop(&engine);
+ ```
+The concept of starting or stopping an engine is only relevant when using the engine with a
+device. Attempting to start or stop an engine that is not associated with a device will result in
+`MA_INVALID_OPERATION`.
-6.3.1.1. Linear Resampling
---------------------------
-The linear resampler is the fastest, but comes at the expense of poorer quality. There is, however, some control over the quality of the linear resampler which
-may make it a suitable option depending on your requirements.
+The master volume of the engine can be controlled with `ma_engine_set_volume()` which takes a
+linear scale, with 0 resulting in silence and anything above 1 resulting in amplification. If you
+prefer decibel based volume control, use `ma_volume_db_to_linear()` to convert from dB to linear.
-The linear resampler performs low-pass filtering before or after downsampling or upsampling, depending on the sample rates you're converting between. When
-decreasing the sample rate, the low-pass filter will be applied before downsampling. When increasing the rate it will be performed after upsampling. By default
-a fourth order low-pass filter will be applied. This can be configured via the `lpfOrder` configuration variable. Setting this to 0 will disable filtering.
+When a sound is spatialized, it is done so relative to a listener. An engine can be configured to
+have multiple listeners which can be configured via the config:
-The low-pass filter has a cutoff frequency which defaults to half the sample rate of the lowest of the input and output sample rates (Nyquist Frequency). This
-can be controlled with the `lpfNyquistFactor` config variable. This defaults to 1, and should be in the range of 0..1, although a value of 0 does not make
-sense and should be avoided. A value of 1 will use the Nyquist Frequency as the cutoff. A value of 0.5 will use half the Nyquist Frequency as the cutoff, etc.
-Values less than 1 will result in more washed out sound due to more of the higher frequencies being removed. This config variable has no impact on performance
-and is a purely perceptual configuration.
+ ```c
+ engineConfig.listenerCount = 2;
+ ```
-The API for the linear resampler is the same as the main resampler API, only it's called `ma_linear_resampler`.
+The maximum number of listeners is restricted to `MA_ENGINE_MAX_LISTENERS`. By default, when a
+sound is spatialized, it will be done so relative to the closest listener. You can also pin a sound
+to a specific listener which will be explained later. Listener's have a position, direction, cone,
+and velocity (for doppler effect). A listener is referenced by an index, the meaning of which is up
+to the caller (the index is 0 based and cannot go beyond the listener count, minus 1). The
+position, direction and velocity are all specified in absolute terms:
+ ```c
+ ma_engine_listener_set_position(&engine, listenerIndex, worldPosX, worldPosY, worldPosZ);
+ ```
-6.3.1.2. Speex Resampling
--------------------------
-The Speex resampler is made up of third party code which is released under the BSD license. Because it is licensed differently to miniaudio, which is public
-domain, it is strictly opt-in and all of it's code is stored in separate files. If you opt-in to the Speex resampler you must consider the license text in it's
-source files. To opt-in, you must first `#include` the following file before the implementation of miniaudio.h:
+The direction of the listener represents it's forward vector. The listener's up vector can also be
+specified and defaults to +1 on the Y axis.
```c
- #include "extras/speex_resampler/ma_speex_resampler.h"
+ ma_engine_listener_set_direction(&engine, listenerIndex, forwardX, forwardY, forwardZ);
+ ma_engine_listener_set_world_up(&engine, listenerIndex, 0, 1, 0);
```
-Both the header and implementation is contained within the same file. The implementation can be included in your program like so:
+The engine supports directional attenuation. The listener can have a cone the controls how sound is
+attenuated based on the listener's direction. When a sound is between the inner and outer cones, it
+will be attenuated between 1 and the cone's outer gain:
```c
- #define MINIAUDIO_SPEEX_RESAMPLER_IMPLEMENTATION
- #include "extras/speex_resampler/ma_speex_resampler.h"
+ ma_engine_listener_set_cone(&engine, listenerIndex, innerAngleInRadians, outerAngleInRadians, outerGain);
```
-Note that even if you opt-in to the Speex backend, miniaudio won't use it unless you explicitly ask for it in the respective config of the object you are
-initializing. If you try to use the Speex resampler without opting in, initialization of the `ma_resampler` object will fail with `MA_NO_BACKEND`.
+When a sound is inside the inner code, no directional attenuation is applied. When the sound is
+outside of the outer cone, the attenuation will be set to `outerGain` in the example above. When
+the sound is in between the inner and outer cones, the attenuation will be interpolated between 1
+and the outer gain.
-The only configuration option to consider with the Speex resampler is the `speex.quality` config variable. This is a value between 0 and 10, with 0 being
-the fastest with the poorest quality and 10 being the slowest with the highest quality. The default value is 3.
+The engine's coordinate system follows the OpenGL coordinate system where positive X points right,
+positive Y points up and negative Z points forward.
+The simplest and least flexible way to play a sound is like so:
+ ```c
+ ma_engine_play_sound(&engine, "my_sound.wav", pGroup);
+ ```
-6.4. General Data Conversion
-----------------------------
-The `ma_data_converter` API can be used to wrap sample format conversion, channel conversion and resampling into one operation. This is what miniaudio uses
-internally to convert between the format requested when the device was initialized and the format of the backend's native device. The API for general data
-conversion is very similar to the resampling API. Create a `ma_data_converter` object like this:
+This is a "fire and forget" style of function. The engine will manage the `ma_sound` object
+internally. When the sound finishes playing, it'll be put up for recycling. For more flexibility
+you'll want to initialize a sound object:
```c
- ma_data_converter_config config = ma_data_converter_config_init(
- inputFormat,
- outputFormat,
- inputChannels,
- outputChannels,
- inputSampleRate,
- outputSampleRate
- );
+ ma_sound sound;
- ma_data_converter converter;
- ma_result result = ma_data_converter_init(&config, &converter);
+ result = ma_sound_init_from_file(&engine, "my_sound.wav", flags, pGroup, NULL, &sound);
if (result != MA_SUCCESS) {
- // An error occurred...
+ return result; // Failed to load sound.
}
```
-In the example above we use `ma_data_converter_config_init()` to initialize the config, however there's many more properties that can be configured, such as
-channel maps and resampling quality. Something like the following may be more suitable depending on your requirements:
-
- ```c
- ma_data_converter_config config = ma_data_converter_config_init_default();
- config.formatIn = inputFormat;
- config.formatOut = outputFormat;
- config.channelsIn = inputChannels;
- config.channelsOut = outputChannels;
- config.sampleRateIn = inputSampleRate;
- config.sampleRateOut = outputSampleRate;
- ma_get_standard_channel_map(ma_standard_channel_map_flac, config.channelCountIn, config.channelMapIn);
- config.resampling.linear.lpfOrder = MA_MAX_FILTER_ORDER;
- ```
+Sounds need to be uninitialized with `ma_sound_uninit()`.
-Do the following to uninitialize the data converter:
+The example above loads a sound from a file. If the resource manager has been disabled you will not
+be able to use this function and instead you'll need to initialize a sound directly from a data
+source:
```c
- ma_data_converter_uninit(&converter);
+ ma_sound sound;
+
+ result = ma_sound_init_from_data_source(&engine, &dataSource, flags, pGroup, &sound);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
```
-The following example shows how data can be processed
+Each `ma_sound` object represents a single instance of the sound. If you want to play the same
+sound multiple times at the same time, you need to initialize a separate `ma_sound` object.
+
+For the most flexibility when initializing sounds, use `ma_sound_init_ex()`. This uses miniaudio's
+standard config/init pattern:
```c
- ma_uint64 frameCountIn = 1000;
- ma_uint64 frameCountOut = 2000;
- ma_result result = ma_data_converter_process_pcm_frames(&converter, pFramesIn, &frameCountIn, pFramesOut, &frameCountOut);
+ ma_sound sound;
+ ma_sound_config soundConfig;
+
+ soundConfig = ma_sound_config_init();
+ soundConfig.pFilePath = NULL; // Set this to load from a file path.
+ soundConfig.pDataSource = NULL; // Set this to initialize from an existing data source.
+ soundConfig.pInitialAttachment = &someNodeInTheNodeGraph;
+ soundConfig.initialAttachmentInputBusIndex = 0;
+ soundConfig.channelsIn = 1;
+ soundConfig.channelsOut = 0; // Set to 0 to use the engine's native channel count.
+
+ result = ma_sound_init_ex(&soundConfig, &sound);
if (result != MA_SUCCESS) {
- // An error occurred...
+ return result;
}
-
- // At this point, frameCountIn contains the number of input frames that were consumed and frameCountOut contains the number
- // of output frames written.
```
-The data converter supports multiple channels and is always interleaved (both input and output). The channel count cannot be changed after initialization.
+In the example above, the sound is being initialized without a file nor a data source. This is
+valid, in which case the sound acts as a node in the middle of the node graph. This means you can
+connect other sounds to this sound and allow it to act like a sound group. Indeed, this is exactly
+what a `ma_sound_group` is.
-Sample rates can be anything other than zero, and are always specified in hertz. They should be set to something like 44100, etc. The sample rate is the only
-configuration property that can be changed after initialization, but only if the `resampling.allowDynamicSampleRate` member of `ma_data_converter_config` is
-set to `MA_TRUE`. To change the sample rate, use `ma_data_converter_set_rate()` or `ma_data_converter_set_rate_ratio()`. The ratio must be in/out. The
-resampling algorithm cannot be changed after initialization.
+When loading a sound, you specify a set of flags that control how the sound is loaded and what
+features are enabled for that sound. When no flags are set, the sound will be fully loaded into
+memory in exactly the same format as how it's stored on the file system. The resource manager will
+allocate a block of memory and then load the file directly into it. When reading audio data, it
+will be decoded dynamically on the fly. In order to save processing time on the audio thread, it
+might be beneficial to pre-decode the sound. You can do this with the `MA_SOUND_FLAG_DECODE` flag:
-Processing always happens on a per PCM frame basis and always assumes interleaved input and output. De-interleaved processing is not supported. To process
-frames, use `ma_data_converter_process_pcm_frames()`. On input, this function takes the number of output frames you can fit in the output buffer and the number
-of input frames contained in the input buffer. On output these variables contain the number of output frames that were written to the output buffer and the
-number of input frames that were consumed in the process. You can pass in NULL for the input buffer in which case it will be treated as an infinitely large
-buffer of zeros. The output buffer can also be NULL, in which case the processing will be treated as seek.
+ ```c
+ ma_sound_init_from_file(&engine, "my_sound.wav", MA_SOUND_FLAG_DECODE, pGroup, NULL, &sound);
+ ```
-Sometimes it's useful to know exactly how many input frames will be required to output a specific number of frames. You can calculate this with
-`ma_data_converter_get_required_input_frame_count()`. Likewise, it's sometimes useful to know exactly how many frames would be output given a certain number of
-input frames. You can do this with `ma_data_converter_get_expected_output_frame_count()`.
+By default, sounds will be loaded synchronously, meaning `ma_sound_init_*()` will not return until
+the sound has been fully loaded. If this is prohibitive you can instead load sounds asynchronously
+by specificying the `MA_SOUND_FLAG_ASYNC` flag:
-Due to the nature of how resampling works, the data converter introduces some latency if resampling is required. This can be retrieved in terms of both the
-input rate and the output rate with `ma_data_converter_get_input_latency()` and `ma_data_converter_get_output_latency()`.
+ ```c
+ ma_sound_init_from_file(&engine, "my_sound.wav", MA_SOUND_FLAG_DECODE | MA_SOUND_FLAG_ASYNC, pGroup, NULL, &sound);
+ ```
+This will result in `ma_sound_init_*()` returning quickly, but the sound won't yet have been fully
+loaded. When you start the sound, it won't output anything until some sound is available. The sound
+will start outputting audio before the sound has been fully decoded when the `MA_SOUND_FLAG_DECODE`
+is specified.
+If you need to wait for an asynchronously loaded sound to be fully loaded, you can use a fence. A
+fence in miniaudio is a simple synchronization mechanism which simply blocks until it's internal
+counter hit's zero. You can specify a fence like so:
-7. Filtering
-============
+ ```c
+ ma_result result;
+ ma_fence fence;
+ ma_sound sounds[4];
-7.1. Biquad Filtering
----------------------
-Biquad filtering is achieved with the `ma_biquad` API. Example:
+ result = ma_fence_init(&fence);
+ if (result != MA_SUCCES) {
+ return result;
+ }
- ```c
- ma_biquad_config config = ma_biquad_config_init(ma_format_f32, channels, b0, b1, b2, a0, a1, a2);
- ma_result result = ma_biquad_init(&config, &biquad);
- if (result != MA_SUCCESS) {
- // Error.
+ // Load some sounds asynchronously.
+ for (int iSound = 0; iSound < 4; iSound += 1) {
+ ma_sound_init_from_file(&engine, mySoundFilesPaths[iSound], MA_SOUND_FLAG_DECODE | MA_SOUND_FLAG_ASYNC, pGroup, &fence, &sounds[iSound]);
}
- ...
+ // ... do some other stuff here in the mean time ...
- ma_biquad_process_pcm_frames(&biquad, pFramesOut, pFramesIn, frameCount);
+ // Wait for all sounds to finish loading.
+ ma_fence_wait(&fence);
```
-Biquad filtering is implemented using transposed direct form 2. The numerator coefficients are b0, b1 and b2, and the denominator coefficients are a0, a1 and
-a2. The a0 coefficient is required and coefficients must not be pre-normalized.
+If loading the entire sound into memory is prohibitive, you can also configure the engine to stream
+the audio data:
-Supported formats are `ma_format_s16` and `ma_format_f32`. If you need to use a different format you need to convert it yourself beforehand. When using
-`ma_format_s16` the biquad filter will use fixed point arithmetic. When using `ma_format_f32`, floating point arithmetic will be used.
+ ```c
+ ma_sound_init_from_file(&engine, "my_sound.wav", MA_SOUND_FLAG_STREAM, pGroup, NULL, &sound);
+ ```
-Input and output frames are always interleaved.
+When streaming sounds, 2 seconds worth of audio data is stored in memory. Although it should work
+fine, it's inefficient to use streaming for short sounds. Streaming is useful for things like music
+tracks in games.
+
+When you initialize a sound, if you specify a sound group the sound will be attached to that group
+automatically. If you set it to NULL, it will be automatically attached to the engine's endpoint.
+If you would instead rather leave the sound unattached by default, you can can specify the
+`MA_SOUND_FLAG_NO_DEFAULT_ATTACHMENT` flag. This is useful if you want to set up a complex node
+graph.
+
+Sounds are not started by default. To start a sound, use `ma_sound_start()`. Stop a sound with
+`ma_sound_stop()`.
+
+Sounds can have their volume controlled with `ma_sound_set_volume()` in the same way as the
+engine's master volume.
+
+Sounds support stereo panning and pitching. Set the pan with `ma_sound_set_pan()`. Setting the pan
+to 0 will result in an unpanned sound. Setting it to -1 will shift everything to the left, whereas
++1 will shift it to the right. The pitch can be controlled with `ma_sound_set_pitch()`. A larger
+value will result in a higher pitch. The pitch must be greater than 0.
-Filtering can be applied in-place by passing in the same pointer for both the input and output buffers, like so:
+The engine supports 3D spatialization of sounds. By default sounds will have spatialization
+enabled, but if a sound does not need to be spatialized it's best to disable it. There are two ways
+to disable spatialization of a sound:
```c
- ma_biquad_process_pcm_frames(&biquad, pMyData, pMyData, frameCount);
+ // Disable spatialization at initialization time via a flag:
+ ma_sound_init_from_file(&engine, "my_sound.wav", MA_SOUND_FLAG_NO_SPATIALIZATION, NULL, NULL, &sound);
+
+ // Dynamically disable or enable spatialization post-initialization:
+ ma_sound_set_spatialization_enabled(&sound, isSpatializationEnabled);
```
-If you need to change the values of the coefficients, but maintain the values in the registers you can do so with `ma_biquad_reinit()`. This is useful if you
-need to change the properties of the filter while keeping the values of registers valid to avoid glitching. Do not use `ma_biquad_init()` for this as it will
-do a full initialization which involves clearing the registers to 0. Note that changing the format or channel count after initialization is invalid and will
-result in an error.
+By default sounds will be spatialized based on the closest listener. If a sound should always be
+spatialized relative to a specific listener it can be pinned to one:
+ ```c
+ ma_sound_set_pinned_listener_index(&sound, listenerIndex);
+ ```
-7.2. Low-Pass Filtering
------------------------
-Low-pass filtering is achieved with the following APIs:
+Like listeners, sounds have a position. By default, the position of a sound is in absolute space,
+but it can be changed to be relative to a listener:
- +---------+------------------------------------------+
- | API | Description |
- +---------+------------------------------------------+
- | ma_lpf1 | First order low-pass filter |
- | ma_lpf2 | Second order low-pass filter |
- | ma_lpf | High order low-pass filter (Butterworth) |
- +---------+------------------------------------------+
+ ```c
+ ma_sound_set_positioning(&sound, ma_positioning_relative);
+ ```
-Low-pass filter example:
+Note that relative positioning of a sound only makes sense if there is either only one listener, or
+the sound is pinned to a specific listener. To set the position of a sound:
```c
- ma_lpf_config config = ma_lpf_config_init(ma_format_f32, channels, sampleRate, cutoffFrequency, order);
- ma_result result = ma_lpf_init(&config, &lpf);
- if (result != MA_SUCCESS) {
- // Error.
- }
+ ma_sound_set_position(&sound, posX, posY, posZ);
+ ```
- ...
+The direction works the same way as a listener and represents the sound's forward direction:
- ma_lpf_process_pcm_frames(&lpf, pFramesOut, pFramesIn, frameCount);
+ ```c
+ ma_sound_set_direction(&sound, forwardX, forwardY, forwardZ);
```
-Supported formats are `ma_format_s16` and` ma_format_f32`. If you need to use a different format you need to convert it yourself beforehand. Input and output
-frames are always interleaved.
+Sound's also have a cone for controlling directional attenuation. This works exactly the same as
+listeners:
+
+ ```c
+ ma_sound_set_cone(&sound, innerAngleInRadians, outerAngleInRadians, outerGain);
+ ```
-Filtering can be applied in-place by passing in the same pointer for both the input and output buffers, like so:
+The velocity of a sound is used for doppler effect and can be set as such:
```c
- ma_lpf_process_pcm_frames(&lpf, pMyData, pMyData, frameCount);
+ ma_sound_set_velocity(&sound, velocityX, velocityY, velocityZ);
```
-The maximum filter order is limited to `MA_MAX_FILTER_ORDER` which is set to 8. If you need more, you can chain first and second order filters together.
+The engine supports different attenuation models which can be configured on a per-sound basis. By
+default the attenuation model is set to `ma_attenuation_model_inverse` which is the equivalent to
+OpenAL's `AL_INVERSE_DISTANCE_CLAMPED`. Configure the attenuation model like so:
```c
- for (iFilter = 0; iFilter < filterCount; iFilter += 1) {
- ma_lpf2_process_pcm_frames(&lpf2[iFilter], pMyData, pMyData, frameCount);
- }
+ ma_sound_set_attenuation_model(&sound, ma_attenuation_model_inverse);
```
-If you need to change the configuration of the filter, but need to maintain the state of internal registers you can do so with `ma_lpf_reinit()`. This may be
-useful if you need to change the sample rate and/or cutoff frequency dynamically while maintaing smooth transitions. Note that changing the format or channel
-count after initialization is invalid and will result in an error.
+The supported attenuation models include the following:
-The `ma_lpf` object supports a configurable order, but if you only need a first order filter you may want to consider using `ma_lpf1`. Likewise, if you only
-need a second order filter you can use `ma_lpf2`. The advantage of this is that they're lighter weight and a bit more efficient.
+ +----------------------------------+----------------------------------------------+
+ | ma_attenuation_model_none | No distance attenuation. |
+ +----------------------------------+----------------------------------------------+
+ | ma_attenuation_model_inverse | Equivalent to `AL_INVERSE_DISTANCE_CLAMPED`. |
+ +----------------------------------+----------------------------------------------+
+ | ma_attenuation_model_linear | Linear attenuation. |
+ +----------------------------------+----------------------------------------------+
+ | ma_attenuation_model_exponential | Exponential attenuation. |
+ +----------------------------------+----------------------------------------------+
-If an even filter order is specified, a series of second order filters will be processed in a chain. If an odd filter order is specified, a first order filter
-will be applied, followed by a series of second order filters in a chain.
+To control how quickly a sound rolls off as it moves away from the listener, you need to configure
+the rolloff:
+ ```c
+ ma_sound_set_rolloff(&sound, rolloff);
+ ```
-7.3. High-Pass Filtering
-------------------------
-High-pass filtering is achieved with the following APIs:
+You can control the minimum and maximum gain to apply from spatialization:
- +---------+-------------------------------------------+
- | API | Description |
- +---------+-------------------------------------------+
- | ma_hpf1 | First order high-pass filter |
- | ma_hpf2 | Second order high-pass filter |
- | ma_hpf | High order high-pass filter (Butterworth) |
- +---------+-------------------------------------------+
+ ```c
+ ma_sound_set_min_gain(&sound, minGain);
+ ma_sound_set_max_gain(&sound, maxGain);
+ ```
-High-pass filters work exactly the same as low-pass filters, only the APIs are called `ma_hpf1`, `ma_hpf2` and `ma_hpf`. See example code for low-pass filters
-for example usage.
+Likewise, in the calculation of attenuation, you can control the minimum and maximum distances for
+the attenuation calculation. This is useful if you want to ensure sounds don't drop below a certain
+volume after the listener moves further away and to have sounds play a maximum volume when the
+listener is within a certain distance:
+ ```c
+ ma_sound_set_min_distance(&sound, minDistance);
+ ma_sound_set_max_distance(&sound, maxDistance);
+ ```
-7.4. Band-Pass Filtering
-------------------------
-Band-pass filtering is achieved with the following APIs:
+The engine's spatialization system supports doppler effect. The doppler factor can be configure on
+a per-sound basis like so:
- +---------+-------------------------------+
- | API | Description |
- +---------+-------------------------------+
- | ma_bpf2 | Second order band-pass filter |
- | ma_bpf | High order band-pass filter |
- +---------+-------------------------------+
+ ```c
+ ma_sound_set_doppler_factor(&sound, dopplerFactor);
+ ```
-Band-pass filters work exactly the same as low-pass filters, only the APIs are called `ma_bpf2` and `ma_hpf`. See example code for low-pass filters for example
-usage. Note that the order for band-pass filters must be an even number which means there is no first order band-pass filter, unlike low-pass and high-pass
-filters.
+You can fade sounds in and out with `ma_sound_set_fade_in_pcm_frames()` and
+`ma_sound_set_fade_in_milliseconds()`. Set the volume to -1 to use the current volume as the
+starting volume:
+ ```c
+ // Fade in over 1 second.
+ ma_sound_set_fade_in_milliseconds(&sound, 0, 1, 1000);
-7.5. Notch Filtering
---------------------
-Notch filtering is achieved with the following APIs:
+ // ... sometime later ...
- +-----------+------------------------------------------+
- | API | Description |
- +-----------+------------------------------------------+
- | ma_notch2 | Second order notching filter |
- +-----------+------------------------------------------+
+ // Fade out over 1 second, starting from the current volume.
+ ma_sound_set_fade_in_milliseconds(&sound, -1, 0, 1000);
+ ```
+By default sounds will start immediately, but sometimes for timing and synchronization purposes it
+can be useful to schedule a sound to start or stop:
-7.6. Peaking EQ Filtering
--------------------------
-Peaking filtering is achieved with the following APIs:
+ ```c
+ // Start the sound in 1 second from now.
+ ma_sound_set_start_time_in_pcm_frames(&sound, ma_engine_get_time(&engine) + (ma_engine_get_sample_rate(&engine) * 1));
- +----------+------------------------------------------+
- | API | Description |
- +----------+------------------------------------------+
- | ma_peak2 | Second order peaking filter |
- +----------+------------------------------------------+
+ // Stop the sound in 2 seconds from now.
+ ma_sound_set_stop_time_in_pcm_frames(&sound, ma_engine_get_time(&engine) + (ma_engine_get_sample_rate(&engine) * 2));
+ ```
+The time is specified in global time which is controlled by the engine. You can get the engine's
+current time with `ma_engine_get_time()`. The engine's global time is incremented automatically as
+audio data is read, but it can be reset with `ma_engine_set_time()` in case it needs to be
+resynchronized for some reason.
-7.7. Low Shelf Filtering
-------------------------
-Low shelf filtering is achieved with the following APIs:
+To determine whether or not a sound is currently playing, use `ma_sound_is_playing()`. This will
+take the scheduled start and stop times into account.
- +-------------+------------------------------------------+
- | API | Description |
- +-------------+------------------------------------------+
- | ma_loshelf2 | Second order low shelf filter |
- +-------------+------------------------------------------+
+Whether or not a sound should loop can be controlled with `ma_sound_set_looping()`. Sounds will not
+be looping by default. Use `ma_sound_is_looping()` to determine whether or not a sound is looping.
-Where a high-pass filter is used to eliminate lower frequencies, a low shelf filter can be used to just turn them down rather than eliminate them entirely.
+Use `ma_sound_at_end()` to determine whether or not a sound is currently at the end. For a looping
+sound this should never return true.
+Internally a sound wraps around a data source. Some APIs exist to control the underlying data
+source, mainly for convenience:
-7.8. High Shelf Filtering
--------------------------
-High shelf filtering is achieved with the following APIs:
+ ```c
+ ma_sound_seek_to_pcm_frame(&sound, frameIndex);
+ ma_sound_get_data_format(&sound, &format, &channels, &sampleRate, pChannelMap, channelMapCapacity);
+ ma_sound_get_cursor_in_pcm_frames(&sound, &cursor);
+ ma_sound_get_length_in_pcm_frames(&sound, &length);
+ ```
- +-------------+------------------------------------------+
- | API | Description |
- +-------------+------------------------------------------+
- | ma_hishelf2 | Second order high shelf filter |
- +-------------+------------------------------------------+
+Sound groups have the same API as sounds, only they are called `ma_sound_group`, and since they do
+not have any notion of a data source, anything relating to a data source is unavailable.
-The high shelf filter has the same API as the low shelf filter, only you would use `ma_hishelf` instead of `ma_loshelf`. Where a low shelf filter is used to
-adjust the volume of low frequencies, the high shelf filter does the same thing for high frequencies.
+Internally, sound data is loaded via the `ma_decoder` API which means by default in only supports
+file formats that have built-in support in miniaudio. You can extend this to support any kind of
+file format through the use of custom decoders. To do this you'll need to use a self-managed
+resource manager and configure it appropriately. See the "Resource Management" section below for
+details on how to set this up.
+6. Resource Management
+======================
+Many programs will want to manage sound resources for things such as reference counting and
+streaming. This is supported by miniaudio via the `ma_resource_manager` API.
+The resource manager is mainly responsible for the following:
-8. Waveform and Noise Generation
-================================
+ * Loading of sound files into memory with reference counting.
+ * Streaming of sound data
-8.1. Waveforms
---------------
-miniaudio supports generation of sine, square, triangle and sawtooth waveforms. This is achieved with the `ma_waveform` API. Example:
+When loading a sound file, the resource manager will give you back a `ma_data_source` compatible
+object called `ma_resource_manager_data_source`. This object can be passed into any
+`ma_data_source` API which is how you can read and seek audio data. When loading a sound file, you
+specify whether or not you want the sound to be fully loaded into memory (and optionally
+pre-decoded) or streamed. When loading into memory, you can also specify whether or not you want
+the data to be loaded asynchronously.
+
+The example below is how you can initialize a resource manager using it's default configuration:
```c
- ma_waveform_config config = ma_waveform_config_init(
- FORMAT,
- CHANNELS,
- SAMPLE_RATE,
- ma_waveform_type_sine,
- amplitude,
- frequency);
+ ma_resource_manager_config config;
+ ma_resource_manager resourceManager;
- ma_waveform waveform;
- ma_result result = ma_waveform_init(&config, &waveform);
+ config = ma_resource_manager_config_init();
+ result = ma_resource_manager_init(&config, &resourceManager);
if (result != MA_SUCCESS) {
- // Error.
+ ma_device_uninit(&device);
+ printf("Failed to initialize the resource manager.");
+ return -1;
}
+ ```
+
+You can configure the format, channels and sample rate of the decoded audio data. By default it
+will use the file's native data format, but you can configure it to use a consistent format. This
+is useful for offloading the cost of data conversion to load time rather than dynamically
+converting at mixing time. To do this, you configure the decoded format, channels and sample rate
+like the code below:
+
+ ```c
+ config = ma_resource_manager_config_init();
+ config.decodedFormat = device.playback.format;
+ config.decodedChannels = device.playback.channels;
+ config.decodedSampleRate = device.sampleRate;
+ ```
+
+In the code above, the resource manager will be configured so that any decoded audio data will be
+pre-converted at load time to the device's native data format. If instead you used defaults and
+the data format of the file did not match the device's data format, you would need to convert the
+data at mixing time which may be prohibitive in high-performance and large scale scenarios like
+games.
+
+Internally the resource manager uses the `ma_decoder` API to load sounds. This means by default it
+only supports decoders that are built into miniaudio. It's possible to support additional encoding
+formats through the use of custom decoders. To do so, pass in your `ma_decoding_backend_vtable`
+vtables into the resource manager config:
+
+ ```c
+ ma_decoding_backend_vtable* pCustomBackendVTables[] =
+ {
+ &g_ma_decoding_backend_vtable_libvorbis,
+ &g_ma_decoding_backend_vtable_libopus
+ };
...
- ma_waveform_read_pcm_frames(&waveform, pOutput, frameCount);
+ resourceManagerConfig.ppCustomDecodingBackendVTables = pCustomBackendVTables;
+ resourceManagerConfig.customDecodingBackendCount = sizeof(pCustomBackendVTables) / sizeof(pCustomBackendVTables[0]);
+ resourceManagerConfig.pCustomDecodingBackendUserData = NULL;
```
-The amplitude, frequency, type, and sample rate can be changed dynamically with `ma_waveform_set_amplitude()`, `ma_waveform_set_frequency()`,
-`ma_waveform_set_type()`, and `ma_waveform_set_sample_rate()` respectively.
+This system can allow you to support any kind of file format. See the "Decoding" section for
+details on how to implement custom decoders. The miniaudio repository includes examples for Opus
+via libopus and libopusfile and Vorbis via libvorbis and libvorbisfile.
-You can invert the waveform by setting the amplitude to a negative value. You can use this to control whether or not a sawtooth has a positive or negative
-ramp, for example.
+Asynchronicity is achieved via a job system. When an operation needs to be performed, such as the
+decoding of a page, a job will be posted to a queue which will then be processed by a job thread.
+By default there will be only one job thread running, but this can be configured, like so:
-Below are the supported waveform types:
+ ```c
+ config = ma_resource_manager_config_init();
+ config.jobThreadCount = MY_JOB_THREAD_COUNT;
+ ```
- +---------------------------+
- | Enum Name |
- +---------------------------+
- | ma_waveform_type_sine |
- | ma_waveform_type_square |
- | ma_waveform_type_triangle |
- | ma_waveform_type_sawtooth |
- +---------------------------+
+By default job threads are managed internally by the resource manager, however you can also self
+manage your job threads if, for example, you want to integrate the job processing into your
+existing job infrastructure, or if you simply don't like the way the resource manager does it. To
+do this, just set the job thread count to 0 and process jobs manually. To process jobs, you first
+need to retrieve a job using `ma_resource_manager_next_job()` and then process it using
+`ma_job_process()`:
+
+ ```c
+ config = ma_resource_manager_config_init();
+ config.jobThreadCount = 0; // Don't manage any job threads internally.
+ config.flags = MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING; // Optional. Makes `ma_resource_manager_next_job()` non-blocking.
+
+ // ... Initialize your custom job threads ...
+ void my_custom_job_thread(...)
+ {
+ for (;;) {
+ ma_job job;
+ ma_result result = ma_resource_manager_next_job(pMyResourceManager, &job);
+ if (result != MA_SUCCESS) {
+ if (result == MA_NOT_DATA_AVAILABLE) {
+ // No jobs are available. Keep going. Will only get this if the resource manager was initialized
+ // with MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING.
+ continue;
+ } else if (result == MA_CANCELLED) {
+ // MA_JOB_TYPE_QUIT was posted. Exit.
+ break;
+ } else {
+ // Some other error occurred.
+ break;
+ }
+ }
+ ma_job_process(&job);
+ }
+ }
+ ```
-8.2. Noise
-----------
-miniaudio supports generation of white, pink and Brownian noise via the `ma_noise` API. Example:
+In the example above, the `MA_JOB_TYPE_QUIT` event is the used as the termination
+indicator, but you can use whatever you would like to terminate the thread. The call to
+`ma_resource_manager_next_job()` is blocking by default, but can be configured to be non-blocking
+by initializing the resource manager with the `MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING` configuration
+flag. Note that the `MA_JOB_TYPE_QUIT` will never be removed from the job queue. This
+is to give every thread the opportunity to catch the event and terminate naturally.
+
+When loading a file, it's sometimes convenient to be able to customize how files are opened and
+read instead of using standard `fopen()`, `fclose()`, etc. which is what miniaudio will use by
+default. This can be done by setting `pVFS` member of the resource manager's config:
```c
- ma_noise_config config = ma_noise_config_init(
- FORMAT,
- CHANNELS,
- ma_noise_type_white,
- SEED,
- amplitude);
+ // Initialize your custom VFS object. See documentation for VFS for information on how to do this.
+ my_custom_vfs vfs = my_custom_vfs_init();
- ma_noise noise;
- ma_result result = ma_noise_init(&config, &noise);
+ config = ma_resource_manager_config_init();
+ config.pVFS = &vfs;
+ ```
+
+This is particularly useful in programs like games where you want to read straight from an archive
+rather than the normal file system. If you do not specify a custom VFS, the resource manager will
+use the operating system's normal file operations. This is default.
+
+To load a sound file and create a data source, call `ma_resource_manager_data_source_init()`. When
+loading a sound you need to specify the file path and options for how the sounds should be loaded.
+By default a sound will be loaded synchronously. The returned data source is owned by the caller
+which means the caller is responsible for the allocation and freeing of the data source. Below is
+an example for initializing a data source:
+
+ ```c
+ ma_resource_manager_data_source dataSource;
+ ma_result result = ma_resource_manager_data_source_init(pResourceManager, pFilePath, flags, &dataSource);
if (result != MA_SUCCESS) {
// Error.
}
- ...
+ // ...
- ma_noise_read_pcm_frames(&noise, pOutput, frameCount);
- ```
+ // A ma_resource_manager_data_source object is compatible with the `ma_data_source` API. To read data, just call
+ // the `ma_data_source_read_pcm_frames()` like you would with any normal data source.
+ result = ma_data_source_read_pcm_frames(&dataSource, pDecodedData, frameCount, &framesRead);
+ if (result != MA_SUCCESS) {
+ // Failed to read PCM frames.
+ }
-The noise API uses simple LCG random number generation. It supports a custom seed which is useful for things like automated testing requiring reproducibility.
-Setting the seed to zero will default to `MA_DEFAULT_LCG_SEED`.
+ // ...
-The amplitude, seed, and type can be changed dynamically with `ma_noise_set_amplitude()`, `ma_noise_set_seed()`, and `ma_noise_set_type()` respectively.
+ ma_resource_manager_data_source_uninit(pResourceManager, &dataSource);
+ ```
-By default, the noise API will use different values for different channels. So, for example, the left side in a stereo stream will be different to the right
-side. To instead have each channel use the same random value, set the `duplicateChannels` member of the noise config to true, like so:
+The `flags` parameter specifies how you want to perform loading of the sound file. It can be a
+combination of the following flags:
- ```c
- config.duplicateChannels = MA_TRUE;
+ ```
+ MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM
+ MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE
+ MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC
+ MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT
```
-Below are the supported noise types.
-
- +------------------------+
- | Enum Name |
- +------------------------+
- | ma_noise_type_white |
- | ma_noise_type_pink |
- | ma_noise_type_brownian |
- +------------------------+
+When no flags are specified (set to 0), the sound will be fully loaded into memory, but not
+decoded, meaning the raw file data will be stored in memory, and then dynamically decoded when
+`ma_data_source_read_pcm_frames()` is called. To instead decode the audio data before storing it in
+memory, use the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE` flag. By default, the sound file will
+be loaded synchronously, meaning `ma_resource_manager_data_source_init()` will only return after
+the entire file has been loaded. This is good for simplicity, but can be prohibitively slow. You
+can instead load the sound asynchronously using the `MA_RESOURCE_MANAGER_DATA_SOURCE_ASYNC` flag.
+This will result in `ma_resource_manager_data_source_init()` returning quickly, but no data will be
+returned by `ma_data_source_read_pcm_frames()` until some data is available. When no data is
+available because the asynchronous decoding hasn't caught up, `MA_BUSY` will be returned by
+`ma_data_source_read_pcm_frames()`.
+
+For large sounds, it's often prohibitive to store the entire file in memory. To mitigate this, you
+can instead stream audio data which you can do by specifying the
+`MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM` flag. When streaming, data will be decoded in 1
+second pages. When a new page needs to be decoded, a job will be posted to the job queue and then
+subsequently processed in a job thread.
+
+For in-memory sounds, reference counting is used to ensure the data is loaded only once. This means
+multiple calls to `ma_resource_manager_data_source_init()` with the same file path will result in
+the file data only being loaded once. Each call to `ma_resource_manager_data_source_init()` must be
+matched up with a call to `ma_resource_manager_data_source_uninit()`. Sometimes it can be useful
+for a program to register self-managed raw audio data and associate it with a file path. Use the
+`ma_resource_manager_register_*()` and `ma_resource_manager_unregister_*()` APIs to do this.
+`ma_resource_manager_register_decoded_data()` is used to associate a pointer to raw, self-managed
+decoded audio data in the specified data format with the specified name. Likewise,
+`ma_resource_manager_register_encoded_data()` is used to associate a pointer to raw self-managed
+encoded audio data (the raw file data) with the specified name. Note that these names need not be
+actual file paths. When `ma_resource_manager_data_source_init()` is called (without the
+`MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM` flag), the resource manager will look for these
+explicitly registered data buffers and, if found, will use it as the backing data for the data
+source. Note that the resource manager does *not* make a copy of this data so it is up to the
+caller to ensure the pointer stays valid for it's lifetime. Use
+`ma_resource_manager_unregister_data()` to unregister the self-managed data. You can also use
+`ma_resource_manager_register_file()` and `ma_resource_manager_unregister_file()` to register and
+unregister a file. It does not make sense to use the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM`
+flag with a self-managed data pointer.
+
+
+6.1. Asynchronous Loading and Synchronization
+---------------------------------------------
+When loading asynchronously, it can be useful to poll whether or not loading has finished. Use
+`ma_resource_manager_data_source_result()` to determine this. For in-memory sounds, this will
+return `MA_SUCCESS` when the file has been *entirely* decoded. If the sound is still being decoded,
+`MA_BUSY` will be returned. Otherwise, some other error code will be returned if the sound failed
+to load. For streaming data sources, `MA_SUCCESS` will be returned when the first page has been
+decoded and the sound is ready to be played. If the first page is still being decoded, `MA_BUSY`
+will be returned. Otherwise, some other error code will be returned if the sound failed to load.
+In addition to polling, you can also use a simple synchronization object called a "fence" to wait
+for asynchronously loaded sounds to finish. This is called `ma_fence`. The advantage to using a
+fence is that it can be used to wait for a group of sounds to finish loading rather than waiting
+for sounds on an individual basis. There are two stages to loading a sound:
+ * Initialization of the internal decoder; and
+ * Completion of decoding of the file (the file is fully decoded)
-9. Audio Buffers
-================
-miniaudio supports reading from a buffer of raw audio data via the `ma_audio_buffer` API. This can read from memory that's managed by the application, but
-can also handle the memory management for you internally. Memory management is flexible and should support most use cases.
+You can specify separate fences for each of the different stages. Waiting for the initialization
+of the internal decoder is important for when you need to know the sample format, channels and
+sample rate of the file.
-Audio buffers are initialised using the standard configuration system used everywhere in miniaudio:
+The example below shows how you could use a fence when loading a number of sounds:
```c
- ma_audio_buffer_config config = ma_audio_buffer_config_init(
- format,
- channels,
- sizeInFrames,
- pExistingData,
- &allocationCallbacks);
+ // This fence will be released when all sounds are finished loading entirely.
+ ma_fence fence;
+ ma_fence_init(&fence);
- ma_audio_buffer buffer;
- result = ma_audio_buffer_init(&config, &buffer);
- if (result != MA_SUCCESS) {
- // Error.
+ // This will be passed into the initialization routine for each sound.
+ ma_resource_manager_pipeline_notifications notifications = ma_resource_manager_pipeline_notifications_init();
+ notifications.done.pFence = &fence;
+
+ // Now load a bunch of sounds:
+ for (iSound = 0; iSound < soundCount; iSound += 1) {
+ ma_resource_manager_data_source_init(pResourceManager, pSoundFilePaths[iSound], flags, ¬ifications, &pSoundSources[iSound]);
}
- ...
+ // ... DO SOMETHING ELSE WHILE SOUNDS ARE LOADING ...
- ma_audio_buffer_uninit(&buffer);
+ // Wait for loading of sounds to finish.
+ ma_fence_wait(&fence);
```
-In the example above, the memory pointed to by `pExistingData` will *not* be copied and is how an application can do self-managed memory allocation. If you
-would rather make a copy of the data, use `ma_audio_buffer_init_copy()`. To uninitialize the buffer, use `ma_audio_buffer_uninit()`.
+In the example above we used a fence for waiting until the entire file has been fully decoded. If
+you only need to wait for the initialization of the internal decoder to complete, you can use the
+`init` member of the `ma_resource_manager_pipeline_notifications` object:
-Sometimes it can be convenient to allocate the memory for the `ma_audio_buffer` structure and the raw audio data in a contiguous block of memory. That is,
-the raw audio data will be located immediately after the `ma_audio_buffer` structure. To do this, use `ma_audio_buffer_alloc_and_init()`:
+ ```c
+ notifications.init.pFence = &fence;
+ ```
+
+If a fence is not appropriate for your situation, you can instead use a callback that is fired on
+an individual sound basis. This is done in a very similar way to fences:
```c
- ma_audio_buffer_config config = ma_audio_buffer_config_init(
- format,
- channels,
- sizeInFrames,
- pExistingData,
- &allocationCallbacks);
+ typedef struct
+ {
+ ma_async_notification_callbacks cb;
+ void* pMyData;
+ } my_notification;
- ma_audio_buffer* pBuffer
- result = ma_audio_buffer_alloc_and_init(&config, &pBuffer);
- if (result != MA_SUCCESS) {
- // Error
+ void my_notification_callback(ma_async_notification* pNotification)
+ {
+ my_notification* pMyNotification = (my_notification*)pNotification;
+
+ // Do something in response to the sound finishing loading.
}
...
- ma_audio_buffer_uninit_and_free(&buffer);
- ```
+ my_notification myCallback;
+ myCallback.cb.onSignal = my_notification_callback;
+ myCallback.pMyData = pMyData;
+
+ ma_resource_manager_pipeline_notifications notifications = ma_resource_manager_pipeline_notifications_init();
+ notifications.done.pNotification = &myCallback;
-If you initialize the buffer with `ma_audio_buffer_alloc_and_init()` you should uninitialize it with `ma_audio_buffer_uninit_and_free()`. In the example above,
-the memory pointed to by `pExistingData` will be copied into the buffer, which is contrary to the behavior of `ma_audio_buffer_init()`.
+ ma_resource_manager_data_source_init(pResourceManager, "my_sound.wav", flags, ¬ifications, &mySound);
+ ```
-An audio buffer has a playback cursor just like a decoder. As you read frames from the buffer, the cursor moves forward. The last parameter (`loop`) can be
-used to determine if the buffer should loop. The return value is the number of frames actually read. If this is less than the number of frames requested it
-means the end has been reached. This should never happen if the `loop` parameter is set to true. If you want to manually loop back to the start, you can do so
-with with `ma_audio_buffer_seek_to_pcm_frame(pAudioBuffer, 0)`. Below is an example for reading data from an audio buffer.
+In the example above we just extend the `ma_async_notification_callbacks` object and pass an
+instantiation into the `ma_resource_manager_pipeline_notifications` in the same way as we did with
+the fence, only we set `pNotification` instead of `pFence`. You can set both of these at the same
+time and they should both work as expected. If using the `pNotification` system, you need to ensure
+your `ma_async_notification_callbacks` object stays valid.
+
+
+
+6.2. Resource Manager Implementation Details
+--------------------------------------------
+Resources are managed in two main ways:
+
+ * By storing the entire sound inside an in-memory buffer (referred to as a data buffer)
+ * By streaming audio data on the fly (referred to as a data stream)
+
+A resource managed data source (`ma_resource_manager_data_source`) encapsulates a data buffer or
+data stream, depending on whether or not the data source was initialized with the
+`MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM` flag. If so, it will make use of a
+`ma_resource_manager_data_stream` object. Otherwise it will use a `ma_resource_manager_data_buffer`
+object. Both of these objects are data sources which means they can be used with any
+`ma_data_source_*()` API.
+
+Another major feature of the resource manager is the ability to asynchronously decode audio files.
+This relieves the audio thread of time-consuming decoding which can negatively affect scalability
+due to the audio thread needing to complete it's work extremely quickly to avoid glitching.
+Asynchronous decoding is achieved through a job system. There is a central multi-producer,
+multi-consumer, fixed-capacity job queue. When some asynchronous work needs to be done, a job is
+posted to the queue which is then read by a job thread. The number of job threads can be
+configured for improved scalability, and job threads can all run in parallel without needing to
+worry about the order of execution (how this is achieved is explained below).
+
+When a sound is being loaded asynchronously, playback can begin before the sound has been fully
+decoded. This enables the application to start playback of the sound quickly, while at the same
+time allowing to resource manager to keep loading in the background. Since there may be less
+threads than the number of sounds being loaded at a given time, a simple scheduling system is used
+to keep decoding time balanced and fair. The resource manager solves this by splitting decoding
+into chunks called pages. By default, each page is 1 second long. When a page has been decoded, a
+new job will be posted to start decoding the next page. By dividing up decoding into pages, an
+individual sound shouldn't ever delay every other sound from having their first page decoded. Of
+course, when loading many sounds at the same time, there will always be an amount of time required
+to process jobs in the queue so in heavy load situations there will still be some delay. To
+determine if a data source is ready to have some frames read, use
+`ma_resource_manager_data_source_get_available_frames()`. This will return the number of frames
+available starting from the current position.
+
+
+6.2.1. Job Queue
+----------------
+The resource manager uses a job queue which is multi-producer, multi-consumer, and fixed-capacity.
+This job queue is not currently lock-free, and instead uses a spinlock to achieve thread-safety.
+Only a fixed number of jobs can be allocated and inserted into the queue which is done through a
+lock-free data structure for allocating an index into a fixed sized array, with reference counting
+for mitigation of the ABA problem. The reference count is 32-bit.
+
+For many types of jobs it's important that they execute in a specific order. In these cases, jobs
+are executed serially. For the resource manager, serial execution of jobs is only required on a
+per-object basis (per data buffer or per data stream). Each of these objects stores an execution
+counter. When a job is posted it is associated with an execution counter. When the job is
+processed, it checks if the execution counter of the job equals the execution counter of the
+owning object and if so, processes the job. If the counters are not equal, the job will be posted
+back onto the job queue for later processing. When the job finishes processing the execution order
+of the main object is incremented. This system means the no matter how many job threads are
+executing, decoding of an individual sound will always get processed serially. The advantage to
+having multiple threads comes into play when loading multiple sounds at the same time.
+
+The resource manager's job queue is not 100% lock-free and will use a spinlock to achieve
+thread-safety for a very small section of code. This is only relevant when the resource manager
+uses more than one job thread. If only using a single job thread, which is the default, the
+lock should never actually wait in practice. The amount of time spent locking should be quite
+short, but it's something to be aware of for those who have pedantic lock-free requirements and
+need to use more than one job thread. There are plans to remove this lock in a future version.
+
+In addition, posting a job will release a semaphore, which on Win32 is implemented with
+`ReleaseSemaphore` and on POSIX platforms via a condition variable:
```c
- ma_uint64 framesRead = ma_audio_buffer_read_pcm_frames(pAudioBuffer, pFramesOut, desiredFrameCount, isLooping);
- if (framesRead < desiredFrameCount) {
- // If not looping, this means the end has been reached. This should never happen in looping mode with valid input.
+ pthread_mutex_lock(&pSemaphore->lock);
+ {
+ pSemaphore->value += 1;
+ pthread_cond_signal(&pSemaphore->cond);
}
+ pthread_mutex_unlock(&pSemaphore->lock);
```
-Sometimes you may want to avoid the cost of data movement between the internal buffer and the output buffer. Instead you can use memory mapping to retrieve a
-pointer to a segment of data:
+Again, this is relevant for those with strict lock-free requirements in the audio thread. To avoid
+this, you can use non-blocking mode (via the `MA_JOB_QUEUE_FLAG_NON_BLOCKING`
+flag) and implement your own job processing routine (see the "Resource Manager" section above for
+details on how to do this).
+
+
+
+6.2.2. Data Buffers
+-------------------
+When the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM` flag is excluded at initialization time, the
+resource manager will try to load the data into an in-memory data buffer. Before doing so, however,
+it will first check if the specified file is already loaded. If so, it will increment a reference
+counter and just use the already loaded data. This saves both time and memory. When the data buffer
+is uninitialized, the reference counter will be decremented. If the counter hits zero, the file
+will be unloaded. This is a detail to keep in mind because it could result in excessive loading and
+unloading of a sound. For example, the following sequence will result in a file be loaded twice,
+once after the other:
```c
- void* pMappedFrames;
- ma_uint64 frameCount = frameCountToTryMapping;
- ma_result result = ma_audio_buffer_map(pAudioBuffer, &pMappedFrames, &frameCount);
- if (result == MA_SUCCESS) {
- // Map was successful. The value in frameCount will be how many frames were _actually_ mapped, which may be
- // less due to the end of the buffer being reached.
- ma_copy_pcm_frames(pFramesOut, pMappedFrames, frameCount, pAudioBuffer->format, pAudioBuffer->channels);
+ ma_resource_manager_data_source_init(pResourceManager, "my_file", ..., &myDataBuffer0); // Refcount = 1. Initial load.
+ ma_resource_manager_data_source_uninit(pResourceManager, &myDataBuffer0); // Refcount = 0. Unloaded.
- // You must unmap the buffer.
- ma_audio_buffer_unmap(pAudioBuffer, frameCount);
- }
+ ma_resource_manager_data_source_init(pResourceManager, "my_file", ..., &myDataBuffer1); // Refcount = 1. Reloaded because previous uninit() unloaded it.
+ ma_resource_manager_data_source_uninit(pResourceManager, &myDataBuffer1); // Refcount = 0. Unloaded.
```
-When you use memory mapping, the read cursor is increment by the frame count passed in to `ma_audio_buffer_unmap()`. If you decide not to process every frame
-you can pass in a value smaller than the value returned by `ma_audio_buffer_map()`. The disadvantage to using memory mapping is that it does not handle looping
-for you. You can determine if the buffer is at the end for the purpose of looping with `ma_audio_buffer_at_end()` or by inspecting the return value of
-`ma_audio_buffer_unmap()` and checking if it equals `MA_AT_END`. You should not treat `MA_AT_END` as an error when returned by `ma_audio_buffer_unmap()`.
+A binary search tree (BST) is used for storing data buffers as it has good balance between
+efficiency and simplicity. The key of the BST is a 64-bit hash of the file path that was passed
+into `ma_resource_manager_data_source_init()`. The advantage of using a hash is that it saves
+memory over storing the entire path, has faster comparisons, and results in a mostly balanced BST
+due to the random nature of the hash. The disadvantage is that file names are case-sensitive. If
+this is an issue, you should normalize your file names to upper- or lower-case before initializing
+your data sources.
+
+When a sound file has not already been loaded and the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC`
+flag is excluded, the file will be decoded synchronously by the calling thread. There are two
+options for controlling how the audio is stored in the data buffer - encoded or decoded. When the
+`MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE` option is excluded, the raw file data will be stored
+in memory. Otherwise the sound will be decoded before storing it in memory. Synchronous loading is
+a very simple and standard process of simply adding an item to the BST, allocating a block of
+memory and then decoding (if `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE` is specified).
+
+When the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC` flag is specified, loading of the data buffer
+is done asynchronously. In this case, a job is posted to the queue to start loading and then the
+function immediately returns, setting an internal result code to `MA_BUSY`. This result code is
+returned when the program calls `ma_resource_manager_data_source_result()`. When decoding has fully
+completed `MA_SUCCESS` will be returned. This can be used to know if loading has fully completed.
+
+When loading asynchronously, a single job is posted to the queue of the type
+`MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_BUFFER_NODE`. This involves making a copy of the file path and
+associating it with job. When the job is processed by the job thread, it will first load the file
+using the VFS associated with the resource manager. When using a custom VFS, it's important that it
+be completely thread-safe because it will be used from one or more job threads at the same time.
+Individual files should only ever be accessed by one thread at a time, however. After opening the
+file via the VFS, the job will determine whether or not the file is being decoded. If not, it
+simply allocates a block of memory and loads the raw file contents into it and returns. On the
+other hand, when the file is being decoded, it will first allocate a decoder on the heap and
+initialize it. Then it will check if the length of the file is known. If so it will allocate a
+block of memory to store the decoded output and initialize it to silence. If the size is unknown,
+it will allocate room for one page. After memory has been allocated, the first page will be
+decoded. If the sound is shorter than a page, the result code will be set to `MA_SUCCESS` and the
+completion event will be signalled and loading is now complete. If, however, there is more to
+decode, a job with the code `MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_BUFFER_NODE` is posted. This job
+will decode the next page and perform the same process if it reaches the end. If there is more to
+decode, the job will post another `MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_BUFFER_NODE` job which will
+keep on happening until the sound has been fully decoded. For sounds of an unknown length, each
+page will be linked together as a linked list. Internally this is implemented via the
+`ma_paged_audio_buffer` object.
+
+
+6.2.3. Data Streams
+-------------------
+Data streams only ever store two pages worth of data for each instance. They are most useful for
+large sounds like music tracks in games that would consume too much memory if fully decoded in
+memory. After every frame from a page has been read, a job will be posted to load the next page
+which is done from the VFS.
+
+For data streams, the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC` flag will determine whether or
+not initialization of the data source waits until the two pages have been decoded. When unset,
+`ma_resource_manager_data_source_init()` will wait until the two pages have been loaded, otherwise
+it will return immediately.
+
+When frames are read from a data stream using `ma_resource_manager_data_source_read_pcm_frames()`,
+`MA_BUSY` will be returned if there are no frames available. If there are some frames available,
+but less than the number requested, `MA_SUCCESS` will be returned, but the actual number of frames
+read will be less than the number requested. Due to the asynchronous nature of data streams,
+seeking is also asynchronous. If the data stream is in the middle of a seek, `MA_BUSY` will be
+returned when trying to read frames.
+
+When `ma_resource_manager_data_source_read_pcm_frames()` results in a page getting fully consumed
+a job is posted to load the next page. This will be posted from the same thread that called
+`ma_resource_manager_data_source_read_pcm_frames()`.
+
+Data streams are uninitialized by posting a job to the queue, but the function won't return until
+that job has been processed. The reason for this is that the caller owns the data stream object and
+therefore miniaudio needs to ensure everything completes before handing back control to the caller.
+Also, if the data stream is uninitialized while pages are in the middle of decoding, they must
+complete before destroying any underlying object and the job system handles this cleanly.
+
+Note that when a new page needs to be loaded, a job will be posted to the resource manager's job
+thread from the audio thread. You must keep in mind the details mentioned in the "Job Queue"
+section above regarding locking when posting an event if you require a strictly lock-free audio
+thread.
+
+
+
+7. Node Graph
+=============
+miniaudio's routing infrastructure follows a node graph paradigm. The idea is that you create a
+node whose outputs are attached to inputs of another node, thereby creating a graph. There are
+different types of nodes, with each node in the graph processing input data to produce output,
+which is then fed through the chain. Each node in the graph can apply their own custom effects. At
+the start of the graph will usually be one or more data source nodes which have no inputs, but
+instead pull their data from a data source. At the end of the graph is an endpoint which represents
+the end of the chain and is where the final output is ultimately extracted from.
+
+Each node has a number of input buses and a number of output buses. An output bus from a node is
+attached to an input bus of another. Multiple nodes can connect their output buses to another
+node's input bus, in which case their outputs will be mixed before processing by the node. Below is
+a diagram that illustrates a hypothetical node graph setup:
+ ```
+ >>>>>>>>>>>>>>>>>>>>>>>>>>>>>> Data flows left to right >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
+
+ +---------------+ +-----------------+
+ | Data Source 1 =----+ +----------+ +----= Low Pass Filter =----+
+ +---------------+ | | =----+ +-----------------+ | +----------+
+ +----= Splitter | +----= ENDPOINT |
+ +---------------+ | | =----+ +-----------------+ | +----------+
+ | Data Source 2 =----+ +----------+ +----= Echo / Delay =----+
+ +---------------+ +-----------------+
+ ```
+In the above graph, it starts with two data sources whose outputs are attached to the input of a
+splitter node. It's at this point that the two data sources are mixed. After mixing, the splitter
+performs it's processing routine and produces two outputs which is simply a duplication of the
+input stream. One output is attached to a low pass filter, whereas the other output is attached to
+a echo/delay. The outputs of the the low pass filter and the echo are attached to the endpoint, and
+since they're both connected to the same input but, they'll be mixed.
+
+Each input bus must be configured to accept the same number of channels, but the number of channels
+used by input buses can be different to the number of channels for output buses in which case
+miniaudio will automatically convert the input data to the output channel count before processing.
+The number of channels of an output bus of one node must match the channel count of the input bus
+it's attached to. The channel counts cannot be changed after the node has been initialized. If you
+attempt to attach an output bus to an input bus with a different channel count, attachment will
+fail.
+
+To use a node graph, you first need to initialize a `ma_node_graph` object. This is essentially a
+container around the entire graph. The `ma_node_graph` object is required for some thread-safety
+issues which will be explained later. A `ma_node_graph` object is initialized using miniaudio's
+standard config/init system:
-10. Ring Buffers
-================
-miniaudio supports lock free (single producer, single consumer) ring buffers which are exposed via the `ma_rb` and `ma_pcm_rb` APIs. The `ma_rb` API operates
-on bytes, whereas the `ma_pcm_rb` operates on PCM frames. They are otherwise identical as `ma_pcm_rb` is just a wrapper around `ma_rb`.
+ ```c
+ ma_node_graph_config nodeGraphConfig = ma_node_graph_config_init(myChannelCount);
-Unlike most other APIs in miniaudio, ring buffers support both interleaved and deinterleaved streams. The caller can also allocate their own backing memory for
-the ring buffer to use internally for added flexibility. Otherwise the ring buffer will manage it's internal memory for you.
+ result = ma_node_graph_init(&nodeGraphConfig, NULL, &nodeGraph); // Second parameter is a pointer to allocation callbacks.
+ if (result != MA_SUCCESS) {
+ // Failed to initialize node graph.
+ }
+ ```
-The examples below use the PCM frame variant of the ring buffer since that's most likely the one you will want to use. To initialize a ring buffer, do
-something like the following:
+When you initialize the node graph, you're specifying the channel count of the endpoint. The
+endpoint is a special node which has one input bus and one output bus, both of which have the
+same channel count, which is specified in the config. Any nodes that connect directly to the
+endpoint must be configured such that their output buses have the same channel count. When you read
+audio data from the node graph, it'll have the channel count you specified in the config. To read
+data from the graph:
```c
- ma_pcm_rb rb;
- ma_result result = ma_pcm_rb_init(FORMAT, CHANNELS, BUFFER_SIZE_IN_FRAMES, NULL, NULL, &rb);
+ ma_uint32 framesRead;
+ result = ma_node_graph_read_pcm_frames(&nodeGraph, pFramesOut, frameCount, &framesRead);
if (result != MA_SUCCESS) {
- // Error
+ // Failed to read data from the node graph.
}
```
-The `ma_pcm_rb_init()` function takes the sample format and channel count as parameters because it's the PCM varient of the ring buffer API. For the regular
-ring buffer that operates on bytes you would call `ma_rb_init()` which leaves these out and just takes the size of the buffer in bytes instead of frames. The
-fourth parameter is an optional pre-allocated buffer and the fifth parameter is a pointer to a `ma_allocation_callbacks` structure for custom memory allocation
-routines. Passing in `NULL` for this results in `MA_MALLOC()` and `MA_FREE()` being used.
+When you read audio data, miniaudio starts at the node graph's endpoint node which then pulls in
+data from it's input attachments, which in turn recusively pull in data from their inputs, and so
+on. At the start of the graph there will be some kind of data source node which will have zero
+inputs and will instead read directly from a data source. The base nodes don't literally need to
+read from a `ma_data_source` object, but they will always have some kind of underlying object that
+sources some kind of audio. The `ma_data_source_node` node can be used to read from a
+`ma_data_source`. Data is always in floating-point format and in the number of channels you
+specified when the graph was initialized. The sample rate is defined by the underlying data sources.
+It's up to you to ensure they use a consistent and appropraite sample rate.
+
+The `ma_node` API is designed to allow custom nodes to be implemented with relative ease, but
+miniaudio includes a few stock nodes for common functionality. This is how you would initialize a
+node which reads directly from a data source (`ma_data_source_node`) which is an example of one
+of the stock nodes that comes with miniaudio:
-Use `ma_pcm_rb_init_ex()` if you need a deinterleaved buffer. The data for each sub-buffer is offset from each other based on the stride. To manage your
-sub-buffers you can use `ma_pcm_rb_get_subbuffer_stride()`, `ma_pcm_rb_get_subbuffer_offset()` and `ma_pcm_rb_get_subbuffer_ptr()`.
+ ```c
+ ma_data_source_node_config config = ma_data_source_node_config_init(pMyDataSource);
-Use `ma_pcm_rb_acquire_read()` and `ma_pcm_rb_acquire_write()` to retrieve a pointer to a section of the ring buffer. You specify the number of frames you
-need, and on output it will set to what was actually acquired. If the read or write pointer is positioned such that the number of frames requested will require
-a loop, it will be clamped to the end of the buffer. Therefore, the number of frames you're given may be less than the number you requested.
+ ma_data_source_node dataSourceNode;
+ result = ma_data_source_node_init(&nodeGraph, &config, NULL, &dataSourceNode);
+ if (result != MA_SUCCESS) {
+ // Failed to create data source node.
+ }
+ ```
-After calling `ma_pcm_rb_acquire_read()` or `ma_pcm_rb_acquire_write()`, you do your work on the buffer and then "commit" it with `ma_pcm_rb_commit_read()` or
-`ma_pcm_rb_commit_write()`. This is where the read/write pointers are updated. When you commit you need to pass in the buffer that was returned by the earlier
-call to `ma_pcm_rb_acquire_read()` or `ma_pcm_rb_acquire_write()` and is only used for validation. The number of frames passed to `ma_pcm_rb_commit_read()` and
-`ma_pcm_rb_commit_write()` is what's used to increment the pointers, and can be less that what was originally requested.
+The data source node will use the output channel count to determine the channel count of the output
+bus. There will be 1 output bus and 0 input buses (data will be drawn directly from the data
+source). The data source must output to floating-point (`ma_format_f32`) or else an error will be
+returned from `ma_data_source_node_init()`.
-If you want to correct for drift between the write pointer and the read pointer you can use a combination of `ma_pcm_rb_pointer_distance()`,
-`ma_pcm_rb_seek_read()` and `ma_pcm_rb_seek_write()`. Note that you can only move the pointers forward, and you should only move the read pointer forward via
-the consumer thread, and the write pointer forward by the producer thread. If there is too much space between the pointers, move the read pointer forward. If
-there is too little space between the pointers, move the write pointer forward.
+By default the node will not be attached to the graph. To do so, use `ma_node_attach_output_bus()`:
-You can use a ring buffer at the byte level instead of the PCM frame level by using the `ma_rb` API. This is exactly the same, only you will use the `ma_rb`
-functions instead of `ma_pcm_rb` and instead of frame counts you will pass around byte counts.
+ ```c
+ result = ma_node_attach_output_bus(&dataSourceNode, 0, ma_node_graph_get_endpoint(&nodeGraph), 0);
+ if (result != MA_SUCCESS) {
+ // Failed to attach node.
+ }
+ ```
-The maximum size of the buffer in bytes is `0x7FFFFFFF-(MA_SIMD_ALIGNMENT-1)` due to the most significant bit being used to encode a loop flag and the internally
-managed buffers always being aligned to MA_SIMD_ALIGNMENT.
+The code above connects the data source node directly to the endpoint. Since the data source node
+has only a single output bus, the index will always be 0. Likewise, the endpoint only has a single
+input bus which means the input bus index will also always be 0.
-Note that the ring buffer is only thread safe when used by a single consumer thread and single producer thread.
+To detach a specific output bus, use `ma_node_detach_output_bus()`. To detach all output buses, use
+`ma_node_detach_all_output_buses()`. If you want to just move the output bus from one attachment to
+another, you do not need to detach first. You can just call `ma_node_attach_output_bus()` and it'll
+deal with it for you.
+Less frequently you may want to create a specialized node. This will be a node where you implement
+your own processing callback to apply a custom effect of some kind. This is similar to initalizing
+one of the stock node types, only this time you need to specify a pointer to a vtable containing a
+pointer to the processing function and the number of input and output buses. Example:
+ ```c
+ static void my_custom_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
+ {
+ // Do some processing of ppFramesIn (one stream of audio data per input bus)
+ const float* pFramesIn_0 = ppFramesIn[0]; // Input bus @ index 0.
+ const float* pFramesIn_1 = ppFramesIn[1]; // Input bus @ index 1.
+ float* pFramesOut_0 = ppFramesOut[0]; // Output bus @ index 0.
+
+ // Do some processing. On input, `pFrameCountIn` will be the number of input frames in each
+ // buffer in `ppFramesIn` and `pFrameCountOut` will be the capacity of each of the buffers
+ // in `ppFramesOut`. On output, `pFrameCountIn` should be set to the number of input frames
+ // your node consumed and `pFrameCountOut` should be set the number of output frames that
+ // were produced.
+ //
+ // You should process as many frames as you can. If your effect consumes input frames at the
+ // same rate as output frames (always the case, unless you're doing resampling), you need
+ // only look at `ppFramesOut` and process that exact number of frames. If you're doing
+ // resampling, you'll need to be sure to set both `pFrameCountIn` and `pFrameCountOut`
+ // properly.
+ }
+
+ static ma_node_vtable my_custom_node_vtable =
+ {
+ my_custom_node_process_pcm_frames, // The function that will be called process your custom node. This is where you'd implement your effect processing.
+ NULL, // Optional. A callback for calculating the number of input frames that are required to process a specified number of output frames.
+ 2, // 2 input buses.
+ 1, // 1 output bus.
+ 0 // Default flags.
+ };
-11. Backends
-============
-The following backends are supported by miniaudio.
+ ...
- +-------------+-----------------------+--------------------------------------------------------+
- | Name | Enum Name | Supported Operating Systems |
- +-------------+-----------------------+--------------------------------------------------------+
- | WASAPI | ma_backend_wasapi | Windows Vista+ |
- | DirectSound | ma_backend_dsound | Windows XP+ |
- | WinMM | ma_backend_winmm | Windows XP+ (may work on older versions, but untested) |
- | Core Audio | ma_backend_coreaudio | macOS, iOS |
- | ALSA | ma_backend_alsa | Linux |
- | PulseAudio | ma_backend_pulseaudio | Cross Platform (disabled on Windows, BSD and Android) |
- | JACK | ma_backend_jack | Cross Platform (disabled on BSD and Android) |
- | sndio | ma_backend_sndio | OpenBSD |
- | audio(4) | ma_backend_audio4 | NetBSD, OpenBSD |
- | OSS | ma_backend_oss | FreeBSD |
- | AAudio | ma_backend_aaudio | Android 8+ |
- | OpenSL ES | ma_backend_opensl | Android (API level 16+) |
- | Web Audio | ma_backend_webaudio | Web (via Emscripten) |
- | Custom | ma_backend_custom | Cross Platform |
- | Null | ma_backend_null | Cross Platform (not used on Web) |
- +-------------+-----------------------+--------------------------------------------------------+
+ // Each bus needs to have a channel count specified. To do this you need to specify the channel
+ // counts in an array and then pass that into the node config.
+ ma_uint32 inputChannels[2]; // Equal in size to the number of input channels specified in the vtable.
+ ma_uint32 outputChannels[1]; // Equal in size to the number of output channels specicied in the vtable.
-Some backends have some nuance details you may want to be aware of.
+ inputChannels[0] = channelsIn;
+ inputChannels[1] = channelsIn;
+ outputChannels[0] = channelsOut;
-11.1. WASAPI
-------------
-- Low-latency shared mode will be disabled when using an application-defined sample rate which is different to the device's native sample rate. To work around
- this, set `wasapi.noAutoConvertSRC` to true in the device config. This is due to IAudioClient3_InitializeSharedAudioStream() failing when the
- `AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM` flag is specified. Setting wasapi.noAutoConvertSRC will result in miniaudio's internal resampler being used instead
- which will in turn enable the use of low-latency shared mode.
+ ma_node_config nodeConfig = ma_node_config_init();
+ nodeConfig.vtable = &my_custom_node_vtable;
+ nodeConfig.pInputChannels = inputChannels;
+ nodeConfig.pOutputChannels = outputChannels;
-11.2. PulseAudio
-----------------
-- If you experience bad glitching/noise on Arch Linux, consider this fix from the Arch wiki:
- https://wiki.archlinux.org/index.php/PulseAudio/Troubleshooting#Glitches,_skips_or_crackling. Alternatively, consider using a different backend such as ALSA.
+ ma_node_base node;
+ result = ma_node_init(&nodeGraph, &nodeConfig, NULL, &node);
+ if (result != MA_SUCCESS) {
+ // Failed to initialize node.
+ }
+ ```
-11.3. Android
--------------
-- To capture audio on Android, remember to add the RECORD_AUDIO permission to your manifest: ``
-- With OpenSL|ES, only a single ma_context can be active at any given time. This is due to a limitation with OpenSL|ES.
-- With AAudio, only default devices are enumerated. This is due to AAudio not having an enumeration API (devices are enumerated through Java). You can however
- perform your own device enumeration through Java and then set the ID in the ma_device_id structure (ma_device_id.aaudio) and pass it to ma_device_init().
-- The backend API will perform resampling where possible. The reason for this as opposed to using miniaudio's built-in resampler is to take advantage of any
- potential device-specific optimizations the driver may implement.
-
-11.4. UWP
----------
-- UWP only supports default playback and capture devices.
-- UWP requires the Microphone capability to be enabled in the application's manifest (Package.appxmanifest):
+When initializing a custom node, as in the code above, you'll normally just place your vtable in
+static space. The number of input and output buses are specified as part of the vtable. If you need
+a variable number of buses on a per-node bases, the vtable should have the relevant bus count set
+to `MA_NODE_BUS_COUNT_UNKNOWN`. In this case, the bus count should be set in the node config:
+
+ ```c
+ static ma_node_vtable my_custom_node_vtable =
+ {
+ my_custom_node_process_pcm_frames, // The function that will be called process your custom node. This is where you'd implement your effect processing.
+ NULL, // Optional. A callback for calculating the number of input frames that are required to process a specified number of output frames.
+ MA_NODE_BUS_COUNT_UNKNOWN, // The number of input buses is determined on a per-node basis.
+ 1, // 1 output bus.
+ 0 // Default flags.
+ };
+
+ ...
+ ma_node_config nodeConfig = ma_node_config_init();
+ nodeConfig.vtable = &my_custom_node_vtable;
+ nodeConfig.inputBusCount = myBusCount; // <-- Since the vtable specifies MA_NODE_BUS_COUNT_UNKNOWN, the input bus count should be set here.
+ nodeConfig.pInputChannels = inputChannels; // <-- Make sure there are nodeConfig.inputBusCount elements in this array.
+ nodeConfig.pOutputChannels = outputChannels; // <-- The vtable specifies 1 output bus, so there must be 1 element in this array.
```
-
- ...
-
-
-
-
+
+In the above example it's important to never set the `inputBusCount` and `outputBusCount` members
+to anything other than their defaults if the vtable specifies an explicit count. They can only be
+set if the vtable specifies MA_NODE_BUS_COUNT_UNKNOWN in the relevant bus count.
+
+Most often you'll want to create a structure to encapsulate your node with some extra data. You
+need to make sure the `ma_node_base` object is your first member of the structure:
+
+ ```c
+ typedef struct
+ {
+ ma_node_base base; // <-- Make sure this is always the first member.
+ float someCustomData;
+ } my_custom_node;
```
-11.5. Web Audio / Emscripten
-----------------------------
-- You cannot use `-std=c*` compiler flags, nor `-ansi`. This only applies to the Emscripten build.
-- The first time a context is initialized it will create a global object called "miniaudio" whose primary purpose is to act as a factory for device objects.
-- Currently the Web Audio backend uses ScriptProcessorNode's, but this may need to change later as they've been deprecated.
-- Google has implemented a policy in their browsers that prevent automatic media output without first receiving some kind of user input. The following web page
- has additional details: https://developers.google.com/web/updates/2017/09/autoplay-policy-changes. Starting the device may fail if you try to start playback
- without first handling some kind of user input.
+By doing this, your object will be compatible with all `ma_node` APIs and you can attach it to the
+graph just like any other node.
+
+In the custom processing callback (`my_custom_node_process_pcm_frames()` in the example above), the
+number of channels for each bus is what was specified by the config when the node was initialized
+with `ma_node_init()`. In addition, all attachments to each of the input buses will have been
+pre-mixed by miniaudio. The config allows you to specify different channel counts for each
+individual input and output bus. It's up to the effect to handle it appropriate, and if it can't,
+return an error in it's initialization routine.
+
+Custom nodes can be assigned some flags to describe their behaviour. These are set via the vtable
+and include the following:
+
+ +-----------------------------------------+---------------------------------------------------+
+ | Flag Name | Description |
+ +-----------------------------------------+---------------------------------------------------+
+ | MA_NODE_FLAG_PASSTHROUGH | Useful for nodes that do not do any kind of audio |
+ | | processing, but are instead used for tracking |
+ | | time, handling events, etc. Also used by the |
+ | | internal endpoint node. It reads directly from |
+ | | the input bus to the output bus. Nodes with this |
+ | | flag must have exactly 1 input bus and 1 output |
+ | | bus, and both buses must have the same channel |
+ | | counts. |
+ +-----------------------------------------+---------------------------------------------------+
+ | MA_NODE_FLAG_CONTINUOUS_PROCESSING | Causes the processing callback to be called even |
+ | | when no data is available to be read from input |
+ | | attachments. This is useful for effects like |
+ | | echos where there will be a tail of audio data |
+ | | that still needs to be processed even when the |
+ | | original data sources have reached their ends. |
+ +-----------------------------------------+---------------------------------------------------+
+ | MA_NODE_FLAG_ALLOW_NULL_INPUT | Used in conjunction with |
+ | | `MA_NODE_FLAG_CONTINUOUS_PROCESSING`. When this |
+ | | is set, the `ppFramesIn` parameter of the |
+ | | processing callback will be set to NULL when |
+ | | there are no input frames are available. When |
+ | | this is unset, silence will be posted to the |
+ | | processing callback. |
+ +-----------------------------------------+---------------------------------------------------+
+ | MA_NODE_FLAG_DIFFERENT_PROCESSING_RATES | Used to tell miniaudio that input and output |
+ | | frames are processed at different rates. You |
+ | | should set this for any nodes that perform |
+ | | resampling. |
+ +-----------------------------------------+---------------------------------------------------+
+ | MA_NODE_FLAG_SILENT_OUTPUT | Used to tell miniaudio that a node produces only |
+ | | silent output. This is useful for nodes where you |
+ | | don't want the output to contribute to the final |
+ | | mix. An example might be if you want split your |
+ | | stream and have one branch be output to a file. |
+ | | When using this flag, you should avoid writing to |
+ | | the output buffer of the node's processing |
+ | | callback because miniaudio will ignore it anyway. |
+ +-----------------------------------------+---------------------------------------------------+
+
+
+If you need to make a copy of an audio stream for effect processing you can use a splitter node
+called `ma_splitter_node`. This takes has 1 input bus and splits the stream into 2 output buses.
+You can use it like this:
+ ```c
+ ma_splitter_node_config splitterNodeConfig = ma_splitter_node_config_init(channelsIn, channelsOut);
+ ma_splitter_node splitterNode;
+ result = ma_splitter_node_init(&nodeGraph, &splitterNodeConfig, NULL, &splitterNode);
+ if (result != MA_SUCCESS) {
+ // Failed to create node.
+ }
-12. Miscellaneous Notes
-=======================
-- Automatic stream routing is enabled on a per-backend basis. Support is explicitly enabled for WASAPI and Core Audio, however other backends such as
- PulseAudio may naturally support it, though not all have been tested.
-- The contents of the output buffer passed into the data callback will always be pre-initialized to silence unless the `noPreZeroedOutputBuffer` config variable
- in `ma_device_config` is set to true, in which case it'll be undefined which will require you to write something to the entire buffer.
-- By default miniaudio will automatically clip samples. This only applies when the playback sample format is configured as `ma_format_f32`. If you are doing
- clipping yourself, you can disable this overhead by setting `noClip` to true in the device config.
-- The sndio backend is currently only enabled on OpenBSD builds.
-- The audio(4) backend is supported on OpenBSD, but you may need to disable sndiod before you can use it.
-- Note that GCC and Clang requires `-msse2`, `-mavx2`, etc. for SIMD optimizations.
-- When compiling with VC6 and earlier, decoding is restricted to files less than 2GB in size. This is due to 64-bit file APIs not being available.
-*/
+ // Attach your output buses to two different input buses (can be on two different nodes).
+ ma_node_attach_output_bus(&splitterNode, 0, ma_node_graph_get_endpoint(&nodeGraph), 0); // Attach directly to the endpoint.
+ ma_node_attach_output_bus(&splitterNode, 1, &myEffectNode, 0); // Attach to input bus 0 of some effect node.
+ ```
-#ifndef miniaudio_h
-#define miniaudio_h
+The volume of an output bus can be configured on a per-bus basis:
-#ifdef __cplusplus
-extern "C" {
-#endif
+ ```c
+ ma_node_set_output_bus_volume(&splitterNode, 0, 0.5f);
+ ma_node_set_output_bus_volume(&splitterNode, 1, 0.5f);
+ ```
-#define MA_STRINGIFY(x) #x
-#define MA_XSTRINGIFY(x) MA_STRINGIFY(x)
+In the code above we're using the splitter node from before and changing the volume of each of the
+copied streams.
-#define MA_VERSION_MAJOR 0
-#define MA_VERSION_MINOR 10
-#define MA_VERSION_REVISION 42
-#define MA_VERSION_STRING MA_XSTRINGIFY(MA_VERSION_MAJOR) "." MA_XSTRINGIFY(MA_VERSION_MINOR) "." MA_XSTRINGIFY(MA_VERSION_REVISION)
+You can start and stop a node with the following:
-#if defined(_MSC_VER) && !defined(__clang__)
- #pragma warning(push)
- #pragma warning(disable:4201) /* nonstandard extension used: nameless struct/union */
- #pragma warning(disable:4214) /* nonstandard extension used: bit field types other than int */
- #pragma warning(disable:4324) /* structure was padded due to alignment specifier */
-#elif defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8)))
- #pragma GCC diagnostic push
- #pragma GCC diagnostic ignored "-Wpedantic" /* For ISO C99 doesn't support unnamed structs/unions [-Wpedantic] */
- #if defined(__clang__)
- #pragma GCC diagnostic ignored "-Wc11-extensions" /* anonymous unions are a C11 extension */
- #endif
-#endif
+ ```c
+ ma_node_set_state(&splitterNode, ma_node_state_started); // The default state.
+ ma_node_set_state(&splitterNode, ma_node_state_stopped);
+ ```
-/* Platform/backend detection. */
-#ifdef _WIN32
- #define MA_WIN32
- #if defined(WINAPI_FAMILY) && (WINAPI_FAMILY == WINAPI_FAMILY_PC_APP || WINAPI_FAMILY == WINAPI_FAMILY_PHONE_APP)
- #define MA_WIN32_UWP
- #else
- #define MA_WIN32_DESKTOP
- #endif
-#else
- #define MA_POSIX
- #include /* Unfortunate #include, but needed for pthread_t, pthread_mutex_t and pthread_cond_t types. */
+By default the node is in a started state, but since it won't be connected to anything won't
+actually be invoked by the node graph until it's connected. When you stop a node, data will not be
+read from any of it's input connections. You can use this property to stop a group of sounds
+atomically.
- #ifdef __unix__
- #define MA_UNIX
- #if defined(__DragonFly__) || defined(__FreeBSD__) || defined(__NetBSD__) || defined(__OpenBSD__)
- #define MA_BSD
- #endif
- #endif
- #ifdef __linux__
- #define MA_LINUX
- #endif
- #ifdef __APPLE__
- #define MA_APPLE
- #endif
- #ifdef __ANDROID__
- #define MA_ANDROID
- #endif
- #ifdef __EMSCRIPTEN__
- #define MA_EMSCRIPTEN
- #endif
-#endif
+You can configure the initial state of a node in it's config:
-#include /* For size_t. */
+ ```c
+ nodeConfig.initialState = ma_node_state_stopped;
+ ```
-/* Sized types. */
-typedef signed char ma_int8;
-typedef unsigned char ma_uint8;
-typedef signed short ma_int16;
-typedef unsigned short ma_uint16;
-typedef signed int ma_int32;
-typedef unsigned int ma_uint32;
-#if defined(_MSC_VER)
- typedef signed __int64 ma_int64;
- typedef unsigned __int64 ma_uint64;
-#else
- #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
- #pragma GCC diagnostic push
- #pragma GCC diagnostic ignored "-Wlong-long"
- #if defined(__clang__)
- #pragma GCC diagnostic ignored "-Wc++11-long-long"
- #endif
- #endif
- typedef signed long long ma_int64;
- typedef unsigned long long ma_uint64;
- #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
- #pragma GCC diagnostic pop
- #endif
-#endif
-#if defined(__LP64__) || defined(_WIN64) || (defined(__x86_64__) && !defined(__ILP32__)) || defined(_M_X64) || defined(__ia64) || defined (_M_IA64) || defined(__aarch64__) || defined(_M_ARM64) || defined(__powerpc64__)
- typedef ma_uint64 ma_uintptr;
-#else
- typedef ma_uint32 ma_uintptr;
-#endif
+Note that for the stock specialized nodes, all of their configs will have a `nodeConfig` member
+which is the config to use with the base node. This is where the initial state can be configured
+for specialized nodes:
-typedef ma_uint8 ma_bool8;
-typedef ma_uint32 ma_bool32;
-#define MA_TRUE 1
-#define MA_FALSE 0
+ ```c
+ dataSourceNodeConfig.nodeConfig.initialState = ma_node_state_stopped;
+ ```
-typedef void* ma_handle;
-typedef void* ma_ptr;
-typedef void (* ma_proc)(void);
+When using a specialized node like `ma_data_source_node` or `ma_splitter_node`, be sure to not
+modify the `vtable` member of the `nodeConfig` object.
-#if defined(_MSC_VER) && !defined(_WCHAR_T_DEFINED)
-typedef ma_uint16 wchar_t;
-#endif
-/* Define NULL for some compilers. */
-#ifndef NULL
-#define NULL 0
-#endif
+7.1. Timing
+-----------
+The node graph supports starting and stopping nodes at scheduled times. This is especially useful
+for data source nodes where you want to get the node set up, but only start playback at a specific
+time. There are two clocks: local and global.
+
+A local clock is per-node, whereas the global clock is per graph. Scheduling starts and stops can
+only be done based on the global clock because the local clock will not be running while the node
+is stopped. The global clocks advances whenever `ma_node_graph_read_pcm_frames()` is called. On the
+other hand, the local clock only advances when the node's processing callback is fired, and is
+advanced based on the output frame count.
+
+To retrieve the global time, use `ma_node_graph_get_time()`. The global time can be set with
+`ma_node_graph_set_time()` which might be useful if you want to do seeking on a global timeline.
+Getting and setting the local time is similar. Use `ma_node_get_time()` to retrieve the local time,
+and `ma_node_set_time()` to set the local time. The global and local times will be advanced by the
+audio thread, so care should be taken to avoid data races. Ideally you should avoid calling these
+outside of the node processing callbacks which are always run on the audio thread.
+
+There is basic support for scheduling the starting and stopping of nodes. You can only schedule one
+start and one stop at a time. This is mainly intended for putting nodes into a started or stopped
+state in a frame-exact manner. Without this mechanism, starting and stopping of a node is limited
+to the resolution of a call to `ma_node_graph_read_pcm_frames()` which would typically be in blocks
+of several milliseconds. The following APIs can be used for scheduling node states:
-#if defined(SIZE_MAX)
- #define MA_SIZE_MAX SIZE_MAX
-#else
- #define MA_SIZE_MAX 0xFFFFFFFF /* When SIZE_MAX is not defined by the standard library just default to the maximum 32-bit unsigned integer. */
-#endif
+ ```c
+ ma_node_set_state_time()
+ ma_node_get_state_time()
+ ```
+The time is absolute and must be based on the global clock. An example is below:
-#ifdef _MSC_VER
- #define MA_INLINE __forceinline
-#elif defined(__GNUC__)
- /*
- I've had a bug report where GCC is emitting warnings about functions possibly not being inlineable. This warning happens when
- the __attribute__((always_inline)) attribute is defined without an "inline" statement. I think therefore there must be some
- case where "__inline__" is not always defined, thus the compiler emitting these warnings. When using -std=c89 or -ansi on the
- command line, we cannot use the "inline" keyword and instead need to use "__inline__". In an attempt to work around this issue
- I am using "__inline__" only when we're compiling in strict ANSI mode.
- */
- #if defined(__STRICT_ANSI__)
- #define MA_INLINE __inline__ __attribute__((always_inline))
- #else
- #define MA_INLINE inline __attribute__((always_inline))
- #endif
-#elif defined(__WATCOMC__)
- #define MA_INLINE __inline
-#else
- #define MA_INLINE
-#endif
+ ```c
+ ma_node_set_state_time(&myNode, ma_node_state_started, sampleRate*1); // Delay starting to 1 second.
+ ma_node_set_state_time(&myNode, ma_node_state_stopped, sampleRate*5); // Delay stopping to 5 seconds.
+ ```
-#if !defined(MA_API)
- #if defined(MA_DLL)
- #if defined(_WIN32)
- #define MA_DLL_IMPORT __declspec(dllimport)
- #define MA_DLL_EXPORT __declspec(dllexport)
- #define MA_DLL_PRIVATE static
- #else
- #if defined(__GNUC__) && __GNUC__ >= 4
- #define MA_DLL_IMPORT __attribute__((visibility("default")))
- #define MA_DLL_EXPORT __attribute__((visibility("default")))
- #define MA_DLL_PRIVATE __attribute__((visibility("hidden")))
- #else
- #define MA_DLL_IMPORT
- #define MA_DLL_EXPORT
- #define MA_DLL_PRIVATE static
- #endif
- #endif
+An example for changing the state using a relative time.
- #if defined(MINIAUDIO_IMPLEMENTATION) || defined(MA_IMPLEMENTATION)
- #define MA_API MA_DLL_EXPORT
- #else
- #define MA_API MA_DLL_IMPORT
- #endif
- #define MA_PRIVATE MA_DLL_PRIVATE
- #else
- #define MA_API extern
- #define MA_PRIVATE static
- #endif
-#endif
+ ```c
+ ma_node_set_state_time(&myNode, ma_node_state_started, sampleRate*1 + ma_node_graph_get_time(&myNodeGraph));
+ ma_node_set_state_time(&myNode, ma_node_state_stopped, sampleRate*5 + ma_node_graph_get_time(&myNodeGraph));
+ ```
-/* SIMD alignment in bytes. Currently set to 64 bytes in preparation for future AVX-512 optimizations. */
-#define MA_SIMD_ALIGNMENT 64
+Note that due to the nature of multi-threading the times may not be 100% exact. If this is an
+issue, consider scheduling state changes from within a processing callback. An idea might be to
+have some kind of passthrough trigger node that is used specifically for tracking time and handling
+events.
-/*
-Logging Levels
-==============
-Log levels are only used to give logging callbacks some context as to the severity of a log message
-so they can do filtering. All log levels will be posted to registered logging callbacks, except for
-MA_LOG_LEVEL_DEBUG which will only get processed if MA_DEBUG_OUTPUT is enabled.
-MA_LOG_LEVEL_DEBUG
- Used for debugging. These log messages are only posted when `MA_DEBUG_OUTPUT` is enabled.
+7.2. Thread Safety and Locking
+------------------------------
+When processing audio, it's ideal not to have any kind of locking in the audio thread. Since it's
+expected that `ma_node_graph_read_pcm_frames()` would be run on the audio thread, it does so
+without the use of any locks. This section discusses the implementation used by miniaudio and goes
+over some of the compromises employed by miniaudio to achieve this goal. Note that the current
+implementation may not be ideal - feedback and critiques are most welcome.
+
+The node graph API is not *entirely* lock-free. Only `ma_node_graph_read_pcm_frames()` is expected
+to be lock-free. Attachment, detachment and uninitialization of nodes use locks to simplify the
+implementation, but are crafted in a way such that such locking is not required when reading audio
+data from the graph. Locking in these areas are achieved by means of spinlocks.
+
+The main complication with keeping `ma_node_graph_read_pcm_frames()` lock-free stems from the fact
+that a node can be uninitialized, and it's memory potentially freed, while in the middle of being
+processed on the audio thread. There are times when the audio thread will be referencing a node,
+which means the uninitialization process of a node needs to make sure it delays returning until the
+audio thread is finished so that control is not handed back to the caller thereby giving them a
+chance to free the node's memory.
+
+When the audio thread is processing a node, it does so by reading from each of the output buses of
+the node. In order for a node to process data for one of it's output buses, it needs to read from
+each of it's input buses, and so on an so forth. It follows that once all output buses of a node
+are detached, the node as a whole will be disconnected and no further processing will occur unless
+it's output buses are reattached, which won't be happening when the node is being uninitialized.
+By having `ma_node_detach_output_bus()` wait until the audio thread is finished with it, we can
+simplify a few things, at the expense of making `ma_node_detach_output_bus()` a bit slower. By
+doing this, the implementation of `ma_node_uninit()` becomes trivial - just detach all output
+nodes, followed by each of the attachments to each of it's input nodes, and then do any final clean
+up.
+
+With the above design, the worst-case scenario is `ma_node_detach_output_bus()` taking as long as
+it takes to process the output bus being detached. This will happen if it's called at just the
+wrong moment where the audio thread has just iterated it and has just started processing. The
+caller of `ma_node_detach_output_bus()` will stall until the audio thread is finished, which
+includes the cost of recursively processing it's inputs. This is the biggest compromise made with
+the approach taken by miniaudio for it's lock-free processing system. The cost of detaching nodes
+earlier in the pipeline (data sources, for example) will be cheaper than the cost of detaching
+higher level nodes, such as some kind of final post-processing endpoint. If you need to do mass
+detachments, detach starting from the lowest level nodes and work your way towards the final
+endpoint node (but don't try detaching the node graph's endpoint). If the audio thread is not
+running, detachment will be fast and detachment in any order will be the same. The reason nodes
+need to wait for their input attachments to complete is due to the potential for desyncs between
+data sources. If the node was to terminate processing mid way through processing it's inputs,
+there's a chance that some of the underlying data sources will have been read, but then others not.
+That will then result in a potential desynchronization when detaching and reattaching higher-level
+nodes. A possible solution to this is to have an option when detaching to terminate processing
+before processing all input attachments which should be fairly simple.
+
+Another compromise, albeit less significant, is locking when attaching and detaching nodes. This
+locking is achieved by means of a spinlock in order to reduce memory overhead. A lock is present
+for each input bus and output bus. When an output bus is connected to an input bus, both the output
+bus and input bus is locked. This locking is specifically for attaching and detaching across
+different threads and does not affect `ma_node_graph_read_pcm_frames()` in any way. The locking and
+unlocking is mostly self-explanatory, but a slightly less intuitive aspect comes into it when
+considering that iterating over attachments must not break as a result of attaching or detaching a
+node while iteration is occuring.
+
+Attaching and detaching are both quite simple. When an output bus of a node is attached to an input
+bus of another node, it's added to a linked list. Basically, an input bus is a linked list, where
+each item in the list is and output bus. We have some intentional (and convenient) restrictions on
+what can done with the linked list in order to simplify the implementation. First of all, whenever
+something needs to iterate over the list, it must do so in a forward direction. Backwards iteration
+is not supported. Also, items can only be added to the start of the list.
+
+The linked list is a doubly-linked list where each item in the list (an output bus) holds a pointer
+to the next item in the list, and another to the previous item. A pointer to the previous item is
+only required for fast detachment of the node - it is never used in iteration. This is an
+important property because it means from the perspective of iteration, attaching and detaching of
+an item can be done with a single atomic assignment. This is exploited by both the attachment and
+detachment process. When attaching the node, the first thing that is done is the setting of the
+local "next" and "previous" pointers of the node. After that, the item is "attached" to the list
+by simply performing an atomic exchange with the head pointer. After that, the node is "attached"
+to the list from the perspective of iteration. Even though the "previous" pointer of the next item
+hasn't yet been set, from the perspective of iteration it's been attached because iteration will
+only be happening in a forward direction which means the "previous" pointer won't actually ever get
+used. The same general process applies to detachment. See `ma_node_attach_output_bus()` and
+`ma_node_detach_output_bus()` for the implementation of this mechanism.
+
+
+
+8. Decoding
+===========
+The `ma_decoder` API is used for reading audio files. Decoders are completely decoupled from
+devices and can be used independently. The following formats are supported:
-MA_LOG_LEVEL_INFO
- Informational logging. Useful for debugging. This will also enable warning and error logs. This
- will never be called from within the data callback.
+ +---------+------------------+----------+
+ | Format | Decoding Backend | Built-In |
+ +---------+------------------+----------+
+ | WAV | dr_wav | Yes |
+ | MP3 | dr_mp3 | Yes |
+ | FLAC | dr_flac | Yes |
+ | Vorbis | stb_vorbis | No |
+ +---------+------------------+----------+
-MA_LOG_LEVEL_WARNING
- Warnings. You should enable this in you development builds and action them when encounted. This
- will also enable error logs. These logs usually indicate a potential problem or
- misconfiguration, but still allow you to keep running. This will never be called from within
- the data callback.
+Vorbis is supported via stb_vorbis which can be enabled by including the header section before the
+implementation of miniaudio, like the following:
-MA_LOG_LEVEL_ERROR
- Error logging. This will be fired when an operation fails and is subsequently aborted. This can
- be fired from within the data callback, in which case the device will be stopped. You should
- always have this log level enabled.
-*/
-#define MA_LOG_LEVEL_DEBUG 4
-#define MA_LOG_LEVEL_INFO 3
-#define MA_LOG_LEVEL_WARNING 2
-#define MA_LOG_LEVEL_ERROR 1
+ ```c
+ #define STB_VORBIS_HEADER_ONLY
+ #include "extras/stb_vorbis.c" // Enables Vorbis decoding.
-/* Deprecated. */
-#define MA_LOG_LEVEL_VERBOSE MA_LOG_LEVEL_DEBUG
+ #define MINIAUDIO_IMPLEMENTATION
+ #include "miniaudio.h"
-/* Deprecated. */
-#ifndef MA_LOG_LEVEL
-#define MA_LOG_LEVEL MA_LOG_LEVEL_ERROR
-#endif
+ // The stb_vorbis implementation must come after the implementation of miniaudio.
+ #undef STB_VORBIS_HEADER_ONLY
+ #include "extras/stb_vorbis.c"
+ ```
-/*
-An annotation for variables which must be used atomically. This doesn't actually do anything - it's
-just used as a way for humans to identify variables that should be used atomically.
-*/
-#define MA_ATOMIC
+A copy of stb_vorbis is included in the "extras" folder in the miniaudio repository (https://github.com/mackron/miniaudio).
-typedef struct ma_context ma_context;
-typedef struct ma_device ma_device;
+Built-in decoders are amalgamated into the implementation section of miniaudio. You can disable the
+built-in decoders by specifying one or more of the following options before the miniaudio
+implementation:
-typedef ma_uint8 ma_channel;
-#define MA_CHANNEL_NONE 0
-#define MA_CHANNEL_MONO 1
-#define MA_CHANNEL_FRONT_LEFT 2
-#define MA_CHANNEL_FRONT_RIGHT 3
-#define MA_CHANNEL_FRONT_CENTER 4
-#define MA_CHANNEL_LFE 5
-#define MA_CHANNEL_BACK_LEFT 6
-#define MA_CHANNEL_BACK_RIGHT 7
-#define MA_CHANNEL_FRONT_LEFT_CENTER 8
-#define MA_CHANNEL_FRONT_RIGHT_CENTER 9
-#define MA_CHANNEL_BACK_CENTER 10
-#define MA_CHANNEL_SIDE_LEFT 11
-#define MA_CHANNEL_SIDE_RIGHT 12
-#define MA_CHANNEL_TOP_CENTER 13
-#define MA_CHANNEL_TOP_FRONT_LEFT 14
-#define MA_CHANNEL_TOP_FRONT_CENTER 15
-#define MA_CHANNEL_TOP_FRONT_RIGHT 16
-#define MA_CHANNEL_TOP_BACK_LEFT 17
-#define MA_CHANNEL_TOP_BACK_CENTER 18
-#define MA_CHANNEL_TOP_BACK_RIGHT 19
-#define MA_CHANNEL_AUX_0 20
-#define MA_CHANNEL_AUX_1 21
-#define MA_CHANNEL_AUX_2 22
-#define MA_CHANNEL_AUX_3 23
-#define MA_CHANNEL_AUX_4 24
-#define MA_CHANNEL_AUX_5 25
-#define MA_CHANNEL_AUX_6 26
-#define MA_CHANNEL_AUX_7 27
-#define MA_CHANNEL_AUX_8 28
-#define MA_CHANNEL_AUX_9 29
-#define MA_CHANNEL_AUX_10 30
-#define MA_CHANNEL_AUX_11 31
-#define MA_CHANNEL_AUX_12 32
-#define MA_CHANNEL_AUX_13 33
-#define MA_CHANNEL_AUX_14 34
-#define MA_CHANNEL_AUX_15 35
-#define MA_CHANNEL_AUX_16 36
-#define MA_CHANNEL_AUX_17 37
-#define MA_CHANNEL_AUX_18 38
-#define MA_CHANNEL_AUX_19 39
-#define MA_CHANNEL_AUX_20 40
-#define MA_CHANNEL_AUX_21 41
-#define MA_CHANNEL_AUX_22 42
-#define MA_CHANNEL_AUX_23 43
-#define MA_CHANNEL_AUX_24 44
-#define MA_CHANNEL_AUX_25 45
-#define MA_CHANNEL_AUX_26 46
-#define MA_CHANNEL_AUX_27 47
-#define MA_CHANNEL_AUX_28 48
-#define MA_CHANNEL_AUX_29 49
-#define MA_CHANNEL_AUX_30 50
-#define MA_CHANNEL_AUX_31 51
-#define MA_CHANNEL_LEFT MA_CHANNEL_FRONT_LEFT
-#define MA_CHANNEL_RIGHT MA_CHANNEL_FRONT_RIGHT
-#define MA_CHANNEL_POSITION_COUNT (MA_CHANNEL_AUX_31 + 1)
-
-
-typedef int ma_result;
-#define MA_SUCCESS 0
-#define MA_ERROR -1 /* A generic error. */
-#define MA_INVALID_ARGS -2
-#define MA_INVALID_OPERATION -3
-#define MA_OUT_OF_MEMORY -4
-#define MA_OUT_OF_RANGE -5
-#define MA_ACCESS_DENIED -6
-#define MA_DOES_NOT_EXIST -7
-#define MA_ALREADY_EXISTS -8
-#define MA_TOO_MANY_OPEN_FILES -9
-#define MA_INVALID_FILE -10
-#define MA_TOO_BIG -11
-#define MA_PATH_TOO_LONG -12
-#define MA_NAME_TOO_LONG -13
-#define MA_NOT_DIRECTORY -14
-#define MA_IS_DIRECTORY -15
-#define MA_DIRECTORY_NOT_EMPTY -16
-#define MA_AT_END -17
-#define MA_NO_SPACE -18
-#define MA_BUSY -19
-#define MA_IO_ERROR -20
-#define MA_INTERRUPT -21
-#define MA_UNAVAILABLE -22
-#define MA_ALREADY_IN_USE -23
-#define MA_BAD_ADDRESS -24
-#define MA_BAD_SEEK -25
-#define MA_BAD_PIPE -26
-#define MA_DEADLOCK -27
-#define MA_TOO_MANY_LINKS -28
-#define MA_NOT_IMPLEMENTED -29
-#define MA_NO_MESSAGE -30
-#define MA_BAD_MESSAGE -31
-#define MA_NO_DATA_AVAILABLE -32
-#define MA_INVALID_DATA -33
-#define MA_TIMEOUT -34
-#define MA_NO_NETWORK -35
-#define MA_NOT_UNIQUE -36
-#define MA_NOT_SOCKET -37
-#define MA_NO_ADDRESS -38
-#define MA_BAD_PROTOCOL -39
-#define MA_PROTOCOL_UNAVAILABLE -40
-#define MA_PROTOCOL_NOT_SUPPORTED -41
-#define MA_PROTOCOL_FAMILY_NOT_SUPPORTED -42
-#define MA_ADDRESS_FAMILY_NOT_SUPPORTED -43
-#define MA_SOCKET_NOT_SUPPORTED -44
-#define MA_CONNECTION_RESET -45
-#define MA_ALREADY_CONNECTED -46
-#define MA_NOT_CONNECTED -47
-#define MA_CONNECTION_REFUSED -48
-#define MA_NO_HOST -49
-#define MA_IN_PROGRESS -50
-#define MA_CANCELLED -51
-#define MA_MEMORY_ALREADY_MAPPED -52
-
-/* General miniaudio-specific errors. */
-#define MA_FORMAT_NOT_SUPPORTED -100
-#define MA_DEVICE_TYPE_NOT_SUPPORTED -101
-#define MA_SHARE_MODE_NOT_SUPPORTED -102
-#define MA_NO_BACKEND -103
-#define MA_NO_DEVICE -104
-#define MA_API_NOT_FOUND -105
-#define MA_INVALID_DEVICE_CONFIG -106
-#define MA_LOOP -107
-
-/* State errors. */
-#define MA_DEVICE_NOT_INITIALIZED -200
-#define MA_DEVICE_ALREADY_INITIALIZED -201
-#define MA_DEVICE_NOT_STARTED -202
-#define MA_DEVICE_NOT_STOPPED -203
-
-/* Operation errors. */
-#define MA_FAILED_TO_INIT_BACKEND -300
-#define MA_FAILED_TO_OPEN_BACKEND_DEVICE -301
-#define MA_FAILED_TO_START_BACKEND_DEVICE -302
-#define MA_FAILED_TO_STOP_BACKEND_DEVICE -303
-
-
-#define MA_MIN_CHANNELS 1
-#ifndef MA_MAX_CHANNELS
-#define MA_MAX_CHANNELS 32
-#endif
+ ```c
+ #define MA_NO_WAV
+ #define MA_NO_MP3
+ #define MA_NO_FLAC
+ ```
+Disabling built-in decoding libraries is useful if you use these libraries independantly of the
+`ma_decoder` API.
-#ifndef MA_MAX_FILTER_ORDER
-#define MA_MAX_FILTER_ORDER 8
-#endif
+A decoder can be initialized from a file with `ma_decoder_init_file()`, a block of memory with
+`ma_decoder_init_memory()`, or from data delivered via callbacks with `ma_decoder_init()`. Here is
+an example for loading a decoder from a file:
-typedef enum
-{
- ma_stream_format_pcm = 0
-} ma_stream_format;
+ ```c
+ ma_decoder decoder;
+ ma_result result = ma_decoder_init_file("MySong.mp3", NULL, &decoder);
+ if (result != MA_SUCCESS) {
+ return false; // An error occurred.
+ }
-typedef enum
-{
- ma_stream_layout_interleaved = 0,
- ma_stream_layout_deinterleaved
-} ma_stream_layout;
+ ...
-typedef enum
-{
- ma_dither_mode_none = 0,
- ma_dither_mode_rectangle,
- ma_dither_mode_triangle
-} ma_dither_mode;
+ ma_decoder_uninit(&decoder);
+ ```
-typedef enum
-{
- /*
- I like to keep these explicitly defined because they're used as a key into a lookup table. When items are
- added to this, make sure there are no gaps and that they're added to the lookup table in ma_get_bytes_per_sample().
- */
- ma_format_unknown = 0, /* Mainly used for indicating an error, but also used as the default for the output format for decoders. */
- ma_format_u8 = 1,
- ma_format_s16 = 2, /* Seems to be the most widely supported format. */
- ma_format_s24 = 3, /* Tightly packed. 3 bytes per sample. */
- ma_format_s32 = 4,
- ma_format_f32 = 5,
- ma_format_count
-} ma_format;
+When initializing a decoder, you can optionally pass in a pointer to a `ma_decoder_config` object
+(the `NULL` argument in the example above) which allows you to configure the output format, channel
+count, sample rate and channel map:
-typedef enum
-{
- /* Standard rates need to be in priority order. */
- ma_standard_sample_rate_48000 = 48000, /* Most common */
- ma_standard_sample_rate_44100 = 44100,
+ ```c
+ ma_decoder_config config = ma_decoder_config_init(ma_format_f32, 2, 48000);
+ ```
- ma_standard_sample_rate_32000 = 32000, /* Lows */
- ma_standard_sample_rate_24000 = 24000,
- ma_standard_sample_rate_22050 = 22050,
+When passing in `NULL` for decoder config in `ma_decoder_init*()`, the output format will be the
+same as that defined by the decoding backend.
- ma_standard_sample_rate_88200 = 88200, /* Highs */
- ma_standard_sample_rate_96000 = 96000,
- ma_standard_sample_rate_176400 = 176400,
- ma_standard_sample_rate_192000 = 192000,
+Data is read from the decoder as PCM frames. This will output the number of PCM frames actually
+read. If this is less than the requested number of PCM frames it means you've reached the end. The
+return value will be `MA_AT_END` if no samples have been read and the end has been reached.
- ma_standard_sample_rate_16000 = 16000, /* Extreme lows */
- ma_standard_sample_rate_11025 = 11250,
- ma_standard_sample_rate_8000 = 8000,
+ ```c
+ ma_result result = ma_decoder_read_pcm_frames(pDecoder, pFrames, framesToRead, &framesRead);
+ if (framesRead < framesToRead) {
+ // Reached the end.
+ }
+ ```
- ma_standard_sample_rate_352800 = 352800, /* Extreme highs */
- ma_standard_sample_rate_384000 = 384000,
+You can also seek to a specific frame like so:
- ma_standard_sample_rate_min = ma_standard_sample_rate_8000,
- ma_standard_sample_rate_max = ma_standard_sample_rate_384000,
- ma_standard_sample_rate_count = 14 /* Need to maintain the count manually. Make sure this is updated if items are added to enum. */
-} ma_standard_sample_rate;
+ ```c
+ ma_result result = ma_decoder_seek_to_pcm_frame(pDecoder, targetFrame);
+ if (result != MA_SUCCESS) {
+ return false; // An error occurred.
+ }
+ ```
-/* These are deprecated. Use ma_standard_sample_rate_min and ma_standard_sample_rate_max. */
-#define MA_MIN_SAMPLE_RATE (ma_uint32)ma_standard_sample_rate_min
-#define MA_MAX_SAMPLE_RATE (ma_uint32)ma_standard_sample_rate_max
+If you want to loop back to the start, you can simply seek back to the first PCM frame:
+ ```c
+ ma_decoder_seek_to_pcm_frame(pDecoder, 0);
+ ```
-typedef enum
-{
- ma_channel_mix_mode_rectangular = 0, /* Simple averaging based on the plane(s) the channel is sitting on. */
- ma_channel_mix_mode_simple, /* Drop excess channels; zeroed out extra channels. */
- ma_channel_mix_mode_custom_weights, /* Use custom weights specified in ma_channel_router_config. */
- ma_channel_mix_mode_planar_blend = ma_channel_mix_mode_rectangular,
- ma_channel_mix_mode_default = ma_channel_mix_mode_rectangular
-} ma_channel_mix_mode;
+When loading a decoder, miniaudio uses a trial and error technique to find the appropriate decoding
+backend. This can be unnecessarily inefficient if the type is already known. In this case you can
+use `encodingFormat` variable in the device config to specify a specific encoding format you want
+to decode:
-typedef enum
-{
- ma_standard_channel_map_microsoft,
- ma_standard_channel_map_alsa,
- ma_standard_channel_map_rfc3551, /* Based off AIFF. */
- ma_standard_channel_map_flac,
- ma_standard_channel_map_vorbis,
- ma_standard_channel_map_sound4, /* FreeBSD's sound(4). */
- ma_standard_channel_map_sndio, /* www.sndio.org/tips.html */
- ma_standard_channel_map_webaudio = ma_standard_channel_map_flac, /* https://webaudio.github.io/web-audio-api/#ChannelOrdering. Only 1, 2, 4 and 6 channels are defined, but can fill in the gaps with logical assumptions. */
- ma_standard_channel_map_default = ma_standard_channel_map_microsoft
-} ma_standard_channel_map;
+ ```c
+ decoderConfig.encodingFormat = ma_encoding_format_wav;
+ ```
-typedef enum
-{
- ma_performance_profile_low_latency = 0,
- ma_performance_profile_conservative
-} ma_performance_profile;
+See the `ma_encoding_format` enum for possible encoding formats.
+The `ma_decoder_init_file()` API will try using the file extension to determine which decoding
+backend to prefer.
-typedef struct
-{
- void* pUserData;
- void* (* onMalloc)(size_t sz, void* pUserData);
- void* (* onRealloc)(void* p, size_t sz, void* pUserData);
- void (* onFree)(void* p, void* pUserData);
-} ma_allocation_callbacks;
-typedef struct
-{
- ma_int32 state;
-} ma_lcg;
+8.1. Custom Decoders
+--------------------
+It's possible to implement a custom decoder and plug it into miniaudio. This is extremely useful
+when you want to use the `ma_decoder` API, but need to support an encoding format that's not one of
+the stock formats supported by miniaudio. This can be put to particularly good use when using the
+`ma_engine` and/or `ma_resource_manager` APIs because they use `ma_decoder` internally. If, for
+example, you wanted to support Opus, you can do so with a custom decoder (there if a reference
+Opus decoder in the "extras" folder of the miniaudio repository which uses libopus + libopusfile).
+A custom decoder must implement a data source. A vtable called `ma_decoding_backend_vtable` needs
+to be implemented which is then passed into the decoder config:
-#ifndef MA_NO_THREADING
-/* Thread priorities should be ordered such that the default priority of the worker thread is 0. */
-typedef enum
-{
- ma_thread_priority_idle = -5,
- ma_thread_priority_lowest = -4,
- ma_thread_priority_low = -3,
- ma_thread_priority_normal = -2,
- ma_thread_priority_high = -1,
- ma_thread_priority_highest = 0,
- ma_thread_priority_realtime = 1,
- ma_thread_priority_default = 0
-} ma_thread_priority;
+ ```c
+ ma_decoding_backend_vtable* pCustomBackendVTables[] =
+ {
+ &g_ma_decoding_backend_vtable_libvorbis,
+ &g_ma_decoding_backend_vtable_libopus
+ };
-/* Spinlocks are 32-bit for compatibility reasons. */
-typedef ma_uint32 ma_spinlock;
+ ...
-#if defined(MA_WIN32)
-typedef ma_handle ma_thread;
-#endif
-#if defined(MA_POSIX)
-typedef pthread_t ma_thread;
-#endif
+ decoderConfig = ma_decoder_config_init_default();
+ decoderConfig.pCustomBackendUserData = NULL;
+ decoderConfig.ppCustomBackendVTables = pCustomBackendVTables;
+ decoderConfig.customBackendCount = sizeof(pCustomBackendVTables) / sizeof(pCustomBackendVTables[0]);
+ ```
-#if defined(MA_WIN32)
-typedef ma_handle ma_mutex;
-#endif
-#if defined(MA_POSIX)
-typedef pthread_mutex_t ma_mutex;
-#endif
+The `ma_decoding_backend_vtable` vtable has the following functions:
-#if defined(MA_WIN32)
-typedef ma_handle ma_event;
-#endif
-#if defined(MA_POSIX)
-typedef struct
-{
- ma_uint32 value;
- pthread_mutex_t lock;
- pthread_cond_t cond;
-} ma_event;
-#endif /* MA_POSIX */
+ ```
+ onInit
+ onInitFile
+ onInitFileW
+ onInitMemory
+ onUninit
+ ```
-#if defined(MA_WIN32)
-typedef ma_handle ma_semaphore;
-#endif
-#if defined(MA_POSIX)
-typedef struct
-{
- int value;
- pthread_mutex_t lock;
- pthread_cond_t cond;
-} ma_semaphore;
-#endif /* MA_POSIX */
-#else
-/* MA_NO_THREADING is set which means threading is disabled. Threading is required by some API families. If any of these are enabled we need to throw an error. */
-#ifndef MA_NO_DEVICE_IO
-#error "MA_NO_THREADING cannot be used without MA_NO_DEVICE_IO";
-#endif
-#endif /* MA_NO_THREADING */
+There are only two functions that must be implemented - `onInit` and `onUninit`. The other
+functions can be implemented for a small optimization for loading from a file path or memory. If
+these are not specified, miniaudio will deal with it for you via a generic implementation.
+When you initialize a custom data source (by implementing the `onInit` function in the vtable) you
+will need to output a pointer to a `ma_data_source` which implements your custom decoder. See the
+section about data sources for details on how to implemen this. Alternatively, see the
+"custom_decoders" example in the miniaudio repository.
-/*
-Retrieves the version of miniaudio as separated integers. Each component can be NULL if it's not required.
-*/
-MA_API void ma_version(ma_uint32* pMajor, ma_uint32* pMinor, ma_uint32* pRevision);
+The `onInit` function takes a pointer to some callbacks for the purpose of reading raw audio data
+from some abitrary source. You'll use these functions to read from the raw data and perform the
+decoding. When you call them, you will pass in the `pReadSeekTellUserData` pointer to the relevant
+parameter.
-/*
-Retrieves the version of miniaudio as a string which can be useful for logging purposes.
-*/
-MA_API const char* ma_version_string(void);
+The `pConfig` parameter in `onInit` can be used to configure the backend if appropriate. It's only
+used as a hint and can be ignored. However, if any of the properties are relevant to your decoder,
+an optimal implementation will handle the relevant properties appropriately.
+If memory allocation is required, it should be done so via the specified allocation callbacks if
+possible (the `pAllocationCallbacks` parameter).
-/**************************************************************************************************************************************************************
+If an error occurs when initializing the decoder, you should leave `ppBackend` unset, or set to
+NULL, and make sure everything is cleaned up appropriately and an appropriate result code returned.
+When multiple custom backends are specified, miniaudio will cycle through the vtables in the order
+they're listed in the array that's passed into the decoder config so it's important that your
+initialization routine is clean.
-Logging
+When a decoder is uninitialized, the `onUninit` callback will be fired which will give you an
+opportunity to clean up and internal data.
-**************************************************************************************************************************************************************/
-#include /* For va_list. */
-#if defined(__has_attribute)
- #if __has_attribute(format)
- #define MA_ATTRIBUTE_FORMAT(fmt, va) __attribute__((format(printf, fmt, va)))
- #endif
-#endif
-#ifndef MA_ATTRIBUTE_FORMAT
-#define MA_ATTRIBUTE_FORMAT(fmt,va)
-#endif
-#ifndef MA_MAX_LOG_CALLBACKS
-#define MA_MAX_LOG_CALLBACKS 4
-#endif
+9. Encoding
+===========
+The `ma_encoding` API is used for writing audio files. The only supported output format is WAV
+which is achieved via dr_wav which is amalgamated into the implementation section of miniaudio.
+This can be disabled by specifying the following option before the implementation of miniaudio:
-typedef void (* ma_log_callback_proc)(void* pUserData, ma_uint32 level, const char* pMessage);
+ ```c
+ #define MA_NO_WAV
+ ```
-typedef struct
-{
- ma_log_callback_proc onLog;
- void* pUserData;
-} ma_log_callback;
+An encoder can be initialized to write to a file with `ma_encoder_init_file()` or from data
+delivered via callbacks with `ma_encoder_init()`. Below is an example for initializing an encoder
+to output to a file.
-MA_API ma_log_callback ma_log_callback_init(ma_log_callback_proc onLog, void* pUserData);
+ ```c
+ ma_encoder_config config = ma_encoder_config_init(ma_encoding_format_wav, FORMAT, CHANNELS, SAMPLE_RATE);
+ ma_encoder encoder;
+ ma_result result = ma_encoder_init_file("my_file.wav", &config, &encoder);
+ if (result != MA_SUCCESS) {
+ // Error
+ }
+ ...
-typedef struct
-{
- ma_log_callback callbacks[MA_MAX_LOG_CALLBACKS];
- ma_uint32 callbackCount;
- ma_allocation_callbacks allocationCallbacks; /* Need to store these persistently because ma_log_postv() might need to allocate a buffer on the heap. */
-#ifndef MA_NO_THREADING
- ma_mutex lock; /* For thread safety just to make it easier and safer for the logging implementation. */
-#endif
-} ma_log;
+ ma_encoder_uninit(&encoder);
+ ```
-MA_API ma_result ma_log_init(const ma_allocation_callbacks* pAllocationCallbacks, ma_log* pLog);
-MA_API void ma_log_uninit(ma_log* pLog);
-MA_API ma_result ma_log_register_callback(ma_log* pLog, ma_log_callback callback);
-MA_API ma_result ma_log_unregister_callback(ma_log* pLog, ma_log_callback callback);
-MA_API ma_result ma_log_post(ma_log* pLog, ma_uint32 level, const char* pMessage);
-MA_API ma_result ma_log_postv(ma_log* pLog, ma_uint32 level, const char* pFormat, va_list args);
-MA_API ma_result ma_log_postf(ma_log* pLog, ma_uint32 level, const char* pFormat, ...) MA_ATTRIBUTE_FORMAT(3, 4);
+When initializing an encoder you must specify a config which is initialized with
+`ma_encoder_config_init()`. Here you must specify the file type, the output sample format, output
+channel count and output sample rate. The following file types are supported:
+ +------------------------+-------------+
+ | Enum | Description |
+ +------------------------+-------------+
+ | ma_encoding_format_wav | WAV |
+ +------------------------+-------------+
-/**************************************************************************************************************************************************************
+If the format, channel count or sample rate is not supported by the output file type an error will
+be returned. The encoder will not perform data conversion so you will need to convert it before
+outputting any audio data. To output audio data, use `ma_encoder_write_pcm_frames()`, like in the
+example below:
-Biquad Filtering
+ ```c
+ framesWritten = ma_encoder_write_pcm_frames(&encoder, pPCMFramesToWrite, framesToWrite);
+ ```
-**************************************************************************************************************************************************************/
-typedef union
-{
- float f32;
- ma_int32 s32;
-} ma_biquad_coefficient;
+Encoders must be uninitialized with `ma_encoder_uninit()`.
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- double b0;
- double b1;
- double b2;
- double a0;
- double a1;
- double a2;
-} ma_biquad_config;
-MA_API ma_biquad_config ma_biquad_config_init(ma_format format, ma_uint32 channels, double b0, double b1, double b2, double a0, double a1, double a2);
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- ma_biquad_coefficient b0;
- ma_biquad_coefficient b1;
- ma_biquad_coefficient b2;
- ma_biquad_coefficient a1;
- ma_biquad_coefficient a2;
- ma_biquad_coefficient r1[MA_MAX_CHANNELS];
- ma_biquad_coefficient r2[MA_MAX_CHANNELS];
-} ma_biquad;
+10. Data Conversion
+===================
+A data conversion API is included with miniaudio which supports the majority of data conversion
+requirements. This supports conversion between sample formats, channel counts (with channel
+mapping) and sample rates.
-MA_API ma_result ma_biquad_init(const ma_biquad_config* pConfig, ma_biquad* pBQ);
-MA_API ma_result ma_biquad_reinit(const ma_biquad_config* pConfig, ma_biquad* pBQ);
-MA_API ma_result ma_biquad_process_pcm_frames(ma_biquad* pBQ, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
-MA_API ma_uint32 ma_biquad_get_latency(const ma_biquad* pBQ);
+10.1. Sample Format Conversion
+------------------------------
+Conversion between sample formats is achieved with the `ma_pcm_*_to_*()`, `ma_pcm_convert()` and
+`ma_convert_pcm_frames_format()` APIs. Use `ma_pcm_*_to_*()` to convert between two specific
+formats. Use `ma_pcm_convert()` to convert based on a `ma_format` variable. Use
+`ma_convert_pcm_frames_format()` to convert PCM frames where you want to specify the frame count
+and channel count as a variable instead of the total sample count.
-/**************************************************************************************************************************************************************
-Low-Pass Filtering
+10.1.1. Dithering
+-----------------
+Dithering can be set using the ditherMode parameter.
-**************************************************************************************************************************************************************/
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- ma_uint32 sampleRate;
- double cutoffFrequency;
- double q;
-} ma_lpf1_config, ma_lpf2_config;
+The different dithering modes include the following, in order of efficiency:
-MA_API ma_lpf1_config ma_lpf1_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency);
-MA_API ma_lpf2_config ma_lpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q);
+ +-----------+--------------------------+
+ | Type | Enum Token |
+ +-----------+--------------------------+
+ | None | ma_dither_mode_none |
+ | Rectangle | ma_dither_mode_rectangle |
+ | Triangle | ma_dither_mode_triangle |
+ +-----------+--------------------------+
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- ma_biquad_coefficient a;
- ma_biquad_coefficient r1[MA_MAX_CHANNELS];
-} ma_lpf1;
+Note that even if the dither mode is set to something other than `ma_dither_mode_none`, it will be
+ignored for conversions where dithering is not needed. Dithering is available for the following
+conversions:
-MA_API ma_result ma_lpf1_init(const ma_lpf1_config* pConfig, ma_lpf1* pLPF);
-MA_API ma_result ma_lpf1_reinit(const ma_lpf1_config* pConfig, ma_lpf1* pLPF);
-MA_API ma_result ma_lpf1_process_pcm_frames(ma_lpf1* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
-MA_API ma_uint32 ma_lpf1_get_latency(const ma_lpf1* pLPF);
+ ```
+ s16 -> u8
+ s24 -> u8
+ s32 -> u8
+ f32 -> u8
+ s24 -> s16
+ s32 -> s16
+ f32 -> s16
+ ```
-typedef struct
-{
- ma_biquad bq; /* The second order low-pass filter is implemented as a biquad filter. */
-} ma_lpf2;
+Note that it is not an error to pass something other than ma_dither_mode_none for conversions where
+dither is not used. It will just be ignored.
-MA_API ma_result ma_lpf2_init(const ma_lpf2_config* pConfig, ma_lpf2* pLPF);
-MA_API ma_result ma_lpf2_reinit(const ma_lpf2_config* pConfig, ma_lpf2* pLPF);
-MA_API ma_result ma_lpf2_process_pcm_frames(ma_lpf2* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
-MA_API ma_uint32 ma_lpf2_get_latency(const ma_lpf2* pLPF);
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- ma_uint32 sampleRate;
- double cutoffFrequency;
- ma_uint32 order; /* If set to 0, will be treated as a passthrough (no filtering will be applied). */
-} ma_lpf_config;
+10.2. Channel Conversion
+------------------------
+Channel conversion is used for channel rearrangement and conversion from one channel count to
+another. The `ma_channel_converter` API is used for channel conversion. Below is an example of
+initializing a simple channel converter which converts from mono to stereo.
-MA_API ma_lpf_config ma_lpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order);
+ ```c
+ ma_channel_converter_config config = ma_channel_converter_config_init(
+ ma_format, // Sample format
+ 1, // Input channels
+ NULL, // Input channel map
+ 2, // Output channels
+ NULL, // Output channel map
+ ma_channel_mix_mode_default); // The mixing algorithm to use when combining channels.
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- ma_uint32 sampleRate;
- ma_uint32 lpf1Count;
- ma_uint32 lpf2Count;
- ma_lpf1 lpf1[1];
- ma_lpf2 lpf2[MA_MAX_FILTER_ORDER/2];
-} ma_lpf;
+ result = ma_channel_converter_init(&config, NULL, &converter);
+ if (result != MA_SUCCESS) {
+ // Error.
+ }
+ ```
-MA_API ma_result ma_lpf_init(const ma_lpf_config* pConfig, ma_lpf* pLPF);
-MA_API ma_result ma_lpf_reinit(const ma_lpf_config* pConfig, ma_lpf* pLPF);
-MA_API ma_result ma_lpf_process_pcm_frames(ma_lpf* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
-MA_API ma_uint32 ma_lpf_get_latency(const ma_lpf* pLPF);
+To perform the conversion simply call `ma_channel_converter_process_pcm_frames()` like so:
+ ```c
+ ma_result result = ma_channel_converter_process_pcm_frames(&converter, pFramesOut, pFramesIn, frameCount);
+ if (result != MA_SUCCESS) {
+ // Error.
+ }
+ ```
-/**************************************************************************************************************************************************************
+It is up to the caller to ensure the output buffer is large enough to accomodate the new PCM
+frames.
-High-Pass Filtering
+Input and output PCM frames are always interleaved. Deinterleaved layouts are not supported.
-**************************************************************************************************************************************************************/
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- ma_uint32 sampleRate;
- double cutoffFrequency;
- double q;
-} ma_hpf1_config, ma_hpf2_config;
-MA_API ma_hpf1_config ma_hpf1_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency);
-MA_API ma_hpf2_config ma_hpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q);
+10.2.1. Channel Mapping
+-----------------------
+In addition to converting from one channel count to another, like the example above, the channel
+converter can also be used to rearrange channels. When initializing the channel converter, you can
+optionally pass in channel maps for both the input and output frames. If the channel counts are the
+same, and each channel map contains the same channel positions with the exception that they're in
+a different order, a simple shuffling of the channels will be performed. If, however, there is not
+a 1:1 mapping of channel positions, or the channel counts differ, the input channels will be mixed
+based on a mixing mode which is specified when initializing the `ma_channel_converter_config`
+object.
+
+When converting from mono to multi-channel, the mono channel is simply copied to each output
+channel. When going the other way around, the audio of each output channel is simply averaged and
+copied to the mono channel.
+
+In more complicated cases blending is used. The `ma_channel_mix_mode_simple` mode will drop excess
+channels and silence extra channels. For example, converting from 4 to 2 channels, the 3rd and 4th
+channels will be dropped, whereas converting from 2 to 4 channels will put silence into the 3rd and
+4th channels.
+
+The `ma_channel_mix_mode_rectangle` mode uses spacial locality based on a rectangle to compute a
+simple distribution between input and output. Imagine sitting in the middle of a room, with
+speakers on the walls representing channel positions. The `MA_CHANNEL_FRONT_LEFT` position can be
+thought of as being in the corner of the front and left walls.
+
+Finally, the `ma_channel_mix_mode_custom_weights` mode can be used to use custom user-defined
+weights. Custom weights can be passed in as the last parameter of
+`ma_channel_converter_config_init()`.
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- ma_biquad_coefficient a;
- ma_biquad_coefficient r1[MA_MAX_CHANNELS];
-} ma_hpf1;
+Predefined channel maps can be retrieved with `ma_channel_map_init_standard()`. This takes a
+`ma_standard_channel_map` enum as it's first parameter, which can be one of the following:
-MA_API ma_result ma_hpf1_init(const ma_hpf1_config* pConfig, ma_hpf1* pHPF);
-MA_API ma_result ma_hpf1_reinit(const ma_hpf1_config* pConfig, ma_hpf1* pHPF);
-MA_API ma_result ma_hpf1_process_pcm_frames(ma_hpf1* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
-MA_API ma_uint32 ma_hpf1_get_latency(const ma_hpf1* pHPF);
+ +-----------------------------------+-----------------------------------------------------------+
+ | Name | Description |
+ +-----------------------------------+-----------------------------------------------------------+
+ | ma_standard_channel_map_default | Default channel map used by miniaudio. See below. |
+ | ma_standard_channel_map_microsoft | Channel map used by Microsoft's bitfield channel maps. |
+ | ma_standard_channel_map_alsa | Default ALSA channel map. |
+ | ma_standard_channel_map_rfc3551 | RFC 3551. Based on AIFF. |
+ | ma_standard_channel_map_flac | FLAC channel map. |
+ | ma_standard_channel_map_vorbis | Vorbis channel map. |
+ | ma_standard_channel_map_sound4 | FreeBSD's sound(4). |
+ | ma_standard_channel_map_sndio | sndio channel map. http://www.sndio.org/tips.html. |
+ | ma_standard_channel_map_webaudio | https://webaudio.github.io/web-audio-api/#ChannelOrdering |
+ +-----------------------------------+-----------------------------------------------------------+
-typedef struct
-{
- ma_biquad bq; /* The second order high-pass filter is implemented as a biquad filter. */
-} ma_hpf2;
+Below are the channel maps used by default in miniaudio (`ma_standard_channel_map_default`):
-MA_API ma_result ma_hpf2_init(const ma_hpf2_config* pConfig, ma_hpf2* pHPF);
-MA_API ma_result ma_hpf2_reinit(const ma_hpf2_config* pConfig, ma_hpf2* pHPF);
-MA_API ma_result ma_hpf2_process_pcm_frames(ma_hpf2* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
-MA_API ma_uint32 ma_hpf2_get_latency(const ma_hpf2* pHPF);
+ +---------------+---------------------------------+
+ | Channel Count | Mapping |
+ +---------------+---------------------------------+
+ | 1 (Mono) | 0: MA_CHANNEL_MONO |
+ +---------------+---------------------------------+
+ | 2 (Stereo) | 0: MA_CHANNEL_FRONT_LEFT
|
+ | | 1: MA_CHANNEL_FRONT_RIGHT |
+ +---------------+---------------------------------+
+ | 3 | 0: MA_CHANNEL_FRONT_LEFT
|
+ | | 1: MA_CHANNEL_FRONT_RIGHT
|
+ | | 2: MA_CHANNEL_FRONT_CENTER |
+ +---------------+---------------------------------+
+ | 4 (Surround) | 0: MA_CHANNEL_FRONT_LEFT
|
+ | | 1: MA_CHANNEL_FRONT_RIGHT
|
+ | | 2: MA_CHANNEL_FRONT_CENTER
|
+ | | 3: MA_CHANNEL_BACK_CENTER |
+ +---------------+---------------------------------+
+ | 5 | 0: MA_CHANNEL_FRONT_LEFT
|
+ | | 1: MA_CHANNEL_FRONT_RIGHT
|
+ | | 2: MA_CHANNEL_FRONT_CENTER
|
+ | | 3: MA_CHANNEL_BACK_LEFT
|
+ | | 4: MA_CHANNEL_BACK_RIGHT |
+ +---------------+---------------------------------+
+ | 6 (5.1) | 0: MA_CHANNEL_FRONT_LEFT
|
+ | | 1: MA_CHANNEL_FRONT_RIGHT
|
+ | | 2: MA_CHANNEL_FRONT_CENTER
|
+ | | 3: MA_CHANNEL_LFE
|
+ | | 4: MA_CHANNEL_SIDE_LEFT
|
+ | | 5: MA_CHANNEL_SIDE_RIGHT |
+ +---------------+---------------------------------+
+ | 7 | 0: MA_CHANNEL_FRONT_LEFT
|
+ | | 1: MA_CHANNEL_FRONT_RIGHT
|
+ | | 2: MA_CHANNEL_FRONT_CENTER
|
+ | | 3: MA_CHANNEL_LFE
|
+ | | 4: MA_CHANNEL_BACK_CENTER
|
+ | | 4: MA_CHANNEL_SIDE_LEFT
|
+ | | 5: MA_CHANNEL_SIDE_RIGHT |
+ +---------------+---------------------------------+
+ | 8 (7.1) | 0: MA_CHANNEL_FRONT_LEFT
|
+ | | 1: MA_CHANNEL_FRONT_RIGHT
|
+ | | 2: MA_CHANNEL_FRONT_CENTER
|
+ | | 3: MA_CHANNEL_LFE
|
+ | | 4: MA_CHANNEL_BACK_LEFT
|
+ | | 5: MA_CHANNEL_BACK_RIGHT
|
+ | | 6: MA_CHANNEL_SIDE_LEFT
|
+ | | 7: MA_CHANNEL_SIDE_RIGHT |
+ +---------------+---------------------------------+
+ | Other | All channels set to 0. This |
+ | | is equivalent to the same |
+ | | mapping as the device. |
+ +---------------+---------------------------------+
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- ma_uint32 sampleRate;
- double cutoffFrequency;
- ma_uint32 order; /* If set to 0, will be treated as a passthrough (no filtering will be applied). */
-} ma_hpf_config;
-MA_API ma_hpf_config ma_hpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order);
+10.3. Resampling
+----------------
+Resampling is achieved with the `ma_resampler` object. To create a resampler object, do something
+like the following:
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- ma_uint32 sampleRate;
- ma_uint32 hpf1Count;
- ma_uint32 hpf2Count;
- ma_hpf1 hpf1[1];
- ma_hpf2 hpf2[MA_MAX_FILTER_ORDER/2];
-} ma_hpf;
+ ```c
+ ma_resampler_config config = ma_resampler_config_init(
+ ma_format_s16,
+ channels,
+ sampleRateIn,
+ sampleRateOut,
+ ma_resample_algorithm_linear);
-MA_API ma_result ma_hpf_init(const ma_hpf_config* pConfig, ma_hpf* pHPF);
-MA_API ma_result ma_hpf_reinit(const ma_hpf_config* pConfig, ma_hpf* pHPF);
-MA_API ma_result ma_hpf_process_pcm_frames(ma_hpf* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
-MA_API ma_uint32 ma_hpf_get_latency(const ma_hpf* pHPF);
+ ma_resampler resampler;
+ ma_result result = ma_resampler_init(&config, &resampler);
+ if (result != MA_SUCCESS) {
+ // An error occurred...
+ }
+ ```
+Do the following to uninitialize the resampler:
-/**************************************************************************************************************************************************************
+ ```c
+ ma_resampler_uninit(&resampler);
+ ```
-Band-Pass Filtering
+The following example shows how data can be processed
-**************************************************************************************************************************************************************/
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- ma_uint32 sampleRate;
- double cutoffFrequency;
- double q;
-} ma_bpf2_config;
+ ```c
+ ma_uint64 frameCountIn = 1000;
+ ma_uint64 frameCountOut = 2000;
+ ma_result result = ma_resampler_process_pcm_frames(&resampler, pFramesIn, &frameCountIn, pFramesOut, &frameCountOut);
+ if (result != MA_SUCCESS) {
+ // An error occurred...
+ }
-MA_API ma_bpf2_config ma_bpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q);
+ // At this point, frameCountIn contains the number of input frames that were consumed and frameCountOut contains the
+ // number of output frames written.
+ ```
-typedef struct
-{
- ma_biquad bq; /* The second order band-pass filter is implemented as a biquad filter. */
-} ma_bpf2;
+To initialize the resampler you first need to set up a config (`ma_resampler_config`) with
+`ma_resampler_config_init()`. You need to specify the sample format you want to use, the number of
+channels, the input and output sample rate, and the algorithm.
-MA_API ma_result ma_bpf2_init(const ma_bpf2_config* pConfig, ma_bpf2* pBPF);
-MA_API ma_result ma_bpf2_reinit(const ma_bpf2_config* pConfig, ma_bpf2* pBPF);
-MA_API ma_result ma_bpf2_process_pcm_frames(ma_bpf2* pBPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
-MA_API ma_uint32 ma_bpf2_get_latency(const ma_bpf2* pBPF);
+The sample format can be either `ma_format_s16` or `ma_format_f32`. If you need a different format
+you will need to perform pre- and post-conversions yourself where necessary. Note that the format
+is the same for both input and output. The format cannot be changed after initialization.
+The resampler supports multiple channels and is always interleaved (both input and output). The
+channel count cannot be changed after initialization.
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- ma_uint32 sampleRate;
- double cutoffFrequency;
- ma_uint32 order; /* If set to 0, will be treated as a passthrough (no filtering will be applied). */
-} ma_bpf_config;
+The sample rates can be anything other than zero, and are always specified in hertz. They should be
+set to something like 44100, etc. The sample rate is the only configuration property that can be
+changed after initialization.
-MA_API ma_bpf_config ma_bpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order);
+The miniaudio resampler has built-in support for the following algorithms:
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- ma_uint32 bpf2Count;
- ma_bpf2 bpf2[MA_MAX_FILTER_ORDER/2];
-} ma_bpf;
+ +-----------+------------------------------+
+ | Algorithm | Enum Token |
+ +-----------+------------------------------+
+ | Linear | ma_resample_algorithm_linear |
+ | Custom | ma_resample_algorithm_custom |
+ +-----------+------------------------------+
-MA_API ma_result ma_bpf_init(const ma_bpf_config* pConfig, ma_bpf* pBPF);
-MA_API ma_result ma_bpf_reinit(const ma_bpf_config* pConfig, ma_bpf* pBPF);
-MA_API ma_result ma_bpf_process_pcm_frames(ma_bpf* pBPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
-MA_API ma_uint32 ma_bpf_get_latency(const ma_bpf* pBPF);
+The algorithm cannot be changed after initialization.
+Processing always happens on a per PCM frame basis and always assumes interleaved input and output.
+De-interleaved processing is not supported. To process frames, use
+`ma_resampler_process_pcm_frames()`. On input, this function takes the number of output frames you
+can fit in the output buffer and the number of input frames contained in the input buffer. On
+output these variables contain the number of output frames that were written to the output buffer
+and the number of input frames that were consumed in the process. You can pass in NULL for the
+input buffer in which case it will be treated as an infinitely large buffer of zeros. The output
+buffer can also be NULL, in which case the processing will be treated as seek.
-/**************************************************************************************************************************************************************
+The sample rate can be changed dynamically on the fly. You can change this with explicit sample
+rates with `ma_resampler_set_rate()` and also with a decimal ratio with
+`ma_resampler_set_rate_ratio()`. The ratio is in/out.
-Notching Filter
+Sometimes it's useful to know exactly how many input frames will be required to output a specific
+number of frames. You can calculate this with `ma_resampler_get_required_input_frame_count()`.
+Likewise, it's sometimes useful to know exactly how many frames would be output given a certain
+number of input frames. You can do this with `ma_resampler_get_expected_output_frame_count()`.
-**************************************************************************************************************************************************************/
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- ma_uint32 sampleRate;
- double q;
- double frequency;
-} ma_notch2_config, ma_notch_config;
+Due to the nature of how resampling works, the resampler introduces some latency. This can be
+retrieved in terms of both the input rate and the output rate with
+`ma_resampler_get_input_latency()` and `ma_resampler_get_output_latency()`.
-MA_API ma_notch2_config ma_notch2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double q, double frequency);
-typedef struct
-{
- ma_biquad bq;
-} ma_notch2;
+10.3.1. Resampling Algorithms
+-----------------------------
+The choice of resampling algorithm depends on your situation and requirements.
-MA_API ma_result ma_notch2_init(const ma_notch2_config* pConfig, ma_notch2* pFilter);
-MA_API ma_result ma_notch2_reinit(const ma_notch2_config* pConfig, ma_notch2* pFilter);
-MA_API ma_result ma_notch2_process_pcm_frames(ma_notch2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
-MA_API ma_uint32 ma_notch2_get_latency(const ma_notch2* pFilter);
+10.3.1.1. Linear Resampling
+---------------------------
+The linear resampler is the fastest, but comes at the expense of poorer quality. There is, however,
+some control over the quality of the linear resampler which may make it a suitable option depending
+on your requirements.
-/**************************************************************************************************************************************************************
+The linear resampler performs low-pass filtering before or after downsampling or upsampling,
+depending on the sample rates you're converting between. When decreasing the sample rate, the
+low-pass filter will be applied before downsampling. When increasing the rate it will be performed
+after upsampling. By default a fourth order low-pass filter will be applied. This can be configured
+via the `lpfOrder` configuration variable. Setting this to 0 will disable filtering.
-Peaking EQ Filter
+The low-pass filter has a cutoff frequency which defaults to half the sample rate of the lowest of
+the input and output sample rates (Nyquist Frequency).
-**************************************************************************************************************************************************************/
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- ma_uint32 sampleRate;
- double gainDB;
- double q;
- double frequency;
-} ma_peak2_config, ma_peak_config;
+The API for the linear resampler is the same as the main resampler API, only it's called
+`ma_linear_resampler`.
-MA_API ma_peak2_config ma_peak2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double q, double frequency);
-typedef struct
-{
- ma_biquad bq;
-} ma_peak2;
+10.3.2. Custom Resamplers
+-------------------------
+You can implement a custom resampler by using the `ma_resample_algorithm_custom` resampling
+algorithm and setting a vtable in the resampler config:
-MA_API ma_result ma_peak2_init(const ma_peak2_config* pConfig, ma_peak2* pFilter);
-MA_API ma_result ma_peak2_reinit(const ma_peak2_config* pConfig, ma_peak2* pFilter);
-MA_API ma_result ma_peak2_process_pcm_frames(ma_peak2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
-MA_API ma_uint32 ma_peak2_get_latency(const ma_peak2* pFilter);
+ ```c
+ ma_resampler_config config = ma_resampler_config_init(..., ma_resample_algorithm_custom);
+ config.pBackendVTable = &g_customResamplerVTable;
+ ```
+Custom resamplers are useful if the stock algorithms are not appropriate for your use case. You
+need to implement the required functions in `ma_resampling_backend_vtable`. Note that not all
+functions in the vtable need to be implemented, but if it's possible to implement, they should be.
-/**************************************************************************************************************************************************************
+You can use the `ma_linear_resampler` object for an example on how to implement the vtable. The
+`onGetHeapSize` callback is used to calculate the size of any internal heap allocation the custom
+resampler will need to make given the supplied config. When you initialize the resampler via the
+`onInit` callback, you'll be given a pointer to a heap allocation which is where you should store
+the heap allocated data. You should not free this data in `onUninit` because miniaudio will manage
+it for you.
-Low Shelf Filter
+The `onProcess` callback is where the actual resampling takes place. On input, `pFrameCountIn`
+points to a variable containing the number of frames in the `pFramesIn` buffer and
+`pFrameCountOut` points to a variable containing the capacity in frames of the `pFramesOut` buffer.
+On output, `pFrameCountIn` should be set to the number of input frames that were fully consumed,
+whereas `pFrameCountOut` should be set to the number of frames that were written to `pFramesOut`.
-**************************************************************************************************************************************************************/
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- ma_uint32 sampleRate;
- double gainDB;
- double shelfSlope;
- double frequency;
-} ma_loshelf2_config, ma_loshelf_config;
+The `onSetRate` callback is optional and is used for dynamically changing the sample rate. If
+dynamic rate changes are not supported, you can set this callback to NULL.
-MA_API ma_loshelf2_config ma_loshelf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double shelfSlope, double frequency);
+The `onGetInputLatency` and `onGetOutputLatency` functions are used for retrieving the latency in
+input and output rates respectively. These can be NULL in which case latency calculations will be
+assumed to be NULL.
-typedef struct
-{
- ma_biquad bq;
-} ma_loshelf2;
+The `onGetRequiredInputFrameCount` callback is used to give miniaudio a hint as to how many input
+frames are required to be available to produce the given number of output frames. Likewise, the
+`onGetExpectedOutputFrameCount` callback is used to determine how many output frames will be
+produced given the specified number of input frames. miniaudio will use these as a hint, but they
+are optional and can be set to NULL if you're unable to implement them.
-MA_API ma_result ma_loshelf2_init(const ma_loshelf2_config* pConfig, ma_loshelf2* pFilter);
-MA_API ma_result ma_loshelf2_reinit(const ma_loshelf2_config* pConfig, ma_loshelf2* pFilter);
-MA_API ma_result ma_loshelf2_process_pcm_frames(ma_loshelf2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
-MA_API ma_uint32 ma_loshelf2_get_latency(const ma_loshelf2* pFilter);
-/**************************************************************************************************************************************************************
+10.4. General Data Conversion
+-----------------------------
+The `ma_data_converter` API can be used to wrap sample format conversion, channel conversion and
+resampling into one operation. This is what miniaudio uses internally to convert between the format
+requested when the device was initialized and the format of the backend's native device. The API
+for general data conversion is very similar to the resampling API. Create a `ma_data_converter`
+object like this:
-High Shelf Filter
+ ```c
+ ma_data_converter_config config = ma_data_converter_config_init(
+ inputFormat,
+ outputFormat,
+ inputChannels,
+ outputChannels,
+ inputSampleRate,
+ outputSampleRate
+ );
-**************************************************************************************************************************************************************/
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- ma_uint32 sampleRate;
- double gainDB;
- double shelfSlope;
- double frequency;
-} ma_hishelf2_config, ma_hishelf_config;
+ ma_data_converter converter;
+ ma_result result = ma_data_converter_init(&config, NULL, &converter);
+ if (result != MA_SUCCESS) {
+ // An error occurred...
+ }
+ ```
-MA_API ma_hishelf2_config ma_hishelf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double shelfSlope, double frequency);
+In the example above we use `ma_data_converter_config_init()` to initialize the config, however
+there's many more properties that can be configured, such as channel maps and resampling quality.
+Something like the following may be more suitable depending on your requirements:
-typedef struct
-{
- ma_biquad bq;
-} ma_hishelf2;
+ ```c
+ ma_data_converter_config config = ma_data_converter_config_init_default();
+ config.formatIn = inputFormat;
+ config.formatOut = outputFormat;
+ config.channelsIn = inputChannels;
+ config.channelsOut = outputChannels;
+ config.sampleRateIn = inputSampleRate;
+ config.sampleRateOut = outputSampleRate;
+ ma_channel_map_init_standard(ma_standard_channel_map_flac, config.channelMapIn, sizeof(config.channelMapIn)/sizeof(config.channelMapIn[0]), config.channelCountIn);
+ config.resampling.linear.lpfOrder = MA_MAX_FILTER_ORDER;
+ ```
-MA_API ma_result ma_hishelf2_init(const ma_hishelf2_config* pConfig, ma_hishelf2* pFilter);
-MA_API ma_result ma_hishelf2_reinit(const ma_hishelf2_config* pConfig, ma_hishelf2* pFilter);
-MA_API ma_result ma_hishelf2_process_pcm_frames(ma_hishelf2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
-MA_API ma_uint32 ma_hishelf2_get_latency(const ma_hishelf2* pFilter);
+Do the following to uninitialize the data converter:
+ ```c
+ ma_data_converter_uninit(&converter, NULL);
+ ```
+The following example shows how data can be processed
-/************************************************************************************************************************************************************
-*************************************************************************************************************************************************************
+ ```c
+ ma_uint64 frameCountIn = 1000;
+ ma_uint64 frameCountOut = 2000;
+ ma_result result = ma_data_converter_process_pcm_frames(&converter, pFramesIn, &frameCountIn, pFramesOut, &frameCountOut);
+ if (result != MA_SUCCESS) {
+ // An error occurred...
+ }
-DATA CONVERSION
-===============
+ // At this point, frameCountIn contains the number of input frames that were consumed and frameCountOut contains the number
+ // of output frames written.
+ ```
-This section contains the APIs for data conversion. You will find everything here for channel mapping, sample format conversion, resampling, etc.
+The data converter supports multiple channels and is always interleaved (both input and output).
+The channel count cannot be changed after initialization.
-*************************************************************************************************************************************************************
-************************************************************************************************************************************************************/
+Sample rates can be anything other than zero, and are always specified in hertz. They should be set
+to something like 44100, etc. The sample rate is the only configuration property that can be
+changed after initialization, but only if the `resampling.allowDynamicSampleRate` member of
+`ma_data_converter_config` is set to `MA_TRUE`. To change the sample rate, use
+`ma_data_converter_set_rate()` or `ma_data_converter_set_rate_ratio()`. The ratio must be in/out.
+The resampling algorithm cannot be changed after initialization.
-/**************************************************************************************************************************************************************
+Processing always happens on a per PCM frame basis and always assumes interleaved input and output.
+De-interleaved processing is not supported. To process frames, use
+`ma_data_converter_process_pcm_frames()`. On input, this function takes the number of output frames
+you can fit in the output buffer and the number of input frames contained in the input buffer. On
+output these variables contain the number of output frames that were written to the output buffer
+and the number of input frames that were consumed in the process. You can pass in NULL for the
+input buffer in which case it will be treated as an infinitely large
+buffer of zeros. The output buffer can also be NULL, in which case the processing will be treated
+as seek.
-Resampling
+Sometimes it's useful to know exactly how many input frames will be required to output a specific
+number of frames. You can calculate this with `ma_data_converter_get_required_input_frame_count()`.
+Likewise, it's sometimes useful to know exactly how many frames would be output given a certain
+number of input frames. You can do this with `ma_data_converter_get_expected_output_frame_count()`.
-**************************************************************************************************************************************************************/
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- ma_uint32 sampleRateIn;
- ma_uint32 sampleRateOut;
- ma_uint32 lpfOrder; /* The low-pass filter order. Setting this to 0 will disable low-pass filtering. */
- double lpfNyquistFactor; /* 0..1. Defaults to 1. 1 = Half the sampling frequency (Nyquist Frequency), 0.5 = Quarter the sampling frequency (half Nyquest Frequency), etc. */
-} ma_linear_resampler_config;
+Due to the nature of how resampling works, the data converter introduces some latency if resampling
+is required. This can be retrieved in terms of both the input rate and the output rate with
+`ma_data_converter_get_input_latency()` and `ma_data_converter_get_output_latency()`.
-MA_API ma_linear_resampler_config ma_linear_resampler_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut);
-typedef struct
-{
- ma_linear_resampler_config config;
- ma_uint32 inAdvanceInt;
- ma_uint32 inAdvanceFrac;
- ma_uint32 inTimeInt;
- ma_uint32 inTimeFrac;
- union
- {
- float f32[MA_MAX_CHANNELS];
- ma_int16 s16[MA_MAX_CHANNELS];
- } x0; /* The previous input frame. */
- union
- {
- float f32[MA_MAX_CHANNELS];
- ma_int16 s16[MA_MAX_CHANNELS];
- } x1; /* The next input frame. */
- ma_lpf lpf;
-} ma_linear_resampler;
-MA_API ma_result ma_linear_resampler_init(const ma_linear_resampler_config* pConfig, ma_linear_resampler* pResampler);
-MA_API void ma_linear_resampler_uninit(ma_linear_resampler* pResampler);
-MA_API ma_result ma_linear_resampler_process_pcm_frames(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut);
-MA_API ma_result ma_linear_resampler_set_rate(ma_linear_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut);
-MA_API ma_result ma_linear_resampler_set_rate_ratio(ma_linear_resampler* pResampler, float ratioInOut);
-MA_API ma_uint64 ma_linear_resampler_get_required_input_frame_count(const ma_linear_resampler* pResampler, ma_uint64 outputFrameCount);
-MA_API ma_uint64 ma_linear_resampler_get_expected_output_frame_count(const ma_linear_resampler* pResampler, ma_uint64 inputFrameCount);
-MA_API ma_uint64 ma_linear_resampler_get_input_latency(const ma_linear_resampler* pResampler);
-MA_API ma_uint64 ma_linear_resampler_get_output_latency(const ma_linear_resampler* pResampler);
+11. Filtering
+=============
-typedef enum
-{
- ma_resample_algorithm_linear = 0, /* Fastest, lowest quality. Optional low-pass filtering. Default. */
- ma_resample_algorithm_speex
-} ma_resample_algorithm;
+11.1. Biquad Filtering
+----------------------
+Biquad filtering is achieved with the `ma_biquad` API. Example:
-typedef struct
-{
- ma_format format; /* Must be either ma_format_f32 or ma_format_s16. */
- ma_uint32 channels;
- ma_uint32 sampleRateIn;
- ma_uint32 sampleRateOut;
- ma_resample_algorithm algorithm;
- struct
- {
- ma_uint32 lpfOrder;
- double lpfNyquistFactor;
- } linear;
- struct
- {
- int quality; /* 0 to 10. Defaults to 3. */
- } speex;
-} ma_resampler_config;
+ ```c
+ ma_biquad_config config = ma_biquad_config_init(ma_format_f32, channels, b0, b1, b2, a0, a1, a2);
+ ma_result result = ma_biquad_init(&config, &biquad);
+ if (result != MA_SUCCESS) {
+ // Error.
+ }
-MA_API ma_resampler_config ma_resampler_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut, ma_resample_algorithm algorithm);
+ ...
-typedef struct
-{
- ma_resampler_config config;
- union
- {
- ma_linear_resampler linear;
- struct
- {
- void* pSpeexResamplerState; /* SpeexResamplerState* */
- } speex;
- } state;
-} ma_resampler;
+ ma_biquad_process_pcm_frames(&biquad, pFramesOut, pFramesIn, frameCount);
+ ```
-/*
-Initializes a new resampler object from a config.
-*/
-MA_API ma_result ma_resampler_init(const ma_resampler_config* pConfig, ma_resampler* pResampler);
+Biquad filtering is implemented using transposed direct form 2. The numerator coefficients are b0,
+b1 and b2, and the denominator coefficients are a0, a1 and a2. The a0 coefficient is required and
+coefficients must not be pre-normalized.
-/*
-Uninitializes a resampler.
-*/
-MA_API void ma_resampler_uninit(ma_resampler* pResampler);
+Supported formats are `ma_format_s16` and `ma_format_f32`. If you need to use a different format
+you need to convert it yourself beforehand. When using `ma_format_s16` the biquad filter will use
+fixed point arithmetic. When using `ma_format_f32`, floating point arithmetic will be used.
-/*
-Converts the given input data.
+Input and output frames are always interleaved.
-Both the input and output frames must be in the format specified in the config when the resampler was initilized.
+Filtering can be applied in-place by passing in the same pointer for both the input and output
+buffers, like so:
-On input, [pFrameCountOut] contains the number of output frames to process. On output it contains the number of output frames that
-were actually processed, which may be less than the requested amount which will happen if there's not enough input data. You can use
-ma_resampler_get_expected_output_frame_count() to know how many output frames will be processed for a given number of input frames.
+ ```c
+ ma_biquad_process_pcm_frames(&biquad, pMyData, pMyData, frameCount);
+ ```
-On input, [pFrameCountIn] contains the number of input frames contained in [pFramesIn]. On output it contains the number of whole
-input frames that were actually processed. You can use ma_resampler_get_required_input_frame_count() to know how many input frames
-you should provide for a given number of output frames. [pFramesIn] can be NULL, in which case zeroes will be used instead.
+If you need to change the values of the coefficients, but maintain the values in the registers you
+can do so with `ma_biquad_reinit()`. This is useful if you need to change the properties of the
+filter while keeping the values of registers valid to avoid glitching. Do not use
+`ma_biquad_init()` for this as it will do a full initialization which involves clearing the
+registers to 0. Note that changing the format or channel count after initialization is invalid and
+will result in an error.
-If [pFramesOut] is NULL, a seek is performed. In this case, if [pFrameCountOut] is not NULL it will seek by the specified number of
-output frames. Otherwise, if [pFramesCountOut] is NULL and [pFrameCountIn] is not NULL, it will seek by the specified number of input
-frames. When seeking, [pFramesIn] is allowed to NULL, in which case the internal timing state will be updated, but no input will be
-processed. In this case, any internal filter state will be updated as if zeroes were passed in.
-It is an error for [pFramesOut] to be non-NULL and [pFrameCountOut] to be NULL.
+11.2. Low-Pass Filtering
+------------------------
+Low-pass filtering is achieved with the following APIs:
-It is an error for both [pFrameCountOut] and [pFrameCountIn] to be NULL.
-*/
-MA_API ma_result ma_resampler_process_pcm_frames(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut);
+ +---------+------------------------------------------+
+ | API | Description |
+ +---------+------------------------------------------+
+ | ma_lpf1 | First order low-pass filter |
+ | ma_lpf2 | Second order low-pass filter |
+ | ma_lpf | High order low-pass filter (Butterworth) |
+ +---------+------------------------------------------+
+Low-pass filter example:
-/*
-Sets the input and output sample sample rate.
-*/
-MA_API ma_result ma_resampler_set_rate(ma_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut);
+ ```c
+ ma_lpf_config config = ma_lpf_config_init(ma_format_f32, channels, sampleRate, cutoffFrequency, order);
+ ma_result result = ma_lpf_init(&config, &lpf);
+ if (result != MA_SUCCESS) {
+ // Error.
+ }
-/*
-Sets the input and output sample rate as a ratio.
+ ...
-The ration is in/out.
-*/
-MA_API ma_result ma_resampler_set_rate_ratio(ma_resampler* pResampler, float ratio);
+ ma_lpf_process_pcm_frames(&lpf, pFramesOut, pFramesIn, frameCount);
+ ```
+Supported formats are `ma_format_s16` and` ma_format_f32`. If you need to use a different format
+you need to convert it yourself beforehand. Input and output frames are always interleaved.
-/*
-Calculates the number of whole input frames that would need to be read from the client in order to output the specified
-number of output frames.
+Filtering can be applied in-place by passing in the same pointer for both the input and output
+buffers, like so:
-The returned value does not include cached input frames. It only returns the number of extra frames that would need to be
-read from the input buffer in order to output the specified number of output frames.
-*/
-MA_API ma_uint64 ma_resampler_get_required_input_frame_count(const ma_resampler* pResampler, ma_uint64 outputFrameCount);
+ ```c
+ ma_lpf_process_pcm_frames(&lpf, pMyData, pMyData, frameCount);
+ ```
-/*
-Calculates the number of whole output frames that would be output after fully reading and consuming the specified number of
-input frames.
-*/
-MA_API ma_uint64 ma_resampler_get_expected_output_frame_count(const ma_resampler* pResampler, ma_uint64 inputFrameCount);
+The maximum filter order is limited to `MA_MAX_FILTER_ORDER` which is set to 8. If you need more,
+you can chain first and second order filters together.
+ ```c
+ for (iFilter = 0; iFilter < filterCount; iFilter += 1) {
+ ma_lpf2_process_pcm_frames(&lpf2[iFilter], pMyData, pMyData, frameCount);
+ }
+ ```
-/*
-Retrieves the latency introduced by the resampler in input frames.
-*/
-MA_API ma_uint64 ma_resampler_get_input_latency(const ma_resampler* pResampler);
+If you need to change the configuration of the filter, but need to maintain the state of internal
+registers you can do so with `ma_lpf_reinit()`. This may be useful if you need to change the sample
+rate and/or cutoff frequency dynamically while maintaing smooth transitions. Note that changing the
+format or channel count after initialization is invalid and will result in an error.
-/*
-Retrieves the latency introduced by the resampler in output frames.
-*/
-MA_API ma_uint64 ma_resampler_get_output_latency(const ma_resampler* pResampler);
+The `ma_lpf` object supports a configurable order, but if you only need a first order filter you
+may want to consider using `ma_lpf1`. Likewise, if you only need a second order filter you can use
+`ma_lpf2`. The advantage of this is that they're lighter weight and a bit more efficient.
+If an even filter order is specified, a series of second order filters will be processed in a
+chain. If an odd filter order is specified, a first order filter will be applied, followed by a
+series of second order filters in a chain.
-/**************************************************************************************************************************************************************
+11.3. High-Pass Filtering
+-------------------------
+High-pass filtering is achieved with the following APIs:
-Channel Conversion
+ +---------+-------------------------------------------+
+ | API | Description |
+ +---------+-------------------------------------------+
+ | ma_hpf1 | First order high-pass filter |
+ | ma_hpf2 | Second order high-pass filter |
+ | ma_hpf | High order high-pass filter (Butterworth) |
+ +---------+-------------------------------------------+
-**************************************************************************************************************************************************************/
-typedef struct
-{
- ma_format format;
- ma_uint32 channelsIn;
- ma_uint32 channelsOut;
- ma_channel channelMapIn[MA_MAX_CHANNELS];
- ma_channel channelMapOut[MA_MAX_CHANNELS];
- ma_channel_mix_mode mixingMode;
- float weights[MA_MAX_CHANNELS][MA_MAX_CHANNELS]; /* [in][out]. Only used when mixingMode is set to ma_channel_mix_mode_custom_weights. */
-} ma_channel_converter_config;
+High-pass filters work exactly the same as low-pass filters, only the APIs are called `ma_hpf1`,
+`ma_hpf2` and `ma_hpf`. See example code for low-pass filters for example usage.
-MA_API ma_channel_converter_config ma_channel_converter_config_init(ma_format format, ma_uint32 channelsIn, const ma_channel* pChannelMapIn, ma_uint32 channelsOut, const ma_channel* pChannelMapOut, ma_channel_mix_mode mixingMode);
-typedef struct
-{
- ma_format format;
- ma_uint32 channelsIn;
- ma_uint32 channelsOut;
- ma_channel channelMapIn[MA_MAX_CHANNELS];
- ma_channel channelMapOut[MA_MAX_CHANNELS];
- ma_channel_mix_mode mixingMode;
- union
- {
- float f32[MA_MAX_CHANNELS][MA_MAX_CHANNELS];
- ma_int32 s16[MA_MAX_CHANNELS][MA_MAX_CHANNELS];
- } weights;
- ma_bool8 isPassthrough;
- ma_bool8 isSimpleShuffle;
- ma_bool8 isSimpleMonoExpansion;
- ma_bool8 isStereoToMono;
- ma_uint8 shuffleTable[MA_MAX_CHANNELS];
-} ma_channel_converter;
+11.4. Band-Pass Filtering
+-------------------------
+Band-pass filtering is achieved with the following APIs:
-MA_API ma_result ma_channel_converter_init(const ma_channel_converter_config* pConfig, ma_channel_converter* pConverter);
-MA_API void ma_channel_converter_uninit(ma_channel_converter* pConverter);
-MA_API ma_result ma_channel_converter_process_pcm_frames(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
+ +---------+-------------------------------+
+ | API | Description |
+ +---------+-------------------------------+
+ | ma_bpf2 | Second order band-pass filter |
+ | ma_bpf | High order band-pass filter |
+ +---------+-------------------------------+
+Band-pass filters work exactly the same as low-pass filters, only the APIs are called `ma_bpf2` and
+`ma_hpf`. See example code for low-pass filters for example usage. Note that the order for
+band-pass filters must be an even number which means there is no first order band-pass filter,
+unlike low-pass and high-pass filters.
-/**************************************************************************************************************************************************************
-Data Conversion
+11.5. Notch Filtering
+---------------------
+Notch filtering is achieved with the following APIs:
-**************************************************************************************************************************************************************/
-typedef struct
-{
- ma_format formatIn;
- ma_format formatOut;
- ma_uint32 channelsIn;
- ma_uint32 channelsOut;
- ma_uint32 sampleRateIn;
- ma_uint32 sampleRateOut;
- ma_channel channelMapIn[MA_MAX_CHANNELS];
- ma_channel channelMapOut[MA_MAX_CHANNELS];
- ma_dither_mode ditherMode;
- ma_channel_mix_mode channelMixMode;
- float channelWeights[MA_MAX_CHANNELS][MA_MAX_CHANNELS]; /* [in][out]. Only used when channelMixMode is set to ma_channel_mix_mode_custom_weights. */
- struct
- {
- ma_resample_algorithm algorithm;
- ma_bool32 allowDynamicSampleRate;
- struct
- {
- ma_uint32 lpfOrder;
- double lpfNyquistFactor;
- } linear;
- struct
- {
- int quality;
- } speex;
- } resampling;
-} ma_data_converter_config;
+ +-----------+------------------------------------------+
+ | API | Description |
+ +-----------+------------------------------------------+
+ | ma_notch2 | Second order notching filter |
+ +-----------+------------------------------------------+
-MA_API ma_data_converter_config ma_data_converter_config_init_default(void);
-MA_API ma_data_converter_config ma_data_converter_config_init(ma_format formatIn, ma_format formatOut, ma_uint32 channelsIn, ma_uint32 channelsOut, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut);
-typedef struct
-{
- ma_data_converter_config config;
- ma_channel_converter channelConverter;
- ma_resampler resampler;
- ma_bool8 hasPreFormatConversion;
- ma_bool8 hasPostFormatConversion;
- ma_bool8 hasChannelConverter;
- ma_bool8 hasResampler;
- ma_bool8 isPassthrough;
-} ma_data_converter;
+11.6. Peaking EQ Filtering
+-------------------------
+Peaking filtering is achieved with the following APIs:
-MA_API ma_result ma_data_converter_init(const ma_data_converter_config* pConfig, ma_data_converter* pConverter);
-MA_API void ma_data_converter_uninit(ma_data_converter* pConverter);
-MA_API ma_result ma_data_converter_process_pcm_frames(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut);
-MA_API ma_result ma_data_converter_set_rate(ma_data_converter* pConverter, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut);
-MA_API ma_result ma_data_converter_set_rate_ratio(ma_data_converter* pConverter, float ratioInOut);
-MA_API ma_uint64 ma_data_converter_get_required_input_frame_count(const ma_data_converter* pConverter, ma_uint64 outputFrameCount);
-MA_API ma_uint64 ma_data_converter_get_expected_output_frame_count(const ma_data_converter* pConverter, ma_uint64 inputFrameCount);
-MA_API ma_uint64 ma_data_converter_get_input_latency(const ma_data_converter* pConverter);
-MA_API ma_uint64 ma_data_converter_get_output_latency(const ma_data_converter* pConverter);
+ +----------+------------------------------------------+
+ | API | Description |
+ +----------+------------------------------------------+
+ | ma_peak2 | Second order peaking filter |
+ +----------+------------------------------------------+
-/************************************************************************************************************************************************************
+11.7. Low Shelf Filtering
+-------------------------
+Low shelf filtering is achieved with the following APIs:
-Format Conversion
+ +-------------+------------------------------------------+
+ | API | Description |
+ +-------------+------------------------------------------+
+ | ma_loshelf2 | Second order low shelf filter |
+ +-------------+------------------------------------------+
-************************************************************************************************************************************************************/
-MA_API void ma_pcm_u8_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
-MA_API void ma_pcm_u8_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
-MA_API void ma_pcm_u8_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
-MA_API void ma_pcm_u8_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
-MA_API void ma_pcm_s16_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
-MA_API void ma_pcm_s16_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
-MA_API void ma_pcm_s16_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
-MA_API void ma_pcm_s16_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
-MA_API void ma_pcm_s24_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
-MA_API void ma_pcm_s24_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
-MA_API void ma_pcm_s24_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
-MA_API void ma_pcm_s24_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
-MA_API void ma_pcm_s32_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
-MA_API void ma_pcm_s32_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
-MA_API void ma_pcm_s32_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
-MA_API void ma_pcm_s32_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
-MA_API void ma_pcm_f32_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
-MA_API void ma_pcm_f32_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
-MA_API void ma_pcm_f32_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
-MA_API void ma_pcm_f32_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
-MA_API void ma_pcm_convert(void* pOut, ma_format formatOut, const void* pIn, ma_format formatIn, ma_uint64 sampleCount, ma_dither_mode ditherMode);
-MA_API void ma_convert_pcm_frames_format(void* pOut, ma_format formatOut, const void* pIn, ma_format formatIn, ma_uint64 frameCount, ma_uint32 channels, ma_dither_mode ditherMode);
+Where a high-pass filter is used to eliminate lower frequencies, a low shelf filter can be used to
+just turn them down rather than eliminate them entirely.
-/*
-Deinterleaves an interleaved buffer.
-*/
-MA_API void ma_deinterleave_pcm_frames(ma_format format, ma_uint32 channels, ma_uint64 frameCount, const void* pInterleavedPCMFrames, void** ppDeinterleavedPCMFrames);
-/*
-Interleaves a group of deinterleaved buffers.
-*/
-MA_API void ma_interleave_pcm_frames(ma_format format, ma_uint32 channels, ma_uint64 frameCount, const void** ppDeinterleavedPCMFrames, void* pInterleavedPCMFrames);
+11.8. High Shelf Filtering
+--------------------------
+High shelf filtering is achieved with the following APIs:
+ +-------------+------------------------------------------+
+ | API | Description |
+ +-------------+------------------------------------------+
+ | ma_hishelf2 | Second order high shelf filter |
+ +-------------+------------------------------------------+
-/************************************************************************************************************************************************************
+The high shelf filter has the same API as the low shelf filter, only you would use `ma_hishelf`
+instead of `ma_loshelf`. Where a low shelf filter is used to adjust the volume of low frequencies,
+the high shelf filter does the same thing for high frequencies.
-Channel Maps
-************************************************************************************************************************************************************/
-/*
-This is used in the shuffle table to indicate that the channel index is undefined and should be ignored.
-*/
-#define MA_CHANNEL_INDEX_NULL 255
-/* Retrieves the channel position of the specified channel based on miniaudio's default channel map. */
-MA_API ma_channel ma_channel_map_get_default_channel(ma_uint32 channelCount, ma_uint32 channelIndex);
-/*
-Retrieves the channel position of the specified channel in the given channel map.
+12. Waveform and Noise Generation
+=================================
-The pChannelMap parameter can be null, in which case miniaudio's default channel map will be assumed.
-*/
-MA_API ma_channel ma_channel_map_get_channel(const ma_channel* pChannelMap, ma_uint32 channelCount, ma_uint32 channelIndex);
+12.1. Waveforms
+---------------
+miniaudio supports generation of sine, square, triangle and sawtooth waveforms. This is achieved
+with the `ma_waveform` API. Example:
-/*
-Initializes a blank channel map.
+ ```c
+ ma_waveform_config config = ma_waveform_config_init(
+ FORMAT,
+ CHANNELS,
+ SAMPLE_RATE,
+ ma_waveform_type_sine,
+ amplitude,
+ frequency);
-When a blank channel map is specified anywhere it indicates that the native channel map should be used.
-*/
-MA_API void ma_channel_map_init_blank(ma_uint32 channels, ma_channel* pChannelMap);
+ ma_waveform waveform;
+ ma_result result = ma_waveform_init(&config, &waveform);
+ if (result != MA_SUCCESS) {
+ // Error.
+ }
-/*
-Helper for retrieving a standard channel map.
+ ...
-The output channel map buffer must have a capacity of at least `channels`.
-*/
-MA_API void ma_get_standard_channel_map(ma_standard_channel_map standardChannelMap, ma_uint32 channels, ma_channel* pChannelMap);
+ ma_waveform_read_pcm_frames(&waveform, pOutput, frameCount);
+ ```
-/*
-Copies a channel map.
+The amplitude, frequency, type, and sample rate can be changed dynamically with
+`ma_waveform_set_amplitude()`, `ma_waveform_set_frequency()`, `ma_waveform_set_type()`, and
+`ma_waveform_set_sample_rate()` respectively.
-Both input and output channel map buffers must have a capacity of at at least `channels`.
-*/
-MA_API void ma_channel_map_copy(ma_channel* pOut, const ma_channel* pIn, ma_uint32 channels);
+You can invert the waveform by setting the amplitude to a negative value. You can use this to
+control whether or not a sawtooth has a positive or negative ramp, for example.
-/*
-Copies a channel map if one is specified, otherwise copies the default channel map.
+Below are the supported waveform types:
-The output buffer must have a capacity of at least `channels`. If not NULL, the input channel map must also have a capacity of at least `channels`.
-*/
-MA_API void ma_channel_map_copy_or_default(ma_channel* pOut, const ma_channel* pIn, ma_uint32 channels);
+ +---------------------------+
+ | Enum Name |
+ +---------------------------+
+ | ma_waveform_type_sine |
+ | ma_waveform_type_square |
+ | ma_waveform_type_triangle |
+ | ma_waveform_type_sawtooth |
+ +---------------------------+
-/*
-Determines whether or not a channel map is valid.
-A blank channel map is valid (all channels set to MA_CHANNEL_NONE). The way a blank channel map is handled is context specific, but
-is usually treated as a passthrough.
+12.2. Noise
+-----------
+miniaudio supports generation of white, pink and Brownian noise via the `ma_noise` API. Example:
-Invalid channel maps:
- - A channel map with no channels
- - A channel map with more than one channel and a mono channel
+ ```c
+ ma_noise_config config = ma_noise_config_init(
+ FORMAT,
+ CHANNELS,
+ ma_noise_type_white,
+ SEED,
+ amplitude);
-The channel map buffer must have a capacity of at least `channels`.
-*/
-MA_API ma_bool32 ma_channel_map_valid(ma_uint32 channels, const ma_channel* pChannelMap);
+ ma_noise noise;
+ ma_result result = ma_noise_init(&config, &noise);
+ if (result != MA_SUCCESS) {
+ // Error.
+ }
-/*
-Helper for comparing two channel maps for equality.
+ ...
-This assumes the channel count is the same between the two.
+ ma_noise_read_pcm_frames(&noise, pOutput, frameCount);
+ ```
-Both channels map buffers must have a capacity of at least `channels`.
-*/
-MA_API ma_bool32 ma_channel_map_equal(ma_uint32 channels, const ma_channel* pChannelMapA, const ma_channel* pChannelMapB);
+The noise API uses simple LCG random number generation. It supports a custom seed which is useful
+for things like automated testing requiring reproducibility. Setting the seed to zero will default
+to `MA_DEFAULT_LCG_SEED`.
-/*
-Helper for determining if a channel map is blank (all channels set to MA_CHANNEL_NONE).
+The amplitude, seed, and type can be changed dynamically with `ma_noise_set_amplitude()`,
+`ma_noise_set_seed()`, and `ma_noise_set_type()` respectively.
-The channel map buffer must have a capacity of at least `channels`.
-*/
-MA_API ma_bool32 ma_channel_map_blank(ma_uint32 channels, const ma_channel* pChannelMap);
+By default, the noise API will use different values for different channels. So, for example, the
+left side in a stereo stream will be different to the right side. To instead have each channel use
+the same random value, set the `duplicateChannels` member of the noise config to true, like so:
-/*
-Helper for determining whether or not a channel is present in the given channel map.
+ ```c
+ config.duplicateChannels = MA_TRUE;
+ ```
-The channel map buffer must have a capacity of at least `channels`.
-*/
-MA_API ma_bool32 ma_channel_map_contains_channel_position(ma_uint32 channels, const ma_channel* pChannelMap, ma_channel channelPosition);
+Below are the supported noise types.
+ +------------------------+
+ | Enum Name |
+ +------------------------+
+ | ma_noise_type_white |
+ | ma_noise_type_pink |
+ | ma_noise_type_brownian |
+ +------------------------+
-/************************************************************************************************************************************************************
-Conversion Helpers
-************************************************************************************************************************************************************/
+13. Audio Buffers
+=================
+miniaudio supports reading from a buffer of raw audio data via the `ma_audio_buffer` API. This can
+read from memory that's managed by the application, but can also handle the memory management for
+you internally. Memory management is flexible and should support most use cases.
-/*
-High-level helper for doing a full format conversion in one go. Returns the number of output frames. Call this with pOut set to NULL to
-determine the required size of the output buffer. frameCountOut should be set to the capacity of pOut. If pOut is NULL, frameCountOut is
-ignored.
+Audio buffers are initialised using the standard configuration system used everywhere in miniaudio:
-A return value of 0 indicates an error.
+ ```c
+ ma_audio_buffer_config config = ma_audio_buffer_config_init(
+ format,
+ channels,
+ sizeInFrames,
+ pExistingData,
+ &allocationCallbacks);
-This function is useful for one-off bulk conversions, but if you're streaming data you should use the ma_data_converter APIs instead.
-*/
-MA_API ma_uint64 ma_convert_frames(void* pOut, ma_uint64 frameCountOut, ma_format formatOut, ma_uint32 channelsOut, ma_uint32 sampleRateOut, const void* pIn, ma_uint64 frameCountIn, ma_format formatIn, ma_uint32 channelsIn, ma_uint32 sampleRateIn);
-MA_API ma_uint64 ma_convert_frames_ex(void* pOut, ma_uint64 frameCountOut, const void* pIn, ma_uint64 frameCountIn, const ma_data_converter_config* pConfig);
+ ma_audio_buffer buffer;
+ result = ma_audio_buffer_init(&config, &buffer);
+ if (result != MA_SUCCESS) {
+ // Error.
+ }
+ ...
-/************************************************************************************************************************************************************
+ ma_audio_buffer_uninit(&buffer);
+ ```
-Ring Buffer
+In the example above, the memory pointed to by `pExistingData` will *not* be copied and is how an
+application can do self-managed memory allocation. If you would rather make a copy of the data, use
+`ma_audio_buffer_init_copy()`. To uninitialize the buffer, use `ma_audio_buffer_uninit()`.
-************************************************************************************************************************************************************/
-typedef struct
-{
- void* pBuffer;
- ma_uint32 subbufferSizeInBytes;
- ma_uint32 subbufferCount;
- ma_uint32 subbufferStrideInBytes;
- MA_ATOMIC ma_uint32 encodedReadOffset; /* Most significant bit is the loop flag. Lower 31 bits contains the actual offset in bytes. Must be used atomically. */
- MA_ATOMIC ma_uint32 encodedWriteOffset; /* Most significant bit is the loop flag. Lower 31 bits contains the actual offset in bytes. Must be used atomically. */
- ma_bool8 ownsBuffer; /* Used to know whether or not miniaudio is responsible for free()-ing the buffer. */
- ma_bool8 clearOnWriteAcquire; /* When set, clears the acquired write buffer before returning from ma_rb_acquire_write(). */
- ma_allocation_callbacks allocationCallbacks;
-} ma_rb;
+Sometimes it can be convenient to allocate the memory for the `ma_audio_buffer` structure and the
+raw audio data in a contiguous block of memory. That is, the raw audio data will be located
+immediately after the `ma_audio_buffer` structure. To do this, use
+`ma_audio_buffer_alloc_and_init()`:
-MA_API ma_result ma_rb_init_ex(size_t subbufferSizeInBytes, size_t subbufferCount, size_t subbufferStrideInBytes, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_rb* pRB);
-MA_API ma_result ma_rb_init(size_t bufferSizeInBytes, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_rb* pRB);
-MA_API void ma_rb_uninit(ma_rb* pRB);
-MA_API void ma_rb_reset(ma_rb* pRB);
-MA_API ma_result ma_rb_acquire_read(ma_rb* pRB, size_t* pSizeInBytes, void** ppBufferOut);
-MA_API ma_result ma_rb_commit_read(ma_rb* pRB, size_t sizeInBytes, void* pBufferOut);
-MA_API ma_result ma_rb_acquire_write(ma_rb* pRB, size_t* pSizeInBytes, void** ppBufferOut);
-MA_API ma_result ma_rb_commit_write(ma_rb* pRB, size_t sizeInBytes, void* pBufferOut);
-MA_API ma_result ma_rb_seek_read(ma_rb* pRB, size_t offsetInBytes);
-MA_API ma_result ma_rb_seek_write(ma_rb* pRB, size_t offsetInBytes);
-MA_API ma_int32 ma_rb_pointer_distance(ma_rb* pRB); /* Returns the distance between the write pointer and the read pointer. Should never be negative for a correct program. Will return the number of bytes that can be read before the read pointer hits the write pointer. */
-MA_API ma_uint32 ma_rb_available_read(ma_rb* pRB);
-MA_API ma_uint32 ma_rb_available_write(ma_rb* pRB);
-MA_API size_t ma_rb_get_subbuffer_size(ma_rb* pRB);
-MA_API size_t ma_rb_get_subbuffer_stride(ma_rb* pRB);
-MA_API size_t ma_rb_get_subbuffer_offset(ma_rb* pRB, size_t subbufferIndex);
-MA_API void* ma_rb_get_subbuffer_ptr(ma_rb* pRB, size_t subbufferIndex, void* pBuffer);
+ ```c
+ ma_audio_buffer_config config = ma_audio_buffer_config_init(
+ format,
+ channels,
+ sizeInFrames,
+ pExistingData,
+ &allocationCallbacks);
+ ma_audio_buffer* pBuffer
+ result = ma_audio_buffer_alloc_and_init(&config, &pBuffer);
+ if (result != MA_SUCCESS) {
+ // Error
+ }
-typedef struct
-{
- ma_rb rb;
- ma_format format;
- ma_uint32 channels;
-} ma_pcm_rb;
+ ...
-MA_API ma_result ma_pcm_rb_init_ex(ma_format format, ma_uint32 channels, ma_uint32 subbufferSizeInFrames, ma_uint32 subbufferCount, ma_uint32 subbufferStrideInFrames, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_pcm_rb* pRB);
-MA_API ma_result ma_pcm_rb_init(ma_format format, ma_uint32 channels, ma_uint32 bufferSizeInFrames, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_pcm_rb* pRB);
-MA_API void ma_pcm_rb_uninit(ma_pcm_rb* pRB);
-MA_API void ma_pcm_rb_reset(ma_pcm_rb* pRB);
-MA_API ma_result ma_pcm_rb_acquire_read(ma_pcm_rb* pRB, ma_uint32* pSizeInFrames, void** ppBufferOut);
-MA_API ma_result ma_pcm_rb_commit_read(ma_pcm_rb* pRB, ma_uint32 sizeInFrames, void* pBufferOut);
-MA_API ma_result ma_pcm_rb_acquire_write(ma_pcm_rb* pRB, ma_uint32* pSizeInFrames, void** ppBufferOut);
-MA_API ma_result ma_pcm_rb_commit_write(ma_pcm_rb* pRB, ma_uint32 sizeInFrames, void* pBufferOut);
-MA_API ma_result ma_pcm_rb_seek_read(ma_pcm_rb* pRB, ma_uint32 offsetInFrames);
-MA_API ma_result ma_pcm_rb_seek_write(ma_pcm_rb* pRB, ma_uint32 offsetInFrames);
-MA_API ma_int32 ma_pcm_rb_pointer_distance(ma_pcm_rb* pRB); /* Return value is in frames. */
-MA_API ma_uint32 ma_pcm_rb_available_read(ma_pcm_rb* pRB);
-MA_API ma_uint32 ma_pcm_rb_available_write(ma_pcm_rb* pRB);
-MA_API ma_uint32 ma_pcm_rb_get_subbuffer_size(ma_pcm_rb* pRB);
-MA_API ma_uint32 ma_pcm_rb_get_subbuffer_stride(ma_pcm_rb* pRB);
-MA_API ma_uint32 ma_pcm_rb_get_subbuffer_offset(ma_pcm_rb* pRB, ma_uint32 subbufferIndex);
-MA_API void* ma_pcm_rb_get_subbuffer_ptr(ma_pcm_rb* pRB, ma_uint32 subbufferIndex, void* pBuffer);
+ ma_audio_buffer_uninit_and_free(&buffer);
+ ```
+If you initialize the buffer with `ma_audio_buffer_alloc_and_init()` you should uninitialize it
+with `ma_audio_buffer_uninit_and_free()`. In the example above, the memory pointed to by
+`pExistingData` will be copied into the buffer, which is contrary to the behavior of
+`ma_audio_buffer_init()`.
-/*
-The idea of the duplex ring buffer is to act as the intermediary buffer when running two asynchronous devices in a duplex set up. The
-capture device writes to it, and then a playback device reads from it.
+An audio buffer has a playback cursor just like a decoder. As you read frames from the buffer, the
+cursor moves forward. The last parameter (`loop`) can be used to determine if the buffer should
+loop. The return value is the number of frames actually read. If this is less than the number of
+frames requested it means the end has been reached. This should never happen if the `loop`
+parameter is set to true. If you want to manually loop back to the start, you can do so with with
+`ma_audio_buffer_seek_to_pcm_frame(pAudioBuffer, 0)`. Below is an example for reading data from an
+audio buffer.
-At the moment this is just a simple naive implementation, but in the future I want to implement some dynamic resampling to seamlessly
-handle desyncs. Note that the API is work in progress and may change at any time in any version.
+ ```c
+ ma_uint64 framesRead = ma_audio_buffer_read_pcm_frames(pAudioBuffer, pFramesOut, desiredFrameCount, isLooping);
+ if (framesRead < desiredFrameCount) {
+ // If not looping, this means the end has been reached. This should never happen in looping mode with valid input.
+ }
+ ```
-The size of the buffer is based on the capture side since that's what'll be written to the buffer. It is based on the capture period size
-in frames. The internal sample rate of the capture device is also needed in order to calculate the size.
-*/
-typedef struct
-{
+Sometimes you may want to avoid the cost of data movement between the internal buffer and the
+output buffer. Instead you can use memory mapping to retrieve a pointer to a segment of data:
+
+ ```c
+ void* pMappedFrames;
+ ma_uint64 frameCount = frameCountToTryMapping;
+ ma_result result = ma_audio_buffer_map(pAudioBuffer, &pMappedFrames, &frameCount);
+ if (result == MA_SUCCESS) {
+ // Map was successful. The value in frameCount will be how many frames were _actually_ mapped, which may be
+ // less due to the end of the buffer being reached.
+ ma_copy_pcm_frames(pFramesOut, pMappedFrames, frameCount, pAudioBuffer->format, pAudioBuffer->channels);
+
+ // You must unmap the buffer.
+ ma_audio_buffer_unmap(pAudioBuffer, frameCount);
+ }
+ ```
+
+When you use memory mapping, the read cursor is increment by the frame count passed in to
+`ma_audio_buffer_unmap()`. If you decide not to process every frame you can pass in a value smaller
+than the value returned by `ma_audio_buffer_map()`. The disadvantage to using memory mapping is
+that it does not handle looping for you. You can determine if the buffer is at the end for the
+purpose of looping with `ma_audio_buffer_at_end()` or by inspecting the return value of
+`ma_audio_buffer_unmap()` and checking if it equals `MA_AT_END`. You should not treat `MA_AT_END`
+as an error when returned by `ma_audio_buffer_unmap()`.
+
+
+
+14. Ring Buffers
+================
+miniaudio supports lock free (single producer, single consumer) ring buffers which are exposed via
+the `ma_rb` and `ma_pcm_rb` APIs. The `ma_rb` API operates on bytes, whereas the `ma_pcm_rb`
+operates on PCM frames. They are otherwise identical as `ma_pcm_rb` is just a wrapper around
+`ma_rb`.
+
+Unlike most other APIs in miniaudio, ring buffers support both interleaved and deinterleaved
+streams. The caller can also allocate their own backing memory for the ring buffer to use
+internally for added flexibility. Otherwise the ring buffer will manage it's internal memory for
+you.
+
+The examples below use the PCM frame variant of the ring buffer since that's most likely the one
+you will want to use. To initialize a ring buffer, do something like the following:
+
+ ```c
ma_pcm_rb rb;
-} ma_duplex_rb;
+ ma_result result = ma_pcm_rb_init(FORMAT, CHANNELS, BUFFER_SIZE_IN_FRAMES, NULL, NULL, &rb);
+ if (result != MA_SUCCESS) {
+ // Error
+ }
+ ```
-MA_API ma_result ma_duplex_rb_init(ma_format captureFormat, ma_uint32 captureChannels, ma_uint32 sampleRate, ma_uint32 captureInternalSampleRate, ma_uint32 captureInternalPeriodSizeInFrames, const ma_allocation_callbacks* pAllocationCallbacks, ma_duplex_rb* pRB);
-MA_API ma_result ma_duplex_rb_uninit(ma_duplex_rb* pRB);
+The `ma_pcm_rb_init()` function takes the sample format and channel count as parameters because
+it's the PCM varient of the ring buffer API. For the regular ring buffer that operates on bytes you
+would call `ma_rb_init()` which leaves these out and just takes the size of the buffer in bytes
+instead of frames. The fourth parameter is an optional pre-allocated buffer and the fifth parameter
+is a pointer to a `ma_allocation_callbacks` structure for custom memory allocation routines.
+Passing in `NULL` for this results in `MA_MALLOC()` and `MA_FREE()` being used.
+
+Use `ma_pcm_rb_init_ex()` if you need a deinterleaved buffer. The data for each sub-buffer is
+offset from each other based on the stride. To manage your sub-buffers you can use
+`ma_pcm_rb_get_subbuffer_stride()`, `ma_pcm_rb_get_subbuffer_offset()` and
+`ma_pcm_rb_get_subbuffer_ptr()`.
+
+Use `ma_pcm_rb_acquire_read()` and `ma_pcm_rb_acquire_write()` to retrieve a pointer to a section
+of the ring buffer. You specify the number of frames you need, and on output it will set to what
+was actually acquired. If the read or write pointer is positioned such that the number of frames
+requested will require a loop, it will be clamped to the end of the buffer. Therefore, the number
+of frames you're given may be less than the number you requested.
+
+After calling `ma_pcm_rb_acquire_read()` or `ma_pcm_rb_acquire_write()`, you do your work on the
+buffer and then "commit" it with `ma_pcm_rb_commit_read()` or `ma_pcm_rb_commit_write()`. This is
+where the read/write pointers are updated. When you commit you need to pass in the buffer that was
+returned by the earlier call to `ma_pcm_rb_acquire_read()` or `ma_pcm_rb_acquire_write()` and is
+only used for validation. The number of frames passed to `ma_pcm_rb_commit_read()` and
+`ma_pcm_rb_commit_write()` is what's used to increment the pointers, and can be less that what was
+originally requested.
+
+If you want to correct for drift between the write pointer and the read pointer you can use a
+combination of `ma_pcm_rb_pointer_distance()`, `ma_pcm_rb_seek_read()` and
+`ma_pcm_rb_seek_write()`. Note that you can only move the pointers forward, and you should only
+move the read pointer forward via the consumer thread, and the write pointer forward by the
+producer thread. If there is too much space between the pointers, move the read pointer forward. If
+there is too little space between the pointers, move the write pointer forward.
+You can use a ring buffer at the byte level instead of the PCM frame level by using the `ma_rb`
+API. This is exactly the same, only you will use the `ma_rb` functions instead of `ma_pcm_rb` and
+instead of frame counts you will pass around byte counts.
-/************************************************************************************************************************************************************
+The maximum size of the buffer in bytes is `0x7FFFFFFF-(MA_SIMD_ALIGNMENT-1)` due to the most
+significant bit being used to encode a loop flag and the internally managed buffers always being
+aligned to `MA_SIMD_ALIGNMENT`.
-Miscellaneous Helpers
+Note that the ring buffer is only thread safe when used by a single consumer thread and single
+producer thread.
-************************************************************************************************************************************************************/
-/*
-Retrieves a human readable description of the given result code.
-*/
-MA_API const char* ma_result_description(ma_result result);
-/*
-malloc(). Calls MA_MALLOC().
-*/
-MA_API void* ma_malloc(size_t sz, const ma_allocation_callbacks* pAllocationCallbacks);
-/*
-realloc(). Calls MA_REALLOC().
-*/
-MA_API void* ma_realloc(void* p, size_t sz, const ma_allocation_callbacks* pAllocationCallbacks);
+15. Backends
+============
+The following backends are supported by miniaudio.
-/*
-free(). Calls MA_FREE().
-*/
-MA_API void ma_free(void* p, const ma_allocation_callbacks* pAllocationCallbacks);
+ +-------------+-----------------------+--------------------------------------------------------+
+ | Name | Enum Name | Supported Operating Systems |
+ +-------------+-----------------------+--------------------------------------------------------+
+ | WASAPI | ma_backend_wasapi | Windows Vista+ |
+ | DirectSound | ma_backend_dsound | Windows XP+ |
+ | WinMM | ma_backend_winmm | Windows XP+ (may work on older versions, but untested) |
+ | Core Audio | ma_backend_coreaudio | macOS, iOS |
+ | ALSA | ma_backend_alsa | Linux |
+ | PulseAudio | ma_backend_pulseaudio | Cross Platform (disabled on Windows, BSD and Android) |
+ | JACK | ma_backend_jack | Cross Platform (disabled on BSD and Android) |
+ | sndio | ma_backend_sndio | OpenBSD |
+ | audio(4) | ma_backend_audio4 | NetBSD, OpenBSD |
+ | OSS | ma_backend_oss | FreeBSD |
+ | AAudio | ma_backend_aaudio | Android 8+ |
+ | OpenSL ES | ma_backend_opensl | Android (API level 16+) |
+ | Web Audio | ma_backend_webaudio | Web (via Emscripten) |
+ | Custom | ma_backend_custom | Cross Platform |
+ | Null | ma_backend_null | Cross Platform (not used on Web) |
+ +-------------+-----------------------+--------------------------------------------------------+
-/*
-Performs an aligned malloc, with the assumption that the alignment is a power of 2.
-*/
-MA_API void* ma_aligned_malloc(size_t sz, size_t alignment, const ma_allocation_callbacks* pAllocationCallbacks);
+Some backends have some nuance details you may want to be aware of.
-/*
-Free's an aligned malloc'd buffer.
-*/
-MA_API void ma_aligned_free(void* p, const ma_allocation_callbacks* pAllocationCallbacks);
+15.1. WASAPI
+------------
+- Low-latency shared mode will be disabled when using an application-defined sample rate which is
+ different to the device's native sample rate. To work around this, set `wasapi.noAutoConvertSRC`
+ to true in the device config. This is due to IAudioClient3_InitializeSharedAudioStream() failing
+ when the `AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM` flag is specified. Setting wasapi.noAutoConvertSRC
+ will result in miniaudio's internal resampler being used instead which will in turn enable the
+ use of low-latency shared mode.
+
+15.2. PulseAudio
+----------------
+- If you experience bad glitching/noise on Arch Linux, consider this fix from the Arch wiki:
+ https://wiki.archlinux.org/index.php/PulseAudio/Troubleshooting#Glitches,_skips_or_crackling.
+ Alternatively, consider using a different backend such as ALSA.
-/*
-Retrieves a friendly name for a format.
-*/
-MA_API const char* ma_get_format_name(ma_format format);
+15.3. Android
+-------------
+- To capture audio on Android, remember to add the RECORD_AUDIO permission to your manifest:
+ ``
+- With OpenSL|ES, only a single ma_context can be active at any given time. This is due to a
+ limitation with OpenSL|ES.
+- With AAudio, only default devices are enumerated. This is due to AAudio not having an enumeration
+ API (devices are enumerated through Java). You can however perform your own device enumeration
+ through Java and then set the ID in the ma_device_id structure (ma_device_id.aaudio) and pass it
+ to ma_device_init().
+- The backend API will perform resampling where possible. The reason for this as opposed to using
+ miniaudio's built-in resampler is to take advantage of any potential device-specific
+ optimizations the driver may implement.
+
+15.4. UWP
+---------
+- UWP only supports default playback and capture devices.
+- UWP requires the Microphone capability to be enabled in the application's manifest (Package.appxmanifest):
-/*
-Blends two frames in floating point format.
-*/
-MA_API void ma_blend_f32(float* pOut, float* pInA, float* pInB, float factor, ma_uint32 channels);
+ ```
+
+ ...
+
+
+
+
+ ```
-/*
-Retrieves the size of a sample in bytes for the given format.
+15.5. Web Audio / Emscripten
+----------------------------
+- You cannot use `-std=c*` compiler flags, nor `-ansi`. This only applies to the Emscripten build.
+- The first time a context is initialized it will create a global object called "miniaudio" whose
+ primary purpose is to act as a factory for device objects.
+- Currently the Web Audio backend uses ScriptProcessorNode's, but this may need to change later as
+ they've been deprecated.
+- Google has implemented a policy in their browsers that prevent automatic media output without
+ first receiving some kind of user input. The following web page has additional details:
+ https://developers.google.com/web/updates/2017/09/autoplay-policy-changes. Starting the device
+ may fail if you try to start playback without first handling some kind of user input.
-This API is efficient and is implemented using a lookup table.
-Thread Safety: SAFE
- This API is pure.
-*/
-MA_API ma_uint32 ma_get_bytes_per_sample(ma_format format);
-static MA_INLINE ma_uint32 ma_get_bytes_per_frame(ma_format format, ma_uint32 channels) { return ma_get_bytes_per_sample(format) * channels; }
-/*
-Converts a log level to a string.
-*/
-MA_API const char* ma_log_level_to_string(ma_uint32 logLevel);
+16. Optimization Tips
+=====================
+16.1. High Level API
+--------------------
+- If a sound does not require doppler or pitch shifting, consider disabling pitching by
+ initializing the sound with the `MA_SOUND_FLAG_NO_PITCH` flag.
+- If a sound does not require spatialization, disable it by initialzing the sound with the
+ `MA_SOUND_FLAG_NO_SPATIALIZATION` flag. It can be renabled again post-initialization with
+ `ma_sound_set_spatialization_enabled()`.
-/************************************************************************************************************************************************************
-*************************************************************************************************************************************************************
-DEVICE I/O
-==========
+17. Miscellaneous Notes
+=======================
+- Automatic stream routing is enabled on a per-backend basis. Support is explicitly enabled for
+ WASAPI and Core Audio, however other backends such as PulseAudio may naturally support it, though
+ not all have been tested.
+- The contents of the output buffer passed into the data callback will always be pre-initialized to
+ silence unless the `noPreSilencedOutputBuffer` config variable in `ma_device_config` is set to
+ true, in which case it'll be undefined which will require you to write something to the entire
+ buffer.
+- By default miniaudio will automatically clip samples. This only applies when the playback sample
+ format is configured as `ma_format_f32`. If you are doing clipping yourself, you can disable this
+ overhead by setting `noClip` to true in the device config.
+- Note that GCC and Clang requires `-msse2`, `-mavx2`, etc. for SIMD optimizations.
+- The sndio backend is currently only enabled on OpenBSD builds.
+- The audio(4) backend is supported on OpenBSD, but you may need to disable sndiod before you can
+ use it.
+- When compiling with VC6 and earlier, decoding is restricted to files less than 2GB in size. This
+ is due to 64-bit file APIs not being available.
+*/
-This section contains the APIs for device playback and capture. Here is where you'll find ma_device_init(), etc.
+#ifndef miniaudio_h
+#define miniaudio_h
-*************************************************************************************************************************************************************
-************************************************************************************************************************************************************/
-#ifndef MA_NO_DEVICE_IO
-/* Some backends are only supported on certain platforms. */
-#if defined(MA_WIN32)
- #define MA_SUPPORT_WASAPI
- #if defined(MA_WIN32_DESKTOP) /* DirectSound and WinMM backends are only supported on desktops. */
- #define MA_SUPPORT_DSOUND
- #define MA_SUPPORT_WINMM
- #define MA_SUPPORT_JACK /* JACK is technically supported on Windows, but I don't know how many people use it in practice... */
- #endif
-#endif
-#if defined(MA_UNIX)
- #if defined(MA_LINUX)
- #if !defined(MA_ANDROID) /* ALSA is not supported on Android. */
- #define MA_SUPPORT_ALSA
- #endif
- #endif
- #if !defined(MA_BSD) && !defined(MA_ANDROID) && !defined(MA_EMSCRIPTEN)
- #define MA_SUPPORT_PULSEAUDIO
- #define MA_SUPPORT_JACK
- #endif
- #if defined(MA_ANDROID)
- #define MA_SUPPORT_AAUDIO
- #define MA_SUPPORT_OPENSL
- #endif
- #if defined(__OpenBSD__) /* <-- Change this to "#if defined(MA_BSD)" to enable sndio on all BSD flavors. */
- #define MA_SUPPORT_SNDIO /* sndio is only supported on OpenBSD for now. May be expanded later if there's demand. */
- #endif
- #if defined(__NetBSD__) || defined(__OpenBSD__)
- #define MA_SUPPORT_AUDIO4 /* Only support audio(4) on platforms with known support. */
- #endif
- #if defined(__FreeBSD__) || defined(__DragonFly__)
- #define MA_SUPPORT_OSS /* Only support OSS on specific platforms with known support. */
- #endif
-#endif
-#if defined(MA_APPLE)
- #define MA_SUPPORT_COREAUDIO
-#endif
-#if defined(MA_EMSCRIPTEN)
- #define MA_SUPPORT_WEBAUDIO
+#ifdef __cplusplus
+extern "C" {
#endif
-/* All platforms should support custom backends. */
-#define MA_SUPPORT_CUSTOM
+#define MA_STRINGIFY(x) #x
+#define MA_XSTRINGIFY(x) MA_STRINGIFY(x)
-/* Explicitly disable the Null backend for Emscripten because it uses a background thread which is not properly supported right now. */
-#if !defined(MA_EMSCRIPTEN)
-#define MA_SUPPORT_NULL
+#define MA_VERSION_MAJOR 0
+#define MA_VERSION_MINOR 11
+#define MA_VERSION_REVISION 8
+#define MA_VERSION_STRING MA_XSTRINGIFY(MA_VERSION_MAJOR) "." MA_XSTRINGIFY(MA_VERSION_MINOR) "." MA_XSTRINGIFY(MA_VERSION_REVISION)
+
+#if defined(_MSC_VER) && !defined(__clang__)
+ #pragma warning(push)
+ #pragma warning(disable:4201) /* nonstandard extension used: nameless struct/union */
+ #pragma warning(disable:4214) /* nonstandard extension used: bit field types other than int */
+ #pragma warning(disable:4324) /* structure was padded due to alignment specifier */
+#elif defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8)))
+ #pragma GCC diagnostic push
+ #pragma GCC diagnostic ignored "-Wpedantic" /* For ISO C99 doesn't support unnamed structs/unions [-Wpedantic] */
+ #if defined(__clang__)
+ #pragma GCC diagnostic ignored "-Wc11-extensions" /* anonymous unions are a C11 extension */
+ #endif
#endif
+
-#if defined(MA_SUPPORT_WASAPI) && !defined(MA_NO_WASAPI) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_WASAPI))
- #define MA_HAS_WASAPI
-#endif
-#if defined(MA_SUPPORT_DSOUND) && !defined(MA_NO_DSOUND) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_DSOUND))
- #define MA_HAS_DSOUND
-#endif
-#if defined(MA_SUPPORT_WINMM) && !defined(MA_NO_WINMM) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_WINMM))
- #define MA_HAS_WINMM
-#endif
-#if defined(MA_SUPPORT_ALSA) && !defined(MA_NO_ALSA) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_ALSA))
- #define MA_HAS_ALSA
-#endif
-#if defined(MA_SUPPORT_PULSEAUDIO) && !defined(MA_NO_PULSEAUDIO) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_PULSEAUDIO))
- #define MA_HAS_PULSEAUDIO
-#endif
-#if defined(MA_SUPPORT_JACK) && !defined(MA_NO_JACK) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_JACK))
- #define MA_HAS_JACK
-#endif
-#if defined(MA_SUPPORT_COREAUDIO) && !defined(MA_NO_COREAUDIO) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_COREAUDIO))
- #define MA_HAS_COREAUDIO
-#endif
-#if defined(MA_SUPPORT_SNDIO) && !defined(MA_NO_SNDIO) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_SNDIO))
- #define MA_HAS_SNDIO
-#endif
-#if defined(MA_SUPPORT_AUDIO4) && !defined(MA_NO_AUDIO4) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_AUDIO4))
- #define MA_HAS_AUDIO4
-#endif
-#if defined(MA_SUPPORT_OSS) && !defined(MA_NO_OSS) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_OSS))
- #define MA_HAS_OSS
-#endif
-#if defined(MA_SUPPORT_AAUDIO) && !defined(MA_NO_AAUDIO) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_AAUDIO))
- #define MA_HAS_AAUDIO
-#endif
-#if defined(MA_SUPPORT_OPENSL) && !defined(MA_NO_OPENSL) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_OPENSL))
- #define MA_HAS_OPENSL
-#endif
-#if defined(MA_SUPPORT_WEBAUDIO) && !defined(MA_NO_WEBAUDIO) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_WEBAUDIO))
- #define MA_HAS_WEBAUDIO
-#endif
-#if defined(MA_SUPPORT_CUSTOM) && !defined(MA_NO_CUSTOM) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_CUSTOM))
- #define MA_HAS_CUSTOM
-#endif
-#if defined(MA_SUPPORT_NULL) && !defined(MA_NO_NULL) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_NULL))
- #define MA_HAS_NULL
+#if defined(__LP64__) || defined(_WIN64) || (defined(__x86_64__) && !defined(__ILP32__)) || defined(_M_X64) || defined(__ia64) || defined(_M_IA64) || defined(__aarch64__) || defined(_M_ARM64) || defined(__powerpc64__)
+ #define MA_SIZEOF_PTR 8
+#else
+ #define MA_SIZEOF_PTR 4
#endif
-#define MA_STATE_UNINITIALIZED 0
-#define MA_STATE_STOPPED 1 /* The device's default state after initialization. */
-#define MA_STATE_STARTED 2 /* The device is started and is requesting and/or delivering audio data. */
-#define MA_STATE_STARTING 3 /* Transitioning from a stopped state to started. */
-#define MA_STATE_STOPPING 4 /* Transitioning from a started state to stopped. */
+#include /* For size_t. */
-#ifdef MA_SUPPORT_WASAPI
-/* We need a IMMNotificationClient object for WASAPI. */
-typedef struct
-{
- void* lpVtbl;
- ma_uint32 counter;
- ma_device* pDevice;
-} ma_IMMNotificationClient;
+/* Sized types. */
+#if defined(MA_USE_STDINT)
+ #include
+ typedef int8_t ma_int8;
+ typedef uint8_t ma_uint8;
+ typedef int16_t ma_int16;
+ typedef uint16_t ma_uint16;
+ typedef int32_t ma_int32;
+ typedef uint32_t ma_uint32;
+ typedef int64_t ma_int64;
+ typedef uint64_t ma_uint64;
+#else
+ typedef signed char ma_int8;
+ typedef unsigned char ma_uint8;
+ typedef signed short ma_int16;
+ typedef unsigned short ma_uint16;
+ typedef signed int ma_int32;
+ typedef unsigned int ma_uint32;
+ #if defined(_MSC_VER) && !defined(__clang__)
+ typedef signed __int64 ma_int64;
+ typedef unsigned __int64 ma_uint64;
+ #else
+ #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
+ #pragma GCC diagnostic push
+ #pragma GCC diagnostic ignored "-Wlong-long"
+ #if defined(__clang__)
+ #pragma GCC diagnostic ignored "-Wc++11-long-long"
+ #endif
+ #endif
+ typedef signed long long ma_int64;
+ typedef unsigned long long ma_uint64;
+ #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
+ #pragma GCC diagnostic pop
+ #endif
+ #endif
+#endif /* MA_USE_STDINT */
+
+#if MA_SIZEOF_PTR == 8
+ typedef ma_uint64 ma_uintptr;
+#else
+ typedef ma_uint32 ma_uintptr;
#endif
-/* Backend enums must be in priority order. */
-typedef enum
-{
- ma_backend_wasapi,
- ma_backend_dsound,
- ma_backend_winmm,
- ma_backend_coreaudio,
- ma_backend_sndio,
- ma_backend_audio4,
- ma_backend_oss,
- ma_backend_pulseaudio,
- ma_backend_alsa,
- ma_backend_jack,
- ma_backend_aaudio,
- ma_backend_opensl,
- ma_backend_webaudio,
- ma_backend_custom, /* <-- Custom backend, with callbacks defined by the context config. */
- ma_backend_null /* <-- Must always be the last item. Lowest priority, and used as the terminator for backend enumeration. */
-} ma_backend;
+typedef ma_uint8 ma_bool8;
+typedef ma_uint32 ma_bool32;
+#define MA_TRUE 1
+#define MA_FALSE 0
-#define MA_BACKEND_COUNT (ma_backend_null+1)
+typedef void* ma_handle;
+typedef void* ma_ptr;
+typedef void (* ma_proc)(void);
+#if defined(_MSC_VER) && !defined(_WCHAR_T_DEFINED)
+typedef ma_uint16 wchar_t;
+#endif
-/*
-The callback for processing audio data from the device.
+/* Define NULL for some compilers. */
+#ifndef NULL
+#define NULL 0
+#endif
-The data callback is fired by miniaudio whenever the device needs to have more data delivered to a playback device, or when a capture device has some data
-available. This is called as soon as the backend asks for more data which means it may be called with inconsistent frame counts. You cannot assume the
-callback will be fired with a consistent frame count.
+#if defined(SIZE_MAX)
+ #define MA_SIZE_MAX SIZE_MAX
+#else
+ #define MA_SIZE_MAX 0xFFFFFFFF /* When SIZE_MAX is not defined by the standard library just default to the maximum 32-bit unsigned integer. */
+#endif
-Parameters
-----------
-pDevice (in)
- A pointer to the relevant device.
+/* Platform/backend detection. */
+#ifdef _WIN32
+ #define MA_WIN32
+ #if defined(WINAPI_FAMILY) && ((defined(WINAPI_FAMILY_PC_APP) && WINAPI_FAMILY == WINAPI_FAMILY_PC_APP) || (defined(WINAPI_FAMILY_PHONE_APP) && WINAPI_FAMILY == WINAPI_FAMILY_PHONE_APP))
+ #define MA_WIN32_UWP
+ #elif defined(WINAPI_FAMILY) && (defined(WINAPI_FAMILY_GAMES) && WINAPI_FAMILY == WINAPI_FAMILY_GAMES)
+ #define MA_WIN32_GDK
+ #else
+ #define MA_WIN32_DESKTOP
+ #endif
+#else
+ #define MA_POSIX
-pOutput (out)
- A pointer to the output buffer that will receive audio data that will later be played back through the speakers. This will be non-null for a playback or
- full-duplex device and null for a capture and loopback device.
+ /*
+ Use the MA_NO_PTHREAD_IN_HEADER option at your own risk. This is intentionally undocumented.
+ You can use this to avoid including pthread.h in the header section. The downside is that it
+ results in some fixed sized structures being declared for the various types that are used in
+ miniaudio. The risk here is that these types might be too small for a given platform. This
+ risk is yours to take and no support will be offered if you enable this option.
+ */
+ #ifndef MA_NO_PTHREAD_IN_HEADER
+ #include /* Unfortunate #include, but needed for pthread_t, pthread_mutex_t and pthread_cond_t types. */
+ typedef pthread_t ma_pthread_t;
+ typedef pthread_mutex_t ma_pthread_mutex_t;
+ typedef pthread_cond_t ma_pthread_cond_t;
+ #else
+ typedef ma_uintptr ma_pthread_t;
+ typedef union ma_pthread_mutex_t { char __data[40]; ma_uint64 __alignment; } ma_pthread_mutex_t;
+ typedef union ma_pthread_cond_t { char __data[48]; ma_uint64 __alignment; } ma_pthread_cond_t;
+ #endif
-pInput (in)
- A pointer to the buffer containing input data from a recording device. This will be non-null for a capture, full-duplex or loopback device and null for a
- playback device.
+ #ifdef __unix__
+ #define MA_UNIX
+ #if defined(__DragonFly__) || defined(__FreeBSD__) || defined(__NetBSD__) || defined(__OpenBSD__)
+ #define MA_BSD
+ #endif
+ #endif
+ #ifdef __linux__
+ #define MA_LINUX
+ #endif
+ #ifdef __APPLE__
+ #define MA_APPLE
+ #endif
+ #ifdef __ANDROID__
+ #define MA_ANDROID
+ #endif
+ #ifdef __EMSCRIPTEN__
+ #define MA_EMSCRIPTEN
+ #endif
+#endif
-frameCount (in)
- The number of PCM frames to process. Note that this will not necessarily be equal to what you requested when you initialized the device. The
- `periodSizeInFrames` and `periodSizeInMilliseconds` members of the device config are just hints, and are not necessarily exactly what you'll get. You must
- not assume this will always be the same value each time the callback is fired.
+#ifdef _MSC_VER
+ #define MA_INLINE __forceinline
+#elif defined(__GNUC__)
+ /*
+ I've had a bug report where GCC is emitting warnings about functions possibly not being inlineable. This warning happens when
+ the __attribute__((always_inline)) attribute is defined without an "inline" statement. I think therefore there must be some
+ case where "__inline__" is not always defined, thus the compiler emitting these warnings. When using -std=c89 or -ansi on the
+ command line, we cannot use the "inline" keyword and instead need to use "__inline__". In an attempt to work around this issue
+ I am using "__inline__" only when we're compiling in strict ANSI mode.
+ */
+ #if defined(__STRICT_ANSI__)
+ #define MA_INLINE __inline__ __attribute__((always_inline))
+ #else
+ #define MA_INLINE inline __attribute__((always_inline))
+ #endif
+#elif defined(__WATCOMC__)
+ #define MA_INLINE __inline
+#else
+ #define MA_INLINE
+#endif
-Remarks
--------
-You cannot stop and start the device from inside the callback or else you'll get a deadlock. You must also not uninitialize the device from inside the
-callback. The following APIs cannot be called from inside the callback:
+#if !defined(MA_API)
+ #if defined(MA_DLL)
+ #if defined(_WIN32)
+ #define MA_DLL_IMPORT __declspec(dllimport)
+ #define MA_DLL_EXPORT __declspec(dllexport)
+ #define MA_DLL_PRIVATE static
+ #else
+ #if defined(__GNUC__) && __GNUC__ >= 4
+ #define MA_DLL_IMPORT __attribute__((visibility("default")))
+ #define MA_DLL_EXPORT __attribute__((visibility("default")))
+ #define MA_DLL_PRIVATE __attribute__((visibility("hidden")))
+ #else
+ #define MA_DLL_IMPORT
+ #define MA_DLL_EXPORT
+ #define MA_DLL_PRIVATE static
+ #endif
+ #endif
- ma_device_init()
- ma_device_init_ex()
- ma_device_uninit()
- ma_device_start()
- ma_device_stop()
+ #if defined(MINIAUDIO_IMPLEMENTATION) || defined(MA_IMPLEMENTATION)
+ #define MA_API MA_DLL_EXPORT
+ #else
+ #define MA_API MA_DLL_IMPORT
+ #endif
+ #define MA_PRIVATE MA_DLL_PRIVATE
+ #else
+ #define MA_API extern
+ #define MA_PRIVATE static
+ #endif
+#endif
-The proper way to stop the device is to call `ma_device_stop()` from a different thread, normally the main application thread.
-*/
-typedef void (* ma_device_callback_proc)(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount);
+/* SIMD alignment in bytes. Currently set to 32 bytes in preparation for future AVX optimizations. */
+#define MA_SIMD_ALIGNMENT 32
-/*
-The callback for when the device has been stopped.
-This will be called when the device is stopped explicitly with `ma_device_stop()` and also called implicitly when the device is stopped through external forces
-such as being unplugged or an internal error occuring.
+/*
+Logging Levels
+==============
+Log levels are only used to give logging callbacks some context as to the severity of a log message
+so they can do filtering. All log levels will be posted to registered logging callbacks. If you
+don't want to output a certain log level you can discriminate against the log level in the callback.
+MA_LOG_LEVEL_DEBUG
+ Used for debugging. Useful for debug and test builds, but should be disabled in release builds.
-Parameters
-----------
-pDevice (in)
- A pointer to the device that has just stopped.
+MA_LOG_LEVEL_INFO
+ Informational logging. Useful for debugging. This will never be called from within the data
+ callback.
+MA_LOG_LEVEL_WARNING
+ Warnings. You should enable this in you development builds and action them when encounted. These
+ logs usually indicate a potential problem or misconfiguration, but still allow you to keep
+ running. This will never be called from within the data callback.
-Remarks
--------
-Do not restart or uninitialize the device from the callback.
+MA_LOG_LEVEL_ERROR
+ Error logging. This will be fired when an operation fails and is subsequently aborted. This can
+ be fired from within the data callback, in which case the device will be stopped. You should
+ always have this log level enabled.
*/
-typedef void (* ma_stop_proc)(ma_device* pDevice);
+typedef enum
+{
+ MA_LOG_LEVEL_DEBUG = 4,
+ MA_LOG_LEVEL_INFO = 3,
+ MA_LOG_LEVEL_WARNING = 2,
+ MA_LOG_LEVEL_ERROR = 1
+} ma_log_level;
/*
-The callback for handling log messages.
+Variables needing to be accessed atomically should be declared with this macro for two reasons:
+ 1) It allows people who read the code to identify a variable as such; and
+ 2) It forces alignment on platforms where it's required or optimal.
-Parameters
-----------
-pContext (in)
- A pointer to the context the log message originated from.
-
-pDevice (in)
- A pointer to the device the log message originate from, if any. This can be null, in which case the message came from the context.
-
-logLevel (in)
- The log level. This can be one of the following:
-
- +----------------------+
- | Log Level |
- +----------------------+
- | MA_LOG_LEVEL_DEBUG |
- | MA_LOG_LEVEL_INFO |
- | MA_LOG_LEVEL_WARNING |
- | MA_LOG_LEVEL_ERROR |
- +----------------------+
-
-message (in)
- The log message.
+Note that for x86/64, alignment is not strictly necessary, but does have some performance
+implications. Where supported by the compiler, alignment will be used, but otherwise if the CPU
+architecture does not require it, it will simply leave it unaligned. This is the case with old
+versions of Visual Studio, which I've confirmed with at least VC6.
+*/
+#if defined (__STDC_VERSION__) && (__STDC_VERSION__ >= 201112L)
+ #include
+ #define MA_ATOMIC(alignment, type) alignas(alignment) type
+#else
+ #if defined(__GNUC__)
+ /* GCC-style compilers. */
+ #define MA_ATOMIC(alignment, type) type __attribute__((aligned(alignment)))
+ #elif defined(_MSC_VER) && _MSC_VER > 1200 /* 1200 = VC6. Alignment not supported, but not necessary because x86 is the only supported target. */
+ /* MSVC. */
+ #define MA_ATOMIC(alignment, type) __declspec(align(alignment)) type
+ #else
+ /* Other compilers. */
+ #define MA_ATOMIC(alignment, type) type
+ #endif
+#endif
+typedef struct ma_context ma_context;
+typedef struct ma_device ma_device;
-Remarks
--------
-Do not modify the state of the device from inside the callback.
-*/
-typedef void (* ma_log_proc)(ma_context* pContext, ma_device* pDevice, ma_uint32 logLevel, const char* message);
+typedef ma_uint8 ma_channel;
+typedef enum
+{
+ MA_CHANNEL_NONE = 0,
+ MA_CHANNEL_MONO = 1,
+ MA_CHANNEL_FRONT_LEFT = 2,
+ MA_CHANNEL_FRONT_RIGHT = 3,
+ MA_CHANNEL_FRONT_CENTER = 4,
+ MA_CHANNEL_LFE = 5,
+ MA_CHANNEL_BACK_LEFT = 6,
+ MA_CHANNEL_BACK_RIGHT = 7,
+ MA_CHANNEL_FRONT_LEFT_CENTER = 8,
+ MA_CHANNEL_FRONT_RIGHT_CENTER = 9,
+ MA_CHANNEL_BACK_CENTER = 10,
+ MA_CHANNEL_SIDE_LEFT = 11,
+ MA_CHANNEL_SIDE_RIGHT = 12,
+ MA_CHANNEL_TOP_CENTER = 13,
+ MA_CHANNEL_TOP_FRONT_LEFT = 14,
+ MA_CHANNEL_TOP_FRONT_CENTER = 15,
+ MA_CHANNEL_TOP_FRONT_RIGHT = 16,
+ MA_CHANNEL_TOP_BACK_LEFT = 17,
+ MA_CHANNEL_TOP_BACK_CENTER = 18,
+ MA_CHANNEL_TOP_BACK_RIGHT = 19,
+ MA_CHANNEL_AUX_0 = 20,
+ MA_CHANNEL_AUX_1 = 21,
+ MA_CHANNEL_AUX_2 = 22,
+ MA_CHANNEL_AUX_3 = 23,
+ MA_CHANNEL_AUX_4 = 24,
+ MA_CHANNEL_AUX_5 = 25,
+ MA_CHANNEL_AUX_6 = 26,
+ MA_CHANNEL_AUX_7 = 27,
+ MA_CHANNEL_AUX_8 = 28,
+ MA_CHANNEL_AUX_9 = 29,
+ MA_CHANNEL_AUX_10 = 30,
+ MA_CHANNEL_AUX_11 = 31,
+ MA_CHANNEL_AUX_12 = 32,
+ MA_CHANNEL_AUX_13 = 33,
+ MA_CHANNEL_AUX_14 = 34,
+ MA_CHANNEL_AUX_15 = 35,
+ MA_CHANNEL_AUX_16 = 36,
+ MA_CHANNEL_AUX_17 = 37,
+ MA_CHANNEL_AUX_18 = 38,
+ MA_CHANNEL_AUX_19 = 39,
+ MA_CHANNEL_AUX_20 = 40,
+ MA_CHANNEL_AUX_21 = 41,
+ MA_CHANNEL_AUX_22 = 42,
+ MA_CHANNEL_AUX_23 = 43,
+ MA_CHANNEL_AUX_24 = 44,
+ MA_CHANNEL_AUX_25 = 45,
+ MA_CHANNEL_AUX_26 = 46,
+ MA_CHANNEL_AUX_27 = 47,
+ MA_CHANNEL_AUX_28 = 48,
+ MA_CHANNEL_AUX_29 = 49,
+ MA_CHANNEL_AUX_30 = 50,
+ MA_CHANNEL_AUX_31 = 51,
+ MA_CHANNEL_LEFT = MA_CHANNEL_FRONT_LEFT,
+ MA_CHANNEL_RIGHT = MA_CHANNEL_FRONT_RIGHT,
+ MA_CHANNEL_POSITION_COUNT = (MA_CHANNEL_AUX_31 + 1)
+} _ma_channel_position; /* Do not use `_ma_channel_position` directly. Use `ma_channel` instead. */
typedef enum
{
- ma_device_type_playback = 1,
- ma_device_type_capture = 2,
- ma_device_type_duplex = ma_device_type_playback | ma_device_type_capture, /* 3 */
- ma_device_type_loopback = 4
-} ma_device_type;
+ MA_SUCCESS = 0,
+ MA_ERROR = -1, /* A generic error. */
+ MA_INVALID_ARGS = -2,
+ MA_INVALID_OPERATION = -3,
+ MA_OUT_OF_MEMORY = -4,
+ MA_OUT_OF_RANGE = -5,
+ MA_ACCESS_DENIED = -6,
+ MA_DOES_NOT_EXIST = -7,
+ MA_ALREADY_EXISTS = -8,
+ MA_TOO_MANY_OPEN_FILES = -9,
+ MA_INVALID_FILE = -10,
+ MA_TOO_BIG = -11,
+ MA_PATH_TOO_LONG = -12,
+ MA_NAME_TOO_LONG = -13,
+ MA_NOT_DIRECTORY = -14,
+ MA_IS_DIRECTORY = -15,
+ MA_DIRECTORY_NOT_EMPTY = -16,
+ MA_AT_END = -17,
+ MA_NO_SPACE = -18,
+ MA_BUSY = -19,
+ MA_IO_ERROR = -20,
+ MA_INTERRUPT = -21,
+ MA_UNAVAILABLE = -22,
+ MA_ALREADY_IN_USE = -23,
+ MA_BAD_ADDRESS = -24,
+ MA_BAD_SEEK = -25,
+ MA_BAD_PIPE = -26,
+ MA_DEADLOCK = -27,
+ MA_TOO_MANY_LINKS = -28,
+ MA_NOT_IMPLEMENTED = -29,
+ MA_NO_MESSAGE = -30,
+ MA_BAD_MESSAGE = -31,
+ MA_NO_DATA_AVAILABLE = -32,
+ MA_INVALID_DATA = -33,
+ MA_TIMEOUT = -34,
+ MA_NO_NETWORK = -35,
+ MA_NOT_UNIQUE = -36,
+ MA_NOT_SOCKET = -37,
+ MA_NO_ADDRESS = -38,
+ MA_BAD_PROTOCOL = -39,
+ MA_PROTOCOL_UNAVAILABLE = -40,
+ MA_PROTOCOL_NOT_SUPPORTED = -41,
+ MA_PROTOCOL_FAMILY_NOT_SUPPORTED = -42,
+ MA_ADDRESS_FAMILY_NOT_SUPPORTED = -43,
+ MA_SOCKET_NOT_SUPPORTED = -44,
+ MA_CONNECTION_RESET = -45,
+ MA_ALREADY_CONNECTED = -46,
+ MA_NOT_CONNECTED = -47,
+ MA_CONNECTION_REFUSED = -48,
+ MA_NO_HOST = -49,
+ MA_IN_PROGRESS = -50,
+ MA_CANCELLED = -51,
+ MA_MEMORY_ALREADY_MAPPED = -52,
+
+ /* General miniaudio-specific errors. */
+ MA_FORMAT_NOT_SUPPORTED = -100,
+ MA_DEVICE_TYPE_NOT_SUPPORTED = -101,
+ MA_SHARE_MODE_NOT_SUPPORTED = -102,
+ MA_NO_BACKEND = -103,
+ MA_NO_DEVICE = -104,
+ MA_API_NOT_FOUND = -105,
+ MA_INVALID_DEVICE_CONFIG = -106,
+ MA_LOOP = -107,
+
+ /* State errors. */
+ MA_DEVICE_NOT_INITIALIZED = -200,
+ MA_DEVICE_ALREADY_INITIALIZED = -201,
+ MA_DEVICE_NOT_STARTED = -202,
+ MA_DEVICE_NOT_STOPPED = -203,
+
+ /* Operation errors. */
+ MA_FAILED_TO_INIT_BACKEND = -300,
+ MA_FAILED_TO_OPEN_BACKEND_DEVICE = -301,
+ MA_FAILED_TO_START_BACKEND_DEVICE = -302,
+ MA_FAILED_TO_STOP_BACKEND_DEVICE = -303
+} ma_result;
+
+
+#define MA_MIN_CHANNELS 1
+#ifndef MA_MAX_CHANNELS
+#define MA_MAX_CHANNELS 254
+#endif
+
+#ifndef MA_MAX_FILTER_ORDER
+#define MA_MAX_FILTER_ORDER 8
+#endif
typedef enum
{
- ma_share_mode_shared = 0,
- ma_share_mode_exclusive
-} ma_share_mode;
+ ma_stream_format_pcm = 0
+} ma_stream_format;
-/* iOS/tvOS/watchOS session categories. */
typedef enum
{
- ma_ios_session_category_default = 0, /* AVAudioSessionCategoryPlayAndRecord with AVAudioSessionCategoryOptionDefaultToSpeaker. */
- ma_ios_session_category_none, /* Leave the session category unchanged. */
- ma_ios_session_category_ambient, /* AVAudioSessionCategoryAmbient */
- ma_ios_session_category_solo_ambient, /* AVAudioSessionCategorySoloAmbient */
- ma_ios_session_category_playback, /* AVAudioSessionCategoryPlayback */
- ma_ios_session_category_record, /* AVAudioSessionCategoryRecord */
- ma_ios_session_category_play_and_record, /* AVAudioSessionCategoryPlayAndRecord */
- ma_ios_session_category_multi_route /* AVAudioSessionCategoryMultiRoute */
-} ma_ios_session_category;
+ ma_stream_layout_interleaved = 0,
+ ma_stream_layout_deinterleaved
+} ma_stream_layout;
-/* iOS/tvOS/watchOS session category options */
typedef enum
{
- ma_ios_session_category_option_mix_with_others = 0x01, /* AVAudioSessionCategoryOptionMixWithOthers */
- ma_ios_session_category_option_duck_others = 0x02, /* AVAudioSessionCategoryOptionDuckOthers */
- ma_ios_session_category_option_allow_bluetooth = 0x04, /* AVAudioSessionCategoryOptionAllowBluetooth */
- ma_ios_session_category_option_default_to_speaker = 0x08, /* AVAudioSessionCategoryOptionDefaultToSpeaker */
- ma_ios_session_category_option_interrupt_spoken_audio_and_mix_with_others = 0x11, /* AVAudioSessionCategoryOptionInterruptSpokenAudioAndMixWithOthers */
- ma_ios_session_category_option_allow_bluetooth_a2dp = 0x20, /* AVAudioSessionCategoryOptionAllowBluetoothA2DP */
- ma_ios_session_category_option_allow_air_play = 0x40, /* AVAudioSessionCategoryOptionAllowAirPlay */
-} ma_ios_session_category_option;
+ ma_dither_mode_none = 0,
+ ma_dither_mode_rectangle,
+ ma_dither_mode_triangle
+} ma_dither_mode;
-/* OpenSL stream types. */
typedef enum
{
- ma_opensl_stream_type_default = 0, /* Leaves the stream type unset. */
- ma_opensl_stream_type_voice, /* SL_ANDROID_STREAM_VOICE */
- ma_opensl_stream_type_system, /* SL_ANDROID_STREAM_SYSTEM */
- ma_opensl_stream_type_ring, /* SL_ANDROID_STREAM_RING */
- ma_opensl_stream_type_media, /* SL_ANDROID_STREAM_MEDIA */
- ma_opensl_stream_type_alarm, /* SL_ANDROID_STREAM_ALARM */
- ma_opensl_stream_type_notification /* SL_ANDROID_STREAM_NOTIFICATION */
-} ma_opensl_stream_type;
+ /*
+ I like to keep these explicitly defined because they're used as a key into a lookup table. When items are
+ added to this, make sure there are no gaps and that they're added to the lookup table in ma_get_bytes_per_sample().
+ */
+ ma_format_unknown = 0, /* Mainly used for indicating an error, but also used as the default for the output format for decoders. */
+ ma_format_u8 = 1,
+ ma_format_s16 = 2, /* Seems to be the most widely supported format. */
+ ma_format_s24 = 3, /* Tightly packed. 3 bytes per sample. */
+ ma_format_s32 = 4,
+ ma_format_f32 = 5,
+ ma_format_count
+} ma_format;
-/* OpenSL recording presets. */
typedef enum
{
- ma_opensl_recording_preset_default = 0, /* Leaves the input preset unset. */
- ma_opensl_recording_preset_generic, /* SL_ANDROID_RECORDING_PRESET_GENERIC */
- ma_opensl_recording_preset_camcorder, /* SL_ANDROID_RECORDING_PRESET_CAMCORDER */
- ma_opensl_recording_preset_voice_recognition, /* SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION */
- ma_opensl_recording_preset_voice_communication, /* SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION */
- ma_opensl_recording_preset_voice_unprocessed /* SL_ANDROID_RECORDING_PRESET_UNPROCESSED */
-} ma_opensl_recording_preset;
+ /* Standard rates need to be in priority order. */
+ ma_standard_sample_rate_48000 = 48000, /* Most common */
+ ma_standard_sample_rate_44100 = 44100,
+
+ ma_standard_sample_rate_32000 = 32000, /* Lows */
+ ma_standard_sample_rate_24000 = 24000,
+ ma_standard_sample_rate_22050 = 22050,
+
+ ma_standard_sample_rate_88200 = 88200, /* Highs */
+ ma_standard_sample_rate_96000 = 96000,
+ ma_standard_sample_rate_176400 = 176400,
+ ma_standard_sample_rate_192000 = 192000,
+
+ ma_standard_sample_rate_16000 = 16000, /* Extreme lows */
+ ma_standard_sample_rate_11025 = 11250,
+ ma_standard_sample_rate_8000 = 8000,
+
+ ma_standard_sample_rate_352800 = 352800, /* Extreme highs */
+ ma_standard_sample_rate_384000 = 384000,
+
+ ma_standard_sample_rate_min = ma_standard_sample_rate_8000,
+ ma_standard_sample_rate_max = ma_standard_sample_rate_384000,
+ ma_standard_sample_rate_count = 14 /* Need to maintain the count manually. Make sure this is updated if items are added to enum. */
+} ma_standard_sample_rate;
+
-/* AAudio usage types. */
typedef enum
{
- ma_aaudio_usage_default = 0, /* Leaves the usage type unset. */
- ma_aaudio_usage_announcement, /* AAUDIO_SYSTEM_USAGE_ANNOUNCEMENT */
- ma_aaudio_usage_emergency, /* AAUDIO_SYSTEM_USAGE_EMERGENCY */
- ma_aaudio_usage_safety, /* AAUDIO_SYSTEM_USAGE_SAFETY */
- ma_aaudio_usage_vehicle_status, /* AAUDIO_SYSTEM_USAGE_VEHICLE_STATUS */
- ma_aaudio_usage_alarm, /* AAUDIO_USAGE_ALARM */
- ma_aaudio_usage_assistance_accessibility, /* AAUDIO_USAGE_ASSISTANCE_ACCESSIBILITY */
- ma_aaudio_usage_assistance_navigation_guidance, /* AAUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE */
- ma_aaudio_usage_assistance_sonification, /* AAUDIO_USAGE_ASSISTANCE_SONIFICATION */
- ma_aaudio_usage_assitant, /* AAUDIO_USAGE_ASSISTANT */
- ma_aaudio_usage_game, /* AAUDIO_USAGE_GAME */
- ma_aaudio_usage_media, /* AAUDIO_USAGE_MEDIA */
- ma_aaudio_usage_notification, /* AAUDIO_USAGE_NOTIFICATION */
- ma_aaudio_usage_notification_event, /* AAUDIO_USAGE_NOTIFICATION_EVENT */
- ma_aaudio_usage_notification_ringtone, /* AAUDIO_USAGE_NOTIFICATION_RINGTONE */
- ma_aaudio_usage_voice_communication, /* AAUDIO_USAGE_VOICE_COMMUNICATION */
- ma_aaudio_usage_voice_communication_signalling /* AAUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING */
-} ma_aaudio_usage;
+ ma_channel_mix_mode_rectangular = 0, /* Simple averaging based on the plane(s) the channel is sitting on. */
+ ma_channel_mix_mode_simple, /* Drop excess channels; zeroed out extra channels. */
+ ma_channel_mix_mode_custom_weights, /* Use custom weights specified in ma_channel_router_config. */
+ ma_channel_mix_mode_default = ma_channel_mix_mode_rectangular
+} ma_channel_mix_mode;
-/* AAudio content types. */
typedef enum
{
- ma_aaudio_content_type_default = 0, /* Leaves the content type unset. */
- ma_aaudio_content_type_movie, /* AAUDIO_CONTENT_TYPE_MOVIE */
- ma_aaudio_content_type_music, /* AAUDIO_CONTENT_TYPE_MUSIC */
- ma_aaudio_content_type_sonification, /* AAUDIO_CONTENT_TYPE_SONIFICATION */
- ma_aaudio_content_type_speech /* AAUDIO_CONTENT_TYPE_SPEECH */
-} ma_aaudio_content_type;
+ ma_standard_channel_map_microsoft,
+ ma_standard_channel_map_alsa,
+ ma_standard_channel_map_rfc3551, /* Based off AIFF. */
+ ma_standard_channel_map_flac,
+ ma_standard_channel_map_vorbis,
+ ma_standard_channel_map_sound4, /* FreeBSD's sound(4). */
+ ma_standard_channel_map_sndio, /* www.sndio.org/tips.html */
+ ma_standard_channel_map_webaudio = ma_standard_channel_map_flac, /* https://webaudio.github.io/web-audio-api/#ChannelOrdering. Only 1, 2, 4 and 6 channels are defined, but can fill in the gaps with logical assumptions. */
+ ma_standard_channel_map_default = ma_standard_channel_map_microsoft
+} ma_standard_channel_map;
-/* AAudio input presets. */
typedef enum
{
- ma_aaudio_input_preset_default = 0, /* Leaves the input preset unset. */
- ma_aaudio_input_preset_generic, /* AAUDIO_INPUT_PRESET_GENERIC */
- ma_aaudio_input_preset_camcorder, /* AAUDIO_INPUT_PRESET_CAMCORDER */
- ma_aaudio_input_preset_unprocessed, /* AAUDIO_INPUT_PRESET_UNPROCESSED */
- ma_aaudio_input_preset_voice_recognition, /* AAUDIO_INPUT_PRESET_VOICE_RECOGNITION */
- ma_aaudio_input_preset_voice_communication, /* AAUDIO_INPUT_PRESET_VOICE_COMMUNICATION */
- ma_aaudio_input_preset_voice_performance /* AAUDIO_INPUT_PRESET_VOICE_PERFORMANCE */
-} ma_aaudio_input_preset;
+ ma_performance_profile_low_latency = 0,
+ ma_performance_profile_conservative
+} ma_performance_profile;
-typedef union
+typedef struct
{
- ma_int64 counter;
- double counterD;
-} ma_timer;
+ void* pUserData;
+ void* (* onMalloc)(size_t sz, void* pUserData);
+ void* (* onRealloc)(void* p, size_t sz, void* pUserData);
+ void (* onFree)(void* p, void* pUserData);
+} ma_allocation_callbacks;
-typedef union
+typedef struct
{
- wchar_t wasapi[64]; /* WASAPI uses a wchar_t string for identification. */
- ma_uint8 dsound[16]; /* DirectSound uses a GUID for identification. */
- /*UINT_PTR*/ ma_uint32 winmm; /* When creating a device, WinMM expects a Win32 UINT_PTR for device identification. In practice it's actually just a UINT. */
- char alsa[256]; /* ALSA uses a name string for identification. */
- char pulse[256]; /* PulseAudio uses a name string for identification. */
- int jack; /* JACK always uses default devices. */
- char coreaudio[256]; /* Core Audio uses a string for identification. */
- char sndio[256]; /* "snd/0", etc. */
- char audio4[256]; /* "/dev/audio", etc. */
- char oss[64]; /* "dev/dsp0", etc. "dev/dsp" for the default device. */
- ma_int32 aaudio; /* AAudio uses a 32-bit integer for identification. */
- ma_uint32 opensl; /* OpenSL|ES uses a 32-bit unsigned integer for identification. */
- char webaudio[32]; /* Web Audio always uses default devices for now, but if this changes it'll be a GUID. */
- union
- {
- int i;
- char s[256];
- void* p;
- } custom; /* The custom backend could be anything. Give them a few options. */
- int nullbackend; /* The null backend uses an integer for device IDs. */
-} ma_device_id;
+ ma_int32 state;
+} ma_lcg;
-typedef struct ma_context_config ma_context_config;
-typedef struct ma_device_config ma_device_config;
-typedef struct ma_backend_callbacks ma_backend_callbacks;
+/* Spinlocks are 32-bit for compatibility reasons. */
+typedef ma_uint32 ma_spinlock;
-#define MA_DATA_FORMAT_FLAG_EXCLUSIVE_MODE (1U << 1) /* If set, this is supported in exclusive mode. Otherwise not natively supported by exclusive mode. */
+#ifndef MA_NO_THREADING
+/* Thread priorities should be ordered such that the default priority of the worker thread is 0. */
+typedef enum
+{
+ ma_thread_priority_idle = -5,
+ ma_thread_priority_lowest = -4,
+ ma_thread_priority_low = -3,
+ ma_thread_priority_normal = -2,
+ ma_thread_priority_high = -1,
+ ma_thread_priority_highest = 0,
+ ma_thread_priority_realtime = 1,
+ ma_thread_priority_default = 0
+} ma_thread_priority;
+
+#if defined(MA_WIN32)
+typedef ma_handle ma_thread;
+#endif
+#if defined(MA_POSIX)
+typedef ma_pthread_t ma_thread;
+#endif
+
+#if defined(MA_WIN32)
+typedef ma_handle ma_mutex;
+#endif
+#if defined(MA_POSIX)
+typedef ma_pthread_mutex_t ma_mutex;
+#endif
+#if defined(MA_WIN32)
+typedef ma_handle ma_event;
+#endif
+#if defined(MA_POSIX)
typedef struct
{
- /* Basic info. This is the only information guaranteed to be filled in during device enumeration. */
- ma_device_id id;
- char name[256];
- ma_bool32 isDefault;
+ ma_uint32 value;
+ ma_pthread_mutex_t lock;
+ ma_pthread_cond_t cond;
+} ma_event;
+#endif /* MA_POSIX */
- /*
- Detailed info. As much of this is filled as possible with ma_context_get_device_info(). Note that you are allowed to initialize
- a device with settings outside of this range, but it just means the data will be converted using miniaudio's data conversion
- pipeline before sending the data to/from the device. Most programs will need to not worry about these values, but it's provided
- here mainly for informational purposes or in the rare case that someone might find it useful.
+#if defined(MA_WIN32)
+typedef ma_handle ma_semaphore;
+#endif
+#if defined(MA_POSIX)
+typedef struct
+{
+ int value;
+ ma_pthread_mutex_t lock;
+ ma_pthread_cond_t cond;
+} ma_semaphore;
+#endif /* MA_POSIX */
+#else
+/* MA_NO_THREADING is set which means threading is disabled. Threading is required by some API families. If any of these are enabled we need to throw an error. */
+#ifndef MA_NO_DEVICE_IO
+#error "MA_NO_THREADING cannot be used without MA_NO_DEVICE_IO";
+#endif
+#endif /* MA_NO_THREADING */
- These will be set to 0 when returned by ma_context_enumerate_devices() or ma_context_get_devices().
- */
- ma_uint32 formatCount;
- ma_format formats[ma_format_count];
- ma_uint32 minChannels;
- ma_uint32 maxChannels;
- ma_uint32 minSampleRate;
- ma_uint32 maxSampleRate;
+/*
+Retrieves the version of miniaudio as separated integers. Each component can be NULL if it's not required.
+*/
+MA_API void ma_version(ma_uint32* pMajor, ma_uint32* pMinor, ma_uint32* pRevision);
- /* Experimental. Don't use these right now. */
- ma_uint32 nativeDataFormatCount;
- struct
- {
- ma_format format; /* Sample format. If set to ma_format_unknown, all sample formats are supported. */
- ma_uint32 channels; /* If set to 0, all channels are supported. */
- ma_uint32 sampleRate; /* If set to 0, all sample rates are supported. */
- ma_uint32 flags; /* A combination of MA_DATA_FORMAT_FLAG_* flags. */
- } nativeDataFormats[/*ma_format_count * ma_standard_sample_rate_count * MA_MAX_CHANNELS*/ 64]; /* Not sure how big to make this. There can be *many* permutations for virtual devices which can support anything. */
-} ma_device_info;
+/*
+Retrieves the version of miniaudio as a string which can be useful for logging purposes.
+*/
+MA_API const char* ma_version_string(void);
-struct ma_device_config
-{
- ma_device_type deviceType;
- ma_uint32 sampleRate;
- ma_uint32 periodSizeInFrames;
- ma_uint32 periodSizeInMilliseconds;
- ma_uint32 periods;
- ma_performance_profile performanceProfile;
- ma_bool8 noPreZeroedOutputBuffer; /* When set to true, the contents of the output buffer passed into the data callback will be left undefined rather than initialized to zero. */
- ma_bool8 noClip; /* When set to true, the contents of the output buffer passed into the data callback will be clipped after returning. Only applies when the playback sample format is f32. */
- ma_device_callback_proc dataCallback;
- ma_stop_proc stopCallback;
- void* pUserData;
- struct
- {
- ma_resample_algorithm algorithm;
- struct
- {
- ma_uint32 lpfOrder;
- } linear;
- struct
- {
- int quality;
- } speex;
- } resampling;
- struct
- {
- const ma_device_id* pDeviceID;
- ma_format format;
- ma_uint32 channels;
- ma_channel channelMap[MA_MAX_CHANNELS];
- ma_channel_mix_mode channelMixMode;
- ma_share_mode shareMode;
- } playback;
- struct
- {
- const ma_device_id* pDeviceID;
- ma_format format;
- ma_uint32 channels;
- ma_channel channelMap[MA_MAX_CHANNELS];
- ma_channel_mix_mode channelMixMode;
- ma_share_mode shareMode;
- } capture;
- struct
- {
- ma_bool8 noAutoConvertSRC; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM. */
- ma_bool8 noDefaultQualitySRC; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY. */
- ma_bool8 noAutoStreamRouting; /* Disables automatic stream routing. */
- ma_bool8 noHardwareOffloading; /* Disables WASAPI's hardware offloading feature. */
- } wasapi;
- struct
- {
- ma_bool32 noMMap; /* Disables MMap mode. */
- ma_bool32 noAutoFormat; /* Opens the ALSA device with SND_PCM_NO_AUTO_FORMAT. */
- ma_bool32 noAutoChannels; /* Opens the ALSA device with SND_PCM_NO_AUTO_CHANNELS. */
- ma_bool32 noAutoResample; /* Opens the ALSA device with SND_PCM_NO_AUTO_RESAMPLE. */
- } alsa;
- struct
- {
- const char* pStreamNamePlayback;
- const char* pStreamNameCapture;
- } pulse;
- struct
- {
- ma_bool32 allowNominalSampleRateChange; /* Desktop only. When enabled, allows changing of the sample rate at the operating system level. */
- } coreaudio;
- struct
- {
- ma_opensl_stream_type streamType;
- ma_opensl_recording_preset recordingPreset;
- } opensl;
- struct
- {
- ma_aaudio_usage usage;
- ma_aaudio_content_type contentType;
- ma_aaudio_input_preset inputPreset;
- } aaudio;
-};
+/**************************************************************************************************************************************************************
+
+Logging
+
+**************************************************************************************************************************************************************/
+#include /* For va_list. */
+
+#if defined(__has_attribute)
+ #if __has_attribute(format)
+ #define MA_ATTRIBUTE_FORMAT(fmt, va) __attribute__((format(printf, fmt, va)))
+ #endif
+#endif
+#ifndef MA_ATTRIBUTE_FORMAT
+#define MA_ATTRIBUTE_FORMAT(fmt,va)
+#endif
+
+#ifndef MA_MAX_LOG_CALLBACKS
+#define MA_MAX_LOG_CALLBACKS 4
+#endif
/*
-The callback for handling device enumeration. This is fired from `ma_context_enumerated_devices()`.
+The callback for handling log messages.
Parameters
----------
-pContext (in)
- A pointer to the context performing the enumeration.
+pUserData (in)
+ The user data pointer that was passed into ma_log_register_callback().
-deviceType (in)
- The type of the device being enumerated. This will always be either `ma_device_type_playback` or `ma_device_type_capture`.
+logLevel (in)
+ The log level. This can be one of the following:
-pInfo (in)
- A pointer to a `ma_device_info` containing the ID and name of the enumerated device. Note that this will not include detailed information about the device,
- only basic information (ID and name). The reason for this is that it would otherwise require opening the backend device to probe for the information which
- is too inefficient.
+ +----------------------+
+ | Log Level |
+ +----------------------+
+ | MA_LOG_LEVEL_DEBUG |
+ | MA_LOG_LEVEL_INFO |
+ | MA_LOG_LEVEL_WARNING |
+ | MA_LOG_LEVEL_ERROR |
+ +----------------------+
-pUserData (in)
- The user data pointer passed into `ma_context_enumerate_devices()`.
-*/
-typedef ma_bool32 (* ma_enum_devices_callback_proc)(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pInfo, void* pUserData);
+pMessage (in)
+ The log message.
-/*
-Describes some basic details about a playback or capture device.
+Remarks
+-------
+Do not modify the state of the device from inside the callback.
*/
+typedef void (* ma_log_callback_proc)(void* pUserData, ma_uint32 level, const char* pMessage);
+
typedef struct
{
- const ma_device_id* pDeviceID;
- ma_share_mode shareMode;
- ma_format format;
- ma_uint32 channels;
- ma_uint32 sampleRate;
- ma_channel channelMap[MA_MAX_CHANNELS];
- ma_uint32 periodSizeInFrames;
- ma_uint32 periodSizeInMilliseconds;
- ma_uint32 periodCount;
-} ma_device_descriptor;
-
-/*
-These are the callbacks required to be implemented for a backend. These callbacks are grouped into two parts: context and device. There is one context
-to many devices. A device is created from a context.
-
-The general flow goes like this:
+ ma_log_callback_proc onLog;
+ void* pUserData;
+} ma_log_callback;
- 1) A context is created with `onContextInit()`
- 1a) Available devices can be enumerated with `onContextEnumerateDevices()` if required.
- 1b) Detailed information about a device can be queried with `onContextGetDeviceInfo()` if required.
- 2) A device is created from the context that was created in the first step using `onDeviceInit()`, and optionally a device ID that was
- selected from device enumeration via `onContextEnumerateDevices()`.
- 3) A device is started or stopped with `onDeviceStart()` / `onDeviceStop()`
- 4) Data is delivered to and from the device by the backend. This is always done based on the native format returned by the prior call
- to `onDeviceInit()`. Conversion between the device's native format and the format requested by the application will be handled by
- miniaudio internally.
+MA_API ma_log_callback ma_log_callback_init(ma_log_callback_proc onLog, void* pUserData);
-Initialization of the context is quite simple. You need to do any necessary initialization of internal objects and then output the
-callbacks defined in this structure.
-Once the context has been initialized you can initialize a device. Before doing so, however, the application may want to know which
-physical devices are available. This is where `onContextEnumerateDevices()` comes in. This is fairly simple. For each device, fire the
-given callback with, at a minimum, the basic information filled out in `ma_device_info`. When the callback returns `MA_FALSE`, enumeration
-needs to stop and the `onContextEnumerateDevices()` function return with a success code.
+typedef struct
+{
+ ma_log_callback callbacks[MA_MAX_LOG_CALLBACKS];
+ ma_uint32 callbackCount;
+ ma_allocation_callbacks allocationCallbacks; /* Need to store these persistently because ma_log_postv() might need to allocate a buffer on the heap. */
+#ifndef MA_NO_THREADING
+ ma_mutex lock; /* For thread safety just to make it easier and safer for the logging implementation. */
+#endif
+} ma_log;
-Detailed device information can be retrieved from a device ID using `onContextGetDeviceInfo()`. This takes as input the device type and ID,
-and on output returns detailed information about the device in `ma_device_info`. The `onContextGetDeviceInfo()` callback must handle the
-case when the device ID is NULL, in which case information about the default device needs to be retrieved.
+MA_API ma_result ma_log_init(const ma_allocation_callbacks* pAllocationCallbacks, ma_log* pLog);
+MA_API void ma_log_uninit(ma_log* pLog);
+MA_API ma_result ma_log_register_callback(ma_log* pLog, ma_log_callback callback);
+MA_API ma_result ma_log_unregister_callback(ma_log* pLog, ma_log_callback callback);
+MA_API ma_result ma_log_post(ma_log* pLog, ma_uint32 level, const char* pMessage);
+MA_API ma_result ma_log_postv(ma_log* pLog, ma_uint32 level, const char* pFormat, va_list args);
+MA_API ma_result ma_log_postf(ma_log* pLog, ma_uint32 level, const char* pFormat, ...) MA_ATTRIBUTE_FORMAT(3, 4);
-Once the context has been created and the device ID retrieved (if using anything other than the default device), the device can be created.
-This is a little bit more complicated than initialization of the context due to it's more complicated configuration. When initializing a
-device, a duplex device may be requested. This means a separate data format needs to be specified for both playback and capture. On input,
-the data format is set to what the application wants. On output it's set to the native format which should match as closely as possible to
-the requested format. The conversion between the format requested by the application and the device's native format will be handled
-internally by miniaudio.
-On input, if the sample format is set to `ma_format_unknown`, the backend is free to use whatever sample format it desires, so long as it's
-supported by miniaudio. When the channel count is set to 0, the backend should use the device's native channel count. The same applies for
-sample rate. For the channel map, the default should be used when `ma_channel_map_blank()` returns true (all channels set to
-`MA_CHANNEL_NONE`). On input, the `periodSizeInFrames` or `periodSizeInMilliseconds` option should always be set. The backend should
-inspect both of these variables. If `periodSizeInFrames` is set, it should take priority, otherwise it needs to be derived from the period
-size in milliseconds (`periodSizeInMilliseconds`) and the sample rate, keeping in mind that the sample rate may be 0, in which case the
-sample rate will need to be determined before calculating the period size in frames. On output, all members of the `ma_device_data_format`
-object should be set to a valid value, except for `periodSizeInMilliseconds` which is optional (`periodSizeInFrames` *must* be set).
+/**************************************************************************************************************************************************************
-Starting and stopping of the device is done with `onDeviceStart()` and `onDeviceStop()` and should be self-explanatory. If the backend uses
-asynchronous reading and writing, `onDeviceStart()` and `onDeviceStop()` should always be implemented.
+Biquad Filtering
-The handling of data delivery between the application and the device is the most complicated part of the process. To make this a bit
-easier, some helper callbacks are available. If the backend uses a blocking read/write style of API, the `onDeviceRead()` and
-`onDeviceWrite()` callbacks can optionally be implemented. These are blocking and work just like reading and writing from a file. If the
-backend uses a callback for data delivery, that callback must call `ma_device_handle_backend_data_callback()` from within it's callback.
-This allows miniaudio to then process any necessary data conversion and then pass it to the miniaudio data callback.
+**************************************************************************************************************************************************************/
+typedef union
+{
+ float f32;
+ ma_int32 s32;
+} ma_biquad_coefficient;
-If the backend requires absolute flexibility with it's data delivery, it can optionally implement the `onDeviceDataLoop()` callback
-which will allow it to implement the logic that will run on the audio thread. This is much more advanced and is completely optional.
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ double b0;
+ double b1;
+ double b2;
+ double a0;
+ double a1;
+ double a2;
+} ma_biquad_config;
-The audio thread should run data delivery logic in a loop while `ma_device_get_state() == MA_STATE_STARTED` and no errors have been
-encounted. Do not start or stop the device here. That will be handled from outside the `onDeviceDataLoop()` callback.
+MA_API ma_biquad_config ma_biquad_config_init(ma_format format, ma_uint32 channels, double b0, double b1, double b2, double a0, double a1, double a2);
-The invocation of the `onDeviceDataLoop()` callback will be handled by miniaudio. When you start the device, miniaudio will fire this
-callback. When the device is stopped, the `ma_device_get_state() == MA_STATE_STARTED` condition will fail and the loop will be terminated
-which will then fall through to the part that stops the device. For an example on how to implement the `onDeviceDataLoop()` callback,
-look at `ma_device_audio_thread__default_read_write()`. Implement the `onDeviceDataLoopWakeup()` callback if you need a mechanism to
-wake up the audio thread.
-*/
-struct ma_backend_callbacks
+typedef struct
{
- ma_result (* onContextInit)(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks);
- ma_result (* onContextUninit)(ma_context* pContext);
- ma_result (* onContextEnumerateDevices)(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData);
- ma_result (* onContextGetDeviceInfo)(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo);
- ma_result (* onDeviceInit)(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture);
- ma_result (* onDeviceUninit)(ma_device* pDevice);
- ma_result (* onDeviceStart)(ma_device* pDevice);
- ma_result (* onDeviceStop)(ma_device* pDevice);
- ma_result (* onDeviceRead)(ma_device* pDevice, void* pFrames, ma_uint32 frameCount, ma_uint32* pFramesRead);
- ma_result (* onDeviceWrite)(ma_device* pDevice, const void* pFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten);
- ma_result (* onDeviceDataLoop)(ma_device* pDevice);
- ma_result (* onDeviceDataLoopWakeup)(ma_device* pDevice);
-};
+ ma_format format;
+ ma_uint32 channels;
+ ma_biquad_coefficient b0;
+ ma_biquad_coefficient b1;
+ ma_biquad_coefficient b2;
+ ma_biquad_coefficient a1;
+ ma_biquad_coefficient a2;
+ ma_biquad_coefficient* pR1;
+ ma_biquad_coefficient* pR2;
-struct ma_context_config
+ /* Memory management. */
+ void* _pHeap;
+ ma_bool32 _ownsHeap;
+} ma_biquad;
+
+MA_API ma_result ma_biquad_get_heap_size(const ma_biquad_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_biquad_init_preallocated(const ma_biquad_config* pConfig, void* pHeap, ma_biquad* pBQ);
+MA_API ma_result ma_biquad_init(const ma_biquad_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_biquad* pBQ);
+MA_API void ma_biquad_uninit(ma_biquad* pBQ, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_biquad_reinit(const ma_biquad_config* pConfig, ma_biquad* pBQ);
+MA_API ma_result ma_biquad_process_pcm_frames(ma_biquad* pBQ, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
+MA_API ma_uint32 ma_biquad_get_latency(const ma_biquad* pBQ);
+
+
+/**************************************************************************************************************************************************************
+
+Low-Pass Filtering
+
+**************************************************************************************************************************************************************/
+typedef struct
{
- ma_log_proc logCallback; /* Legacy logging callback. Will be removed in version 0.11. */
- ma_log* pLog;
- ma_thread_priority threadPriority;
- size_t threadStackSize;
- void* pUserData;
- ma_allocation_callbacks allocationCallbacks;
- struct
- {
- ma_bool32 useVerboseDeviceEnumeration;
- } alsa;
- struct
- {
- const char* pApplicationName;
- const char* pServerName;
- ma_bool32 tryAutoSpawn; /* Enables autospawning of the PulseAudio daemon if necessary. */
- } pulse;
- struct
- {
- ma_ios_session_category sessionCategory;
- ma_uint32 sessionCategoryOptions;
- ma_bool32 noAudioSessionActivate; /* iOS only. When set to true, does not perform an explicit [[AVAudioSession sharedInstace] setActive:true] on initialization. */
- ma_bool32 noAudioSessionDeactivate; /* iOS only. When set to true, does not perform an explicit [[AVAudioSession sharedInstace] setActive:false] on uninitialization. */
- } coreaudio;
- struct
- {
- const char* pClientName;
- ma_bool32 tryStartServer;
- } jack;
- ma_backend_callbacks custom;
-};
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ double cutoffFrequency;
+ double q;
+} ma_lpf1_config, ma_lpf2_config;
+
+MA_API ma_lpf1_config ma_lpf1_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency);
+MA_API ma_lpf2_config ma_lpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q);
-/* WASAPI specific structure for some commands which must run on a common thread due to bugs in WASAPI. */
typedef struct
{
- int code;
- ma_event* pEvent; /* This will be signalled when the event is complete. */
- union
- {
- struct
- {
- int _unused;
- } quit;
- struct
- {
- ma_device_type deviceType;
- void* pAudioClient;
- void** ppAudioClientService;
- ma_result* pResult; /* The result from creating the audio client service. */
- } createAudioClient;
- struct
- {
- ma_device* pDevice;
- ma_device_type deviceType;
- } releaseAudioClient;
- } data;
-} ma_context_command__wasapi;
+ ma_format format;
+ ma_uint32 channels;
+ ma_biquad_coefficient a;
+ ma_biquad_coefficient* pR1;
-struct ma_context
+ /* Memory management. */
+ void* _pHeap;
+ ma_bool32 _ownsHeap;
+} ma_lpf1;
+
+MA_API ma_result ma_lpf1_get_heap_size(const ma_lpf1_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_lpf1_init_preallocated(const ma_lpf1_config* pConfig, void* pHeap, ma_lpf1* pLPF);
+MA_API ma_result ma_lpf1_init(const ma_lpf1_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_lpf1* pLPF);
+MA_API void ma_lpf1_uninit(ma_lpf1* pLPF, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_lpf1_reinit(const ma_lpf1_config* pConfig, ma_lpf1* pLPF);
+MA_API ma_result ma_lpf1_process_pcm_frames(ma_lpf1* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
+MA_API ma_uint32 ma_lpf1_get_latency(const ma_lpf1* pLPF);
+
+typedef struct
{
- ma_backend_callbacks callbacks;
- ma_backend backend; /* DirectSound, ALSA, etc. */
- ma_log* pLog;
- ma_log log; /* Only used if the log is owned by the context. The pLog member will be set to &log in this case. */
- ma_log_proc logCallback; /* Legacy callback. Will be removed in version 0.11. */
- ma_thread_priority threadPriority;
- size_t threadStackSize;
- void* pUserData;
- ma_allocation_callbacks allocationCallbacks;
- ma_mutex deviceEnumLock; /* Used to make ma_context_get_devices() thread safe. */
- ma_mutex deviceInfoLock; /* Used to make ma_context_get_device_info() thread safe. */
- ma_uint32 deviceInfoCapacity; /* Total capacity of pDeviceInfos. */
- ma_uint32 playbackDeviceInfoCount;
- ma_uint32 captureDeviceInfoCount;
- ma_device_info* pDeviceInfos; /* Playback devices first, then capture. */
+ ma_biquad bq; /* The second order low-pass filter is implemented as a biquad filter. */
+} ma_lpf2;
- union
- {
-#ifdef MA_SUPPORT_WASAPI
- struct
- {
- ma_thread commandThread;
- ma_mutex commandLock;
- ma_semaphore commandSem;
- ma_uint32 commandIndex;
- ma_uint32 commandCount;
- ma_context_command__wasapi commands[4];
- } wasapi;
-#endif
-#ifdef MA_SUPPORT_DSOUND
- struct
- {
- ma_handle hDSoundDLL;
- ma_proc DirectSoundCreate;
- ma_proc DirectSoundEnumerateA;
- ma_proc DirectSoundCaptureCreate;
- ma_proc DirectSoundCaptureEnumerateA;
- } dsound;
-#endif
-#ifdef MA_SUPPORT_WINMM
- struct
- {
- ma_handle hWinMM;
- ma_proc waveOutGetNumDevs;
- ma_proc waveOutGetDevCapsA;
- ma_proc waveOutOpen;
- ma_proc waveOutClose;
- ma_proc waveOutPrepareHeader;
- ma_proc waveOutUnprepareHeader;
- ma_proc waveOutWrite;
- ma_proc waveOutReset;
- ma_proc waveInGetNumDevs;
- ma_proc waveInGetDevCapsA;
- ma_proc waveInOpen;
- ma_proc waveInClose;
- ma_proc waveInPrepareHeader;
- ma_proc waveInUnprepareHeader;
- ma_proc waveInAddBuffer;
- ma_proc waveInStart;
- ma_proc waveInReset;
- } winmm;
-#endif
-#ifdef MA_SUPPORT_ALSA
- struct
- {
- ma_handle asoundSO;
- ma_proc snd_pcm_open;
- ma_proc snd_pcm_close;
- ma_proc snd_pcm_hw_params_sizeof;
- ma_proc snd_pcm_hw_params_any;
- ma_proc snd_pcm_hw_params_set_format;
- ma_proc snd_pcm_hw_params_set_format_first;
- ma_proc snd_pcm_hw_params_get_format_mask;
- ma_proc snd_pcm_hw_params_set_channels;
- ma_proc snd_pcm_hw_params_set_channels_near;
- ma_proc snd_pcm_hw_params_set_channels_minmax;
- ma_proc snd_pcm_hw_params_set_rate_resample;
- ma_proc snd_pcm_hw_params_set_rate;
- ma_proc snd_pcm_hw_params_set_rate_near;
- ma_proc snd_pcm_hw_params_set_buffer_size_near;
- ma_proc snd_pcm_hw_params_set_periods_near;
- ma_proc snd_pcm_hw_params_set_access;
- ma_proc snd_pcm_hw_params_get_format;
- ma_proc snd_pcm_hw_params_get_channels;
- ma_proc snd_pcm_hw_params_get_channels_min;
- ma_proc snd_pcm_hw_params_get_channels_max;
- ma_proc snd_pcm_hw_params_get_rate;
- ma_proc snd_pcm_hw_params_get_rate_min;
- ma_proc snd_pcm_hw_params_get_rate_max;
- ma_proc snd_pcm_hw_params_get_buffer_size;
- ma_proc snd_pcm_hw_params_get_periods;
- ma_proc snd_pcm_hw_params_get_access;
- ma_proc snd_pcm_hw_params_test_format;
- ma_proc snd_pcm_hw_params_test_channels;
- ma_proc snd_pcm_hw_params_test_rate;
- ma_proc snd_pcm_hw_params;
- ma_proc snd_pcm_sw_params_sizeof;
- ma_proc snd_pcm_sw_params_current;
- ma_proc snd_pcm_sw_params_get_boundary;
- ma_proc snd_pcm_sw_params_set_avail_min;
- ma_proc snd_pcm_sw_params_set_start_threshold;
- ma_proc snd_pcm_sw_params_set_stop_threshold;
- ma_proc snd_pcm_sw_params;
- ma_proc snd_pcm_format_mask_sizeof;
- ma_proc snd_pcm_format_mask_test;
- ma_proc snd_pcm_get_chmap;
- ma_proc snd_pcm_state;
- ma_proc snd_pcm_prepare;
- ma_proc snd_pcm_start;
- ma_proc snd_pcm_drop;
- ma_proc snd_pcm_drain;
- ma_proc snd_pcm_reset;
- ma_proc snd_device_name_hint;
- ma_proc snd_device_name_get_hint;
- ma_proc snd_card_get_index;
- ma_proc snd_device_name_free_hint;
- ma_proc snd_pcm_mmap_begin;
- ma_proc snd_pcm_mmap_commit;
- ma_proc snd_pcm_recover;
- ma_proc snd_pcm_readi;
- ma_proc snd_pcm_writei;
- ma_proc snd_pcm_avail;
- ma_proc snd_pcm_avail_update;
- ma_proc snd_pcm_wait;
- ma_proc snd_pcm_nonblock;
- ma_proc snd_pcm_info;
- ma_proc snd_pcm_info_sizeof;
- ma_proc snd_pcm_info_get_name;
- ma_proc snd_pcm_poll_descriptors;
- ma_proc snd_pcm_poll_descriptors_count;
- ma_proc snd_pcm_poll_descriptors_revents;
- ma_proc snd_config_update_free_global;
+MA_API ma_result ma_lpf2_get_heap_size(const ma_lpf2_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_lpf2_init_preallocated(const ma_lpf2_config* pConfig, void* pHeap, ma_lpf2* pHPF);
+MA_API ma_result ma_lpf2_init(const ma_lpf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_lpf2* pLPF);
+MA_API void ma_lpf2_uninit(ma_lpf2* pLPF, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_lpf2_reinit(const ma_lpf2_config* pConfig, ma_lpf2* pLPF);
+MA_API ma_result ma_lpf2_process_pcm_frames(ma_lpf2* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
+MA_API ma_uint32 ma_lpf2_get_latency(const ma_lpf2* pLPF);
- ma_mutex internalDeviceEnumLock;
- ma_bool32 useVerboseDeviceEnumeration;
- } alsa;
-#endif
-#ifdef MA_SUPPORT_PULSEAUDIO
- struct
- {
- ma_handle pulseSO;
- ma_proc pa_mainloop_new;
- ma_proc pa_mainloop_free;
- ma_proc pa_mainloop_quit;
- ma_proc pa_mainloop_get_api;
- ma_proc pa_mainloop_iterate;
- ma_proc pa_mainloop_wakeup;
- ma_proc pa_threaded_mainloop_new;
- ma_proc pa_threaded_mainloop_free;
- ma_proc pa_threaded_mainloop_start;
- ma_proc pa_threaded_mainloop_stop;
- ma_proc pa_threaded_mainloop_lock;
- ma_proc pa_threaded_mainloop_unlock;
- ma_proc pa_threaded_mainloop_wait;
- ma_proc pa_threaded_mainloop_signal;
- ma_proc pa_threaded_mainloop_accept;
- ma_proc pa_threaded_mainloop_get_retval;
- ma_proc pa_threaded_mainloop_get_api;
- ma_proc pa_threaded_mainloop_in_thread;
- ma_proc pa_threaded_mainloop_set_name;
- ma_proc pa_context_new;
- ma_proc pa_context_unref;
- ma_proc pa_context_connect;
- ma_proc pa_context_disconnect;
- ma_proc pa_context_set_state_callback;
- ma_proc pa_context_get_state;
- ma_proc pa_context_get_sink_info_list;
- ma_proc pa_context_get_source_info_list;
- ma_proc pa_context_get_sink_info_by_name;
- ma_proc pa_context_get_source_info_by_name;
- ma_proc pa_operation_unref;
- ma_proc pa_operation_get_state;
- ma_proc pa_channel_map_init_extend;
- ma_proc pa_channel_map_valid;
- ma_proc pa_channel_map_compatible;
- ma_proc pa_stream_new;
- ma_proc pa_stream_unref;
- ma_proc pa_stream_connect_playback;
- ma_proc pa_stream_connect_record;
- ma_proc pa_stream_disconnect;
- ma_proc pa_stream_get_state;
- ma_proc pa_stream_get_sample_spec;
- ma_proc pa_stream_get_channel_map;
- ma_proc pa_stream_get_buffer_attr;
- ma_proc pa_stream_set_buffer_attr;
- ma_proc pa_stream_get_device_name;
- ma_proc pa_stream_set_write_callback;
- ma_proc pa_stream_set_read_callback;
- ma_proc pa_stream_set_suspended_callback;
- ma_proc pa_stream_is_suspended;
- ma_proc pa_stream_flush;
- ma_proc pa_stream_drain;
- ma_proc pa_stream_is_corked;
- ma_proc pa_stream_cork;
- ma_proc pa_stream_trigger;
- ma_proc pa_stream_begin_write;
- ma_proc pa_stream_write;
- ma_proc pa_stream_peek;
- ma_proc pa_stream_drop;
- ma_proc pa_stream_writable_size;
- ma_proc pa_stream_readable_size;
- /*pa_mainloop**/ ma_ptr pMainLoop;
- /*pa_context**/ ma_ptr pPulseContext;
- } pulse;
-#endif
-#ifdef MA_SUPPORT_JACK
- struct
- {
- ma_handle jackSO;
- ma_proc jack_client_open;
- ma_proc jack_client_close;
- ma_proc jack_client_name_size;
- ma_proc jack_set_process_callback;
- ma_proc jack_set_buffer_size_callback;
- ma_proc jack_on_shutdown;
- ma_proc jack_get_sample_rate;
- ma_proc jack_get_buffer_size;
- ma_proc jack_get_ports;
- ma_proc jack_activate;
- ma_proc jack_deactivate;
- ma_proc jack_connect;
- ma_proc jack_port_register;
- ma_proc jack_port_name;
- ma_proc jack_port_get_buffer;
- ma_proc jack_free;
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ double cutoffFrequency;
+ ma_uint32 order; /* If set to 0, will be treated as a passthrough (no filtering will be applied). */
+} ma_lpf_config;
- char* pClientName;
- ma_bool32 tryStartServer;
- } jack;
-#endif
-#ifdef MA_SUPPORT_COREAUDIO
- struct
- {
- ma_handle hCoreFoundation;
- ma_proc CFStringGetCString;
- ma_proc CFRelease;
+MA_API ma_lpf_config ma_lpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order);
- ma_handle hCoreAudio;
- ma_proc AudioObjectGetPropertyData;
- ma_proc AudioObjectGetPropertyDataSize;
- ma_proc AudioObjectSetPropertyData;
- ma_proc AudioObjectAddPropertyListener;
- ma_proc AudioObjectRemovePropertyListener;
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ ma_uint32 lpf1Count;
+ ma_uint32 lpf2Count;
+ ma_lpf1* pLPF1;
+ ma_lpf2* pLPF2;
- ma_handle hAudioUnit; /* Could possibly be set to AudioToolbox on later versions of macOS. */
- ma_proc AudioComponentFindNext;
- ma_proc AudioComponentInstanceDispose;
- ma_proc AudioComponentInstanceNew;
- ma_proc AudioOutputUnitStart;
- ma_proc AudioOutputUnitStop;
- ma_proc AudioUnitAddPropertyListener;
- ma_proc AudioUnitGetPropertyInfo;
- ma_proc AudioUnitGetProperty;
- ma_proc AudioUnitSetProperty;
- ma_proc AudioUnitInitialize;
- ma_proc AudioUnitRender;
+ /* Memory management. */
+ void* _pHeap;
+ ma_bool32 _ownsHeap;
+} ma_lpf;
- /*AudioComponent*/ ma_ptr component;
- ma_bool32 noAudioSessionDeactivate; /* For tracking whether or not the iOS audio session should be explicitly deactivated. Set from the config in ma_context_init__coreaudio(). */
- } coreaudio;
-#endif
-#ifdef MA_SUPPORT_SNDIO
- struct
- {
- ma_handle sndioSO;
- ma_proc sio_open;
- ma_proc sio_close;
- ma_proc sio_setpar;
- ma_proc sio_getpar;
- ma_proc sio_getcap;
- ma_proc sio_start;
- ma_proc sio_stop;
- ma_proc sio_read;
- ma_proc sio_write;
- ma_proc sio_onmove;
- ma_proc sio_nfds;
- ma_proc sio_pollfd;
- ma_proc sio_revents;
- ma_proc sio_eof;
- ma_proc sio_setvol;
- ma_proc sio_onvol;
- ma_proc sio_initpar;
- } sndio;
-#endif
-#ifdef MA_SUPPORT_AUDIO4
- struct
- {
- int _unused;
- } audio4;
-#endif
-#ifdef MA_SUPPORT_OSS
- struct
- {
- int versionMajor;
- int versionMinor;
- } oss;
-#endif
-#ifdef MA_SUPPORT_AAUDIO
- struct
- {
- ma_handle hAAudio; /* libaaudio.so */
- ma_proc AAudio_createStreamBuilder;
- ma_proc AAudioStreamBuilder_delete;
- ma_proc AAudioStreamBuilder_setDeviceId;
- ma_proc AAudioStreamBuilder_setDirection;
- ma_proc AAudioStreamBuilder_setSharingMode;
- ma_proc AAudioStreamBuilder_setFormat;
- ma_proc AAudioStreamBuilder_setChannelCount;
- ma_proc AAudioStreamBuilder_setSampleRate;
- ma_proc AAudioStreamBuilder_setBufferCapacityInFrames;
- ma_proc AAudioStreamBuilder_setFramesPerDataCallback;
- ma_proc AAudioStreamBuilder_setDataCallback;
- ma_proc AAudioStreamBuilder_setErrorCallback;
- ma_proc AAudioStreamBuilder_setPerformanceMode;
- ma_proc AAudioStreamBuilder_setUsage;
- ma_proc AAudioStreamBuilder_setContentType;
- ma_proc AAudioStreamBuilder_setInputPreset;
- ma_proc AAudioStreamBuilder_openStream;
- ma_proc AAudioStream_close;
- ma_proc AAudioStream_getState;
- ma_proc AAudioStream_waitForStateChange;
- ma_proc AAudioStream_getFormat;
- ma_proc AAudioStream_getChannelCount;
- ma_proc AAudioStream_getSampleRate;
- ma_proc AAudioStream_getBufferCapacityInFrames;
- ma_proc AAudioStream_getFramesPerDataCallback;
- ma_proc AAudioStream_getFramesPerBurst;
- ma_proc AAudioStream_requestStart;
- ma_proc AAudioStream_requestStop;
- } aaudio;
-#endif
-#ifdef MA_SUPPORT_OPENSL
- struct
- {
- ma_handle libOpenSLES;
- ma_handle SL_IID_ENGINE;
- ma_handle SL_IID_AUDIOIODEVICECAPABILITIES;
- ma_handle SL_IID_ANDROIDSIMPLEBUFFERQUEUE;
- ma_handle SL_IID_RECORD;
- ma_handle SL_IID_PLAY;
- ma_handle SL_IID_OUTPUTMIX;
- ma_handle SL_IID_ANDROIDCONFIGURATION;
- ma_proc slCreateEngine;
- } opensl;
-#endif
-#ifdef MA_SUPPORT_WEBAUDIO
- struct
- {
- int _unused;
- } webaudio;
-#endif
-#ifdef MA_SUPPORT_NULL
- struct
- {
- int _unused;
- } null_backend;
-#endif
- };
+MA_API ma_result ma_lpf_get_heap_size(const ma_lpf_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_lpf_init_preallocated(const ma_lpf_config* pConfig, void* pHeap, ma_lpf* pLPF);
+MA_API ma_result ma_lpf_init(const ma_lpf_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_lpf* pLPF);
+MA_API void ma_lpf_uninit(ma_lpf* pLPF, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_lpf_reinit(const ma_lpf_config* pConfig, ma_lpf* pLPF);
+MA_API ma_result ma_lpf_process_pcm_frames(ma_lpf* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
+MA_API ma_uint32 ma_lpf_get_latency(const ma_lpf* pLPF);
- union
- {
-#ifdef MA_WIN32
- struct
- {
- /*HMODULE*/ ma_handle hOle32DLL;
- ma_proc CoInitializeEx;
- ma_proc CoUninitialize;
- ma_proc CoCreateInstance;
- ma_proc CoTaskMemFree;
- ma_proc PropVariantClear;
- ma_proc StringFromGUID2;
- /*HMODULE*/ ma_handle hUser32DLL;
- ma_proc GetForegroundWindow;
- ma_proc GetDesktopWindow;
+/**************************************************************************************************************************************************************
- /*HMODULE*/ ma_handle hAdvapi32DLL;
- ma_proc RegOpenKeyExA;
- ma_proc RegCloseKey;
- ma_proc RegQueryValueExA;
- } win32;
-#endif
-#ifdef MA_POSIX
- struct
- {
- ma_handle pthreadSO;
- ma_proc pthread_create;
- ma_proc pthread_join;
- ma_proc pthread_mutex_init;
- ma_proc pthread_mutex_destroy;
- ma_proc pthread_mutex_lock;
- ma_proc pthread_mutex_unlock;
- ma_proc pthread_cond_init;
- ma_proc pthread_cond_destroy;
- ma_proc pthread_cond_wait;
- ma_proc pthread_cond_signal;
- ma_proc pthread_attr_init;
- ma_proc pthread_attr_destroy;
- ma_proc pthread_attr_setschedpolicy;
- ma_proc pthread_attr_getschedparam;
- ma_proc pthread_attr_setschedparam;
- } posix;
-#endif
- int _unused;
- };
-};
+High-Pass Filtering
-struct ma_device
+**************************************************************************************************************************************************************/
+typedef struct
{
- ma_context* pContext;
- ma_device_type type;
+ ma_format format;
+ ma_uint32 channels;
ma_uint32 sampleRate;
- MA_ATOMIC ma_uint32 state; /* The state of the device is variable and can change at any time on any thread. Must be used atomically. */
- ma_device_callback_proc onData; /* Set once at initialization time and should not be changed after. */
- ma_stop_proc onStop; /* Set once at initialization time and should not be changed after. */
- void* pUserData; /* Application defined data. */
- ma_mutex startStopLock;
- ma_event wakeupEvent;
- ma_event startEvent;
- ma_event stopEvent;
- ma_thread thread;
- ma_result workResult; /* This is set by the worker thread after it's finished doing a job. */
- ma_bool8 isOwnerOfContext; /* When set to true, uninitializing the device will also uninitialize the context. Set to true when NULL is passed into ma_device_init(). */
- ma_bool8 noPreZeroedOutputBuffer;
- ma_bool8 noClip;
- MA_ATOMIC float masterVolumeFactor; /* Linear 0..1. Can be read and written simultaneously by different threads. Must be used atomically. */
- ma_duplex_rb duplexRB; /* Intermediary buffer for duplex device on asynchronous backends. */
- struct
- {
- ma_resample_algorithm algorithm;
- struct
- {
- ma_uint32 lpfOrder;
- } linear;
- struct
- {
- int quality;
- } speex;
- } resampling;
- struct
- {
- ma_device_id id; /* If using an explicit device, will be set to a copy of the ID used for initialization. Otherwise cleared to 0. */
- char name[256]; /* Maybe temporary. Likely to be replaced with a query API. */
- ma_share_mode shareMode; /* Set to whatever was passed in when the device was initialized. */
- ma_format format;
- ma_uint32 channels;
- ma_channel channelMap[MA_MAX_CHANNELS];
- ma_format internalFormat;
- ma_uint32 internalChannels;
- ma_uint32 internalSampleRate;
- ma_channel internalChannelMap[MA_MAX_CHANNELS];
- ma_uint32 internalPeriodSizeInFrames;
- ma_uint32 internalPeriods;
- ma_channel_mix_mode channelMixMode;
- ma_data_converter converter;
- } playback;
- struct
- {
- ma_device_id id; /* If using an explicit device, will be set to a copy of the ID used for initialization. Otherwise cleared to 0. */
- char name[256]; /* Maybe temporary. Likely to be replaced with a query API. */
- ma_share_mode shareMode; /* Set to whatever was passed in when the device was initialized. */
- ma_format format;
- ma_uint32 channels;
- ma_channel channelMap[MA_MAX_CHANNELS];
- ma_format internalFormat;
- ma_uint32 internalChannels;
- ma_uint32 internalSampleRate;
- ma_channel internalChannelMap[MA_MAX_CHANNELS];
- ma_uint32 internalPeriodSizeInFrames;
- ma_uint32 internalPeriods;
- ma_channel_mix_mode channelMixMode;
- ma_data_converter converter;
- } capture;
+ double cutoffFrequency;
+ double q;
+} ma_hpf1_config, ma_hpf2_config;
- union
- {
-#ifdef MA_SUPPORT_WASAPI
- struct
- {
- /*IAudioClient**/ ma_ptr pAudioClientPlayback;
- /*IAudioClient**/ ma_ptr pAudioClientCapture;
- /*IAudioRenderClient**/ ma_ptr pRenderClient;
- /*IAudioCaptureClient**/ ma_ptr pCaptureClient;
- /*IMMDeviceEnumerator**/ ma_ptr pDeviceEnumerator; /* Used for IMMNotificationClient notifications. Required for detecting default device changes. */
- ma_IMMNotificationClient notificationClient;
- /*HANDLE*/ ma_handle hEventPlayback; /* Auto reset. Initialized to signaled. */
- /*HANDLE*/ ma_handle hEventCapture; /* Auto reset. Initialized to unsignaled. */
- ma_uint32 actualPeriodSizeInFramesPlayback; /* Value from GetBufferSize(). internalPeriodSizeInFrames is not set to the _actual_ buffer size when low-latency shared mode is being used due to the way the IAudioClient3 API works. */
- ma_uint32 actualPeriodSizeInFramesCapture;
- ma_uint32 originalPeriodSizeInFrames;
- ma_uint32 originalPeriodSizeInMilliseconds;
- ma_uint32 originalPeriods;
- ma_performance_profile originalPerformanceProfile;
- ma_uint32 periodSizeInFramesPlayback;
- ma_uint32 periodSizeInFramesCapture;
- MA_ATOMIC ma_bool32 isStartedCapture; /* Can be read and written simultaneously across different threads. Must be used atomically, and must be 32-bit. */
- MA_ATOMIC ma_bool32 isStartedPlayback; /* Can be read and written simultaneously across different threads. Must be used atomically, and must be 32-bit. */
- ma_bool8 noAutoConvertSRC; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM. */
- ma_bool8 noDefaultQualitySRC; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY. */
- ma_bool8 noHardwareOffloading;
- ma_bool8 allowCaptureAutoStreamRouting;
- ma_bool8 allowPlaybackAutoStreamRouting;
- ma_bool8 isDetachedPlayback;
- ma_bool8 isDetachedCapture;
- } wasapi;
-#endif
-#ifdef MA_SUPPORT_DSOUND
- struct
- {
- /*LPDIRECTSOUND*/ ma_ptr pPlayback;
- /*LPDIRECTSOUNDBUFFER*/ ma_ptr pPlaybackPrimaryBuffer;
- /*LPDIRECTSOUNDBUFFER*/ ma_ptr pPlaybackBuffer;
- /*LPDIRECTSOUNDCAPTURE*/ ma_ptr pCapture;
- /*LPDIRECTSOUNDCAPTUREBUFFER*/ ma_ptr pCaptureBuffer;
- } dsound;
-#endif
-#ifdef MA_SUPPORT_WINMM
- struct
- {
- /*HWAVEOUT*/ ma_handle hDevicePlayback;
- /*HWAVEIN*/ ma_handle hDeviceCapture;
- /*HANDLE*/ ma_handle hEventPlayback;
- /*HANDLE*/ ma_handle hEventCapture;
- ma_uint32 fragmentSizeInFrames;
- ma_uint32 iNextHeaderPlayback; /* [0,periods). Used as an index into pWAVEHDRPlayback. */
- ma_uint32 iNextHeaderCapture; /* [0,periods). Used as an index into pWAVEHDRCapture. */
- ma_uint32 headerFramesConsumedPlayback; /* The number of PCM frames consumed in the buffer in pWAVEHEADER[iNextHeader]. */
- ma_uint32 headerFramesConsumedCapture; /* ^^^ */
- /*WAVEHDR**/ ma_uint8* pWAVEHDRPlayback; /* One instantiation for each period. */
- /*WAVEHDR**/ ma_uint8* pWAVEHDRCapture; /* One instantiation for each period. */
- ma_uint8* pIntermediaryBufferPlayback;
- ma_uint8* pIntermediaryBufferCapture;
- ma_uint8* _pHeapData; /* Used internally and is used for the heap allocated data for the intermediary buffer and the WAVEHDR structures. */
- } winmm;
-#endif
-#ifdef MA_SUPPORT_ALSA
- struct
- {
- /*snd_pcm_t**/ ma_ptr pPCMPlayback;
- /*snd_pcm_t**/ ma_ptr pPCMCapture;
- /*struct pollfd**/ void* pPollDescriptorsPlayback;
- /*struct pollfd**/ void* pPollDescriptorsCapture;
- int pollDescriptorCountPlayback;
- int pollDescriptorCountCapture;
- int wakeupfdPlayback; /* eventfd for waking up from poll() when the playback device is stopped. */
- int wakeupfdCapture; /* eventfd for waking up from poll() when the capture device is stopped. */
- ma_bool8 isUsingMMapPlayback;
- ma_bool8 isUsingMMapCapture;
- } alsa;
-#endif
-#ifdef MA_SUPPORT_PULSEAUDIO
- struct
- {
- /*pa_stream**/ ma_ptr pStreamPlayback;
- /*pa_stream**/ ma_ptr pStreamCapture;
- } pulse;
-#endif
-#ifdef MA_SUPPORT_JACK
- struct
- {
- /*jack_client_t**/ ma_ptr pClient;
- /*jack_port_t**/ ma_ptr pPortsPlayback[MA_MAX_CHANNELS];
- /*jack_port_t**/ ma_ptr pPortsCapture[MA_MAX_CHANNELS];
- float* pIntermediaryBufferPlayback; /* Typed as a float because JACK is always floating point. */
- float* pIntermediaryBufferCapture;
- } jack;
-#endif
-#ifdef MA_SUPPORT_COREAUDIO
- struct
- {
- ma_uint32 deviceObjectIDPlayback;
- ma_uint32 deviceObjectIDCapture;
- /*AudioUnit*/ ma_ptr audioUnitPlayback;
- /*AudioUnit*/ ma_ptr audioUnitCapture;
- /*AudioBufferList**/ ma_ptr pAudioBufferList; /* Only used for input devices. */
- ma_uint32 audioBufferCapInFrames; /* Only used for input devices. The capacity in frames of each buffer in pAudioBufferList. */
- ma_event stopEvent;
- ma_uint32 originalPeriodSizeInFrames;
- ma_uint32 originalPeriodSizeInMilliseconds;
- ma_uint32 originalPeriods;
- ma_performance_profile originalPerformanceProfile;
- ma_bool32 isDefaultPlaybackDevice;
- ma_bool32 isDefaultCaptureDevice;
- ma_bool32 isSwitchingPlaybackDevice; /* <-- Set to true when the default device has changed and miniaudio is in the process of switching. */
- ma_bool32 isSwitchingCaptureDevice; /* <-- Set to true when the default device has changed and miniaudio is in the process of switching. */
- void* pRouteChangeHandler; /* Only used on mobile platforms. Obj-C object for handling route changes. */
- } coreaudio;
-#endif
-#ifdef MA_SUPPORT_SNDIO
- struct
- {
- ma_ptr handlePlayback;
- ma_ptr handleCapture;
- ma_bool32 isStartedPlayback;
- ma_bool32 isStartedCapture;
- } sndio;
-#endif
-#ifdef MA_SUPPORT_AUDIO4
- struct
- {
- int fdPlayback;
- int fdCapture;
- } audio4;
-#endif
-#ifdef MA_SUPPORT_OSS
- struct
- {
- int fdPlayback;
- int fdCapture;
- } oss;
-#endif
-#ifdef MA_SUPPORT_AAUDIO
- struct
- {
- /*AAudioStream**/ ma_ptr pStreamPlayback;
- /*AAudioStream**/ ma_ptr pStreamCapture;
- } aaudio;
-#endif
-#ifdef MA_SUPPORT_OPENSL
- struct
- {
- /*SLObjectItf*/ ma_ptr pOutputMixObj;
- /*SLOutputMixItf*/ ma_ptr pOutputMix;
- /*SLObjectItf*/ ma_ptr pAudioPlayerObj;
- /*SLPlayItf*/ ma_ptr pAudioPlayer;
- /*SLObjectItf*/ ma_ptr pAudioRecorderObj;
- /*SLRecordItf*/ ma_ptr pAudioRecorder;
- /*SLAndroidSimpleBufferQueueItf*/ ma_ptr pBufferQueuePlayback;
- /*SLAndroidSimpleBufferQueueItf*/ ma_ptr pBufferQueueCapture;
- ma_bool32 isDrainingCapture;
- ma_bool32 isDrainingPlayback;
- ma_uint32 currentBufferIndexPlayback;
- ma_uint32 currentBufferIndexCapture;
- ma_uint8* pBufferPlayback; /* This is malloc()'d and is used for storing audio data. Typed as ma_uint8 for easy offsetting. */
- ma_uint8* pBufferCapture;
- } opensl;
-#endif
-#ifdef MA_SUPPORT_WEBAUDIO
- struct
- {
- int indexPlayback; /* We use a factory on the JavaScript side to manage devices and use an index for JS/C interop. */
- int indexCapture;
- } webaudio;
-#endif
-#ifdef MA_SUPPORT_NULL
- struct
- {
- ma_thread deviceThread;
- ma_event operationEvent;
- ma_event operationCompletionEvent;
- ma_semaphore operationSemaphore;
- ma_uint32 operation;
- ma_result operationResult;
- ma_timer timer;
- double priorRunTime;
- ma_uint32 currentPeriodFramesRemainingPlayback;
- ma_uint32 currentPeriodFramesRemainingCapture;
- ma_uint64 lastProcessedFramePlayback;
- ma_uint64 lastProcessedFrameCapture;
- MA_ATOMIC ma_bool32 isStarted; /* Read and written by multiple threads. Must be used atomically, and must be 32-bit for compiler compatibility. */
- } null_device;
-#endif
- };
-};
-#if defined(_MSC_VER) && !defined(__clang__)
- #pragma warning(pop)
-#elif defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8)))
- #pragma GCC diagnostic pop /* For ISO C99 doesn't support unnamed structs/unions [-Wpedantic] */
-#endif
+MA_API ma_hpf1_config ma_hpf1_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency);
+MA_API ma_hpf2_config ma_hpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q);
-/*
-Initializes a `ma_context_config` object.
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ ma_biquad_coefficient a;
+ ma_biquad_coefficient* pR1;
+ /* Memory management. */
+ void* _pHeap;
+ ma_bool32 _ownsHeap;
+} ma_hpf1;
-Return Value
-------------
-A `ma_context_config` initialized to defaults.
+MA_API ma_result ma_hpf1_get_heap_size(const ma_hpf1_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_hpf1_init_preallocated(const ma_hpf1_config* pConfig, void* pHeap, ma_hpf1* pLPF);
+MA_API ma_result ma_hpf1_init(const ma_hpf1_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hpf1* pHPF);
+MA_API void ma_hpf1_uninit(ma_hpf1* pHPF, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_hpf1_reinit(const ma_hpf1_config* pConfig, ma_hpf1* pHPF);
+MA_API ma_result ma_hpf1_process_pcm_frames(ma_hpf1* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
+MA_API ma_uint32 ma_hpf1_get_latency(const ma_hpf1* pHPF);
+typedef struct
+{
+ ma_biquad bq; /* The second order high-pass filter is implemented as a biquad filter. */
+} ma_hpf2;
-Remarks
--------
-You must always use this to initialize the default state of the `ma_context_config` object. Not using this will result in your program breaking when miniaudio
-is updated and new members are added to `ma_context_config`. It also sets logical defaults.
+MA_API ma_result ma_hpf2_get_heap_size(const ma_hpf2_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_hpf2_init_preallocated(const ma_hpf2_config* pConfig, void* pHeap, ma_hpf2* pHPF);
+MA_API ma_result ma_hpf2_init(const ma_hpf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hpf2* pHPF);
+MA_API void ma_hpf2_uninit(ma_hpf2* pHPF, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_hpf2_reinit(const ma_hpf2_config* pConfig, ma_hpf2* pHPF);
+MA_API ma_result ma_hpf2_process_pcm_frames(ma_hpf2* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
+MA_API ma_uint32 ma_hpf2_get_latency(const ma_hpf2* pHPF);
-You can override members of the returned object by changing it's members directly.
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ double cutoffFrequency;
+ ma_uint32 order; /* If set to 0, will be treated as a passthrough (no filtering will be applied). */
+} ma_hpf_config;
-See Also
---------
-ma_context_init()
-*/
-MA_API ma_context_config ma_context_config_init(void);
+MA_API ma_hpf_config ma_hpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order);
-/*
-Initializes a context.
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ ma_uint32 hpf1Count;
+ ma_uint32 hpf2Count;
+ ma_hpf1* pHPF1;
+ ma_hpf2* pHPF2;
-The context is used for selecting and initializing an appropriate backend and to represent the backend at a more global level than that of an individual
-device. There is one context to many devices, and a device is created from a context. A context is required to enumerate devices.
+ /* Memory management. */
+ void* _pHeap;
+ ma_bool32 _ownsHeap;
+} ma_hpf;
+MA_API ma_result ma_hpf_get_heap_size(const ma_hpf_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_hpf_init_preallocated(const ma_hpf_config* pConfig, void* pHeap, ma_hpf* pLPF);
+MA_API ma_result ma_hpf_init(const ma_hpf_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hpf* pHPF);
+MA_API void ma_hpf_uninit(ma_hpf* pHPF, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_hpf_reinit(const ma_hpf_config* pConfig, ma_hpf* pHPF);
+MA_API ma_result ma_hpf_process_pcm_frames(ma_hpf* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
+MA_API ma_uint32 ma_hpf_get_latency(const ma_hpf* pHPF);
-Parameters
-----------
-backends (in, optional)
- A list of backends to try initializing, in priority order. Can be NULL, in which case it uses default priority order.
-backendCount (in, optional)
- The number of items in `backend`. Ignored if `backend` is NULL.
+/**************************************************************************************************************************************************************
-pConfig (in, optional)
- The context configuration.
+Band-Pass Filtering
-pContext (in)
- A pointer to the context object being initialized.
+**************************************************************************************************************************************************************/
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ double cutoffFrequency;
+ double q;
+} ma_bpf2_config;
+MA_API ma_bpf2_config ma_bpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q);
-Return Value
-------------
-MA_SUCCESS if successful; any other error code otherwise.
+typedef struct
+{
+ ma_biquad bq; /* The second order band-pass filter is implemented as a biquad filter. */
+} ma_bpf2;
+MA_API ma_result ma_bpf2_get_heap_size(const ma_bpf2_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_bpf2_init_preallocated(const ma_bpf2_config* pConfig, void* pHeap, ma_bpf2* pBPF);
+MA_API ma_result ma_bpf2_init(const ma_bpf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_bpf2* pBPF);
+MA_API void ma_bpf2_uninit(ma_bpf2* pBPF, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_bpf2_reinit(const ma_bpf2_config* pConfig, ma_bpf2* pBPF);
+MA_API ma_result ma_bpf2_process_pcm_frames(ma_bpf2* pBPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
+MA_API ma_uint32 ma_bpf2_get_latency(const ma_bpf2* pBPF);
-Thread Safety
--------------
-Unsafe. Do not call this function across multiple threads as some backends read and write to global state.
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ double cutoffFrequency;
+ ma_uint32 order; /* If set to 0, will be treated as a passthrough (no filtering will be applied). */
+} ma_bpf_config;
-Remarks
--------
-When `backends` is NULL, the default priority order will be used. Below is a list of backends in priority order:
+MA_API ma_bpf_config ma_bpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order);
- |-------------|-----------------------|--------------------------------------------------------|
- | Name | Enum Name | Supported Operating Systems |
- |-------------|-----------------------|--------------------------------------------------------|
- | WASAPI | ma_backend_wasapi | Windows Vista+ |
- | DirectSound | ma_backend_dsound | Windows XP+ |
- | WinMM | ma_backend_winmm | Windows XP+ (may work on older versions, but untested) |
- | Core Audio | ma_backend_coreaudio | macOS, iOS |
- | ALSA | ma_backend_alsa | Linux |
- | PulseAudio | ma_backend_pulseaudio | Cross Platform (disabled on Windows, BSD and Android) |
- | JACK | ma_backend_jack | Cross Platform (disabled on BSD and Android) |
- | sndio | ma_backend_sndio | OpenBSD |
- | audio(4) | ma_backend_audio4 | NetBSD, OpenBSD |
- | OSS | ma_backend_oss | FreeBSD |
- | AAudio | ma_backend_aaudio | Android 8+ |
- | OpenSL|ES | ma_backend_opensl | Android (API level 16+) |
- | Web Audio | ma_backend_webaudio | Web (via Emscripten) |
- | Null | ma_backend_null | Cross Platform (not used on Web) |
- |-------------|-----------------------|--------------------------------------------------------|
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 bpf2Count;
+ ma_bpf2* pBPF2;
-The context can be configured via the `pConfig` argument. The config object is initialized with `ma_context_config_init()`. Individual configuration settings
-can then be set directly on the structure. Below are the members of the `ma_context_config` object.
+ /* Memory management. */
+ void* _pHeap;
+ ma_bool32 _ownsHeap;
+} ma_bpf;
- pLog
- A pointer to the `ma_log` to post log messages to. Can be NULL if the application does not
- require logging. See the `ma_log` API for details on how to use the logging system.
+MA_API ma_result ma_bpf_get_heap_size(const ma_bpf_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_bpf_init_preallocated(const ma_bpf_config* pConfig, void* pHeap, ma_bpf* pBPF);
+MA_API ma_result ma_bpf_init(const ma_bpf_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_bpf* pBPF);
+MA_API void ma_bpf_uninit(ma_bpf* pBPF, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_bpf_reinit(const ma_bpf_config* pConfig, ma_bpf* pBPF);
+MA_API ma_result ma_bpf_process_pcm_frames(ma_bpf* pBPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
+MA_API ma_uint32 ma_bpf_get_latency(const ma_bpf* pBPF);
- threadPriority
- The desired priority to use for the audio thread. Allowable values include the following:
- |--------------------------------------|
- | Thread Priority |
- |--------------------------------------|
- | ma_thread_priority_idle |
- | ma_thread_priority_lowest |
- | ma_thread_priority_low |
- | ma_thread_priority_normal |
- | ma_thread_priority_high |
- | ma_thread_priority_highest (default) |
- | ma_thread_priority_realtime |
- | ma_thread_priority_default |
- |--------------------------------------|
+/**************************************************************************************************************************************************************
- pUserData
- A pointer to application-defined data. This can be accessed from the context object directly such as `context.pUserData`.
+Notching Filter
- allocationCallbacks
- Structure containing custom allocation callbacks. Leaving this at defaults will cause it to use MA_MALLOC, MA_REALLOC and MA_FREE. These allocation
- callbacks will be used for anything tied to the context, including devices.
+**************************************************************************************************************************************************************/
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ double q;
+ double frequency;
+} ma_notch2_config, ma_notch_config;
- alsa.useVerboseDeviceEnumeration
- ALSA will typically enumerate many different devices which can be intrusive and not user-friendly. To combat this, miniaudio will enumerate only unique
- card/device pairs by default. The problem with this is that you lose a bit of flexibility and control. Setting alsa.useVerboseDeviceEnumeration makes
- it so the ALSA backend includes all devices. Defaults to false.
+MA_API ma_notch2_config ma_notch2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double q, double frequency);
- pulse.pApplicationName
- PulseAudio only. The application name to use when initializing the PulseAudio context with `pa_context_new()`.
+typedef struct
+{
+ ma_biquad bq;
+} ma_notch2;
- pulse.pServerName
- PulseAudio only. The name of the server to connect to with `pa_context_connect()`.
+MA_API ma_result ma_notch2_get_heap_size(const ma_notch2_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_notch2_init_preallocated(const ma_notch2_config* pConfig, void* pHeap, ma_notch2* pFilter);
+MA_API ma_result ma_notch2_init(const ma_notch2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_notch2* pFilter);
+MA_API void ma_notch2_uninit(ma_notch2* pFilter, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_notch2_reinit(const ma_notch2_config* pConfig, ma_notch2* pFilter);
+MA_API ma_result ma_notch2_process_pcm_frames(ma_notch2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
+MA_API ma_uint32 ma_notch2_get_latency(const ma_notch2* pFilter);
- pulse.tryAutoSpawn
- PulseAudio only. Whether or not to try automatically starting the PulseAudio daemon. Defaults to false. If you set this to true, keep in mind that
- miniaudio uses a trial and error method to find the most appropriate backend, and this will result in the PulseAudio daemon starting which may be
- intrusive for the end user.
- coreaudio.sessionCategory
- iOS only. The session category to use for the shared AudioSession instance. Below is a list of allowable values and their Core Audio equivalents.
+/**************************************************************************************************************************************************************
- |-----------------------------------------|-------------------------------------|
- | miniaudio Token | Core Audio Token |
- |-----------------------------------------|-------------------------------------|
- | ma_ios_session_category_ambient | AVAudioSessionCategoryAmbient |
- | ma_ios_session_category_solo_ambient | AVAudioSessionCategorySoloAmbient |
- | ma_ios_session_category_playback | AVAudioSessionCategoryPlayback |
- | ma_ios_session_category_record | AVAudioSessionCategoryRecord |
- | ma_ios_session_category_play_and_record | AVAudioSessionCategoryPlayAndRecord |
- | ma_ios_session_category_multi_route | AVAudioSessionCategoryMultiRoute |
- | ma_ios_session_category_none | AVAudioSessionCategoryAmbient |
- | ma_ios_session_category_default | AVAudioSessionCategoryAmbient |
- |-----------------------------------------|-------------------------------------|
+Peaking EQ Filter
- coreaudio.sessionCategoryOptions
- iOS only. Session category options to use with the shared AudioSession instance. Below is a list of allowable values and their Core Audio equivalents.
+**************************************************************************************************************************************************************/
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ double gainDB;
+ double q;
+ double frequency;
+} ma_peak2_config, ma_peak_config;
- |---------------------------------------------------------------------------|------------------------------------------------------------------|
- | miniaudio Token | Core Audio Token |
- |---------------------------------------------------------------------------|------------------------------------------------------------------|
- | ma_ios_session_category_option_mix_with_others | AVAudioSessionCategoryOptionMixWithOthers |
- | ma_ios_session_category_option_duck_others | AVAudioSessionCategoryOptionDuckOthers |
- | ma_ios_session_category_option_allow_bluetooth | AVAudioSessionCategoryOptionAllowBluetooth |
- | ma_ios_session_category_option_default_to_speaker | AVAudioSessionCategoryOptionDefaultToSpeaker |
- | ma_ios_session_category_option_interrupt_spoken_audio_and_mix_with_others | AVAudioSessionCategoryOptionInterruptSpokenAudioAndMixWithOthers |
- | ma_ios_session_category_option_allow_bluetooth_a2dp | AVAudioSessionCategoryOptionAllowBluetoothA2DP |
- | ma_ios_session_category_option_allow_air_play | AVAudioSessionCategoryOptionAllowAirPlay |
- |---------------------------------------------------------------------------|------------------------------------------------------------------|
+MA_API ma_peak2_config ma_peak2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double q, double frequency);
- jack.pClientName
- The name of the client to pass to `jack_client_open()`.
+typedef struct
+{
+ ma_biquad bq;
+} ma_peak2;
- jack.tryStartServer
- Whether or not to try auto-starting the JACK server. Defaults to false.
+MA_API ma_result ma_peak2_get_heap_size(const ma_peak2_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_peak2_init_preallocated(const ma_peak2_config* pConfig, void* pHeap, ma_peak2* pFilter);
+MA_API ma_result ma_peak2_init(const ma_peak2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_peak2* pFilter);
+MA_API void ma_peak2_uninit(ma_peak2* pFilter, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_peak2_reinit(const ma_peak2_config* pConfig, ma_peak2* pFilter);
+MA_API ma_result ma_peak2_process_pcm_frames(ma_peak2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
+MA_API ma_uint32 ma_peak2_get_latency(const ma_peak2* pFilter);
-It is recommended that only a single context is active at any given time because it's a bulky data structure which performs run-time linking for the
-relevant backends every time it's initialized.
+/**************************************************************************************************************************************************************
-The location of the context cannot change throughout it's lifetime. Consider allocating the `ma_context` object with `malloc()` if this is an issue. The
-reason for this is that a pointer to the context is stored in the `ma_device` structure.
+Low Shelf Filter
+**************************************************************************************************************************************************************/
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ double gainDB;
+ double shelfSlope;
+ double frequency;
+} ma_loshelf2_config, ma_loshelf_config;
-Example 1 - Default Initialization
-----------------------------------
-The example below shows how to initialize the context using the default configuration.
+MA_API ma_loshelf2_config ma_loshelf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double shelfSlope, double frequency);
-```c
-ma_context context;
-ma_result result = ma_context_init(NULL, 0, NULL, &context);
-if (result != MA_SUCCESS) {
- // Error.
-}
-```
+typedef struct
+{
+ ma_biquad bq;
+} ma_loshelf2;
+MA_API ma_result ma_loshelf2_get_heap_size(const ma_loshelf2_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_loshelf2_init_preallocated(const ma_loshelf2_config* pConfig, void* pHeap, ma_loshelf2* pFilter);
+MA_API ma_result ma_loshelf2_init(const ma_loshelf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_loshelf2* pFilter);
+MA_API void ma_loshelf2_uninit(ma_loshelf2* pFilter, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_loshelf2_reinit(const ma_loshelf2_config* pConfig, ma_loshelf2* pFilter);
+MA_API ma_result ma_loshelf2_process_pcm_frames(ma_loshelf2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
+MA_API ma_uint32 ma_loshelf2_get_latency(const ma_loshelf2* pFilter);
-Example 2 - Custom Configuration
---------------------------------
-The example below shows how to initialize the context using custom backend priorities and a custom configuration. In this hypothetical example, the program
-wants to prioritize ALSA over PulseAudio on Linux. They also want to avoid using the WinMM backend on Windows because it's latency is too high. They also
-want an error to be returned if no valid backend is available which they achieve by excluding the Null backend.
-For the configuration, the program wants to capture any log messages so they can, for example, route it to a log file and user interface.
+/**************************************************************************************************************************************************************
-```c
-ma_backend backends[] = {
- ma_backend_alsa,
- ma_backend_pulseaudio,
- ma_backend_wasapi,
- ma_backend_dsound
-};
+High Shelf Filter
-ma_context_config config = ma_context_config_init();
-config.logCallback = my_log_callback;
-config.pUserData = pMyUserData;
+**************************************************************************************************************************************************************/
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ double gainDB;
+ double shelfSlope;
+ double frequency;
+} ma_hishelf2_config, ma_hishelf_config;
-ma_context context;
-ma_result result = ma_context_init(backends, sizeof(backends)/sizeof(backends[0]), &config, &context);
-if (result != MA_SUCCESS) {
- // Error.
- if (result == MA_NO_BACKEND) {
- // Couldn't find an appropriate backend.
- }
-}
-```
+MA_API ma_hishelf2_config ma_hishelf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double shelfSlope, double frequency);
+typedef struct
+{
+ ma_biquad bq;
+} ma_hishelf2;
-See Also
---------
-ma_context_config_init()
-ma_context_uninit()
-*/
-MA_API ma_result ma_context_init(const ma_backend backends[], ma_uint32 backendCount, const ma_context_config* pConfig, ma_context* pContext);
+MA_API ma_result ma_hishelf2_get_heap_size(const ma_hishelf2_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_hishelf2_init_preallocated(const ma_hishelf2_config* pConfig, void* pHeap, ma_hishelf2* pFilter);
+MA_API ma_result ma_hishelf2_init(const ma_hishelf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hishelf2* pFilter);
+MA_API void ma_hishelf2_uninit(ma_hishelf2* pFilter, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_hishelf2_reinit(const ma_hishelf2_config* pConfig, ma_hishelf2* pFilter);
+MA_API ma_result ma_hishelf2_process_pcm_frames(ma_hishelf2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
+MA_API ma_uint32 ma_hishelf2_get_latency(const ma_hishelf2* pFilter);
-/*
-Uninitializes a context.
-Return Value
-------------
-MA_SUCCESS if successful; any other error code otherwise.
+/*
+Delay
+*/
+typedef struct
+{
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ ma_uint32 delayInFrames;
+ ma_bool32 delayStart; /* Set to true to delay the start of the output; false otherwise. */
+ float wet; /* 0..1. Default = 1. */
+ float dry; /* 0..1. Default = 1. */
+ float decay; /* 0..1. Default = 0 (no feedback). Feedback decay. Use this for echo. */
+} ma_delay_config;
+MA_API ma_delay_config ma_delay_config_init(ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 delayInFrames, float decay);
-Thread Safety
--------------
-Unsafe. Do not call this function across multiple threads as some backends read and write to global state.
+typedef struct
+{
+ ma_delay_config config;
+ ma_uint32 cursor; /* Feedback is written to this cursor. Always equal or in front of the read cursor. */
+ ma_uint32 bufferSizeInFrames; /* The maximum of config.startDelayInFrames and config.feedbackDelayInFrames. */
+ float* pBuffer;
+} ma_delay;
-Remarks
--------
-Results are undefined if you call this while any device created by this context is still active.
+MA_API ma_result ma_delay_init(const ma_delay_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_delay* pDelay);
+MA_API void ma_delay_uninit(ma_delay* pDelay, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_delay_process_pcm_frames(ma_delay* pDelay, void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount);
+MA_API void ma_delay_set_wet(ma_delay* pDelay, float value);
+MA_API float ma_delay_get_wet(const ma_delay* pDelay);
+MA_API void ma_delay_set_dry(ma_delay* pDelay, float value);
+MA_API float ma_delay_get_dry(const ma_delay* pDelay);
+MA_API void ma_delay_set_decay(ma_delay* pDelay, float value);
+MA_API float ma_delay_get_decay(const ma_delay* pDelay);
-See Also
---------
-ma_context_init()
-*/
-MA_API ma_result ma_context_uninit(ma_context* pContext);
+/* Gainer for smooth volume changes. */
+typedef struct
+{
+ ma_uint32 channels;
+ ma_uint32 smoothTimeInFrames;
+} ma_gainer_config;
-/*
-Retrieves the size of the ma_context object.
+MA_API ma_gainer_config ma_gainer_config_init(ma_uint32 channels, ma_uint32 smoothTimeInFrames);
-This is mainly for the purpose of bindings to know how much memory to allocate.
-*/
-MA_API size_t ma_context_sizeof(void);
-/*
-Retrieves a pointer to the log object associated with this context.
+typedef struct
+{
+ ma_gainer_config config;
+ ma_uint32 t;
+ float* pOldGains;
+ float* pNewGains;
+ /* Memory management. */
+ void* _pHeap;
+ ma_bool32 _ownsHeap;
+} ma_gainer;
-Remarks
--------
-Pass the returned pointer to `ma_log_post()`, `ma_log_postv()` or `ma_log_postf()` to post a log
-message.
+MA_API ma_result ma_gainer_get_heap_size(const ma_gainer_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_gainer_init_preallocated(const ma_gainer_config* pConfig, void* pHeap, ma_gainer* pGainer);
+MA_API ma_result ma_gainer_init(const ma_gainer_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_gainer* pGainer);
+MA_API void ma_gainer_uninit(ma_gainer* pGainer, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_gainer_process_pcm_frames(ma_gainer* pGainer, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
+MA_API ma_result ma_gainer_set_gain(ma_gainer* pGainer, float newGain);
+MA_API ma_result ma_gainer_set_gains(ma_gainer* pGainer, float* pNewGains);
-Return Value
-------------
-A pointer to the `ma_log` object that the context uses to post log messages. If some error occurs,
-NULL will be returned.
-*/
-MA_API ma_log* ma_context_get_log(ma_context* pContext);
-/*
-Enumerates over every device (both playback and capture).
+/* Stereo panner. */
+typedef enum
+{
+ ma_pan_mode_balance = 0, /* Does not blend one side with the other. Technically just a balance. Compatible with other popular audio engines and therefore the default. */
+ ma_pan_mode_pan /* A true pan. The sound from one side will "move" to the other side and blend with it. */
+} ma_pan_mode;
-This is a lower-level enumeration function to the easier to use `ma_context_get_devices()`. Use `ma_context_enumerate_devices()` if you would rather not incur
-an internal heap allocation, or it simply suits your code better.
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ ma_pan_mode mode;
+ float pan;
+} ma_panner_config;
-Note that this only retrieves the ID and name/description of the device. The reason for only retrieving basic information is that it would otherwise require
-opening the backend device in order to probe it for more detailed information which can be inefficient. Consider using `ma_context_get_device_info()` for this,
-but don't call it from within the enumeration callback.
+MA_API ma_panner_config ma_panner_config_init(ma_format format, ma_uint32 channels);
-Returning false from the callback will stop enumeration. Returning true will continue enumeration.
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ ma_pan_mode mode;
+ float pan; /* -1..1 where 0 is no pan, -1 is left side, +1 is right side. Defaults to 0. */
+} ma_panner;
-Parameters
-----------
-pContext (in)
- A pointer to the context performing the enumeration.
+MA_API ma_result ma_panner_init(const ma_panner_config* pConfig, ma_panner* pPanner);
+MA_API ma_result ma_panner_process_pcm_frames(ma_panner* pPanner, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
+MA_API void ma_panner_set_mode(ma_panner* pPanner, ma_pan_mode mode);
+MA_API ma_pan_mode ma_panner_get_mode(const ma_panner* pPanner);
+MA_API void ma_panner_set_pan(ma_panner* pPanner, float pan);
+MA_API float ma_panner_get_pan(const ma_panner* pPanner);
-callback (in)
- The callback to fire for each enumerated device.
-pUserData (in)
- A pointer to application-defined data passed to the callback.
+/* Fader. */
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+} ma_fader_config;
-Return Value
-------------
-MA_SUCCESS if successful; any other error code otherwise.
+MA_API ma_fader_config ma_fader_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate);
+typedef struct
+{
+ ma_fader_config config;
+ float volumeBeg; /* If volumeBeg and volumeEnd is equal to 1, no fading happens (ma_fader_process_pcm_frames() will run as a passthrough). */
+ float volumeEnd;
+ ma_uint64 lengthInFrames; /* The total length of the fade. */
+ ma_uint64 cursorInFrames; /* The current time in frames. Incremented by ma_fader_process_pcm_frames(). */
+} ma_fader;
-Thread Safety
--------------
-Safe. This is guarded using a simple mutex lock.
+MA_API ma_result ma_fader_init(const ma_fader_config* pConfig, ma_fader* pFader);
+MA_API ma_result ma_fader_process_pcm_frames(ma_fader* pFader, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
+MA_API void ma_fader_get_data_format(const ma_fader* pFader, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate);
+MA_API void ma_fader_set_fade(ma_fader* pFader, float volumeBeg, float volumeEnd, ma_uint64 lengthInFrames);
+MA_API float ma_fader_get_current_volume(ma_fader* pFader);
-Remarks
--------
-Do _not_ assume the first enumerated device of a given type is the default device.
-Some backends and platforms may only support default playback and capture devices.
+/* Spatializer. */
+typedef struct
+{
+ float x;
+ float y;
+ float z;
+} ma_vec3f;
-In general, you should not do anything complicated from within the callback. In particular, do not try initializing a device from within the callback. Also,
-do not try to call `ma_context_get_device_info()` from within the callback.
+typedef enum
+{
+ ma_attenuation_model_none, /* No distance attenuation and no spatialization. */
+ ma_attenuation_model_inverse, /* Equivalent to OpenAL's AL_INVERSE_DISTANCE_CLAMPED. */
+ ma_attenuation_model_linear, /* Linear attenuation. Equivalent to OpenAL's AL_LINEAR_DISTANCE_CLAMPED. */
+ ma_attenuation_model_exponential /* Exponential attenuation. Equivalent to OpenAL's AL_EXPONENT_DISTANCE_CLAMPED. */
+} ma_attenuation_model;
-Consider using `ma_context_get_devices()` for a simpler and safer API, albeit at the expense of an internal heap allocation.
+typedef enum
+{
+ ma_positioning_absolute,
+ ma_positioning_relative
+} ma_positioning;
+
+typedef enum
+{
+ ma_handedness_right,
+ ma_handedness_left
+} ma_handedness;
-Example 1 - Simple Enumeration
-------------------------------
-ma_bool32 ma_device_enum_callback(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pInfo, void* pUserData)
+typedef struct
{
- printf("Device Name: %s\n", pInfo->name);
- return MA_TRUE;
-}
+ ma_uint32 channelsOut;
+ ma_channel* pChannelMapOut;
+ ma_handedness handedness; /* Defaults to right. Forward is -1 on the Z axis. In a left handed system, forward is +1 on the Z axis. */
+ float coneInnerAngleInRadians;
+ float coneOuterAngleInRadians;
+ float coneOuterGain;
+ float speedOfSound;
+ ma_vec3f worldUp;
+} ma_spatializer_listener_config;
-ma_result result = ma_context_enumerate_devices(&context, my_device_enum_callback, pMyUserData);
-if (result != MA_SUCCESS) {
- // Error.
-}
+MA_API ma_spatializer_listener_config ma_spatializer_listener_config_init(ma_uint32 channelsOut);
-See Also
---------
-ma_context_get_devices()
-*/
-MA_API ma_result ma_context_enumerate_devices(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData);
+typedef struct
+{
+ ma_spatializer_listener_config config;
+ ma_vec3f position; /* The absolute position of the listener. */
+ ma_vec3f direction; /* The direction the listener is facing. The world up vector is config.worldUp. */
+ ma_vec3f velocity;
+ ma_bool32 isEnabled;
+
+ /* Memory management. */
+ ma_bool32 _ownsHeap;
+ void* _pHeap;
+} ma_spatializer_listener;
+
+MA_API ma_result ma_spatializer_listener_get_heap_size(const ma_spatializer_listener_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_spatializer_listener_init_preallocated(const ma_spatializer_listener_config* pConfig, void* pHeap, ma_spatializer_listener* pListener);
+MA_API ma_result ma_spatializer_listener_init(const ma_spatializer_listener_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_spatializer_listener* pListener);
+MA_API void ma_spatializer_listener_uninit(ma_spatializer_listener* pListener, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_channel* ma_spatializer_listener_get_channel_map(ma_spatializer_listener* pListener);
+MA_API void ma_spatializer_listener_set_cone(ma_spatializer_listener* pListener, float innerAngleInRadians, float outerAngleInRadians, float outerGain);
+MA_API void ma_spatializer_listener_get_cone(const ma_spatializer_listener* pListener, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain);
+MA_API void ma_spatializer_listener_set_position(ma_spatializer_listener* pListener, float x, float y, float z);
+MA_API ma_vec3f ma_spatializer_listener_get_position(const ma_spatializer_listener* pListener);
+MA_API void ma_spatializer_listener_set_direction(ma_spatializer_listener* pListener, float x, float y, float z);
+MA_API ma_vec3f ma_spatializer_listener_get_direction(const ma_spatializer_listener* pListener);
+MA_API void ma_spatializer_listener_set_velocity(ma_spatializer_listener* pListener, float x, float y, float z);
+MA_API ma_vec3f ma_spatializer_listener_get_velocity(const ma_spatializer_listener* pListener);
+MA_API void ma_spatializer_listener_set_speed_of_sound(ma_spatializer_listener* pListener, float speedOfSound);
+MA_API float ma_spatializer_listener_get_speed_of_sound(const ma_spatializer_listener* pListener);
+MA_API void ma_spatializer_listener_set_world_up(ma_spatializer_listener* pListener, float x, float y, float z);
+MA_API ma_vec3f ma_spatializer_listener_get_world_up(const ma_spatializer_listener* pListener);
+MA_API void ma_spatializer_listener_set_enabled(ma_spatializer_listener* pListener, ma_bool32 isEnabled);
+MA_API ma_bool32 ma_spatializer_listener_is_enabled(const ma_spatializer_listener* pListener);
-/*
-Retrieves basic information about every active playback and/or capture device.
-This function will allocate memory internally for the device lists and return a pointer to them through the `ppPlaybackDeviceInfos` and `ppCaptureDeviceInfos`
-parameters. If you do not want to incur the overhead of these allocations consider using `ma_context_enumerate_devices()` which will instead use a callback.
+typedef struct
+{
+ ma_uint32 channelsIn;
+ ma_uint32 channelsOut;
+ ma_channel* pChannelMapIn;
+ ma_attenuation_model attenuationModel;
+ ma_positioning positioning;
+ ma_handedness handedness; /* Defaults to right. Forward is -1 on the Z axis. In a left handed system, forward is +1 on the Z axis. */
+ float minGain;
+ float maxGain;
+ float minDistance;
+ float maxDistance;
+ float rolloff;
+ float coneInnerAngleInRadians;
+ float coneOuterAngleInRadians;
+ float coneOuterGain;
+ float dopplerFactor; /* Set to 0 to disable doppler effect. */
+ float directionalAttenuationFactor; /* Set to 0 to disable directional attenuation. */
+ ma_uint32 gainSmoothTimeInFrames; /* When the gain of a channel changes during spatialization, the transition will be linearly interpolated over this number of frames. */
+} ma_spatializer_config;
+
+MA_API ma_spatializer_config ma_spatializer_config_init(ma_uint32 channelsIn, ma_uint32 channelsOut);
-Parameters
-----------
-pContext (in)
- A pointer to the context performing the enumeration.
+typedef struct
+{
+ ma_uint32 channelsIn;
+ ma_uint32 channelsOut;
+ ma_channel* pChannelMapIn;
+ ma_attenuation_model attenuationModel;
+ ma_positioning positioning;
+ ma_handedness handedness; /* Defaults to right. Forward is -1 on the Z axis. In a left handed system, forward is +1 on the Z axis. */
+ float minGain;
+ float maxGain;
+ float minDistance;
+ float maxDistance;
+ float rolloff;
+ float coneInnerAngleInRadians;
+ float coneOuterAngleInRadians;
+ float coneOuterGain;
+ float dopplerFactor; /* Set to 0 to disable doppler effect. */
+ float directionalAttenuationFactor; /* Set to 0 to disable directional attenuation. */
+ ma_uint32 gainSmoothTimeInFrames; /* When the gain of a channel changes during spatialization, the transition will be linearly interpolated over this number of frames. */
+ ma_vec3f position;
+ ma_vec3f direction;
+ ma_vec3f velocity; /* For doppler effect. */
+ float dopplerPitch; /* Will be updated by ma_spatializer_process_pcm_frames() and can be used by higher level functions to apply a pitch shift for doppler effect. */
+ ma_gainer gainer; /* For smooth gain transitions. */
+ float* pNewChannelGainsOut; /* An offset of _pHeap. Used by ma_spatializer_process_pcm_frames() to store new channel gains. The number of elements in this array is equal to config.channelsOut. */
+
+ /* Memory management. */
+ void* _pHeap;
+ ma_bool32 _ownsHeap;
+} ma_spatializer;
+
+MA_API ma_result ma_spatializer_get_heap_size(const ma_spatializer_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_spatializer_init_preallocated(const ma_spatializer_config* pConfig, void* pHeap, ma_spatializer* pSpatializer);
+MA_API ma_result ma_spatializer_init(const ma_spatializer_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_spatializer* pSpatializer);
+MA_API void ma_spatializer_uninit(ma_spatializer* pSpatializer, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_spatializer_process_pcm_frames(ma_spatializer* pSpatializer, ma_spatializer_listener* pListener, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
+MA_API ma_uint32 ma_spatializer_get_input_channels(const ma_spatializer* pSpatializer);
+MA_API ma_uint32 ma_spatializer_get_output_channels(const ma_spatializer* pSpatializer);
+MA_API void ma_spatializer_set_attenuation_model(ma_spatializer* pSpatializer, ma_attenuation_model attenuationModel);
+MA_API ma_attenuation_model ma_spatializer_get_attenuation_model(const ma_spatializer* pSpatializer);
+MA_API void ma_spatializer_set_positioning(ma_spatializer* pSpatializer, ma_positioning positioning);
+MA_API ma_positioning ma_spatializer_get_positioning(const ma_spatializer* pSpatializer);
+MA_API void ma_spatializer_set_rolloff(ma_spatializer* pSpatializer, float rolloff);
+MA_API float ma_spatializer_get_rolloff(const ma_spatializer* pSpatializer);
+MA_API void ma_spatializer_set_min_gain(ma_spatializer* pSpatializer, float minGain);
+MA_API float ma_spatializer_get_min_gain(const ma_spatializer* pSpatializer);
+MA_API void ma_spatializer_set_max_gain(ma_spatializer* pSpatializer, float maxGain);
+MA_API float ma_spatializer_get_max_gain(const ma_spatializer* pSpatializer);
+MA_API void ma_spatializer_set_min_distance(ma_spatializer* pSpatializer, float minDistance);
+MA_API float ma_spatializer_get_min_distance(const ma_spatializer* pSpatializer);
+MA_API void ma_spatializer_set_max_distance(ma_spatializer* pSpatializer, float maxDistance);
+MA_API float ma_spatializer_get_max_distance(const ma_spatializer* pSpatializer);
+MA_API void ma_spatializer_set_cone(ma_spatializer* pSpatializer, float innerAngleInRadians, float outerAngleInRadians, float outerGain);
+MA_API void ma_spatializer_get_cone(const ma_spatializer* pSpatializer, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain);
+MA_API void ma_spatializer_set_doppler_factor(ma_spatializer* pSpatializer, float dopplerFactor);
+MA_API float ma_spatializer_get_doppler_factor(const ma_spatializer* pSpatializer);
+MA_API void ma_spatializer_set_directional_attenuation_factor(ma_spatializer* pSpatializer, float directionalAttenuationFactor);
+MA_API float ma_spatializer_get_directional_attenuation_factor(const ma_spatializer* pSpatializer);
+MA_API void ma_spatializer_set_position(ma_spatializer* pSpatializer, float x, float y, float z);
+MA_API ma_vec3f ma_spatializer_get_position(const ma_spatializer* pSpatializer);
+MA_API void ma_spatializer_set_direction(ma_spatializer* pSpatializer, float x, float y, float z);
+MA_API ma_vec3f ma_spatializer_get_direction(const ma_spatializer* pSpatializer);
+MA_API void ma_spatializer_set_velocity(ma_spatializer* pSpatializer, float x, float y, float z);
+MA_API ma_vec3f ma_spatializer_get_velocity(const ma_spatializer* pSpatializer);
+MA_API void ma_spatializer_get_relative_position_and_direction(const ma_spatializer* pSpatializer, const ma_spatializer_listener* pListener, ma_vec3f* pRelativePos, ma_vec3f* pRelativeDir);
-ppPlaybackDeviceInfos (out)
- A pointer to a pointer that will receive the address of a buffer containing the list of `ma_device_info` structures for playback devices.
-pPlaybackDeviceCount (out)
- A pointer to an unsigned integer that will receive the number of playback devices.
-ppCaptureDeviceInfos (out)
- A pointer to a pointer that will receive the address of a buffer containing the list of `ma_device_info` structures for capture devices.
+/************************************************************************************************************************************************************
+*************************************************************************************************************************************************************
-pCaptureDeviceCount (out)
- A pointer to an unsigned integer that will receive the number of capture devices.
+DATA CONVERSION
+===============
+This section contains the APIs for data conversion. You will find everything here for channel mapping, sample format conversion, resampling, etc.
-Return Value
-------------
-MA_SUCCESS if successful; any other error code otherwise.
+*************************************************************************************************************************************************************
+************************************************************************************************************************************************************/
+/**************************************************************************************************************************************************************
-Thread Safety
--------------
-Unsafe. Since each call to this function invalidates the pointers from the previous call, you should not be calling this simultaneously across multiple
-threads. Instead, you need to make a copy of the returned data with your own higher level synchronization.
+Resampling
+**************************************************************************************************************************************************************/
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRateIn;
+ ma_uint32 sampleRateOut;
+ ma_uint32 lpfOrder; /* The low-pass filter order. Setting this to 0 will disable low-pass filtering. */
+ double lpfNyquistFactor; /* 0..1. Defaults to 1. 1 = Half the sampling frequency (Nyquist Frequency), 0.5 = Quarter the sampling frequency (half Nyquest Frequency), etc. */
+} ma_linear_resampler_config;
-Remarks
--------
-It is _not_ safe to assume the first device in the list is the default device.
+MA_API ma_linear_resampler_config ma_linear_resampler_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut);
-You can pass in NULL for the playback or capture lists in which case they'll be ignored.
+typedef struct
+{
+ ma_linear_resampler_config config;
+ ma_uint32 inAdvanceInt;
+ ma_uint32 inAdvanceFrac;
+ ma_uint32 inTimeInt;
+ ma_uint32 inTimeFrac;
+ union
+ {
+ float* f32;
+ ma_int16* s16;
+ } x0; /* The previous input frame. */
+ union
+ {
+ float* f32;
+ ma_int16* s16;
+ } x1; /* The next input frame. */
+ ma_lpf lpf;
-The returned pointers will become invalid upon the next call this this function, or when the context is uninitialized. Do not free the returned pointers.
+ /* Memory management. */
+ void* _pHeap;
+ ma_bool32 _ownsHeap;
+} ma_linear_resampler;
+MA_API ma_result ma_linear_resampler_get_heap_size(const ma_linear_resampler_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_linear_resampler_init_preallocated(const ma_linear_resampler_config* pConfig, void* pHeap, ma_linear_resampler* pResampler);
+MA_API ma_result ma_linear_resampler_init(const ma_linear_resampler_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_linear_resampler* pResampler);
+MA_API void ma_linear_resampler_uninit(ma_linear_resampler* pResampler, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_linear_resampler_process_pcm_frames(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut);
+MA_API ma_result ma_linear_resampler_set_rate(ma_linear_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut);
+MA_API ma_result ma_linear_resampler_set_rate_ratio(ma_linear_resampler* pResampler, float ratioInOut);
+MA_API ma_uint64 ma_linear_resampler_get_input_latency(const ma_linear_resampler* pResampler);
+MA_API ma_uint64 ma_linear_resampler_get_output_latency(const ma_linear_resampler* pResampler);
+MA_API ma_result ma_linear_resampler_get_required_input_frame_count(const ma_linear_resampler* pResampler, ma_uint64 outputFrameCount, ma_uint64* pInputFrameCount);
+MA_API ma_result ma_linear_resampler_get_expected_output_frame_count(const ma_linear_resampler* pResampler, ma_uint64 inputFrameCount, ma_uint64* pOutputFrameCount);
-See Also
---------
-ma_context_get_devices()
-*/
-MA_API ma_result ma_context_get_devices(ma_context* pContext, ma_device_info** ppPlaybackDeviceInfos, ma_uint32* pPlaybackDeviceCount, ma_device_info** ppCaptureDeviceInfos, ma_uint32* pCaptureDeviceCount);
-/*
-Retrieves information about a device of the given type, with the specified ID and share mode.
+typedef struct ma_resampler_config ma_resampler_config;
+typedef void ma_resampling_backend;
+typedef struct
+{
+ ma_result (* onGetHeapSize )(void* pUserData, const ma_resampler_config* pConfig, size_t* pHeapSizeInBytes);
+ ma_result (* onInit )(void* pUserData, const ma_resampler_config* pConfig, void* pHeap, ma_resampling_backend** ppBackend);
+ void (* onUninit )(void* pUserData, ma_resampling_backend* pBackend, const ma_allocation_callbacks* pAllocationCallbacks);
+ ma_result (* onProcess )(void* pUserData, ma_resampling_backend* pBackend, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut);
+ ma_result (* onSetRate )(void* pUserData, ma_resampling_backend* pBackend, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut); /* Optional. Rate changes will be disabled. */
+ ma_uint64 (* onGetInputLatency )(void* pUserData, const ma_resampling_backend* pBackend); /* Optional. Latency will be reported as 0. */
+ ma_uint64 (* onGetOutputLatency )(void* pUserData, const ma_resampling_backend* pBackend); /* Optional. Latency will be reported as 0. */
+ ma_result (* onGetRequiredInputFrameCount )(void* pUserData, const ma_resampling_backend* pBackend, ma_uint64 outputFrameCount, ma_uint64* pInputFrameCount); /* Optional. Latency mitigation will be disabled. */
+ ma_result (* onGetExpectedOutputFrameCount)(void* pUserData, const ma_resampling_backend* pBackend, ma_uint64 inputFrameCount, ma_uint64* pOutputFrameCount); /* Optional. Latency mitigation will be disabled. */
+} ma_resampling_backend_vtable;
-Parameters
-----------
-pContext (in)
- A pointer to the context performing the query.
+typedef enum
+{
+ ma_resample_algorithm_linear = 0, /* Fastest, lowest quality. Optional low-pass filtering. Default. */
+ ma_resample_algorithm_custom,
+} ma_resample_algorithm;
-deviceType (in)
- The type of the device being queried. Must be either `ma_device_type_playback` or `ma_device_type_capture`.
+struct ma_resampler_config
+{
+ ma_format format; /* Must be either ma_format_f32 or ma_format_s16. */
+ ma_uint32 channels;
+ ma_uint32 sampleRateIn;
+ ma_uint32 sampleRateOut;
+ ma_resample_algorithm algorithm; /* When set to ma_resample_algorithm_custom, pBackendVTable will be used. */
+ ma_resampling_backend_vtable* pBackendVTable;
+ void* pBackendUserData;
+ struct
+ {
+ ma_uint32 lpfOrder;
+ } linear;
+};
-pDeviceID (in)
- The ID of the device being queried.
+MA_API ma_resampler_config ma_resampler_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut, ma_resample_algorithm algorithm);
-shareMode (in)
- The share mode to query for device capabilities. This should be set to whatever you're intending on using when initializing the device. If you're unsure,
- set this to `ma_share_mode_shared`.
+typedef struct
+{
+ ma_resampling_backend* pBackend;
+ ma_resampling_backend_vtable* pBackendVTable;
+ void* pBackendUserData;
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRateIn;
+ ma_uint32 sampleRateOut;
+ union
+ {
+ ma_linear_resampler linear;
+ } state; /* State for stock resamplers so we can avoid a malloc. For stock resamplers, pBackend will point here. */
-pDeviceInfo (out)
- A pointer to the `ma_device_info` structure that will receive the device information.
+ /* Memory management. */
+ void* _pHeap;
+ ma_bool32 _ownsHeap;
+} ma_resampler;
+MA_API ma_result ma_resampler_get_heap_size(const ma_resampler_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_resampler_init_preallocated(const ma_resampler_config* pConfig, void* pHeap, ma_resampler* pResampler);
-Return Value
-------------
-MA_SUCCESS if successful; any other error code otherwise.
+/*
+Initializes a new resampler object from a config.
+*/
+MA_API ma_result ma_resampler_init(const ma_resampler_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_resampler* pResampler);
+/*
+Uninitializes a resampler.
+*/
+MA_API void ma_resampler_uninit(ma_resampler* pResampler, const ma_allocation_callbacks* pAllocationCallbacks);
-Thread Safety
--------------
-Safe. This is guarded using a simple mutex lock.
+/*
+Converts the given input data.
+Both the input and output frames must be in the format specified in the config when the resampler was initilized.
-Remarks
--------
-Do _not_ call this from within the `ma_context_enumerate_devices()` callback.
+On input, [pFrameCountOut] contains the number of output frames to process. On output it contains the number of output frames that
+were actually processed, which may be less than the requested amount which will happen if there's not enough input data. You can use
+ma_resampler_get_expected_output_frame_count() to know how many output frames will be processed for a given number of input frames.
-It's possible for a device to have different information and capabilities depending on whether or not it's opened in shared or exclusive mode. For example, in
-shared mode, WASAPI always uses floating point samples for mixing, but in exclusive mode it can be anything. Therefore, this function allows you to specify
-which share mode you want information for. Note that not all backends and devices support shared or exclusive mode, in which case this function will fail if
-the requested share mode is unsupported.
+On input, [pFrameCountIn] contains the number of input frames contained in [pFramesIn]. On output it contains the number of whole
+input frames that were actually processed. You can use ma_resampler_get_required_input_frame_count() to know how many input frames
+you should provide for a given number of output frames. [pFramesIn] can be NULL, in which case zeroes will be used instead.
-This leaves pDeviceInfo unmodified in the result of an error.
+If [pFramesOut] is NULL, a seek is performed. In this case, if [pFrameCountOut] is not NULL it will seek by the specified number of
+output frames. Otherwise, if [pFramesCountOut] is NULL and [pFrameCountIn] is not NULL, it will seek by the specified number of input
+frames. When seeking, [pFramesIn] is allowed to NULL, in which case the internal timing state will be updated, but no input will be
+processed. In this case, any internal filter state will be updated as if zeroes were passed in.
+
+It is an error for [pFramesOut] to be non-NULL and [pFrameCountOut] to be NULL.
+
+It is an error for both [pFrameCountOut] and [pFrameCountIn] to be NULL.
*/
-MA_API ma_result ma_context_get_device_info(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo);
+MA_API ma_result ma_resampler_process_pcm_frames(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut);
+
/*
-Determines if the given context supports loopback mode.
+Sets the input and output sample sample rate.
+*/
+MA_API ma_result ma_resampler_set_rate(ma_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut);
+/*
+Sets the input and output sample rate as a ratio.
-Parameters
-----------
-pContext (in)
- A pointer to the context getting queried.
+The ration is in/out.
+*/
+MA_API ma_result ma_resampler_set_rate_ratio(ma_resampler* pResampler, float ratio);
+/*
+Retrieves the latency introduced by the resampler in input frames.
+*/
+MA_API ma_uint64 ma_resampler_get_input_latency(const ma_resampler* pResampler);
-Return Value
-------------
-MA_TRUE if the context supports loopback mode; MA_FALSE otherwise.
+/*
+Retrieves the latency introduced by the resampler in output frames.
*/
-MA_API ma_bool32 ma_context_is_loopback_supported(ma_context* pContext);
+MA_API ma_uint64 ma_resampler_get_output_latency(const ma_resampler* pResampler);
+/*
+Calculates the number of whole input frames that would need to be read from the client in order to output the specified
+number of output frames.
+The returned value does not include cached input frames. It only returns the number of extra frames that would need to be
+read from the input buffer in order to output the specified number of output frames.
+*/
+MA_API ma_result ma_resampler_get_required_input_frame_count(const ma_resampler* pResampler, ma_uint64 outputFrameCount, ma_uint64* pInputFrameCount);
/*
-Initializes a device config with default settings.
+Calculates the number of whole output frames that would be output after fully reading and consuming the specified number of
+input frames.
+*/
+MA_API ma_result ma_resampler_get_expected_output_frame_count(const ma_resampler* pResampler, ma_uint64 inputFrameCount, ma_uint64* pOutputFrameCount);
-Parameters
-----------
-deviceType (in)
- The type of the device this config is being initialized for. This must set to one of the following:
+/**************************************************************************************************************************************************************
- |-------------------------|
- | Device Type |
- |-------------------------|
- | ma_device_type_playback |
- | ma_device_type_capture |
- | ma_device_type_duplex |
- | ma_device_type_loopback |
- |-------------------------|
+Channel Conversion
+**************************************************************************************************************************************************************/
+typedef enum
+{
+ ma_channel_conversion_path_unknown,
+ ma_channel_conversion_path_passthrough,
+ ma_channel_conversion_path_mono_out, /* Converting to mono. */
+ ma_channel_conversion_path_mono_in, /* Converting from mono. */
+ ma_channel_conversion_path_shuffle, /* Simple shuffle. Will use this when all channels are present in both input and output channel maps, but just in a different order. */
+ ma_channel_conversion_path_weights /* Blended based on weights. */
+} ma_channel_conversion_path;
-Return Value
-------------
-A new device config object with default settings. You will typically want to adjust the config after this function returns. See remarks.
+typedef enum
+{
+ ma_mono_expansion_mode_duplicate = 0, /* The default. */
+ ma_mono_expansion_mode_average, /* Average the mono channel across all channels. */
+ ma_mono_expansion_mode_stereo_only, /* Duplicate to the left and right channels only and ignore the others. */
+ ma_mono_expansion_mode_default = ma_mono_expansion_mode_duplicate
+} ma_mono_expansion_mode;
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channelsIn;
+ ma_uint32 channelsOut;
+ const ma_channel* pChannelMapIn;
+ const ma_channel* pChannelMapOut;
+ ma_channel_mix_mode mixingMode;
+ float** ppWeights; /* [in][out]. Only used when mixingMode is set to ma_channel_mix_mode_custom_weights. */
+} ma_channel_converter_config;
-Thread Safety
--------------
-Safe.
+MA_API ma_channel_converter_config ma_channel_converter_config_init(ma_format format, ma_uint32 channelsIn, const ma_channel* pChannelMapIn, ma_uint32 channelsOut, const ma_channel* pChannelMapOut, ma_channel_mix_mode mixingMode);
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channelsIn;
+ ma_uint32 channelsOut;
+ ma_channel_mix_mode mixingMode;
+ ma_channel_conversion_path conversionPath;
+ ma_channel* pChannelMapIn;
+ ma_channel* pChannelMapOut;
+ ma_uint8* pShuffleTable; /* Indexed by output channel index. */
+ union
+ {
+ float** f32;
+ ma_int32** s16;
+ } weights; /* [in][out] */
-Callback Safety
----------------
-Safe, but don't try initializing a device in a callback.
+ /* Memory management. */
+ void* _pHeap;
+ ma_bool32 _ownsHeap;
+} ma_channel_converter;
+MA_API ma_result ma_channel_converter_get_heap_size(const ma_channel_converter_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_channel_converter_init_preallocated(const ma_channel_converter_config* pConfig, void* pHeap, ma_channel_converter* pConverter);
+MA_API ma_result ma_channel_converter_init(const ma_channel_converter_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_channel_converter* pConverter);
+MA_API void ma_channel_converter_uninit(ma_channel_converter* pConverter, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_channel_converter_process_pcm_frames(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
+MA_API ma_result ma_channel_converter_get_input_channel_map(const ma_channel_converter* pConverter, ma_channel* pChannelMap, size_t channelMapCap);
+MA_API ma_result ma_channel_converter_get_output_channel_map(const ma_channel_converter* pConverter, ma_channel* pChannelMap, size_t channelMapCap);
-Remarks
--------
-The returned config will be initialized to defaults. You will normally want to customize a few variables before initializing the device. See Example 1 for a
-typical configuration which sets the sample format, channel count, sample rate, data callback and user data. These are usually things you will want to change
-before initializing the device.
-See `ma_device_init()` for details on specific configuration options.
+/**************************************************************************************************************************************************************
+Data Conversion
-Example 1 - Simple Configuration
---------------------------------
-The example below is what a program will typically want to configure for each device at a minimum. Notice how `ma_device_config_init()` is called first, and
-then the returned object is modified directly. This is important because it ensures that your program continues to work as new configuration options are added
-to the `ma_device_config` structure.
+**************************************************************************************************************************************************************/
+typedef struct
+{
+ ma_format formatIn;
+ ma_format formatOut;
+ ma_uint32 channelsIn;
+ ma_uint32 channelsOut;
+ ma_uint32 sampleRateIn;
+ ma_uint32 sampleRateOut;
+ ma_channel* pChannelMapIn;
+ ma_channel* pChannelMapOut;
+ ma_dither_mode ditherMode;
+ ma_channel_mix_mode channelMixMode;
+ float** ppChannelWeights; /* [in][out]. Only used when mixingMode is set to ma_channel_mix_mode_custom_weights. */
+ ma_bool32 allowDynamicSampleRate;
+ ma_resampler_config resampling;
+} ma_data_converter_config;
-```c
-ma_device_config config = ma_device_config_init(ma_device_type_playback);
-config.playback.format = ma_format_f32;
-config.playback.channels = 2;
-config.sampleRate = 48000;
-config.dataCallback = ma_data_callback;
-config.pUserData = pMyUserData;
-```
+MA_API ma_data_converter_config ma_data_converter_config_init_default(void);
+MA_API ma_data_converter_config ma_data_converter_config_init(ma_format formatIn, ma_format formatOut, ma_uint32 channelsIn, ma_uint32 channelsOut, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut);
-See Also
---------
-ma_device_init()
-ma_device_init_ex()
-*/
-MA_API ma_device_config ma_device_config_init(ma_device_type deviceType);
+typedef enum
+{
+ ma_data_converter_execution_path_passthrough, /* No conversion. */
+ ma_data_converter_execution_path_format_only, /* Only format conversion. */
+ ma_data_converter_execution_path_channels_only, /* Only channel conversion. */
+ ma_data_converter_execution_path_resample_only, /* Only resampling. */
+ ma_data_converter_execution_path_resample_first, /* All conversions, but resample as the first step. */
+ ma_data_converter_execution_path_channels_first /* All conversions, but channels as the first step. */
+} ma_data_converter_execution_path;
+typedef struct
+{
+ ma_format formatIn;
+ ma_format formatOut;
+ ma_uint32 channelsIn;
+ ma_uint32 channelsOut;
+ ma_uint32 sampleRateIn;
+ ma_uint32 sampleRateOut;
+ ma_dither_mode ditherMode;
+ ma_data_converter_execution_path executionPath; /* The execution path the data converter will follow when processing. */
+ ma_channel_converter channelConverter;
+ ma_resampler resampler;
+ ma_bool8 hasPreFormatConversion;
+ ma_bool8 hasPostFormatConversion;
+ ma_bool8 hasChannelConverter;
+ ma_bool8 hasResampler;
+ ma_bool8 isPassthrough;
-/*
-Initializes a device.
+ /* Memory management. */
+ ma_bool8 _ownsHeap;
+ void* _pHeap;
+} ma_data_converter;
-A device represents a physical audio device. The idea is you send or receive audio data from the device to either play it back through a speaker, or capture it
-from a microphone. Whether or not you should send or receive data from the device (or both) depends on the type of device you are initializing which can be
-playback, capture, full-duplex or loopback. (Note that loopback mode is only supported on select backends.) Sending and receiving audio data to and from the
-device is done via a callback which is fired by miniaudio at periodic time intervals.
+MA_API ma_result ma_data_converter_get_heap_size(const ma_data_converter_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_data_converter_init_preallocated(const ma_data_converter_config* pConfig, void* pHeap, ma_data_converter* pConverter);
+MA_API ma_result ma_data_converter_init(const ma_data_converter_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_converter* pConverter);
+MA_API void ma_data_converter_uninit(ma_data_converter* pConverter, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_data_converter_process_pcm_frames(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut);
+MA_API ma_result ma_data_converter_set_rate(ma_data_converter* pConverter, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut);
+MA_API ma_result ma_data_converter_set_rate_ratio(ma_data_converter* pConverter, float ratioInOut);
+MA_API ma_uint64 ma_data_converter_get_input_latency(const ma_data_converter* pConverter);
+MA_API ma_uint64 ma_data_converter_get_output_latency(const ma_data_converter* pConverter);
+MA_API ma_result ma_data_converter_get_required_input_frame_count(const ma_data_converter* pConverter, ma_uint64 outputFrameCount, ma_uint64* pInputFrameCount);
+MA_API ma_result ma_data_converter_get_expected_output_frame_count(const ma_data_converter* pConverter, ma_uint64 inputFrameCount, ma_uint64* pOutputFrameCount);
+MA_API ma_result ma_data_converter_get_input_channel_map(const ma_data_converter* pConverter, ma_channel* pChannelMap, size_t channelMapCap);
+MA_API ma_result ma_data_converter_get_output_channel_map(const ma_data_converter* pConverter, ma_channel* pChannelMap, size_t channelMapCap);
-The frequency at which data is delivered to and from a device depends on the size of it's period. The size of the period can be defined in terms of PCM frames
-or milliseconds, whichever is more convenient. Generally speaking, the smaller the period, the lower the latency at the expense of higher CPU usage and
-increased risk of glitching due to the more frequent and granular data deliver intervals. The size of a period will depend on your requirements, but
-miniaudio's defaults should work fine for most scenarios. If you're building a game you should leave this fairly small, whereas if you're building a simple
-media player you can make it larger. Note that the period size you request is actually just a hint - miniaudio will tell the backend what you want, but the
-backend is ultimately responsible for what it gives you. You cannot assume you will get exactly what you ask for.
-When delivering data to and from a device you need to make sure it's in the correct format which you can set through the device configuration. You just set the
-format that you want to use and miniaudio will perform all of the necessary conversion for you internally. When delivering data to and from the callback you
-can assume the format is the same as what you requested when you initialized the device. See Remarks for more details on miniaudio's data conversion pipeline.
+/************************************************************************************************************************************************************
+Format Conversion
-Parameters
-----------
-pContext (in, optional)
- A pointer to the context that owns the device. This can be null, in which case it creates a default context internally.
+************************************************************************************************************************************************************/
+MA_API void ma_pcm_u8_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
+MA_API void ma_pcm_u8_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
+MA_API void ma_pcm_u8_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
+MA_API void ma_pcm_u8_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
+MA_API void ma_pcm_s16_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
+MA_API void ma_pcm_s16_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
+MA_API void ma_pcm_s16_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
+MA_API void ma_pcm_s16_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
+MA_API void ma_pcm_s24_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
+MA_API void ma_pcm_s24_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
+MA_API void ma_pcm_s24_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
+MA_API void ma_pcm_s24_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
+MA_API void ma_pcm_s32_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
+MA_API void ma_pcm_s32_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
+MA_API void ma_pcm_s32_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
+MA_API void ma_pcm_s32_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
+MA_API void ma_pcm_f32_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
+MA_API void ma_pcm_f32_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
+MA_API void ma_pcm_f32_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
+MA_API void ma_pcm_f32_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
+MA_API void ma_pcm_convert(void* pOut, ma_format formatOut, const void* pIn, ma_format formatIn, ma_uint64 sampleCount, ma_dither_mode ditherMode);
+MA_API void ma_convert_pcm_frames_format(void* pOut, ma_format formatOut, const void* pIn, ma_format formatIn, ma_uint64 frameCount, ma_uint32 channels, ma_dither_mode ditherMode);
-pConfig (in)
- A pointer to the device configuration. Cannot be null. See remarks for details.
+/*
+Deinterleaves an interleaved buffer.
+*/
+MA_API void ma_deinterleave_pcm_frames(ma_format format, ma_uint32 channels, ma_uint64 frameCount, const void* pInterleavedPCMFrames, void** ppDeinterleavedPCMFrames);
-pDevice (out)
- A pointer to the device object being initialized.
+/*
+Interleaves a group of deinterleaved buffers.
+*/
+MA_API void ma_interleave_pcm_frames(ma_format format, ma_uint32 channels, ma_uint64 frameCount, const void** ppDeinterleavedPCMFrames, void* pInterleavedPCMFrames);
-Return Value
-------------
-MA_SUCCESS if successful; any other error code otherwise.
+/************************************************************************************************************************************************************
+Channel Maps
-Thread Safety
--------------
-Unsafe. It is not safe to call this function simultaneously for different devices because some backends depend on and mutate global state. The same applies to
-calling this at the same time as `ma_device_uninit()`.
+************************************************************************************************************************************************************/
+/*
+This is used in the shuffle table to indicate that the channel index is undefined and should be ignored.
+*/
+#define MA_CHANNEL_INDEX_NULL 255
+/*
+Retrieves the channel position of the specified channel in the given channel map.
-Callback Safety
----------------
-Unsafe. It is not safe to call this inside any callback.
+The pChannelMap parameter can be null, in which case miniaudio's default channel map will be assumed.
+*/
+MA_API ma_channel ma_channel_map_get_channel(const ma_channel* pChannelMap, ma_uint32 channelCount, ma_uint32 channelIndex);
+/*
+Initializes a blank channel map.
-Remarks
--------
-Setting `pContext` to NULL will result in miniaudio creating a default context internally and is equivalent to passing in a context initialized like so:
+When a blank channel map is specified anywhere it indicates that the native channel map should be used.
+*/
+MA_API void ma_channel_map_init_blank(ma_channel* pChannelMap, ma_uint32 channels);
- ```c
- ma_context_init(NULL, 0, NULL, &context);
- ```
+/*
+Helper for retrieving a standard channel map.
-Do not set `pContext` to NULL if you are needing to open multiple devices. You can, however, use NULL when initializing the first device, and then use
-device.pContext for the initialization of other devices.
+The output channel map buffer must have a capacity of at least `channelMapCap`.
+*/
+MA_API void ma_channel_map_init_standard(ma_standard_channel_map standardChannelMap, ma_channel* pChannelMap, size_t channelMapCap, ma_uint32 channels);
-The device can be configured via the `pConfig` argument. The config object is initialized with `ma_device_config_init()`. Individual configuration settings can
-then be set directly on the structure. Below are the members of the `ma_device_config` object.
+/*
+Copies a channel map.
- deviceType
- Must be `ma_device_type_playback`, `ma_device_type_capture`, `ma_device_type_duplex` of `ma_device_type_loopback`.
+Both input and output channel map buffers must have a capacity of at at least `channels`.
+*/
+MA_API void ma_channel_map_copy(ma_channel* pOut, const ma_channel* pIn, ma_uint32 channels);
- sampleRate
- The sample rate, in hertz. The most common sample rates are 48000 and 44100. Setting this to 0 will use the device's native sample rate.
+/*
+Copies a channel map if one is specified, otherwise copies the default channel map.
- periodSizeInFrames
- The desired size of a period in PCM frames. If this is 0, `periodSizeInMilliseconds` will be used instead. If both are 0 the default buffer size will
- be used depending on the selected performance profile. This value affects latency. See below for details.
+The output buffer must have a capacity of at least `channels`. If not NULL, the input channel map must also have a capacity of at least `channels`.
+*/
+MA_API void ma_channel_map_copy_or_default(ma_channel* pOut, size_t channelMapCapOut, const ma_channel* pIn, ma_uint32 channels);
- periodSizeInMilliseconds
- The desired size of a period in milliseconds. If this is 0, `periodSizeInFrames` will be used instead. If both are 0 the default buffer size will be
- used depending on the selected performance profile. The value affects latency. See below for details.
- periods
- The number of periods making up the device's entire buffer. The total buffer size is `periodSizeInFrames` or `periodSizeInMilliseconds` multiplied by
- this value. This is just a hint as backends will be the ones who ultimately decide how your periods will be configured.
+/*
+Determines whether or not a channel map is valid.
- performanceProfile
- A hint to miniaudio as to the performance requirements of your program. Can be either `ma_performance_profile_low_latency` (default) or
- `ma_performance_profile_conservative`. This mainly affects the size of default buffers and can usually be left at it's default value.
+A blank channel map is valid (all channels set to MA_CHANNEL_NONE). The way a blank channel map is handled is context specific, but
+is usually treated as a passthrough.
- noPreZeroedOutputBuffer
- When set to true, the contents of the output buffer passed into the data callback will be left undefined. When set to false (default), the contents of
- the output buffer will be cleared the zero. You can use this to avoid the overhead of zeroing out the buffer if you can guarantee that your data
- callback will write to every sample in the output buffer, or if you are doing your own clearing.
+Invalid channel maps:
+ - A channel map with no channels
+ - A channel map with more than one channel and a mono channel
- noClip
- When set to true, the contents of the output buffer passed into the data callback will be clipped after returning. When set to false (default), the
- contents of the output buffer are left alone after returning and it will be left up to the backend itself to decide whether or not the clip. This only
- applies when the playback sample format is f32.
+The channel map buffer must have a capacity of at least `channels`.
+*/
+MA_API ma_bool32 ma_channel_map_is_valid(const ma_channel* pChannelMap, ma_uint32 channels);
- dataCallback
- The callback to fire whenever data is ready to be delivered to or from the device.
+/*
+Helper for comparing two channel maps for equality.
- stopCallback
- The callback to fire whenever the device has stopped, either explicitly via `ma_device_stop()`, or implicitly due to things like the device being
- disconnected.
+This assumes the channel count is the same between the two.
- pUserData
- The user data pointer to use with the device. You can access this directly from the device object like `device.pUserData`.
+Both channels map buffers must have a capacity of at least `channels`.
+*/
+MA_API ma_bool32 ma_channel_map_is_equal(const ma_channel* pChannelMapA, const ma_channel* pChannelMapB, ma_uint32 channels);
- resampling.algorithm
- The resampling algorithm to use when miniaudio needs to perform resampling between the rate specified by `sampleRate` and the device's native rate. The
- default value is `ma_resample_algorithm_linear`, and the quality can be configured with `resampling.linear.lpfOrder`.
+/*
+Helper for determining if a channel map is blank (all channels set to MA_CHANNEL_NONE).
- resampling.linear.lpfOrder
- The linear resampler applies a low-pass filter as part of it's procesing for anti-aliasing. This setting controls the order of the filter. The higher
- the value, the better the quality, in general. Setting this to 0 will disable low-pass filtering altogether. The maximum value is
- `MA_MAX_FILTER_ORDER`. The default value is `min(4, MA_MAX_FILTER_ORDER)`.
+The channel map buffer must have a capacity of at least `channels`.
+*/
+MA_API ma_bool32 ma_channel_map_is_blank(const ma_channel* pChannelMap, ma_uint32 channels);
- playback.pDeviceID
- A pointer to a `ma_device_id` structure containing the ID of the playback device to initialize. Setting this NULL (default) will use the system's
- default playback device. Retrieve the device ID from the `ma_device_info` structure, which can be retrieved using device enumeration.
+/*
+Helper for determining whether or not a channel is present in the given channel map.
- playback.format
- The sample format to use for playback. When set to `ma_format_unknown` the device's native format will be used. This can be retrieved after
- initialization from the device object directly with `device.playback.format`.
+The channel map buffer must have a capacity of at least `channels`.
+*/
+MA_API ma_bool32 ma_channel_map_contains_channel_position(ma_uint32 channels, const ma_channel* pChannelMap, ma_channel channelPosition);
- playback.channels
- The number of channels to use for playback. When set to 0 the device's native channel count will be used. This can be retrieved after initialization
- from the device object directly with `device.playback.channels`.
- playback.channelMap
- The channel map to use for playback. When left empty, the device's native channel map will be used. This can be retrieved after initialization from the
- device object direct with `device.playback.channelMap`.
+/************************************************************************************************************************************************************
- playback.shareMode
- The preferred share mode to use for playback. Can be either `ma_share_mode_shared` (default) or `ma_share_mode_exclusive`. Note that if you specify
- exclusive mode, but it's not supported by the backend, initialization will fail. You can then fall back to shared mode if desired by changing this to
- ma_share_mode_shared and reinitializing.
+Conversion Helpers
- capture.pDeviceID
- A pointer to a `ma_device_id` structure containing the ID of the capture device to initialize. Setting this NULL (default) will use the system's
- default capture device. Retrieve the device ID from the `ma_device_info` structure, which can be retrieved using device enumeration.
+************************************************************************************************************************************************************/
- capture.format
- The sample format to use for capture. When set to `ma_format_unknown` the device's native format will be used. This can be retrieved after
- initialization from the device object directly with `device.capture.format`.
+/*
+High-level helper for doing a full format conversion in one go. Returns the number of output frames. Call this with pOut set to NULL to
+determine the required size of the output buffer. frameCountOut should be set to the capacity of pOut. If pOut is NULL, frameCountOut is
+ignored.
- capture.channels
- The number of channels to use for capture. When set to 0 the device's native channel count will be used. This can be retrieved after initialization
- from the device object directly with `device.capture.channels`.
+A return value of 0 indicates an error.
- capture.channelMap
- The channel map to use for capture. When left empty, the device's native channel map will be used. This can be retrieved after initialization from the
- device object direct with `device.capture.channelMap`.
+This function is useful for one-off bulk conversions, but if you're streaming data you should use the ma_data_converter APIs instead.
+*/
+MA_API ma_uint64 ma_convert_frames(void* pOut, ma_uint64 frameCountOut, ma_format formatOut, ma_uint32 channelsOut, ma_uint32 sampleRateOut, const void* pIn, ma_uint64 frameCountIn, ma_format formatIn, ma_uint32 channelsIn, ma_uint32 sampleRateIn);
+MA_API ma_uint64 ma_convert_frames_ex(void* pOut, ma_uint64 frameCountOut, const void* pIn, ma_uint64 frameCountIn, const ma_data_converter_config* pConfig);
- capture.shareMode
- The preferred share mode to use for capture. Can be either `ma_share_mode_shared` (default) or `ma_share_mode_exclusive`. Note that if you specify
- exclusive mode, but it's not supported by the backend, initialization will fail. You can then fall back to shared mode if desired by changing this to
- ma_share_mode_shared and reinitializing.
- wasapi.noAutoConvertSRC
- WASAPI only. When set to true, disables WASAPI's automatic resampling and forces the use of miniaudio's resampler. Defaults to false.
+/************************************************************************************************************************************************************
- wasapi.noDefaultQualitySRC
- WASAPI only. Only used when `wasapi.noAutoConvertSRC` is set to false. When set to true, disables the use of `AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY`.
- You should usually leave this set to false, which is the default.
+Ring Buffer
- wasapi.noAutoStreamRouting
- WASAPI only. When set to true, disables automatic stream routing on the WASAPI backend. Defaults to false.
+************************************************************************************************************************************************************/
+typedef struct
+{
+ void* pBuffer;
+ ma_uint32 subbufferSizeInBytes;
+ ma_uint32 subbufferCount;
+ ma_uint32 subbufferStrideInBytes;
+ MA_ATOMIC(4, ma_uint32) encodedReadOffset; /* Most significant bit is the loop flag. Lower 31 bits contains the actual offset in bytes. Must be used atomically. */
+ MA_ATOMIC(4, ma_uint32) encodedWriteOffset; /* Most significant bit is the loop flag. Lower 31 bits contains the actual offset in bytes. Must be used atomically. */
+ ma_bool8 ownsBuffer; /* Used to know whether or not miniaudio is responsible for free()-ing the buffer. */
+ ma_bool8 clearOnWriteAcquire; /* When set, clears the acquired write buffer before returning from ma_rb_acquire_write(). */
+ ma_allocation_callbacks allocationCallbacks;
+} ma_rb;
- wasapi.noHardwareOffloading
- WASAPI only. When set to true, disables the use of WASAPI's hardware offloading feature. Defaults to false.
+MA_API ma_result ma_rb_init_ex(size_t subbufferSizeInBytes, size_t subbufferCount, size_t subbufferStrideInBytes, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_rb* pRB);
+MA_API ma_result ma_rb_init(size_t bufferSizeInBytes, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_rb* pRB);
+MA_API void ma_rb_uninit(ma_rb* pRB);
+MA_API void ma_rb_reset(ma_rb* pRB);
+MA_API ma_result ma_rb_acquire_read(ma_rb* pRB, size_t* pSizeInBytes, void** ppBufferOut);
+MA_API ma_result ma_rb_commit_read(ma_rb* pRB, size_t sizeInBytes);
+MA_API ma_result ma_rb_acquire_write(ma_rb* pRB, size_t* pSizeInBytes, void** ppBufferOut);
+MA_API ma_result ma_rb_commit_write(ma_rb* pRB, size_t sizeInBytes);
+MA_API ma_result ma_rb_seek_read(ma_rb* pRB, size_t offsetInBytes);
+MA_API ma_result ma_rb_seek_write(ma_rb* pRB, size_t offsetInBytes);
+MA_API ma_int32 ma_rb_pointer_distance(ma_rb* pRB); /* Returns the distance between the write pointer and the read pointer. Should never be negative for a correct program. Will return the number of bytes that can be read before the read pointer hits the write pointer. */
+MA_API ma_uint32 ma_rb_available_read(ma_rb* pRB);
+MA_API ma_uint32 ma_rb_available_write(ma_rb* pRB);
+MA_API size_t ma_rb_get_subbuffer_size(ma_rb* pRB);
+MA_API size_t ma_rb_get_subbuffer_stride(ma_rb* pRB);
+MA_API size_t ma_rb_get_subbuffer_offset(ma_rb* pRB, size_t subbufferIndex);
+MA_API void* ma_rb_get_subbuffer_ptr(ma_rb* pRB, size_t subbufferIndex, void* pBuffer);
- alsa.noMMap
- ALSA only. When set to true, disables MMap mode. Defaults to false.
- alsa.noAutoFormat
- ALSA only. When set to true, disables ALSA's automatic format conversion by including the SND_PCM_NO_AUTO_FORMAT flag. Defaults to false.
+typedef struct
+{
+ ma_rb rb;
+ ma_format format;
+ ma_uint32 channels;
+} ma_pcm_rb;
- alsa.noAutoChannels
- ALSA only. When set to true, disables ALSA's automatic channel conversion by including the SND_PCM_NO_AUTO_CHANNELS flag. Defaults to false.
+MA_API ma_result ma_pcm_rb_init_ex(ma_format format, ma_uint32 channels, ma_uint32 subbufferSizeInFrames, ma_uint32 subbufferCount, ma_uint32 subbufferStrideInFrames, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_pcm_rb* pRB);
+MA_API ma_result ma_pcm_rb_init(ma_format format, ma_uint32 channels, ma_uint32 bufferSizeInFrames, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_pcm_rb* pRB);
+MA_API void ma_pcm_rb_uninit(ma_pcm_rb* pRB);
+MA_API void ma_pcm_rb_reset(ma_pcm_rb* pRB);
+MA_API ma_result ma_pcm_rb_acquire_read(ma_pcm_rb* pRB, ma_uint32* pSizeInFrames, void** ppBufferOut);
+MA_API ma_result ma_pcm_rb_commit_read(ma_pcm_rb* pRB, ma_uint32 sizeInFrames);
+MA_API ma_result ma_pcm_rb_acquire_write(ma_pcm_rb* pRB, ma_uint32* pSizeInFrames, void** ppBufferOut);
+MA_API ma_result ma_pcm_rb_commit_write(ma_pcm_rb* pRB, ma_uint32 sizeInFrames);
+MA_API ma_result ma_pcm_rb_seek_read(ma_pcm_rb* pRB, ma_uint32 offsetInFrames);
+MA_API ma_result ma_pcm_rb_seek_write(ma_pcm_rb* pRB, ma_uint32 offsetInFrames);
+MA_API ma_int32 ma_pcm_rb_pointer_distance(ma_pcm_rb* pRB); /* Return value is in frames. */
+MA_API ma_uint32 ma_pcm_rb_available_read(ma_pcm_rb* pRB);
+MA_API ma_uint32 ma_pcm_rb_available_write(ma_pcm_rb* pRB);
+MA_API ma_uint32 ma_pcm_rb_get_subbuffer_size(ma_pcm_rb* pRB);
+MA_API ma_uint32 ma_pcm_rb_get_subbuffer_stride(ma_pcm_rb* pRB);
+MA_API ma_uint32 ma_pcm_rb_get_subbuffer_offset(ma_pcm_rb* pRB, ma_uint32 subbufferIndex);
+MA_API void* ma_pcm_rb_get_subbuffer_ptr(ma_pcm_rb* pRB, ma_uint32 subbufferIndex, void* pBuffer);
- alsa.noAutoResample
- ALSA only. When set to true, disables ALSA's automatic resampling by including the SND_PCM_NO_AUTO_RESAMPLE flag. Defaults to false.
- pulse.pStreamNamePlayback
- PulseAudio only. Sets the stream name for playback.
+/*
+The idea of the duplex ring buffer is to act as the intermediary buffer when running two asynchronous devices in a duplex set up. The
+capture device writes to it, and then a playback device reads from it.
- pulse.pStreamNameCapture
- PulseAudio only. Sets the stream name for capture.
+At the moment this is just a simple naive implementation, but in the future I want to implement some dynamic resampling to seamlessly
+handle desyncs. Note that the API is work in progress and may change at any time in any version.
- coreaudio.allowNominalSampleRateChange
- Core Audio only. Desktop only. When enabled, allows the sample rate of the device to be changed at the operating system level. This
- is disabled by default in order to prevent intrusive changes to the user's system. This is useful if you want to use a sample rate
- that is known to be natively supported by the hardware thereby avoiding the cost of resampling. When set to true, miniaudio will
- find the closest match between the sample rate requested in the device config and the sample rates natively supported by the
- hardware. When set to false, the sample rate currently set by the operating system will always be used.
+The size of the buffer is based on the capture side since that's what'll be written to the buffer. It is based on the capture period size
+in frames. The internal sample rate of the capture device is also needed in order to calculate the size.
+*/
+typedef struct
+{
+ ma_pcm_rb rb;
+} ma_duplex_rb;
+MA_API ma_result ma_duplex_rb_init(ma_format captureFormat, ma_uint32 captureChannels, ma_uint32 sampleRate, ma_uint32 captureInternalSampleRate, ma_uint32 captureInternalPeriodSizeInFrames, const ma_allocation_callbacks* pAllocationCallbacks, ma_duplex_rb* pRB);
+MA_API ma_result ma_duplex_rb_uninit(ma_duplex_rb* pRB);
-Once initialized, the device's config is immutable. If you need to change the config you will need to initialize a new device.
-After initializing the device it will be in a stopped state. To start it, use `ma_device_start()`.
+/************************************************************************************************************************************************************
-If both `periodSizeInFrames` and `periodSizeInMilliseconds` are set to zero, it will default to `MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY` or
-`MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE`, depending on whether or not `performanceProfile` is set to `ma_performance_profile_low_latency` or
-`ma_performance_profile_conservative`.
+Miscellaneous Helpers
-If you request exclusive mode and the backend does not support it an error will be returned. For robustness, you may want to first try initializing the device
-in exclusive mode, and then fall back to shared mode if required. Alternatively you can just request shared mode (the default if you leave it unset in the
-config) which is the most reliable option. Some backends do not have a practical way of choosing whether or not the device should be exclusive or not (ALSA,
-for example) in which case it just acts as a hint. Unless you have special requirements you should try avoiding exclusive mode as it's intrusive to the user.
-Starting with Windows 10, miniaudio will use low-latency shared mode where possible which may make exclusive mode unnecessary.
+************************************************************************************************************************************************************/
+/*
+Retrieves a human readable description of the given result code.
+*/
+MA_API const char* ma_result_description(ma_result result);
-When sending or receiving data to/from a device, miniaudio will internally perform a format conversion to convert between the format specified by the config
-and the format used internally by the backend. If you pass in 0 for the sample format, channel count, sample rate _and_ channel map, data transmission will run
-on an optimized pass-through fast path. You can retrieve the format, channel count and sample rate by inspecting the `playback/capture.format`,
-`playback/capture.channels` and `sampleRate` members of the device object.
+/*
+malloc()
+*/
+MA_API void* ma_malloc(size_t sz, const ma_allocation_callbacks* pAllocationCallbacks);
-When compiling for UWP you must ensure you call this function on the main UI thread because the operating system may need to present the user with a message
-asking for permissions. Please refer to the official documentation for ActivateAudioInterfaceAsync() for more information.
+/*
+calloc()
+*/
+MA_API void* ma_calloc(size_t sz, const ma_allocation_callbacks* pAllocationCallbacks);
-ALSA Specific: When initializing the default device, requesting shared mode will try using the "dmix" device for playback and the "dsnoop" device for capture.
-If these fail it will try falling back to the "hw" device.
+/*
+realloc()
+*/
+MA_API void* ma_realloc(void* p, size_t sz, const ma_allocation_callbacks* pAllocationCallbacks);
+/*
+free()
+*/
+MA_API void ma_free(void* p, const ma_allocation_callbacks* pAllocationCallbacks);
-Example 1 - Simple Initialization
----------------------------------
-This example shows how to initialize a simple playback device using a standard configuration. If you are just needing to do simple playback from the default
-playback device this is usually all you need.
+/*
+Performs an aligned malloc, with the assumption that the alignment is a power of 2.
+*/
+MA_API void* ma_aligned_malloc(size_t sz, size_t alignment, const ma_allocation_callbacks* pAllocationCallbacks);
-```c
-ma_device_config config = ma_device_config_init(ma_device_type_playback);
-config.playback.format = ma_format_f32;
-config.playback.channels = 2;
-config.sampleRate = 48000;
-config.dataCallback = ma_data_callback;
-config.pMyUserData = pMyUserData;
+/*
+Free's an aligned malloc'd buffer.
+*/
+MA_API void ma_aligned_free(void* p, const ma_allocation_callbacks* pAllocationCallbacks);
-ma_device device;
-ma_result result = ma_device_init(NULL, &config, &device);
-if (result != MA_SUCCESS) {
- // Error
-}
-```
+/*
+Retrieves a friendly name for a format.
+*/
+MA_API const char* ma_get_format_name(ma_format format);
+/*
+Blends two frames in floating point format.
+*/
+MA_API void ma_blend_f32(float* pOut, float* pInA, float* pInB, float factor, ma_uint32 channels);
-Example 2 - Advanced Initialization
------------------------------------
-This example shows how you might do some more advanced initialization. In this hypothetical example we want to control the latency by setting the buffer size
-and period count. We also want to allow the user to be able to choose which device to output from which means we need a context so we can perform device
-enumeration.
+/*
+Retrieves the size of a sample in bytes for the given format.
-```c
-ma_context context;
-ma_result result = ma_context_init(NULL, 0, NULL, &context);
-if (result != MA_SUCCESS) {
- // Error
-}
+This API is efficient and is implemented using a lookup table.
-ma_device_info* pPlaybackDeviceInfos;
-ma_uint32 playbackDeviceCount;
-result = ma_context_get_devices(&context, &pPlaybackDeviceInfos, &playbackDeviceCount, NULL, NULL);
-if (result != MA_SUCCESS) {
- // Error
-}
+Thread Safety: SAFE
+ This API is pure.
+*/
+MA_API ma_uint32 ma_get_bytes_per_sample(ma_format format);
+static MA_INLINE ma_uint32 ma_get_bytes_per_frame(ma_format format, ma_uint32 channels) { return ma_get_bytes_per_sample(format) * channels; }
-// ... choose a device from pPlaybackDeviceInfos ...
+/*
+Converts a log level to a string.
+*/
+MA_API const char* ma_log_level_to_string(ma_uint32 logLevel);
-ma_device_config config = ma_device_config_init(ma_device_type_playback);
-config.playback.pDeviceID = pMyChosenDeviceID; // <-- Get this from the `id` member of one of the `ma_device_info` objects returned by ma_context_get_devices().
-config.playback.format = ma_format_f32;
-config.playback.channels = 2;
-config.sampleRate = 48000;
-config.dataCallback = ma_data_callback;
-config.pUserData = pMyUserData;
-config.periodSizeInMilliseconds = 10;
-config.periods = 3;
-ma_device device;
-result = ma_device_init(&context, &config, &device);
-if (result != MA_SUCCESS) {
- // Error
-}
-```
-See Also
---------
-ma_device_config_init()
-ma_device_uninit()
-ma_device_start()
-ma_context_init()
-ma_context_get_devices()
-ma_context_enumerate_devices()
+/************************************************************************************************************************************************************
+
+Synchronization
+
+************************************************************************************************************************************************************/
+/*
+Locks a spinlock.
*/
-MA_API ma_result ma_device_init(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice);
+MA_API ma_result ma_spinlock_lock(volatile ma_spinlock* pSpinlock);
/*
-Initializes a device without a context, with extra parameters for controlling the configuration of the internal self-managed context.
+Locks a spinlock, but does not yield() when looping.
+*/
+MA_API ma_result ma_spinlock_lock_noyield(volatile ma_spinlock* pSpinlock);
-This is the same as `ma_device_init()`, only instead of a context being passed in, the parameters from `ma_context_init()` are passed in instead. This function
-allows you to configure the internally created context.
+/*
+Unlocks a spinlock.
+*/
+MA_API ma_result ma_spinlock_unlock(volatile ma_spinlock* pSpinlock);
-Parameters
-----------
-backends (in, optional)
- A list of backends to try initializing, in priority order. Can be NULL, in which case it uses default priority order.
+#ifndef MA_NO_THREADING
-backendCount (in, optional)
- The number of items in `backend`. Ignored if `backend` is NULL.
+/*
+Creates a mutex.
-pContextConfig (in, optional)
- The context configuration.
+A mutex must be created from a valid context. A mutex is initially unlocked.
+*/
+MA_API ma_result ma_mutex_init(ma_mutex* pMutex);
-pConfig (in)
- A pointer to the device configuration. Cannot be null. See remarks for details.
+/*
+Deletes a mutex.
+*/
+MA_API void ma_mutex_uninit(ma_mutex* pMutex);
-pDevice (out)
- A pointer to the device object being initialized.
+/*
+Locks a mutex with an infinite timeout.
+*/
+MA_API void ma_mutex_lock(ma_mutex* pMutex);
+/*
+Unlocks a mutex.
+*/
+MA_API void ma_mutex_unlock(ma_mutex* pMutex);
-Return Value
-------------
-MA_SUCCESS if successful; any other error code otherwise.
+/*
+Initializes an auto-reset event.
+*/
+MA_API ma_result ma_event_init(ma_event* pEvent);
-Thread Safety
--------------
-Unsafe. It is not safe to call this function simultaneously for different devices because some backends depend on and mutate global state. The same applies to
-calling this at the same time as `ma_device_uninit()`.
+/*
+Uninitializes an auto-reset event.
+*/
+MA_API void ma_event_uninit(ma_event* pEvent);
+/*
+Waits for the specified auto-reset event to become signalled.
+*/
+MA_API ma_result ma_event_wait(ma_event* pEvent);
-Callback Safety
----------------
-Unsafe. It is not safe to call this inside any callback.
+/*
+Signals the specified auto-reset event.
+*/
+MA_API ma_result ma_event_signal(ma_event* pEvent);
+#endif /* MA_NO_THREADING */
-Remarks
--------
-You only need to use this function if you want to configure the context differently to it's defaults. You should never use this function if you want to manage
-your own context.
+/*
+Fence
+=====
+This locks while the counter is larger than 0. Counter can be incremented and decremented by any
+thread, but care needs to be taken when waiting. It is possible for one thread to acquire the
+fence just as another thread returns from ma_fence_wait().
+
+The idea behind a fence is to allow you to wait for a group of operations to complete. When an
+operation starts, the counter is incremented which locks the fence. When the operation completes,
+the fence will be released which decrements the counter. ma_fence_wait() will block until the
+counter hits zero.
+
+If threading is disabled, ma_fence_wait() will spin on the counter.
+*/
+typedef struct
+{
+#ifndef MA_NO_THREADING
+ ma_event e;
+#endif
+ ma_uint32 counter;
+} ma_fence;
-See the documentation for `ma_context_init()` for information on the different context configuration options.
+MA_API ma_result ma_fence_init(ma_fence* pFence);
+MA_API void ma_fence_uninit(ma_fence* pFence);
+MA_API ma_result ma_fence_acquire(ma_fence* pFence); /* Increment counter. */
+MA_API ma_result ma_fence_release(ma_fence* pFence); /* Decrement counter. */
+MA_API ma_result ma_fence_wait(ma_fence* pFence); /* Wait for counter to reach 0. */
-See Also
---------
-ma_device_init()
-ma_device_uninit()
-ma_device_config_init()
-ma_context_init()
-*/
-MA_API ma_result ma_device_init_ex(const ma_backend backends[], ma_uint32 backendCount, const ma_context_config* pContextConfig, const ma_device_config* pConfig, ma_device* pDevice);
/*
-Uninitializes a device.
+Notification callback for asynchronous operations.
+*/
+typedef void ma_async_notification;
-This will explicitly stop the device. You do not need to call `ma_device_stop()` beforehand, but it's harmless if you do.
+typedef struct
+{
+ void (* onSignal)(ma_async_notification* pNotification);
+} ma_async_notification_callbacks;
+MA_API ma_result ma_async_notification_signal(ma_async_notification* pNotification);
-Parameters
-----------
-pDevice (in)
- A pointer to the device to stop.
+/*
+Simple polling notification.
-Return Value
-------------
-Nothing
+This just sets a variable when the notification has been signalled which is then polled with ma_async_notification_poll_is_signalled()
+*/
+typedef struct
+{
+ ma_async_notification_callbacks cb;
+ ma_bool32 signalled;
+} ma_async_notification_poll;
-
-Thread Safety
--------------
-Unsafe. As soon as this API is called the device should be considered undefined.
-
-
-Callback Safety
----------------
-Unsafe. It is not safe to call this inside any callback. Doing this will result in a deadlock.
-
-
-See Also
---------
-ma_device_init()
-ma_device_stop()
-*/
-MA_API void ma_device_uninit(ma_device* pDevice);
+MA_API ma_result ma_async_notification_poll_init(ma_async_notification_poll* pNotificationPoll);
+MA_API ma_bool32 ma_async_notification_poll_is_signalled(const ma_async_notification_poll* pNotificationPoll);
/*
-Retrieves a pointer to the context that owns the given device.
-*/
-MA_API ma_context* ma_device_get_context(ma_device* pDevice);
+Event Notification
-/*
-Helper function for retrieving the log object associated with the context that owns this device.
+This uses an ma_event. If threading is disabled (MA_NO_THREADING), initialization will fail.
*/
-MA_API ma_log* ma_device_get_log(ma_device* pDevice);
-
-
-/*
-Starts the device. For playback devices this begins playback. For capture devices it begins recording.
-
-Use `ma_device_stop()` to stop the device.
+typedef struct
+{
+ ma_async_notification_callbacks cb;
+#ifndef MA_NO_THREADING
+ ma_event e;
+#endif
+} ma_async_notification_event;
+MA_API ma_result ma_async_notification_event_init(ma_async_notification_event* pNotificationEvent);
+MA_API ma_result ma_async_notification_event_uninit(ma_async_notification_event* pNotificationEvent);
+MA_API ma_result ma_async_notification_event_wait(ma_async_notification_event* pNotificationEvent);
+MA_API ma_result ma_async_notification_event_signal(ma_async_notification_event* pNotificationEvent);
-Parameters
-----------
-pDevice (in)
- A pointer to the device to start.
-Return Value
-------------
-MA_SUCCESS if successful; any other error code otherwise.
+/************************************************************************************************************************************************************
-Thread Safety
--------------
-Safe. It's safe to call this from any thread with the exception of the callback thread.
+Job Queue
+************************************************************************************************************************************************************/
-Callback Safety
----------------
-Unsafe. It is not safe to call this inside any callback.
+/*
+Slot Allocator
+--------------
+The idea of the slot allocator is for it to be used in conjunction with a fixed sized buffer. You use the slot allocator to allocator an index that can be used
+as the insertion point for an object.
+Slots are reference counted to help mitigate the ABA problem in the lock-free queue we use for tracking jobs.
-Remarks
--------
-For a playback device, this will retrieve an initial chunk of audio data from the client before returning. The reason for this is to ensure there is valid
-audio data in the buffer, which needs to be done before the device begins playback.
+The slot index is stored in the low 32 bits. The reference counter is stored in the high 32 bits:
-This API waits until the backend device has been started for real by the worker thread. It also waits on a mutex for thread-safety.
+ +-----------------+-----------------+
+ | 32 Bits | 32 Bits |
+ +-----------------+-----------------+
+ | Reference Count | Slot Index |
+ +-----------------+-----------------+
+*/
+typedef struct
+{
+ ma_uint32 capacity; /* The number of slots to make available. */
+} ma_slot_allocator_config;
-Do not call this in any callback.
+MA_API ma_slot_allocator_config ma_slot_allocator_config_init(ma_uint32 capacity);
-See Also
---------
-ma_device_stop()
-*/
-MA_API ma_result ma_device_start(ma_device* pDevice);
+typedef struct
+{
+ MA_ATOMIC(4, ma_uint32) bitfield; /* Must be used atomically because the allocation and freeing routines need to make copies of this which must never be optimized away by the compiler. */
+} ma_slot_allocator_group;
-/*
-Stops the device. For playback devices this stops playback. For capture devices it stops recording.
+typedef struct
+{
+ ma_slot_allocator_group* pGroups; /* Slots are grouped in chunks of 32. */
+ ma_uint32* pSlots; /* 32 bits for reference counting for ABA mitigation. */
+ ma_uint32 count; /* Allocation count. */
+ ma_uint32 capacity;
-Use `ma_device_start()` to start the device again.
+ /* Memory management. */
+ ma_bool32 _ownsHeap;
+ void* _pHeap;
+} ma_slot_allocator;
+MA_API ma_result ma_slot_allocator_get_heap_size(const ma_slot_allocator_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_slot_allocator_init_preallocated(const ma_slot_allocator_config* pConfig, void* pHeap, ma_slot_allocator* pAllocator);
+MA_API ma_result ma_slot_allocator_init(const ma_slot_allocator_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_slot_allocator* pAllocator);
+MA_API void ma_slot_allocator_uninit(ma_slot_allocator* pAllocator, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_slot_allocator_alloc(ma_slot_allocator* pAllocator, ma_uint64* pSlot);
+MA_API ma_result ma_slot_allocator_free(ma_slot_allocator* pAllocator, ma_uint64 slot);
-Parameters
-----------
-pDevice (in)
- A pointer to the device to stop.
+typedef struct ma_job ma_job;
-Return Value
-------------
-MA_SUCCESS if successful; any other error code otherwise.
+/*
+Callback for processing a job. Each job type will have their own processing callback which will be
+called by ma_job_process().
+*/
+typedef ma_result (* ma_job_proc)(ma_job* pJob);
+/* When a job type is added here an callback needs to be added go "g_jobVTable" in the implementation section. */
+typedef enum
+{
+ /* Miscellaneous. */
+ MA_JOB_TYPE_QUIT = 0,
+ MA_JOB_TYPE_CUSTOM,
-Thread Safety
--------------
-Safe. It's safe to call this from any thread with the exception of the callback thread.
+ /* Resource Manager. */
+ MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_BUFFER_NODE,
+ MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_BUFFER_NODE,
+ MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_BUFFER_NODE,
+ MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_BUFFER,
+ MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_BUFFER,
+ MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_STREAM,
+ MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_STREAM,
+ MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_STREAM,
+ MA_JOB_TYPE_RESOURCE_MANAGER_SEEK_DATA_STREAM,
+ /* Device. */
+ MA_JOB_TYPE_DEVICE_AAUDIO_REROUTE,
-Callback Safety
----------------
-Unsafe. It is not safe to call this inside any callback. Doing this will result in a deadlock.
+ /* Count. Must always be last. */
+ MA_JOB_TYPE_COUNT
+} ma_job_type;
+struct ma_job
+{
+ union
+ {
+ struct
+ {
+ ma_uint16 code; /* Job type. */
+ ma_uint16 slot; /* Index into a ma_slot_allocator. */
+ ma_uint32 refcount;
+ } breakup;
+ ma_uint64 allocation;
+ } toc; /* 8 bytes. We encode the job code into the slot allocation data to save space. */
+ MA_ATOMIC(8, ma_uint64) next; /* refcount + slot for the next item. Does not include the job code. */
+ ma_uint32 order; /* Execution order. Used to create a data dependency and ensure a job is executed in order. Usage is contextual depending on the job type. */
-Remarks
--------
-This API needs to wait on the worker thread to stop the backend device properly before returning. It also waits on a mutex for thread-safety. In addition, some
-backends need to wait for the device to finish playback/recording of the current fragment which can take some time (usually proportionate to the buffer size
-that was specified at initialization time).
+ union
+ {
+ /* Miscellaneous. */
+ struct
+ {
+ ma_job_proc proc;
+ ma_uintptr data0;
+ ma_uintptr data1;
+ } custom;
-Backends are required to either pause the stream in-place or drain the buffer if pausing is not possible. The reason for this is that stopping the device and
-the resuming it with ma_device_start() (which you might do when your program loses focus) may result in a situation where those samples are never output to the
-speakers or received from the microphone which can in turn result in de-syncs.
+ /* Resource Manager */
+ union
+ {
+ struct
+ {
+ /*ma_resource_manager**/ void* pResourceManager;
+ /*ma_resource_manager_data_buffer_node**/ void* pDataBufferNode;
+ char* pFilePath;
+ wchar_t* pFilePathW;
+ ma_bool32 decode; /* When set to true, the data buffer will be decoded. Otherwise it'll be encoded and will use a decoder for the connector. */
+ ma_async_notification* pInitNotification; /* Signalled when the data buffer has been initialized and the format/channels/rate can be retrieved. */
+ ma_async_notification* pDoneNotification; /* Signalled when the data buffer has been fully decoded. Will be passed through to MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_BUFFER_NODE when decoding. */
+ ma_fence* pInitFence; /* Released when initialization of the decoder is complete. */
+ ma_fence* pDoneFence; /* Released if initialization of the decoder fails. Passed through to PAGE_DATA_BUFFER_NODE untouched if init is successful. */
+ } loadDataBufferNode;
+ struct
+ {
+ /*ma_resource_manager**/ void* pResourceManager;
+ /*ma_resource_manager_data_buffer_node**/ void* pDataBufferNode;
+ ma_async_notification* pDoneNotification;
+ ma_fence* pDoneFence;
+ } freeDataBufferNode;
+ struct
+ {
+ /*ma_resource_manager**/ void* pResourceManager;
+ /*ma_resource_manager_data_buffer_node**/ void* pDataBufferNode;
+ /*ma_decoder**/ void* pDecoder;
+ ma_async_notification* pDoneNotification; /* Signalled when the data buffer has been fully decoded. */
+ ma_fence* pDoneFence; /* Passed through from LOAD_DATA_BUFFER_NODE and released when the data buffer completes decoding or an error occurs. */
+ } pageDataBufferNode;
+
+ struct
+ {
+ /*ma_resource_manager_data_buffer**/ void* pDataBuffer;
+ ma_async_notification* pInitNotification; /* Signalled when the data buffer has been initialized and the format/channels/rate can be retrieved. */
+ ma_async_notification* pDoneNotification; /* Signalled when the data buffer has been fully decoded. */
+ ma_fence* pInitFence; /* Released when the data buffer has been initialized and the format/channels/rate can be retrieved. */
+ ma_fence* pDoneFence; /* Released when the data buffer has been fully decoded. */
+ } loadDataBuffer;
+ struct
+ {
+ /*ma_resource_manager_data_buffer**/ void* pDataBuffer;
+ ma_async_notification* pDoneNotification;
+ ma_fence* pDoneFence;
+ } freeDataBuffer;
-Do not call this in any callback.
+ struct
+ {
+ /*ma_resource_manager_data_stream**/ void* pDataStream;
+ char* pFilePath; /* Allocated when the job is posted, freed by the job thread after loading. */
+ wchar_t* pFilePathW; /* ^ As above ^. Only used if pFilePath is NULL. */
+ ma_uint64 initialSeekPoint;
+ ma_async_notification* pInitNotification; /* Signalled after the first two pages have been decoded and frames can be read from the stream. */
+ ma_fence* pInitFence;
+ } loadDataStream;
+ struct
+ {
+ /*ma_resource_manager_data_stream**/ void* pDataStream;
+ ma_async_notification* pDoneNotification;
+ ma_fence* pDoneFence;
+ } freeDataStream;
+ struct
+ {
+ /*ma_resource_manager_data_stream**/ void* pDataStream;
+ ma_uint32 pageIndex; /* The index of the page to decode into. */
+ } pageDataStream;
+ struct
+ {
+ /*ma_resource_manager_data_stream**/ void* pDataStream;
+ ma_uint64 frameIndex;
+ } seekDataStream;
+ } resourceManager;
-This will be called implicitly by `ma_device_uninit()`.
+ /* Device. */
+ union
+ {
+ union
+ {
+ struct
+ {
+ /*ma_device**/ void* pDevice;
+ /*ma_device_type*/ ma_uint32 deviceType;
+ } reroute;
+ } aaudio;
+ } device;
+ } data;
+};
+MA_API ma_job ma_job_init(ma_uint16 code);
+MA_API ma_result ma_job_process(ma_job* pJob);
-See Also
---------
-ma_device_start()
-*/
-MA_API ma_result ma_device_stop(ma_device* pDevice);
/*
-Determines whether or not the device is started.
-
+When set, ma_job_queue_next() will not wait and no semaphore will be signaled in
+ma_job_queue_post(). ma_job_queue_next() will return MA_NO_DATA_AVAILABLE if nothing is available.
-Parameters
-----------
-pDevice (in)
- A pointer to the device whose start state is being retrieved.
+This flag should always be used for platforms that do not support multithreading.
+*/
+typedef enum
+{
+ MA_JOB_QUEUE_FLAG_NON_BLOCKING = 0x00000001
+} ma_job_queue_flags;
+typedef struct
+{
+ ma_uint32 flags;
+ ma_uint32 capacity; /* The maximum number of jobs that can fit in the queue at a time. */
+} ma_job_queue_config;
-Return Value
-------------
-True if the device is started, false otherwise.
+MA_API ma_job_queue_config ma_job_queue_config_init(ma_uint32 flags, ma_uint32 capacity);
-Thread Safety
--------------
-Safe. If another thread calls `ma_device_start()` or `ma_device_stop()` at this same time as this function is called, there's a very small chance the return
-value will be out of sync.
+typedef struct
+{
+ ma_uint32 flags; /* Flags passed in at initialization time. */
+ ma_uint32 capacity; /* The maximum number of jobs that can fit in the queue at a time. Set by the config. */
+ MA_ATOMIC(8, ma_uint64) head; /* The first item in the list. Required for removing from the top of the list. */
+ MA_ATOMIC(8, ma_uint64) tail; /* The last item in the list. Required for appending to the end of the list. */
+#ifndef MA_NO_THREADING
+ ma_semaphore sem; /* Only used when MA_JOB_QUEUE_FLAG_NON_BLOCKING is unset. */
+#endif
+ ma_slot_allocator allocator;
+ ma_job* pJobs;
+#ifndef MA_USE_EXPERIMENTAL_LOCK_FREE_JOB_QUEUE
+ ma_spinlock lock;
+#endif
+ /* Memory management. */
+ void* _pHeap;
+ ma_bool32 _ownsHeap;
+} ma_job_queue;
-Callback Safety
----------------
-Safe. This is implemented as a simple accessor.
+MA_API ma_result ma_job_queue_get_heap_size(const ma_job_queue_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_job_queue_init_preallocated(const ma_job_queue_config* pConfig, void* pHeap, ma_job_queue* pQueue);
+MA_API ma_result ma_job_queue_init(const ma_job_queue_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_job_queue* pQueue);
+MA_API void ma_job_queue_uninit(ma_job_queue* pQueue, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_job_queue_post(ma_job_queue* pQueue, const ma_job* pJob);
+MA_API ma_result ma_job_queue_next(ma_job_queue* pQueue, ma_job* pJob); /* Returns MA_CANCELLED if the next job is a quit job. */
-See Also
---------
-ma_device_start()
-ma_device_stop()
-*/
-MA_API ma_bool32 ma_device_is_started(const ma_device* pDevice);
+/************************************************************************************************************************************************************
+*************************************************************************************************************************************************************
-/*
-Retrieves the state of the device.
+DEVICE I/O
+==========
+This section contains the APIs for device playback and capture. Here is where you'll find ma_device_init(), etc.
-Parameters
-----------
-pDevice (in)
- A pointer to the device whose state is being retrieved.
+*************************************************************************************************************************************************************
+************************************************************************************************************************************************************/
+#ifndef MA_NO_DEVICE_IO
+/* Some backends are only supported on certain platforms. */
+#if defined(MA_WIN32)
+ #define MA_SUPPORT_WASAPI
+ #if defined(MA_WIN32_DESKTOP) /* DirectSound and WinMM backends are only supported on desktops. */
+ #define MA_SUPPORT_DSOUND
+ #define MA_SUPPORT_WINMM
+ #define MA_SUPPORT_JACK /* JACK is technically supported on Windows, but I don't know how many people use it in practice... */
+ #endif
+#endif
+#if defined(MA_UNIX)
+ #if defined(MA_LINUX)
+ #if !defined(MA_ANDROID) /* ALSA is not supported on Android. */
+ #define MA_SUPPORT_ALSA
+ #endif
+ #endif
+ #if !defined(MA_BSD) && !defined(MA_ANDROID) && !defined(MA_EMSCRIPTEN)
+ #define MA_SUPPORT_PULSEAUDIO
+ #define MA_SUPPORT_JACK
+ #endif
+ #if defined(MA_ANDROID)
+ #define MA_SUPPORT_AAUDIO
+ #define MA_SUPPORT_OPENSL
+ #endif
+ #if defined(__OpenBSD__) /* <-- Change this to "#if defined(MA_BSD)" to enable sndio on all BSD flavors. */
+ #define MA_SUPPORT_SNDIO /* sndio is only supported on OpenBSD for now. May be expanded later if there's demand. */
+ #endif
+ #if defined(__NetBSD__) || defined(__OpenBSD__)
+ #define MA_SUPPORT_AUDIO4 /* Only support audio(4) on platforms with known support. */
+ #endif
+ #if defined(__FreeBSD__) || defined(__DragonFly__)
+ #define MA_SUPPORT_OSS /* Only support OSS on specific platforms with known support. */
+ #endif
+#endif
+#if defined(MA_APPLE)
+ #define MA_SUPPORT_COREAUDIO
+#endif
+#if defined(MA_EMSCRIPTEN)
+ #define MA_SUPPORT_WEBAUDIO
+#endif
+/* All platforms should support custom backends. */
+#define MA_SUPPORT_CUSTOM
-Return Value
-------------
-The current state of the device. The return value will be one of the following:
+/* Explicitly disable the Null backend for Emscripten because it uses a background thread which is not properly supported right now. */
+#if !defined(MA_EMSCRIPTEN)
+#define MA_SUPPORT_NULL
+#endif
- +------------------------+------------------------------------------------------------------------------+
- | MA_STATE_UNINITIALIZED | Will only be returned if the device is in the middle of initialization. |
- +------------------------+------------------------------------------------------------------------------+
- | MA_STATE_STOPPED | The device is stopped. The initial state of the device after initialization. |
- +------------------------+------------------------------------------------------------------------------+
- | MA_STATE_STARTED | The device started and requesting and/or delivering audio data. |
- +------------------------+------------------------------------------------------------------------------+
- | MA_STATE_STARTING | The device is in the process of starting. |
- +------------------------+------------------------------------------------------------------------------+
- | MA_STATE_STOPPING | The device is in the process of stopping. |
- +------------------------+------------------------------------------------------------------------------+
+#if defined(MA_SUPPORT_WASAPI) && !defined(MA_NO_WASAPI) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_WASAPI))
+ #define MA_HAS_WASAPI
+#endif
+#if defined(MA_SUPPORT_DSOUND) && !defined(MA_NO_DSOUND) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_DSOUND))
+ #define MA_HAS_DSOUND
+#endif
+#if defined(MA_SUPPORT_WINMM) && !defined(MA_NO_WINMM) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_WINMM))
+ #define MA_HAS_WINMM
+#endif
+#if defined(MA_SUPPORT_ALSA) && !defined(MA_NO_ALSA) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_ALSA))
+ #define MA_HAS_ALSA
+#endif
+#if defined(MA_SUPPORT_PULSEAUDIO) && !defined(MA_NO_PULSEAUDIO) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_PULSEAUDIO))
+ #define MA_HAS_PULSEAUDIO
+#endif
+#if defined(MA_SUPPORT_JACK) && !defined(MA_NO_JACK) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_JACK))
+ #define MA_HAS_JACK
+#endif
+#if defined(MA_SUPPORT_COREAUDIO) && !defined(MA_NO_COREAUDIO) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_COREAUDIO))
+ #define MA_HAS_COREAUDIO
+#endif
+#if defined(MA_SUPPORT_SNDIO) && !defined(MA_NO_SNDIO) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_SNDIO))
+ #define MA_HAS_SNDIO
+#endif
+#if defined(MA_SUPPORT_AUDIO4) && !defined(MA_NO_AUDIO4) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_AUDIO4))
+ #define MA_HAS_AUDIO4
+#endif
+#if defined(MA_SUPPORT_OSS) && !defined(MA_NO_OSS) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_OSS))
+ #define MA_HAS_OSS
+#endif
+#if defined(MA_SUPPORT_AAUDIO) && !defined(MA_NO_AAUDIO) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_AAUDIO))
+ #define MA_HAS_AAUDIO
+#endif
+#if defined(MA_SUPPORT_OPENSL) && !defined(MA_NO_OPENSL) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_OPENSL))
+ #define MA_HAS_OPENSL
+#endif
+#if defined(MA_SUPPORT_WEBAUDIO) && !defined(MA_NO_WEBAUDIO) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_WEBAUDIO))
+ #define MA_HAS_WEBAUDIO
+#endif
+#if defined(MA_SUPPORT_CUSTOM) && !defined(MA_NO_CUSTOM) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_CUSTOM))
+ #define MA_HAS_CUSTOM
+#endif
+#if defined(MA_SUPPORT_NULL) && !defined(MA_NO_NULL) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_NULL))
+ #define MA_HAS_NULL
+#endif
-Thread Safety
--------------
-Safe. This is implemented as a simple accessor. Note that if the device is started or stopped at the same time as this function is called,
-there's a possibility the return value could be out of sync. See remarks.
+typedef enum
+{
+ ma_device_state_uninitialized = 0,
+ ma_device_state_stopped = 1, /* The device's default state after initialization. */
+ ma_device_state_started = 2, /* The device is started and is requesting and/or delivering audio data. */
+ ma_device_state_starting = 3, /* Transitioning from a stopped state to started. */
+ ma_device_state_stopping = 4 /* Transitioning from a started state to stopped. */
+} ma_device_state;
+#ifdef MA_SUPPORT_WASAPI
+/* We need a IMMNotificationClient object for WASAPI. */
+typedef struct
+{
+ void* lpVtbl;
+ ma_uint32 counter;
+ ma_device* pDevice;
+} ma_IMMNotificationClient;
+#endif
-Callback Safety
----------------
-Safe. This is implemented as a simple accessor.
+/* Backend enums must be in priority order. */
+typedef enum
+{
+ ma_backend_wasapi,
+ ma_backend_dsound,
+ ma_backend_winmm,
+ ma_backend_coreaudio,
+ ma_backend_sndio,
+ ma_backend_audio4,
+ ma_backend_oss,
+ ma_backend_pulseaudio,
+ ma_backend_alsa,
+ ma_backend_jack,
+ ma_backend_aaudio,
+ ma_backend_opensl,
+ ma_backend_webaudio,
+ ma_backend_custom, /* <-- Custom backend, with callbacks defined by the context config. */
+ ma_backend_null /* <-- Must always be the last item. Lowest priority, and used as the terminator for backend enumeration. */
+} ma_backend;
+#define MA_BACKEND_COUNT (ma_backend_null+1)
-Remarks
--------
-The general flow of a devices state goes like this:
- ```
- ma_device_init() -> MA_STATE_UNINITIALIZED -> MA_STATE_STOPPED
- ma_device_start() -> MA_STATE_STARTING -> MA_STATE_STARTED
- ma_device_stop() -> MA_STATE_STOPPING -> MA_STATE_STOPPED
- ```
+/*
+Device job thread. This is used by backends that require asynchronous processing of certain
+operations. It is not used by all backends.
-When the state of the device is changed with `ma_device_start()` or `ma_device_stop()` at this same time as this function is called, the
-value returned by this function could potentially be out of sync. If this is significant to your program you need to implement your own
-synchronization.
+The device job thread is made up of a thread and a job queue. You can post a job to the thread with
+ma_device_job_thread_post(). The thread will do the processing of the job.
*/
-MA_API ma_uint32 ma_device_get_state(const ma_device* pDevice);
+typedef struct
+{
+ ma_bool32 noThread; /* Set this to true if you want to process jobs yourself. */
+ ma_uint32 jobQueueCapacity;
+ ma_uint32 jobQueueFlags;
+} ma_device_job_thread_config;
+MA_API ma_device_job_thread_config ma_device_job_thread_config_init(void);
-/*
-Sets the master volume factor for the device.
+typedef struct
+{
+ ma_thread thread;
+ ma_job_queue jobQueue;
+ ma_bool32 _hasThread;
+} ma_device_job_thread;
-The volume factor must be between 0 (silence) and 1 (full volume). Use `ma_device_set_master_gain_db()` to use decibel notation, where 0 is full volume and
-values less than 0 decreases the volume.
+MA_API ma_result ma_device_job_thread_init(const ma_device_job_thread_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_device_job_thread* pJobThread);
+MA_API void ma_device_job_thread_uninit(ma_device_job_thread* pJobThread, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_device_job_thread_post(ma_device_job_thread* pJobThread, const ma_job* pJob);
+MA_API ma_result ma_device_job_thread_next(ma_device_job_thread* pJobThread, ma_job* pJob);
-Parameters
-----------
-pDevice (in)
- A pointer to the device whose volume is being set.
-volume (in)
- The new volume factor. Must be within the range of [0, 1].
+/* Device notification types. */
+typedef enum
+{
+ ma_device_notification_type_started,
+ ma_device_notification_type_stopped,
+ ma_device_notification_type_rerouted,
+ ma_device_notification_type_interruption_began,
+ ma_device_notification_type_interruption_ended
+} ma_device_notification_type;
+typedef struct
+{
+ ma_device* pDevice;
+ ma_device_notification_type type;
+ union
+ {
+ struct
+ {
+ int _unused;
+ } started;
+ struct
+ {
+ int _unused;
+ } stopped;
+ struct
+ {
+ int _unused;
+ } rerouted;
+ struct
+ {
+ int _unused;
+ } interruption;
+ } data;
+} ma_device_notification;
-Return Value
-------------
-MA_SUCCESS if the volume was set successfully.
-MA_INVALID_ARGS if pDevice is NULL.
-MA_INVALID_ARGS if the volume factor is not within the range of [0, 1].
+/*
+The notification callback for when the application should be notified of a change to the device.
+This callback is used for notifying the application of changes such as when the device has started,
+stopped, rerouted or an interruption has occurred. Note that not all backends will post all
+notification types. For example, some backends will perform automatic stream routing without any
+kind of notification to the host program which means miniaudio will never know about it and will
+never be able to fire the rerouted notification. You should keep this in mind when designing your
+program.
-Thread Safety
--------------
-Safe. This just sets a local member of the device object.
+The stopped notification will *not* get fired when a device is rerouted.
-Callback Safety
----------------
-Safe. If you set the volume in the data callback, that data written to the output buffer will have the new volume applied.
+Parameters
+----------
+pNotification (in)
+ A pointer to a structure containing information about the event. Use the `pDevice` member of
+ this object to retrieve the relevant device. The `type` member can be used to discriminate
+ against each of the notification types.
Remarks
-------
-This applies the volume factor across all channels.
+Do not restart or uninitialize the device from the callback.
-This does not change the operating system's volume. It only affects the volume for the given `ma_device` object's audio stream.
+Not all notifications will be triggered by all backends, however the started and stopped events
+should be reliable for all backends. Some backends do not have a good way to detect device
+stoppages due to unplugging the device which may result in the stopped callback not getting
+fired. This has been observed with at least one BSD variant.
+The rerouted notification is fired *after* the reroute has occurred. The stopped notification will
+*not* get fired when a device is rerouted. The following backends are known to do automatic stream
+rerouting, but do not have a way to be notified of the change:
-See Also
---------
-ma_device_get_master_volume()
-ma_device_set_master_volume_gain_db()
-ma_device_get_master_volume_gain_db()
+ * DirectSound
+
+The interruption notifications are used on mobile platforms for detecting when audio is interrupted
+due to things like an incoming phone call. Currently this is only implemented on iOS. None of the
+Android backends will report this notification.
*/
-MA_API ma_result ma_device_set_master_volume(ma_device* pDevice, float volume);
+typedef void (* ma_device_notification_proc)(const ma_device_notification* pNotification);
+
/*
-Retrieves the master volume factor for the device.
+The callback for processing audio data from the device.
+
+The data callback is fired by miniaudio whenever the device needs to have more data delivered to a playback device, or when a capture device has some data
+available. This is called as soon as the backend asks for more data which means it may be called with inconsistent frame counts. You cannot assume the
+callback will be fired with a consistent frame count.
Parameters
----------
pDevice (in)
- A pointer to the device whose volume factor is being retrieved.
-
-pVolume (in)
- A pointer to the variable that will receive the volume factor. The returned value will be in the range of [0, 1].
-
-
-Return Value
-------------
-MA_SUCCESS if successful.
-MA_INVALID_ARGS if pDevice is NULL.
-MA_INVALID_ARGS if pVolume is NULL.
-
+ A pointer to the relevant device.
-Thread Safety
--------------
-Safe. This just a simple member retrieval.
+pOutput (out)
+ A pointer to the output buffer that will receive audio data that will later be played back through the speakers. This will be non-null for a playback or
+ full-duplex device and null for a capture and loopback device.
+pInput (in)
+ A pointer to the buffer containing input data from a recording device. This will be non-null for a capture, full-duplex or loopback device and null for a
+ playback device.
-Callback Safety
----------------
-Safe.
+frameCount (in)
+ The number of PCM frames to process. Note that this will not necessarily be equal to what you requested when you initialized the device. The
+ `periodSizeInFrames` and `periodSizeInMilliseconds` members of the device config are just hints, and are not necessarily exactly what you'll get. You must
+ not assume this will always be the same value each time the callback is fired.
Remarks
-------
-If an error occurs, `*pVolume` will be set to 0.
+You cannot stop and start the device from inside the callback or else you'll get a deadlock. You must also not uninitialize the device from inside the
+callback. The following APIs cannot be called from inside the callback:
+ ma_device_init()
+ ma_device_init_ex()
+ ma_device_uninit()
+ ma_device_start()
+ ma_device_stop()
-See Also
---------
-ma_device_set_master_volume()
-ma_device_set_master_volume_gain_db()
-ma_device_get_master_volume_gain_db()
+The proper way to stop the device is to call `ma_device_stop()` from a different thread, normally the main application thread.
*/
-MA_API ma_result ma_device_get_master_volume(ma_device* pDevice, float* pVolume);
-
-/*
-Sets the master volume for the device as gain in decibels.
-
-A gain of 0 is full volume, whereas a gain of < 0 will decrease the volume.
-
-
-Parameters
-----------
-pDevice (in)
- A pointer to the device whose gain is being set.
-
-gainDB (in)
- The new volume as gain in decibels. Must be less than or equal to 0, where 0 is full volume and anything less than 0 decreases the volume.
-
-
-Return Value
-------------
-MA_SUCCESS if the volume was set successfully.
-MA_INVALID_ARGS if pDevice is NULL.
-MA_INVALID_ARGS if the gain is > 0.
-
-
-Thread Safety
--------------
-Safe. This just sets a local member of the device object.
-
-
-Callback Safety
----------------
-Safe. If you set the volume in the data callback, that data written to the output buffer will have the new volume applied.
+typedef void (* ma_device_data_proc)(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount);
-Remarks
--------
-This applies the gain across all channels.
-This does not change the operating system's volume. It only affects the volume for the given `ma_device` object's audio stream.
+/*
+DEPRECATED. Use ma_device_notification_proc instead.
-See Also
---------
-ma_device_get_master_volume_gain_db()
-ma_device_set_master_volume()
-ma_device_get_master_volume()
-*/
-MA_API ma_result ma_device_set_master_gain_db(ma_device* pDevice, float gainDB);
+The callback for when the device has been stopped.
-/*
-Retrieves the master gain in decibels.
+This will be called when the device is stopped explicitly with `ma_device_stop()` and also called implicitly when the device is stopped through external forces
+such as being unplugged or an internal error occuring.
Parameters
----------
pDevice (in)
- A pointer to the device whose gain is being retrieved.
-
-pGainDB (in)
- A pointer to the variable that will receive the gain in decibels. The returned value will be <= 0.
-
-
-Return Value
-------------
-MA_SUCCESS if successful.
-MA_INVALID_ARGS if pDevice is NULL.
-MA_INVALID_ARGS if pGainDB is NULL.
-
-
-Thread Safety
--------------
-Safe. This just a simple member retrieval.
-
-
-Callback Safety
----------------
-Safe.
+ A pointer to the device that has just stopped.
Remarks
-------
-If an error occurs, `*pGainDB` will be set to 0.
-
-
-See Also
---------
-ma_device_set_master_volume_gain_db()
-ma_device_set_master_volume()
-ma_device_get_master_volume()
+Do not restart or uninitialize the device from the callback.
*/
-MA_API ma_result ma_device_get_master_gain_db(ma_device* pDevice, float* pGainDB);
-
-
-/*
-Called from the data callback of asynchronous backends to allow miniaudio to process the data and fire the miniaudio data callback.
+typedef void (* ma_stop_proc)(ma_device* pDevice); /* DEPRECATED. Use ma_device_notification_proc instead. */
+typedef enum
+{
+ ma_device_type_playback = 1,
+ ma_device_type_capture = 2,
+ ma_device_type_duplex = ma_device_type_playback | ma_device_type_capture, /* 3 */
+ ma_device_type_loopback = 4
+} ma_device_type;
-Parameters
-----------
-pDevice (in)
- A pointer to device whose processing the data callback.
-
-pOutput (out)
- A pointer to the buffer that will receive the output PCM frame data. On a playback device this must not be NULL. On a duplex device
- this can be NULL, in which case pInput must not be NULL.
-
-pInput (in)
- A pointer to the buffer containing input PCM frame data. On a capture device this must not be NULL. On a duplex device this can be
- NULL, in which case `pOutput` must not be NULL.
-
-frameCount (in)
- The number of frames being processed.
-
-
-Return Value
-------------
-MA_SUCCESS if successful; any other result code otherwise.
-
-
-Thread Safety
--------------
-This function should only ever be called from the internal data callback of the backend. It is safe to call this simultaneously between a
-playback and capture device in duplex setups.
-
-
-Callback Safety
----------------
-Do not call this from the miniaudio data callback. It should only ever be called from the internal data callback of the backend.
-
-
-Remarks
--------
-If both `pOutput` and `pInput` are NULL, and error will be returned. In duplex scenarios, both `pOutput` and `pInput` can be non-NULL, in
-which case `pInput` will be processed first, followed by `pOutput`.
-
-If you are implementing a custom backend, and that backend uses a callback for data delivery, you'll need to call this from inside that
-callback.
-*/
-MA_API ma_result ma_device_handle_backend_data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount);
+typedef enum
+{
+ ma_share_mode_shared = 0,
+ ma_share_mode_exclusive
+} ma_share_mode;
+/* iOS/tvOS/watchOS session categories. */
+typedef enum
+{
+ ma_ios_session_category_default = 0, /* AVAudioSessionCategoryPlayAndRecord with AVAudioSessionCategoryOptionDefaultToSpeaker. */
+ ma_ios_session_category_none, /* Leave the session category unchanged. */
+ ma_ios_session_category_ambient, /* AVAudioSessionCategoryAmbient */
+ ma_ios_session_category_solo_ambient, /* AVAudioSessionCategorySoloAmbient */
+ ma_ios_session_category_playback, /* AVAudioSessionCategoryPlayback */
+ ma_ios_session_category_record, /* AVAudioSessionCategoryRecord */
+ ma_ios_session_category_play_and_record, /* AVAudioSessionCategoryPlayAndRecord */
+ ma_ios_session_category_multi_route /* AVAudioSessionCategoryMultiRoute */
+} ma_ios_session_category;
-/*
-Calculates an appropriate buffer size from a descriptor, native sample rate and performance profile.
+/* iOS/tvOS/watchOS session category options */
+typedef enum
+{
+ ma_ios_session_category_option_mix_with_others = 0x01, /* AVAudioSessionCategoryOptionMixWithOthers */
+ ma_ios_session_category_option_duck_others = 0x02, /* AVAudioSessionCategoryOptionDuckOthers */
+ ma_ios_session_category_option_allow_bluetooth = 0x04, /* AVAudioSessionCategoryOptionAllowBluetooth */
+ ma_ios_session_category_option_default_to_speaker = 0x08, /* AVAudioSessionCategoryOptionDefaultToSpeaker */
+ ma_ios_session_category_option_interrupt_spoken_audio_and_mix_with_others = 0x11, /* AVAudioSessionCategoryOptionInterruptSpokenAudioAndMixWithOthers */
+ ma_ios_session_category_option_allow_bluetooth_a2dp = 0x20, /* AVAudioSessionCategoryOptionAllowBluetoothA2DP */
+ ma_ios_session_category_option_allow_air_play = 0x40, /* AVAudioSessionCategoryOptionAllowAirPlay */
+} ma_ios_session_category_option;
-This function is used by backends for helping determine an appropriately sized buffer to use with
-the device depending on the values of `periodSizeInFrames` and `periodSizeInMilliseconds` in the
-`pDescriptor` object. Since buffer size calculations based on time depends on the sample rate, a
-best guess at the device's native sample rate is also required which is where `nativeSampleRate`
-comes in. In addition, the performance profile is also needed for cases where both the period size
-in frames and milliseconds are both zero.
+/* OpenSL stream types. */
+typedef enum
+{
+ ma_opensl_stream_type_default = 0, /* Leaves the stream type unset. */
+ ma_opensl_stream_type_voice, /* SL_ANDROID_STREAM_VOICE */
+ ma_opensl_stream_type_system, /* SL_ANDROID_STREAM_SYSTEM */
+ ma_opensl_stream_type_ring, /* SL_ANDROID_STREAM_RING */
+ ma_opensl_stream_type_media, /* SL_ANDROID_STREAM_MEDIA */
+ ma_opensl_stream_type_alarm, /* SL_ANDROID_STREAM_ALARM */
+ ma_opensl_stream_type_notification /* SL_ANDROID_STREAM_NOTIFICATION */
+} ma_opensl_stream_type;
+/* OpenSL recording presets. */
+typedef enum
+{
+ ma_opensl_recording_preset_default = 0, /* Leaves the input preset unset. */
+ ma_opensl_recording_preset_generic, /* SL_ANDROID_RECORDING_PRESET_GENERIC */
+ ma_opensl_recording_preset_camcorder, /* SL_ANDROID_RECORDING_PRESET_CAMCORDER */
+ ma_opensl_recording_preset_voice_recognition, /* SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION */
+ ma_opensl_recording_preset_voice_communication, /* SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION */
+ ma_opensl_recording_preset_voice_unprocessed /* SL_ANDROID_RECORDING_PRESET_UNPROCESSED */
+} ma_opensl_recording_preset;
-Parameters
-----------
-pDescriptor (in)
- A pointer to device descriptor whose `periodSizeInFrames` and `periodSizeInMilliseconds` members
- will be used for the calculation of the buffer size.
+/* AAudio usage types. */
+typedef enum
+{
+ ma_aaudio_usage_default = 0, /* Leaves the usage type unset. */
+ ma_aaudio_usage_announcement, /* AAUDIO_SYSTEM_USAGE_ANNOUNCEMENT */
+ ma_aaudio_usage_emergency, /* AAUDIO_SYSTEM_USAGE_EMERGENCY */
+ ma_aaudio_usage_safety, /* AAUDIO_SYSTEM_USAGE_SAFETY */
+ ma_aaudio_usage_vehicle_status, /* AAUDIO_SYSTEM_USAGE_VEHICLE_STATUS */
+ ma_aaudio_usage_alarm, /* AAUDIO_USAGE_ALARM */
+ ma_aaudio_usage_assistance_accessibility, /* AAUDIO_USAGE_ASSISTANCE_ACCESSIBILITY */
+ ma_aaudio_usage_assistance_navigation_guidance, /* AAUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE */
+ ma_aaudio_usage_assistance_sonification, /* AAUDIO_USAGE_ASSISTANCE_SONIFICATION */
+ ma_aaudio_usage_assitant, /* AAUDIO_USAGE_ASSISTANT */
+ ma_aaudio_usage_game, /* AAUDIO_USAGE_GAME */
+ ma_aaudio_usage_media, /* AAUDIO_USAGE_MEDIA */
+ ma_aaudio_usage_notification, /* AAUDIO_USAGE_NOTIFICATION */
+ ma_aaudio_usage_notification_event, /* AAUDIO_USAGE_NOTIFICATION_EVENT */
+ ma_aaudio_usage_notification_ringtone, /* AAUDIO_USAGE_NOTIFICATION_RINGTONE */
+ ma_aaudio_usage_voice_communication, /* AAUDIO_USAGE_VOICE_COMMUNICATION */
+ ma_aaudio_usage_voice_communication_signalling /* AAUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING */
+} ma_aaudio_usage;
-nativeSampleRate (in)
- The device's native sample rate. This is only ever used when the `periodSizeInFrames` member of
- `pDescriptor` is zero. In this case, `periodSizeInMilliseconds` will be used instead, in which
- case a sample rate is required to convert to a size in frames.
+/* AAudio content types. */
+typedef enum
+{
+ ma_aaudio_content_type_default = 0, /* Leaves the content type unset. */
+ ma_aaudio_content_type_movie, /* AAUDIO_CONTENT_TYPE_MOVIE */
+ ma_aaudio_content_type_music, /* AAUDIO_CONTENT_TYPE_MUSIC */
+ ma_aaudio_content_type_sonification, /* AAUDIO_CONTENT_TYPE_SONIFICATION */
+ ma_aaudio_content_type_speech /* AAUDIO_CONTENT_TYPE_SPEECH */
+} ma_aaudio_content_type;
-performanceProfile (in)
- When both the `periodSizeInFrames` and `periodSizeInMilliseconds` members of `pDescriptor` are
- zero, miniaudio will fall back to a buffer size based on the performance profile. The profile
- to use for this calculation is determine by this parameter.
+/* AAudio input presets. */
+typedef enum
+{
+ ma_aaudio_input_preset_default = 0, /* Leaves the input preset unset. */
+ ma_aaudio_input_preset_generic, /* AAUDIO_INPUT_PRESET_GENERIC */
+ ma_aaudio_input_preset_camcorder, /* AAUDIO_INPUT_PRESET_CAMCORDER */
+ ma_aaudio_input_preset_unprocessed, /* AAUDIO_INPUT_PRESET_UNPROCESSED */
+ ma_aaudio_input_preset_voice_recognition, /* AAUDIO_INPUT_PRESET_VOICE_RECOGNITION */
+ ma_aaudio_input_preset_voice_communication, /* AAUDIO_INPUT_PRESET_VOICE_COMMUNICATION */
+ ma_aaudio_input_preset_voice_performance /* AAUDIO_INPUT_PRESET_VOICE_PERFORMANCE */
+} ma_aaudio_input_preset;
-Return Value
-------------
-The calculated buffer size in frames.
+typedef union
+{
+ ma_int64 counter;
+ double counterD;
+} ma_timer;
+typedef union
+{
+ wchar_t wasapi[64]; /* WASAPI uses a wchar_t string for identification. */
+ ma_uint8 dsound[16]; /* DirectSound uses a GUID for identification. */
+ /*UINT_PTR*/ ma_uint32 winmm; /* When creating a device, WinMM expects a Win32 UINT_PTR for device identification. In practice it's actually just a UINT. */
+ char alsa[256]; /* ALSA uses a name string for identification. */
+ char pulse[256]; /* PulseAudio uses a name string for identification. */
+ int jack; /* JACK always uses default devices. */
+ char coreaudio[256]; /* Core Audio uses a string for identification. */
+ char sndio[256]; /* "snd/0", etc. */
+ char audio4[256]; /* "/dev/audio", etc. */
+ char oss[64]; /* "dev/dsp0", etc. "dev/dsp" for the default device. */
+ ma_int32 aaudio; /* AAudio uses a 32-bit integer for identification. */
+ ma_uint32 opensl; /* OpenSL|ES uses a 32-bit unsigned integer for identification. */
+ char webaudio[32]; /* Web Audio always uses default devices for now, but if this changes it'll be a GUID. */
+ union
+ {
+ int i;
+ char s[256];
+ void* p;
+ } custom; /* The custom backend could be anything. Give them a few options. */
+ int nullbackend; /* The null backend uses an integer for device IDs. */
+} ma_device_id;
-Thread Safety
--------------
-This is safe so long as nothing modifies `pDescriptor` at the same time. However, this function
-should only ever be called from within the backend's device initialization routine and therefore
-shouldn't have any multithreading concerns.
+typedef struct ma_context_config ma_context_config;
+typedef struct ma_device_config ma_device_config;
+typedef struct ma_backend_callbacks ma_backend_callbacks;
-Callback Safety
----------------
-This is safe to call within the data callback, but there is no reason to ever do this.
+#define MA_DATA_FORMAT_FLAG_EXCLUSIVE_MODE (1U << 1) /* If set, this is supported in exclusive mode. Otherwise not natively supported by exclusive mode. */
+#ifndef MA_MAX_DEVICE_NAME_LENGTH
+#define MA_MAX_DEVICE_NAME_LENGTH 255
+#endif
-Remarks
--------
-If `nativeSampleRate` is zero, this function will fall back to `pDescriptor->sampleRate`. If that
-is also zero, `MA_DEFAULT_SAMPLE_RATE` will be used instead.
-*/
-MA_API ma_uint32 ma_calculate_buffer_size_in_frames_from_descriptor(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile);
+typedef struct
+{
+ /* Basic info. This is the only information guaranteed to be filled in during device enumeration. */
+ ma_device_id id;
+ char name[MA_MAX_DEVICE_NAME_LENGTH + 1]; /* +1 for null terminator. */
+ ma_bool32 isDefault;
+ ma_uint32 nativeDataFormatCount;
+ struct
+ {
+ ma_format format; /* Sample format. If set to ma_format_unknown, all sample formats are supported. */
+ ma_uint32 channels; /* If set to 0, all channels are supported. */
+ ma_uint32 sampleRate; /* If set to 0, all sample rates are supported. */
+ ma_uint32 flags; /* A combination of MA_DATA_FORMAT_FLAG_* flags. */
+ } nativeDataFormats[/*ma_format_count * ma_standard_sample_rate_count * MA_MAX_CHANNELS*/ 64]; /* Not sure how big to make this. There can be *many* permutations for virtual devices which can support anything. */
+} ma_device_info;
+struct ma_device_config
+{
+ ma_device_type deviceType;
+ ma_uint32 sampleRate;
+ ma_uint32 periodSizeInFrames;
+ ma_uint32 periodSizeInMilliseconds;
+ ma_uint32 periods;
+ ma_performance_profile performanceProfile;
+ ma_bool8 noPreSilencedOutputBuffer; /* When set to true, the contents of the output buffer passed into the data callback will be left undefined rather than initialized to silence. */
+ ma_bool8 noClip; /* When set to true, the contents of the output buffer passed into the data callback will be clipped after returning. Only applies when the playback sample format is f32. */
+ ma_bool8 noDisableDenormals; /* Do not disable denormals when firing the data callback. */
+ ma_bool8 noFixedSizedCallback; /* Disables strict fixed-sized data callbacks. Setting this to true will result in the period size being treated only as a hint to the backend. This is an optimization for those who don't need fixed sized callbacks. */
+ ma_device_data_proc dataCallback;
+ ma_device_notification_proc notificationCallback;
+ ma_stop_proc stopCallback;
+ void* pUserData;
+ ma_resampler_config resampling;
+ struct
+ {
+ const ma_device_id* pDeviceID;
+ ma_format format;
+ ma_uint32 channels;
+ ma_channel* pChannelMap;
+ ma_channel_mix_mode channelMixMode;
+ ma_share_mode shareMode;
+ } playback;
+ struct
+ {
+ const ma_device_id* pDeviceID;
+ ma_format format;
+ ma_uint32 channels;
+ ma_channel* pChannelMap;
+ ma_channel_mix_mode channelMixMode;
+ ma_share_mode shareMode;
+ } capture;
-/*
-Retrieves a friendly name for a backend.
-*/
-MA_API const char* ma_get_backend_name(ma_backend backend);
+ struct
+ {
+ ma_bool8 noAutoConvertSRC; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM. */
+ ma_bool8 noDefaultQualitySRC; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY. */
+ ma_bool8 noAutoStreamRouting; /* Disables automatic stream routing. */
+ ma_bool8 noHardwareOffloading; /* Disables WASAPI's hardware offloading feature. */
+ } wasapi;
+ struct
+ {
+ ma_bool32 noMMap; /* Disables MMap mode. */
+ ma_bool32 noAutoFormat; /* Opens the ALSA device with SND_PCM_NO_AUTO_FORMAT. */
+ ma_bool32 noAutoChannels; /* Opens the ALSA device with SND_PCM_NO_AUTO_CHANNELS. */
+ ma_bool32 noAutoResample; /* Opens the ALSA device with SND_PCM_NO_AUTO_RESAMPLE. */
+ } alsa;
+ struct
+ {
+ const char* pStreamNamePlayback;
+ const char* pStreamNameCapture;
+ } pulse;
+ struct
+ {
+ ma_bool32 allowNominalSampleRateChange; /* Desktop only. When enabled, allows changing of the sample rate at the operating system level. */
+ } coreaudio;
+ struct
+ {
+ ma_opensl_stream_type streamType;
+ ma_opensl_recording_preset recordingPreset;
+ } opensl;
+ struct
+ {
+ ma_aaudio_usage usage;
+ ma_aaudio_content_type contentType;
+ ma_aaudio_input_preset inputPreset;
+ ma_bool32 noAutoStartAfterReroute;
+ } aaudio;
+};
-/*
-Determines whether or not the given backend is available by the compilation environment.
-*/
-MA_API ma_bool32 ma_is_backend_enabled(ma_backend backend);
/*
-Retrieves compile-time enabled backends.
+The callback for handling device enumeration. This is fired from `ma_context_enumerated_devices()`.
Parameters
----------
-pBackends (out, optional)
- A pointer to the buffer that will receive the enabled backends. Set to NULL to retrieve the backend count. Setting
- the capacity of the buffer to `MA_BUFFER_COUNT` will guarantee it's large enough for all backends.
-
-backendCap (in)
- The capacity of the `pBackends` buffer.
-
-pBackendCount (out)
- A pointer to the variable that will receive the enabled backend count.
-
-
-Return Value
-------------
-MA_SUCCESS if successful.
-MA_INVALID_ARGS if `pBackendCount` is NULL.
-MA_NO_SPACE if the capacity of `pBackends` is not large enough.
-
-If `MA_NO_SPACE` is returned, the `pBackends` buffer will be filled with `*pBackendCount` values.
-
-
-Thread Safety
--------------
-Safe.
-
-
-Callback Safety
----------------
-Safe.
-
-
-Remarks
--------
-If you want to retrieve the number of backends so you can determine the capacity of `pBackends` buffer, you can call
-this function with `pBackends` set to NULL.
-
-This will also enumerate the null backend. If you don't want to include this you need to check for `ma_backend_null`
-when you enumerate over the returned backends and handle it appropriately. Alternatively, you can disable it at
-compile time with `MA_NO_NULL`.
-
-The returned backends are determined based on compile time settings, not the platform it's currently running on. For
-example, PulseAudio will be returned if it was enabled at compile time, even when the user doesn't actually have
-PulseAudio installed.
-
-
-Example 1
----------
-The example below retrieves the enabled backend count using a fixed sized buffer allocated on the stack. The buffer is
-given a capacity of `MA_BACKEND_COUNT` which will guarantee it'll be large enough to store all available backends.
-Since `MA_BACKEND_COUNT` is always a relatively small value, this should be suitable for most scenarios.
-
-```
-ma_backend enabledBackends[MA_BACKEND_COUNT];
-size_t enabledBackendCount;
-
-result = ma_get_enabled_backends(enabledBackends, MA_BACKEND_COUNT, &enabledBackendCount);
-if (result != MA_SUCCESS) {
- // Failed to retrieve enabled backends. Should never happen in this example since all inputs are valid.
-}
-```
-
-
-See Also
---------
-ma_is_backend_enabled()
-*/
-MA_API ma_result ma_get_enabled_backends(ma_backend* pBackends, size_t backendCap, size_t* pBackendCount);
-
-/*
-Determines whether or not loopback mode is support by a backend.
-*/
-MA_API ma_bool32 ma_is_loopback_supported(ma_backend backend);
-
-#endif /* MA_NO_DEVICE_IO */
-
-
-#ifndef MA_NO_THREADING
-
-/*
-Locks a spinlock.
-*/
-MA_API ma_result ma_spinlock_lock(volatile ma_spinlock* pSpinlock);
-
-/*
-Locks a spinlock, but does not yield() when looping.
-*/
-MA_API ma_result ma_spinlock_lock_noyield(volatile ma_spinlock* pSpinlock);
-
-/*
-Unlocks a spinlock.
-*/
-MA_API ma_result ma_spinlock_unlock(volatile ma_spinlock* pSpinlock);
+pContext (in)
+ A pointer to the context performing the enumeration.
+deviceType (in)
+ The type of the device being enumerated. This will always be either `ma_device_type_playback` or `ma_device_type_capture`.
-/*
-Creates a mutex.
+pInfo (in)
+ A pointer to a `ma_device_info` containing the ID and name of the enumerated device. Note that this will not include detailed information about the device,
+ only basic information (ID and name). The reason for this is that it would otherwise require opening the backend device to probe for the information which
+ is too inefficient.
-A mutex must be created from a valid context. A mutex is initially unlocked.
+pUserData (in)
+ The user data pointer passed into `ma_context_enumerate_devices()`.
*/
-MA_API ma_result ma_mutex_init(ma_mutex* pMutex);
+typedef ma_bool32 (* ma_enum_devices_callback_proc)(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pInfo, void* pUserData);
-/*
-Deletes a mutex.
-*/
-MA_API void ma_mutex_uninit(ma_mutex* pMutex);
/*
-Locks a mutex with an infinite timeout.
+Describes some basic details about a playback or capture device.
*/
-MA_API void ma_mutex_lock(ma_mutex* pMutex);
+typedef struct
+{
+ const ma_device_id* pDeviceID;
+ ma_share_mode shareMode;
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ ma_channel channelMap[MA_MAX_CHANNELS];
+ ma_uint32 periodSizeInFrames;
+ ma_uint32 periodSizeInMilliseconds;
+ ma_uint32 periodCount;
+} ma_device_descriptor;
/*
-Unlocks a mutex.
-*/
-MA_API void ma_mutex_unlock(ma_mutex* pMutex);
-
+These are the callbacks required to be implemented for a backend. These callbacks are grouped into two parts: context and device. There is one context
+to many devices. A device is created from a context.
-/*
-Initializes an auto-reset event.
-*/
-MA_API ma_result ma_event_init(ma_event* pEvent);
+The general flow goes like this:
-/*
-Uninitializes an auto-reset event.
-*/
-MA_API void ma_event_uninit(ma_event* pEvent);
+ 1) A context is created with `onContextInit()`
+ 1a) Available devices can be enumerated with `onContextEnumerateDevices()` if required.
+ 1b) Detailed information about a device can be queried with `onContextGetDeviceInfo()` if required.
+ 2) A device is created from the context that was created in the first step using `onDeviceInit()`, and optionally a device ID that was
+ selected from device enumeration via `onContextEnumerateDevices()`.
+ 3) A device is started or stopped with `onDeviceStart()` / `onDeviceStop()`
+ 4) Data is delivered to and from the device by the backend. This is always done based on the native format returned by the prior call
+ to `onDeviceInit()`. Conversion between the device's native format and the format requested by the application will be handled by
+ miniaudio internally.
-/*
-Waits for the specified auto-reset event to become signalled.
-*/
-MA_API ma_result ma_event_wait(ma_event* pEvent);
+Initialization of the context is quite simple. You need to do any necessary initialization of internal objects and then output the
+callbacks defined in this structure.
-/*
-Signals the specified auto-reset event.
-*/
-MA_API ma_result ma_event_signal(ma_event* pEvent);
-#endif /* MA_NO_THREADING */
+Once the context has been initialized you can initialize a device. Before doing so, however, the application may want to know which
+physical devices are available. This is where `onContextEnumerateDevices()` comes in. This is fairly simple. For each device, fire the
+given callback with, at a minimum, the basic information filled out in `ma_device_info`. When the callback returns `MA_FALSE`, enumeration
+needs to stop and the `onContextEnumerateDevices()` function returns with a success code.
+Detailed device information can be retrieved from a device ID using `onContextGetDeviceInfo()`. This takes as input the device type and ID,
+and on output returns detailed information about the device in `ma_device_info`. The `onContextGetDeviceInfo()` callback must handle the
+case when the device ID is NULL, in which case information about the default device needs to be retrieved.
-/************************************************************************************************************************************************************
+Once the context has been created and the device ID retrieved (if using anything other than the default device), the device can be created.
+This is a little bit more complicated than initialization of the context due to it's more complicated configuration. When initializing a
+device, a duplex device may be requested. This means a separate data format needs to be specified for both playback and capture. On input,
+the data format is set to what the application wants. On output it's set to the native format which should match as closely as possible to
+the requested format. The conversion between the format requested by the application and the device's native format will be handled
+internally by miniaudio.
-Utiltities
+On input, if the sample format is set to `ma_format_unknown`, the backend is free to use whatever sample format it desires, so long as it's
+supported by miniaudio. When the channel count is set to 0, the backend should use the device's native channel count. The same applies for
+sample rate. For the channel map, the default should be used when `ma_channel_map_is_blank()` returns true (all channels set to
+`MA_CHANNEL_NONE`). On input, the `periodSizeInFrames` or `periodSizeInMilliseconds` option should always be set. The backend should
+inspect both of these variables. If `periodSizeInFrames` is set, it should take priority, otherwise it needs to be derived from the period
+size in milliseconds (`periodSizeInMilliseconds`) and the sample rate, keeping in mind that the sample rate may be 0, in which case the
+sample rate will need to be determined before calculating the period size in frames. On output, all members of the `ma_device_data_format`
+object should be set to a valid value, except for `periodSizeInMilliseconds` which is optional (`periodSizeInFrames` *must* be set).
-************************************************************************************************************************************************************/
+Starting and stopping of the device is done with `onDeviceStart()` and `onDeviceStop()` and should be self-explanatory. If the backend uses
+asynchronous reading and writing, `onDeviceStart()` and `onDeviceStop()` should always be implemented.
-/*
-Adjust buffer size based on a scaling factor.
+The handling of data delivery between the application and the device is the most complicated part of the process. To make this a bit
+easier, some helper callbacks are available. If the backend uses a blocking read/write style of API, the `onDeviceRead()` and
+`onDeviceWrite()` callbacks can optionally be implemented. These are blocking and work just like reading and writing from a file. If the
+backend uses a callback for data delivery, that callback must call `ma_device_handle_backend_data_callback()` from within it's callback.
+This allows miniaudio to then process any necessary data conversion and then pass it to the miniaudio data callback.
-This just multiplies the base size by the scaling factor, making sure it's a size of at least 1.
-*/
-MA_API ma_uint32 ma_scale_buffer_size(ma_uint32 baseBufferSize, float scale);
+If the backend requires absolute flexibility with it's data delivery, it can optionally implement the `onDeviceDataLoop()` callback
+which will allow it to implement the logic that will run on the audio thread. This is much more advanced and is completely optional.
-/*
-Calculates a buffer size in milliseconds from the specified number of frames and sample rate.
-*/
-MA_API ma_uint32 ma_calculate_buffer_size_in_milliseconds_from_frames(ma_uint32 bufferSizeInFrames, ma_uint32 sampleRate);
+The audio thread should run data delivery logic in a loop while `ma_device_get_state() == ma_device_state_started` and no errors have been
+encounted. Do not start or stop the device here. That will be handled from outside the `onDeviceDataLoop()` callback.
-/*
-Calculates a buffer size in frames from the specified number of milliseconds and sample rate.
-*/
-MA_API ma_uint32 ma_calculate_buffer_size_in_frames_from_milliseconds(ma_uint32 bufferSizeInMilliseconds, ma_uint32 sampleRate);
+The invocation of the `onDeviceDataLoop()` callback will be handled by miniaudio. When you start the device, miniaudio will fire this
+callback. When the device is stopped, the `ma_device_get_state() == ma_device_state_started` condition will fail and the loop will be terminated
+which will then fall through to the part that stops the device. For an example on how to implement the `onDeviceDataLoop()` callback,
+look at `ma_device_audio_thread__default_read_write()`. Implement the `onDeviceDataLoopWakeup()` callback if you need a mechanism to
+wake up the audio thread.
-/*
-Copies PCM frames from one buffer to another.
+If the backend supports an optimized retrieval of device information from an initialized `ma_device` object, it should implement the
+`onDeviceGetInfo()` callback. This is optional, in which case it will fall back to `onContextGetDeviceInfo()` which is less efficient.
*/
-MA_API void ma_copy_pcm_frames(void* dst, const void* src, ma_uint64 frameCount, ma_format format, ma_uint32 channels);
+struct ma_backend_callbacks
+{
+ ma_result (* onContextInit)(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks);
+ ma_result (* onContextUninit)(ma_context* pContext);
+ ma_result (* onContextEnumerateDevices)(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData);
+ ma_result (* onContextGetDeviceInfo)(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo);
+ ma_result (* onDeviceInit)(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture);
+ ma_result (* onDeviceUninit)(ma_device* pDevice);
+ ma_result (* onDeviceStart)(ma_device* pDevice);
+ ma_result (* onDeviceStop)(ma_device* pDevice);
+ ma_result (* onDeviceRead)(ma_device* pDevice, void* pFrames, ma_uint32 frameCount, ma_uint32* pFramesRead);
+ ma_result (* onDeviceWrite)(ma_device* pDevice, const void* pFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten);
+ ma_result (* onDeviceDataLoop)(ma_device* pDevice);
+ ma_result (* onDeviceDataLoopWakeup)(ma_device* pDevice);
+ ma_result (* onDeviceGetInfo)(ma_device* pDevice, ma_device_type type, ma_device_info* pDeviceInfo);
+};
-/*
-Copies silent frames into the given buffer.
-
-Remarks
--------
-For all formats except `ma_format_u8`, the output buffer will be filled with 0. For `ma_format_u8` it will be filled with 128. The reason for this is that it
-makes more sense for the purpose of mixing to initialize it to the center point.
-*/
-MA_API void ma_silence_pcm_frames(void* p, ma_uint64 frameCount, ma_format format, ma_uint32 channels);
-static MA_INLINE void ma_zero_pcm_frames(void* p, ma_uint64 frameCount, ma_format format, ma_uint32 channels) { ma_silence_pcm_frames(p, frameCount, format, channels); }
-
-
-/*
-Offsets a pointer by the specified number of PCM frames.
-*/
-MA_API void* ma_offset_pcm_frames_ptr(void* p, ma_uint64 offsetInFrames, ma_format format, ma_uint32 channels);
-MA_API const void* ma_offset_pcm_frames_const_ptr(const void* p, ma_uint64 offsetInFrames, ma_format format, ma_uint32 channels);
-static MA_INLINE float* ma_offset_pcm_frames_ptr_f32(float* p, ma_uint64 offsetInFrames, ma_uint32 channels) { return (float*)ma_offset_pcm_frames_ptr((void*)p, offsetInFrames, ma_format_f32, channels); }
-static MA_INLINE const float* ma_offset_pcm_frames_const_ptr_f32(const float* p, ma_uint64 offsetInFrames, ma_uint32 channels) { return (const float*)ma_offset_pcm_frames_const_ptr((const void*)p, offsetInFrames, ma_format_f32, channels); }
-
-
-/*
-Clips f32 samples.
-*/
-MA_API void ma_clip_samples_f32(float* p, ma_uint64 sampleCount);
-static MA_INLINE void ma_clip_pcm_frames_f32(float* p, ma_uint64 frameCount, ma_uint32 channels) { ma_clip_samples_f32(p, frameCount*channels); }
-
-/*
-Helper for applying a volume factor to samples.
-
-Note that the source and destination buffers can be the same, in which case it'll perform the operation in-place.
-*/
-MA_API void ma_copy_and_apply_volume_factor_u8(ma_uint8* pSamplesOut, const ma_uint8* pSamplesIn, ma_uint64 sampleCount, float factor);
-MA_API void ma_copy_and_apply_volume_factor_s16(ma_int16* pSamplesOut, const ma_int16* pSamplesIn, ma_uint64 sampleCount, float factor);
-MA_API void ma_copy_and_apply_volume_factor_s24(void* pSamplesOut, const void* pSamplesIn, ma_uint64 sampleCount, float factor);
-MA_API void ma_copy_and_apply_volume_factor_s32(ma_int32* pSamplesOut, const ma_int32* pSamplesIn, ma_uint64 sampleCount, float factor);
-MA_API void ma_copy_and_apply_volume_factor_f32(float* pSamplesOut, const float* pSamplesIn, ma_uint64 sampleCount, float factor);
-
-MA_API void ma_apply_volume_factor_u8(ma_uint8* pSamples, ma_uint64 sampleCount, float factor);
-MA_API void ma_apply_volume_factor_s16(ma_int16* pSamples, ma_uint64 sampleCount, float factor);
-MA_API void ma_apply_volume_factor_s24(void* pSamples, ma_uint64 sampleCount, float factor);
-MA_API void ma_apply_volume_factor_s32(ma_int32* pSamples, ma_uint64 sampleCount, float factor);
-MA_API void ma_apply_volume_factor_f32(float* pSamples, ma_uint64 sampleCount, float factor);
-
-MA_API void ma_copy_and_apply_volume_factor_pcm_frames_u8(ma_uint8* pPCMFramesOut, const ma_uint8* pPCMFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor);
-MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s16(ma_int16* pPCMFramesOut, const ma_int16* pPCMFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor);
-MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s24(void* pPCMFramesOut, const void* pPCMFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor);
-MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s32(ma_int32* pPCMFramesOut, const ma_int32* pPCMFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor);
-MA_API void ma_copy_and_apply_volume_factor_pcm_frames_f32(float* pPCMFramesOut, const float* pPCMFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor);
-MA_API void ma_copy_and_apply_volume_factor_pcm_frames(void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount, ma_format format, ma_uint32 channels, float factor);
-
-MA_API void ma_apply_volume_factor_pcm_frames_u8(ma_uint8* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor);
-MA_API void ma_apply_volume_factor_pcm_frames_s16(ma_int16* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor);
-MA_API void ma_apply_volume_factor_pcm_frames_s24(void* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor);
-MA_API void ma_apply_volume_factor_pcm_frames_s32(ma_int32* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor);
-MA_API void ma_apply_volume_factor_pcm_frames_f32(float* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor);
-MA_API void ma_apply_volume_factor_pcm_frames(void* pFrames, ma_uint64 frameCount, ma_format format, ma_uint32 channels, float factor);
-
-
-/*
-Helper for converting a linear factor to gain in decibels.
-*/
-MA_API float ma_factor_to_gain_db(float factor);
-
-/*
-Helper for converting gain in decibels to a linear factor.
-*/
-MA_API float ma_gain_db_to_factor(float gain);
-
-
-typedef void ma_data_source;
-
-typedef struct
-{
- ma_result (* onRead)(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
- ma_result (* onSeek)(ma_data_source* pDataSource, ma_uint64 frameIndex);
- ma_result (* onMap)(ma_data_source* pDataSource, void** ppFramesOut, ma_uint64* pFrameCount); /* Returns MA_AT_END if the end has been reached. This should be considered successful. */
- ma_result (* onUnmap)(ma_data_source* pDataSource, ma_uint64 frameCount);
- ma_result (* onGetDataFormat)(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate);
- ma_result (* onGetCursor)(ma_data_source* pDataSource, ma_uint64* pCursor);
- ma_result (* onGetLength)(ma_data_source* pDataSource, ma_uint64* pLength);
-} ma_data_source_vtable, ma_data_source_callbacks; /* TODO: Remove ma_data_source_callbacks in version 0.11. */
-
-typedef ma_data_source* (* ma_data_source_get_next_proc)(ma_data_source* pDataSource);
-
-typedef struct
-{
- const ma_data_source_vtable* vtable; /* Can be null, which is useful for proxies. */
-} ma_data_source_config;
-
-MA_API ma_data_source_config ma_data_source_config_init(void);
-
-
-typedef struct
-{
- ma_data_source_callbacks cb; /* TODO: Remove this. */
-
- /* Variables below are placeholder and not yet used. */
- const ma_data_source_vtable* vtable;
- ma_uint64 rangeBegInFrames;
- ma_uint64 rangeEndInFrames; /* Set to -1 for unranged (default). */
- ma_uint64 loopBegInFrames; /* Relative to rangeBegInFrames. */
- ma_uint64 loopEndInFrames; /* Relative to rangeBegInFrames. Set to -1 for the end of the range. */
- ma_data_source* pCurrent; /* When non-NULL, the data source being initialized will act as a proxy and will route all operations to pCurrent. Used in conjunction with pNext/onGetNext for seamless chaining. */
- ma_data_source* pNext; /* When set to NULL, onGetNext will be used. */
- ma_data_source_get_next_proc onGetNext; /* Will be used when pNext is NULL. If both are NULL, no next will be used. */
-} ma_data_source_base;
-
-MA_API ma_result ma_data_source_init(const ma_data_source_config* pConfig, ma_data_source* pDataSource);
-MA_API void ma_data_source_uninit(ma_data_source* pDataSource);
-MA_API ma_result ma_data_source_read_pcm_frames(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead, ma_bool32 loop); /* Must support pFramesOut = NULL in which case a forward seek should be performed. */
-MA_API ma_result ma_data_source_seek_pcm_frames(ma_data_source* pDataSource, ma_uint64 frameCount, ma_uint64* pFramesSeeked, ma_bool32 loop); /* Can only seek forward. Equivalent to ma_data_source_read_pcm_frames(pDataSource, NULL, frameCount); */
-MA_API ma_result ma_data_source_seek_to_pcm_frame(ma_data_source* pDataSource, ma_uint64 frameIndex);
-MA_API ma_result ma_data_source_map(ma_data_source* pDataSource, void** ppFramesOut, ma_uint64* pFrameCount); /* Returns MA_NOT_IMPLEMENTED if mapping is not supported. */
-MA_API ma_result ma_data_source_unmap(ma_data_source* pDataSource, ma_uint64 frameCount); /* Returns MA_AT_END if the end has been reached. */
-MA_API ma_result ma_data_source_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate);
-MA_API ma_result ma_data_source_get_cursor_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pCursor);
-MA_API ma_result ma_data_source_get_length_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pLength); /* Returns MA_NOT_IMPLEMENTED if the length is unknown or cannot be determined. Decoders can return this. */
-#if defined(MA_EXPERIMENTAL__DATA_LOOPING_AND_CHAINING)
-MA_API ma_result ma_data_source_set_range_in_pcm_frames(ma_data_source* pDataSource, ma_uint64 rangeBegInFrames, ma_uint64 rangeEndInFrames);
-MA_API void ma_data_source_get_range_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pRangeBegInFrames, ma_uint64* pRangeEndInFrames);
-MA_API ma_result ma_data_source_set_loop_point_in_pcm_frames(ma_data_source* pDataSource, ma_uint64 loopBegInFrames, ma_uint64 loopEndInFrames);
-MA_API void ma_data_source_get_loop_point_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pLoopBegInFrames, ma_uint64* pLoopEndInFrames);
-MA_API ma_result ma_data_source_set_current(ma_data_source* pDataSource, ma_data_source* pCurrentDataSource);
-MA_API ma_data_source* ma_data_source_get_current(ma_data_source* pDataSource);
-MA_API ma_result ma_data_source_set_next(ma_data_source* pDataSource, ma_data_source* pNextDataSource);
-MA_API ma_data_source* ma_data_source_get_next(ma_data_source* pDataSource);
-MA_API ma_result ma_data_source_set_next_callback(ma_data_source* pDataSource, ma_data_source_get_next_proc onGetNext);
-MA_API ma_data_source_get_next_proc ma_data_source_get_next_callback(ma_data_source* pDataSource);
-#endif
-
-
-typedef struct
-{
- ma_data_source_base ds;
- ma_format format;
- ma_uint32 channels;
- ma_uint64 cursor;
- ma_uint64 sizeInFrames;
- const void* pData;
-} ma_audio_buffer_ref;
-
-MA_API ma_result ma_audio_buffer_ref_init(ma_format format, ma_uint32 channels, const void* pData, ma_uint64 sizeInFrames, ma_audio_buffer_ref* pAudioBufferRef);
-MA_API void ma_audio_buffer_ref_uninit(ma_audio_buffer_ref* pAudioBufferRef);
-MA_API ma_result ma_audio_buffer_ref_set_data(ma_audio_buffer_ref* pAudioBufferRef, const void* pData, ma_uint64 sizeInFrames);
-MA_API ma_uint64 ma_audio_buffer_ref_read_pcm_frames(ma_audio_buffer_ref* pAudioBufferRef, void* pFramesOut, ma_uint64 frameCount, ma_bool32 loop);
-MA_API ma_result ma_audio_buffer_ref_seek_to_pcm_frame(ma_audio_buffer_ref* pAudioBufferRef, ma_uint64 frameIndex);
-MA_API ma_result ma_audio_buffer_ref_map(ma_audio_buffer_ref* pAudioBufferRef, void** ppFramesOut, ma_uint64* pFrameCount);
-MA_API ma_result ma_audio_buffer_ref_unmap(ma_audio_buffer_ref* pAudioBufferRef, ma_uint64 frameCount); /* Returns MA_AT_END if the end has been reached. This should be considered successful. */
-MA_API ma_bool32 ma_audio_buffer_ref_at_end(const ma_audio_buffer_ref* pAudioBufferRef);
-MA_API ma_result ma_audio_buffer_ref_get_cursor_in_pcm_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pCursor);
-MA_API ma_result ma_audio_buffer_ref_get_length_in_pcm_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pLength);
-MA_API ma_result ma_audio_buffer_ref_get_available_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pAvailableFrames);
-
-
-
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- ma_uint64 sizeInFrames;
- const void* pData; /* If set to NULL, will allocate a block of memory for you. */
- ma_allocation_callbacks allocationCallbacks;
-} ma_audio_buffer_config;
-
-MA_API ma_audio_buffer_config ma_audio_buffer_config_init(ma_format format, ma_uint32 channels, ma_uint64 sizeInFrames, const void* pData, const ma_allocation_callbacks* pAllocationCallbacks);
-
-typedef struct
+struct ma_context_config
{
- ma_audio_buffer_ref ref;
+ ma_log* pLog;
+ ma_thread_priority threadPriority;
+ size_t threadStackSize;
+ void* pUserData;
ma_allocation_callbacks allocationCallbacks;
- ma_bool32 ownsData; /* Used to control whether or not miniaudio owns the data buffer. If set to true, pData will be freed in ma_audio_buffer_uninit(). */
- ma_uint8 _pExtraData[1]; /* For allocating a buffer with the memory located directly after the other memory of the structure. */
-} ma_audio_buffer;
-
-MA_API ma_result ma_audio_buffer_init(const ma_audio_buffer_config* pConfig, ma_audio_buffer* pAudioBuffer);
-MA_API ma_result ma_audio_buffer_init_copy(const ma_audio_buffer_config* pConfig, ma_audio_buffer* pAudioBuffer);
-MA_API ma_result ma_audio_buffer_alloc_and_init(const ma_audio_buffer_config* pConfig, ma_audio_buffer** ppAudioBuffer); /* Always copies the data. Doesn't make sense to use this otherwise. Use ma_audio_buffer_uninit_and_free() to uninit. */
-MA_API void ma_audio_buffer_uninit(ma_audio_buffer* pAudioBuffer);
-MA_API void ma_audio_buffer_uninit_and_free(ma_audio_buffer* pAudioBuffer);
-MA_API ma_uint64 ma_audio_buffer_read_pcm_frames(ma_audio_buffer* pAudioBuffer, void* pFramesOut, ma_uint64 frameCount, ma_bool32 loop);
-MA_API ma_result ma_audio_buffer_seek_to_pcm_frame(ma_audio_buffer* pAudioBuffer, ma_uint64 frameIndex);
-MA_API ma_result ma_audio_buffer_map(ma_audio_buffer* pAudioBuffer, void** ppFramesOut, ma_uint64* pFrameCount);
-MA_API ma_result ma_audio_buffer_unmap(ma_audio_buffer* pAudioBuffer, ma_uint64 frameCount); /* Returns MA_AT_END if the end has been reached. This should be considered successful. */
-MA_API ma_bool32 ma_audio_buffer_at_end(const ma_audio_buffer* pAudioBuffer);
-MA_API ma_result ma_audio_buffer_get_cursor_in_pcm_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pCursor);
-MA_API ma_result ma_audio_buffer_get_length_in_pcm_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pLength);
-MA_API ma_result ma_audio_buffer_get_available_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pAvailableFrames);
-
-
-
-/************************************************************************************************************************************************************
-
-VFS
-===
-
-The VFS object (virtual file system) is what's used to customize file access. This is useful in cases where stdio FILE* based APIs may not be entirely
-appropriate for a given situation.
-
-************************************************************************************************************************************************************/
-typedef void ma_vfs;
-typedef ma_handle ma_vfs_file;
-
-#define MA_OPEN_MODE_READ 0x00000001
-#define MA_OPEN_MODE_WRITE 0x00000002
-
-typedef enum
-{
- ma_seek_origin_start,
- ma_seek_origin_current,
- ma_seek_origin_end /* Not used by decoders. */
-} ma_seek_origin;
-
-typedef struct
-{
- ma_uint64 sizeInBytes;
-} ma_file_info;
-
-typedef struct
-{
- ma_result (* onOpen) (ma_vfs* pVFS, const char* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile);
- ma_result (* onOpenW)(ma_vfs* pVFS, const wchar_t* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile);
- ma_result (* onClose)(ma_vfs* pVFS, ma_vfs_file file);
- ma_result (* onRead) (ma_vfs* pVFS, ma_vfs_file file, void* pDst, size_t sizeInBytes, size_t* pBytesRead);
- ma_result (* onWrite)(ma_vfs* pVFS, ma_vfs_file file, const void* pSrc, size_t sizeInBytes, size_t* pBytesWritten);
- ma_result (* onSeek) (ma_vfs* pVFS, ma_vfs_file file, ma_int64 offset, ma_seek_origin origin);
- ma_result (* onTell) (ma_vfs* pVFS, ma_vfs_file file, ma_int64* pCursor);
- ma_result (* onInfo) (ma_vfs* pVFS, ma_vfs_file file, ma_file_info* pInfo);
-} ma_vfs_callbacks;
-
-MA_API ma_result ma_vfs_open(ma_vfs* pVFS, const char* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile);
-MA_API ma_result ma_vfs_open_w(ma_vfs* pVFS, const wchar_t* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile);
-MA_API ma_result ma_vfs_close(ma_vfs* pVFS, ma_vfs_file file);
-MA_API ma_result ma_vfs_read(ma_vfs* pVFS, ma_vfs_file file, void* pDst, size_t sizeInBytes, size_t* pBytesRead);
-MA_API ma_result ma_vfs_write(ma_vfs* pVFS, ma_vfs_file file, const void* pSrc, size_t sizeInBytes, size_t* pBytesWritten);
-MA_API ma_result ma_vfs_seek(ma_vfs* pVFS, ma_vfs_file file, ma_int64 offset, ma_seek_origin origin);
-MA_API ma_result ma_vfs_tell(ma_vfs* pVFS, ma_vfs_file file, ma_int64* pCursor);
-MA_API ma_result ma_vfs_info(ma_vfs* pVFS, ma_vfs_file file, ma_file_info* pInfo);
-MA_API ma_result ma_vfs_open_and_read_file(ma_vfs* pVFS, const char* pFilePath, void** ppData, size_t* pSize, const ma_allocation_callbacks* pAllocationCallbacks);
-
-typedef struct
-{
- ma_vfs_callbacks cb;
- ma_allocation_callbacks allocationCallbacks; /* Only used for the wchar_t version of open() on non-Windows platforms. */
-} ma_default_vfs;
-
-MA_API ma_result ma_default_vfs_init(ma_default_vfs* pVFS, const ma_allocation_callbacks* pAllocationCallbacks);
-
-
-
-typedef ma_result (* ma_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead, size_t* pBytesRead);
-typedef ma_result (* ma_seek_proc)(void* pUserData, ma_int64 offset, ma_seek_origin origin);
-typedef ma_result (* ma_tell_proc)(void* pUserData, ma_int64* pCursor);
-
-
-
-#if !defined(MA_NO_DECODING) || !defined(MA_NO_ENCODING)
-typedef enum
-{
- ma_resource_format_wav
-} ma_resource_format;
-
-typedef enum
-{
- ma_encoding_format_unknown = 0,
- ma_encoding_format_wav,
- ma_encoding_format_flac,
- ma_encoding_format_mp3,
- ma_encoding_format_vorbis
-} ma_encoding_format;
-#endif
-
-/************************************************************************************************************************************************************
-
-Decoding
-========
-
-Decoders are independent of the main device API. Decoding APIs can be called freely inside the device's data callback, but they are not thread safe unless
-you do your own synchronization.
-
-************************************************************************************************************************************************************/
-#ifndef MA_NO_DECODING
-typedef struct ma_decoder ma_decoder;
-
-
-typedef struct
-{
- ma_format preferredFormat;
-} ma_decoding_backend_config;
-
-MA_API ma_decoding_backend_config ma_decoding_backend_config_init(ma_format preferredFormat);
-
-
-typedef struct
-{
- ma_result (* onInit )(void* pUserData, ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend);
- ma_result (* onInitFile )(void* pUserData, const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend); /* Optional. */
- ma_result (* onInitFileW )(void* pUserData, const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend); /* Optional. */
- ma_result (* onInitMemory )(void* pUserData, const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend); /* Optional. */
- void (* onUninit )(void* pUserData, ma_data_source* pBackend, const ma_allocation_callbacks* pAllocationCallbacks);
- ma_result (* onGetChannelMap)(void* pUserData, ma_data_source* pBackend, ma_channel* pChannelMap, size_t channelMapCap);
-} ma_decoding_backend_vtable;
-
-
-/* TODO: Convert read and seek to be consistent with the VFS API (ma_result return value, bytes read moved to an output parameter). */
-typedef size_t (* ma_decoder_read_proc)(ma_decoder* pDecoder, void* pBufferOut, size_t bytesToRead); /* Returns the number of bytes read. */
-typedef ma_bool32 (* ma_decoder_seek_proc)(ma_decoder* pDecoder, ma_int64 byteOffset, ma_seek_origin origin);
-typedef ma_result (* ma_decoder_tell_proc)(ma_decoder* pDecoder, ma_int64* pCursor);
+ struct
+ {
+ ma_bool32 useVerboseDeviceEnumeration;
+ } alsa;
+ struct
+ {
+ const char* pApplicationName;
+ const char* pServerName;
+ ma_bool32 tryAutoSpawn; /* Enables autospawning of the PulseAudio daemon if necessary. */
+ } pulse;
+ struct
+ {
+ ma_ios_session_category sessionCategory;
+ ma_uint32 sessionCategoryOptions;
+ ma_bool32 noAudioSessionActivate; /* iOS only. When set to true, does not perform an explicit [[AVAudioSession sharedInstace] setActive:true] on initialization. */
+ ma_bool32 noAudioSessionDeactivate; /* iOS only. When set to true, does not perform an explicit [[AVAudioSession sharedInstace] setActive:false] on uninitialization. */
+ } coreaudio;
+ struct
+ {
+ const char* pClientName;
+ ma_bool32 tryStartServer;
+ } jack;
+ ma_backend_callbacks custom;
+};
+/* WASAPI specific structure for some commands which must run on a common thread due to bugs in WASAPI. */
typedef struct
{
- ma_format format; /* Set to 0 or ma_format_unknown to use the stream's internal format. */
- ma_uint32 channels; /* Set to 0 to use the stream's internal channels. */
- ma_uint32 sampleRate; /* Set to 0 to use the stream's internal sample rate. */
- ma_channel channelMap[MA_MAX_CHANNELS];
- ma_channel_mix_mode channelMixMode;
- ma_dither_mode ditherMode;
- struct
+ int code;
+ ma_event* pEvent; /* This will be signalled when the event is complete. */
+ union
{
- ma_resample_algorithm algorithm;
struct
{
- ma_uint32 lpfOrder;
- } linear;
+ int _unused;
+ } quit;
struct
{
- int quality;
- } speex;
- } resampling;
- ma_allocation_callbacks allocationCallbacks;
- ma_encoding_format encodingFormat;
- ma_decoding_backend_vtable** ppCustomBackendVTables;
- ma_uint32 customBackendCount;
- void* pCustomBackendUserData;
-} ma_decoder_config;
+ ma_device_type deviceType;
+ void* pAudioClient;
+ void** ppAudioClientService;
+ ma_result* pResult; /* The result from creating the audio client service. */
+ } createAudioClient;
+ struct
+ {
+ ma_device* pDevice;
+ ma_device_type deviceType;
+ } releaseAudioClient;
+ } data;
+} ma_context_command__wasapi;
-struct ma_decoder
+struct ma_context
{
- ma_data_source_base ds;
- ma_data_source* pBackend; /* The decoding backend we'll be pulling data from. */
- const ma_decoding_backend_vtable* pBackendVTable; /* The vtable for the decoding backend. This needs to be stored so we can access the onUninit() callback. */
- void* pBackendUserData;
- ma_decoder_read_proc onRead;
- ma_decoder_seek_proc onSeek;
- ma_decoder_tell_proc onTell;
+ ma_backend_callbacks callbacks;
+ ma_backend backend; /* DirectSound, ALSA, etc. */
+ ma_log* pLog;
+ ma_log log; /* Only used if the log is owned by the context. The pLog member will be set to &log in this case. */
+ ma_thread_priority threadPriority;
+ size_t threadStackSize;
void* pUserData;
- ma_uint64 readPointerInPCMFrames; /* In output sample rate. Used for keeping track of how many frames are available for decoding. */
- ma_format outputFormat;
- ma_uint32 outputChannels;
- ma_uint32 outputSampleRate;
- ma_channel outputChannelMap[MA_MAX_CHANNELS];
- ma_data_converter converter; /* <-- Data conversion is achieved by running frames through this. */
ma_allocation_callbacks allocationCallbacks;
+ ma_mutex deviceEnumLock; /* Used to make ma_context_get_devices() thread safe. */
+ ma_mutex deviceInfoLock; /* Used to make ma_context_get_device_info() thread safe. */
+ ma_uint32 deviceInfoCapacity; /* Total capacity of pDeviceInfos. */
+ ma_uint32 playbackDeviceInfoCount;
+ ma_uint32 captureDeviceInfoCount;
+ ma_device_info* pDeviceInfos; /* Playback devices first, then capture. */
+
union
{
+#ifdef MA_SUPPORT_WASAPI
struct
{
- ma_vfs* pVFS;
- ma_vfs_file file;
- } vfs;
+ ma_thread commandThread;
+ ma_mutex commandLock;
+ ma_semaphore commandSem;
+ ma_uint32 commandIndex;
+ ma_uint32 commandCount;
+ ma_context_command__wasapi commands[4];
+ } wasapi;
+#endif
+#ifdef MA_SUPPORT_DSOUND
struct
{
- const ma_uint8* pData;
- size_t dataSize;
- size_t currentReadPos;
- } memory; /* Only used for decoders that were opened against a block of memory. */
- } data;
-};
-
-MA_API ma_decoder_config ma_decoder_config_init(ma_format outputFormat, ma_uint32 outputChannels, ma_uint32 outputSampleRate);
-MA_API ma_decoder_config ma_decoder_config_init_default(void);
-
-MA_API ma_result ma_decoder_init(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_memory(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_vfs(ma_vfs* pVFS, const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_vfs_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_file(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_file_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-
-/*
-Uninitializes a decoder.
-*/
-MA_API ma_result ma_decoder_uninit(ma_decoder* pDecoder);
-
-/*
-Retrieves the current position of the read cursor in PCM frames.
-*/
-MA_API ma_result ma_decoder_get_cursor_in_pcm_frames(ma_decoder* pDecoder, ma_uint64* pCursor);
-
-/*
-Retrieves the length of the decoder in PCM frames.
-
-Do not call this on streams of an undefined length, such as internet radio.
-
-If the length is unknown or an error occurs, 0 will be returned.
-
-This will always return 0 for Vorbis decoders. This is due to a limitation with stb_vorbis in push mode which is what miniaudio
-uses internally.
-
-For MP3's, this will decode the entire file. Do not call this in time critical scenarios.
-
-This function is not thread safe without your own synchronization.
-*/
-MA_API ma_uint64 ma_decoder_get_length_in_pcm_frames(ma_decoder* pDecoder);
+ ma_handle hDSoundDLL;
+ ma_proc DirectSoundCreate;
+ ma_proc DirectSoundEnumerateA;
+ ma_proc DirectSoundCaptureCreate;
+ ma_proc DirectSoundCaptureEnumerateA;
+ } dsound;
+#endif
+#ifdef MA_SUPPORT_WINMM
+ struct
+ {
+ ma_handle hWinMM;
+ ma_proc waveOutGetNumDevs;
+ ma_proc waveOutGetDevCapsA;
+ ma_proc waveOutOpen;
+ ma_proc waveOutClose;
+ ma_proc waveOutPrepareHeader;
+ ma_proc waveOutUnprepareHeader;
+ ma_proc waveOutWrite;
+ ma_proc waveOutReset;
+ ma_proc waveInGetNumDevs;
+ ma_proc waveInGetDevCapsA;
+ ma_proc waveInOpen;
+ ma_proc waveInClose;
+ ma_proc waveInPrepareHeader;
+ ma_proc waveInUnprepareHeader;
+ ma_proc waveInAddBuffer;
+ ma_proc waveInStart;
+ ma_proc waveInReset;
+ } winmm;
+#endif
+#ifdef MA_SUPPORT_ALSA
+ struct
+ {
+ ma_handle asoundSO;
+ ma_proc snd_pcm_open;
+ ma_proc snd_pcm_close;
+ ma_proc snd_pcm_hw_params_sizeof;
+ ma_proc snd_pcm_hw_params_any;
+ ma_proc snd_pcm_hw_params_set_format;
+ ma_proc snd_pcm_hw_params_set_format_first;
+ ma_proc snd_pcm_hw_params_get_format_mask;
+ ma_proc snd_pcm_hw_params_set_channels;
+ ma_proc snd_pcm_hw_params_set_channels_near;
+ ma_proc snd_pcm_hw_params_set_channels_minmax;
+ ma_proc snd_pcm_hw_params_set_rate_resample;
+ ma_proc snd_pcm_hw_params_set_rate;
+ ma_proc snd_pcm_hw_params_set_rate_near;
+ ma_proc snd_pcm_hw_params_set_buffer_size_near;
+ ma_proc snd_pcm_hw_params_set_periods_near;
+ ma_proc snd_pcm_hw_params_set_access;
+ ma_proc snd_pcm_hw_params_get_format;
+ ma_proc snd_pcm_hw_params_get_channels;
+ ma_proc snd_pcm_hw_params_get_channels_min;
+ ma_proc snd_pcm_hw_params_get_channels_max;
+ ma_proc snd_pcm_hw_params_get_rate;
+ ma_proc snd_pcm_hw_params_get_rate_min;
+ ma_proc snd_pcm_hw_params_get_rate_max;
+ ma_proc snd_pcm_hw_params_get_buffer_size;
+ ma_proc snd_pcm_hw_params_get_periods;
+ ma_proc snd_pcm_hw_params_get_access;
+ ma_proc snd_pcm_hw_params_test_format;
+ ma_proc snd_pcm_hw_params_test_channels;
+ ma_proc snd_pcm_hw_params_test_rate;
+ ma_proc snd_pcm_hw_params;
+ ma_proc snd_pcm_sw_params_sizeof;
+ ma_proc snd_pcm_sw_params_current;
+ ma_proc snd_pcm_sw_params_get_boundary;
+ ma_proc snd_pcm_sw_params_set_avail_min;
+ ma_proc snd_pcm_sw_params_set_start_threshold;
+ ma_proc snd_pcm_sw_params_set_stop_threshold;
+ ma_proc snd_pcm_sw_params;
+ ma_proc snd_pcm_format_mask_sizeof;
+ ma_proc snd_pcm_format_mask_test;
+ ma_proc snd_pcm_get_chmap;
+ ma_proc snd_pcm_state;
+ ma_proc snd_pcm_prepare;
+ ma_proc snd_pcm_start;
+ ma_proc snd_pcm_drop;
+ ma_proc snd_pcm_drain;
+ ma_proc snd_pcm_reset;
+ ma_proc snd_device_name_hint;
+ ma_proc snd_device_name_get_hint;
+ ma_proc snd_card_get_index;
+ ma_proc snd_device_name_free_hint;
+ ma_proc snd_pcm_mmap_begin;
+ ma_proc snd_pcm_mmap_commit;
+ ma_proc snd_pcm_recover;
+ ma_proc snd_pcm_readi;
+ ma_proc snd_pcm_writei;
+ ma_proc snd_pcm_avail;
+ ma_proc snd_pcm_avail_update;
+ ma_proc snd_pcm_wait;
+ ma_proc snd_pcm_nonblock;
+ ma_proc snd_pcm_info;
+ ma_proc snd_pcm_info_sizeof;
+ ma_proc snd_pcm_info_get_name;
+ ma_proc snd_pcm_poll_descriptors;
+ ma_proc snd_pcm_poll_descriptors_count;
+ ma_proc snd_pcm_poll_descriptors_revents;
+ ma_proc snd_config_update_free_global;
-/*
-Reads PCM frames from the given decoder.
-
-This is not thread safe without your own synchronization.
-*/
-MA_API ma_uint64 ma_decoder_read_pcm_frames(ma_decoder* pDecoder, void* pFramesOut, ma_uint64 frameCount);
-
-/*
-Seeks to a PCM frame based on it's absolute index.
-
-This is not thread safe without your own synchronization.
-*/
-MA_API ma_result ma_decoder_seek_to_pcm_frame(ma_decoder* pDecoder, ma_uint64 frameIndex);
-
-/*
-Retrieves the number of frames that can be read before reaching the end.
-
-This calls `ma_decoder_get_length_in_pcm_frames()` so you need to be aware of the rules for that function, in
-particular ensuring you do not call it on streams of an undefined length, such as internet radio.
-
-If the total length of the decoder cannot be retrieved, such as with Vorbis decoders, `MA_NOT_IMPLEMENTED` will be
-returned.
-*/
-MA_API ma_result ma_decoder_get_available_frames(ma_decoder* pDecoder, ma_uint64* pAvailableFrames);
-
-/*
-Helper for opening and decoding a file into a heap allocated block of memory. Free the returned pointer with ma_free(). On input,
-pConfig should be set to what you want. On output it will be set to what you got.
-*/
-MA_API ma_result ma_decode_from_vfs(ma_vfs* pVFS, const char* pFilePath, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut);
-MA_API ma_result ma_decode_file(const char* pFilePath, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut);
-MA_API ma_result ma_decode_memory(const void* pData, size_t dataSize, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut);
-
-
-
-
-/*
-DEPRECATED
-
-Set the "encodingFormat" variable in the decoder config instead:
-
- decoderConfig.encodingFormat = ma_encoding_format_wav;
-
-These functions will be removed in version 0.11.
-*/
-MA_API ma_result ma_decoder_init_wav(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_flac(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_mp3(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_vorbis(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_memory_wav(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_memory_flac(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_memory_mp3(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_memory_vorbis(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_vfs_wav(ma_vfs* pVFS, const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_vfs_flac(ma_vfs* pVFS, const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_vfs_mp3(ma_vfs* pVFS, const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_vfs_vorbis(ma_vfs* pVFS, const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_vfs_wav_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_vfs_flac_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_vfs_mp3_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_vfs_vorbis_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_file_wav(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_file_flac(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_file_mp3(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_file_vorbis(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_file_wav_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_file_flac_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_file_mp3_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-MA_API ma_result ma_decoder_init_file_vorbis_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
-
-#endif /* MA_NO_DECODING */
-
-
-/************************************************************************************************************************************************************
-
-Encoding
-========
-
-Encoders do not perform any format conversion for you. If your target format does not support the format, and error will be returned.
-
-************************************************************************************************************************************************************/
-#ifndef MA_NO_ENCODING
-typedef struct ma_encoder ma_encoder;
-
-typedef size_t (* ma_encoder_write_proc) (ma_encoder* pEncoder, const void* pBufferIn, size_t bytesToWrite); /* Returns the number of bytes written. */
-typedef ma_bool32 (* ma_encoder_seek_proc) (ma_encoder* pEncoder, int byteOffset, ma_seek_origin origin);
-typedef ma_result (* ma_encoder_init_proc) (ma_encoder* pEncoder);
-typedef void (* ma_encoder_uninit_proc) (ma_encoder* pEncoder);
-typedef ma_uint64 (* ma_encoder_write_pcm_frames_proc)(ma_encoder* pEncoder, const void* pFramesIn, ma_uint64 frameCount);
-
-typedef struct
-{
- ma_resource_format resourceFormat;
- ma_format format;
- ma_uint32 channels;
- ma_uint32 sampleRate;
- ma_allocation_callbacks allocationCallbacks;
-} ma_encoder_config;
-
-MA_API ma_encoder_config ma_encoder_config_init(ma_resource_format resourceFormat, ma_format format, ma_uint32 channels, ma_uint32 sampleRate);
-
-struct ma_encoder
-{
- ma_encoder_config config;
- ma_encoder_write_proc onWrite;
- ma_encoder_seek_proc onSeek;
- ma_encoder_init_proc onInit;
- ma_encoder_uninit_proc onUninit;
- ma_encoder_write_pcm_frames_proc onWritePCMFrames;
- void* pUserData;
- void* pInternalEncoder; /* <-- The drwav/drflac/stb_vorbis/etc. objects. */
- void* pFile; /* FILE*. Only used when initialized with ma_encoder_init_file(). */
-};
-
-MA_API ma_result ma_encoder_init(ma_encoder_write_proc onWrite, ma_encoder_seek_proc onSeek, void* pUserData, const ma_encoder_config* pConfig, ma_encoder* pEncoder);
-MA_API ma_result ma_encoder_init_file(const char* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder);
-MA_API ma_result ma_encoder_init_file_w(const wchar_t* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder);
-MA_API void ma_encoder_uninit(ma_encoder* pEncoder);
-MA_API ma_uint64 ma_encoder_write_pcm_frames(ma_encoder* pEncoder, const void* pFramesIn, ma_uint64 frameCount);
-
-#endif /* MA_NO_ENCODING */
-
-
-/************************************************************************************************************************************************************
-
-Generation
-
-************************************************************************************************************************************************************/
-#ifndef MA_NO_GENERATION
-typedef enum
-{
- ma_waveform_type_sine,
- ma_waveform_type_square,
- ma_waveform_type_triangle,
- ma_waveform_type_sawtooth
-} ma_waveform_type;
-
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- ma_uint32 sampleRate;
- ma_waveform_type type;
- double amplitude;
- double frequency;
-} ma_waveform_config;
-
-MA_API ma_waveform_config ma_waveform_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_waveform_type type, double amplitude, double frequency);
-
-typedef struct
-{
- ma_data_source_base ds;
- ma_waveform_config config;
- double advance;
- double time;
-} ma_waveform;
-
-MA_API ma_result ma_waveform_init(const ma_waveform_config* pConfig, ma_waveform* pWaveform);
-MA_API void ma_waveform_uninit(ma_waveform* pWaveform);
-MA_API ma_uint64 ma_waveform_read_pcm_frames(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount);
-MA_API ma_result ma_waveform_seek_to_pcm_frame(ma_waveform* pWaveform, ma_uint64 frameIndex);
-MA_API ma_result ma_waveform_set_amplitude(ma_waveform* pWaveform, double amplitude);
-MA_API ma_result ma_waveform_set_frequency(ma_waveform* pWaveform, double frequency);
-MA_API ma_result ma_waveform_set_type(ma_waveform* pWaveform, ma_waveform_type type);
-MA_API ma_result ma_waveform_set_sample_rate(ma_waveform* pWaveform, ma_uint32 sampleRate);
-
-typedef enum
-{
- ma_noise_type_white,
- ma_noise_type_pink,
- ma_noise_type_brownian
-} ma_noise_type;
-
-typedef struct
-{
- ma_format format;
- ma_uint32 channels;
- ma_noise_type type;
- ma_int32 seed;
- double amplitude;
- ma_bool32 duplicateChannels;
-} ma_noise_config;
-
-MA_API ma_noise_config ma_noise_config_init(ma_format format, ma_uint32 channels, ma_noise_type type, ma_int32 seed, double amplitude);
-
-typedef struct
-{
- ma_data_source_vtable ds;
- ma_noise_config config;
- ma_lcg lcg;
- union
- {
+ ma_mutex internalDeviceEnumLock;
+ ma_bool32 useVerboseDeviceEnumeration;
+ } alsa;
+#endif
+#ifdef MA_SUPPORT_PULSEAUDIO
struct
{
- double bin[MA_MAX_CHANNELS][16];
- double accumulation[MA_MAX_CHANNELS];
- ma_uint32 counter[MA_MAX_CHANNELS];
- } pink;
+ ma_handle pulseSO;
+ ma_proc pa_mainloop_new;
+ ma_proc pa_mainloop_free;
+ ma_proc pa_mainloop_quit;
+ ma_proc pa_mainloop_get_api;
+ ma_proc pa_mainloop_iterate;
+ ma_proc pa_mainloop_wakeup;
+ ma_proc pa_threaded_mainloop_new;
+ ma_proc pa_threaded_mainloop_free;
+ ma_proc pa_threaded_mainloop_start;
+ ma_proc pa_threaded_mainloop_stop;
+ ma_proc pa_threaded_mainloop_lock;
+ ma_proc pa_threaded_mainloop_unlock;
+ ma_proc pa_threaded_mainloop_wait;
+ ma_proc pa_threaded_mainloop_signal;
+ ma_proc pa_threaded_mainloop_accept;
+ ma_proc pa_threaded_mainloop_get_retval;
+ ma_proc pa_threaded_mainloop_get_api;
+ ma_proc pa_threaded_mainloop_in_thread;
+ ma_proc pa_threaded_mainloop_set_name;
+ ma_proc pa_context_new;
+ ma_proc pa_context_unref;
+ ma_proc pa_context_connect;
+ ma_proc pa_context_disconnect;
+ ma_proc pa_context_set_state_callback;
+ ma_proc pa_context_get_state;
+ ma_proc pa_context_get_sink_info_list;
+ ma_proc pa_context_get_source_info_list;
+ ma_proc pa_context_get_sink_info_by_name;
+ ma_proc pa_context_get_source_info_by_name;
+ ma_proc pa_operation_unref;
+ ma_proc pa_operation_get_state;
+ ma_proc pa_channel_map_init_extend;
+ ma_proc pa_channel_map_valid;
+ ma_proc pa_channel_map_compatible;
+ ma_proc pa_stream_new;
+ ma_proc pa_stream_unref;
+ ma_proc pa_stream_connect_playback;
+ ma_proc pa_stream_connect_record;
+ ma_proc pa_stream_disconnect;
+ ma_proc pa_stream_get_state;
+ ma_proc pa_stream_get_sample_spec;
+ ma_proc pa_stream_get_channel_map;
+ ma_proc pa_stream_get_buffer_attr;
+ ma_proc pa_stream_set_buffer_attr;
+ ma_proc pa_stream_get_device_name;
+ ma_proc pa_stream_set_write_callback;
+ ma_proc pa_stream_set_read_callback;
+ ma_proc pa_stream_set_suspended_callback;
+ ma_proc pa_stream_set_moved_callback;
+ ma_proc pa_stream_is_suspended;
+ ma_proc pa_stream_flush;
+ ma_proc pa_stream_drain;
+ ma_proc pa_stream_is_corked;
+ ma_proc pa_stream_cork;
+ ma_proc pa_stream_trigger;
+ ma_proc pa_stream_begin_write;
+ ma_proc pa_stream_write;
+ ma_proc pa_stream_peek;
+ ma_proc pa_stream_drop;
+ ma_proc pa_stream_writable_size;
+ ma_proc pa_stream_readable_size;
+
+ /*pa_mainloop**/ ma_ptr pMainLoop;
+ /*pa_context**/ ma_ptr pPulseContext;
+ char* pApplicationName; /* Set when the context is initialized. Used by devices for their local pa_context objects. */
+ char* pServerName; /* Set when the context is initialized. Used by devices for their local pa_context objects. */
+ } pulse;
+#endif
+#ifdef MA_SUPPORT_JACK
struct
{
- double accumulation[MA_MAX_CHANNELS];
- } brownian;
- } state;
-} ma_noise;
-
-MA_API ma_result ma_noise_init(const ma_noise_config* pConfig, ma_noise* pNoise);
-MA_API void ma_noise_uninit(ma_noise* pNoise);
-MA_API ma_uint64 ma_noise_read_pcm_frames(ma_noise* pNoise, void* pFramesOut, ma_uint64 frameCount);
-MA_API ma_result ma_noise_set_amplitude(ma_noise* pNoise, double amplitude);
-MA_API ma_result ma_noise_set_seed(ma_noise* pNoise, ma_int32 seed);
-MA_API ma_result ma_noise_set_type(ma_noise* pNoise, ma_noise_type type);
-
-#endif /* MA_NO_GENERATION */
+ ma_handle jackSO;
+ ma_proc jack_client_open;
+ ma_proc jack_client_close;
+ ma_proc jack_client_name_size;
+ ma_proc jack_set_process_callback;
+ ma_proc jack_set_buffer_size_callback;
+ ma_proc jack_on_shutdown;
+ ma_proc jack_get_sample_rate;
+ ma_proc jack_get_buffer_size;
+ ma_proc jack_get_ports;
+ ma_proc jack_activate;
+ ma_proc jack_deactivate;
+ ma_proc jack_connect;
+ ma_proc jack_port_register;
+ ma_proc jack_port_name;
+ ma_proc jack_port_get_buffer;
+ ma_proc jack_free;
-#ifdef __cplusplus
-}
+ char* pClientName;
+ ma_bool32 tryStartServer;
+ } jack;
#endif
-#endif /* miniaudio_h */
-
-
-
-/************************************************************************************************************************************************************
-*************************************************************************************************************************************************************
-
-IMPLEMENTATION
-
-*************************************************************************************************************************************************************
-************************************************************************************************************************************************************/
-#if defined(MINIAUDIO_IMPLEMENTATION) || defined(MA_IMPLEMENTATION)
-#ifndef miniaudio_c
-#define miniaudio_c
+#ifdef MA_SUPPORT_COREAUDIO
+ struct
+ {
+ ma_handle hCoreFoundation;
+ ma_proc CFStringGetCString;
+ ma_proc CFRelease;
-#include
-#include /* For INT_MAX */
-#include /* sin(), etc. */
+ ma_handle hCoreAudio;
+ ma_proc AudioObjectGetPropertyData;
+ ma_proc AudioObjectGetPropertyDataSize;
+ ma_proc AudioObjectSetPropertyData;
+ ma_proc AudioObjectAddPropertyListener;
+ ma_proc AudioObjectRemovePropertyListener;
-#include
-#include
-#if !defined(_MSC_VER) && !defined(__DMC__)
- #include /* For strcasecmp(). */
- #include /* For wcslen(), wcsrtombs() */
-#endif
+ ma_handle hAudioUnit; /* Could possibly be set to AudioToolbox on later versions of macOS. */
+ ma_proc AudioComponentFindNext;
+ ma_proc AudioComponentInstanceDispose;
+ ma_proc AudioComponentInstanceNew;
+ ma_proc AudioOutputUnitStart;
+ ma_proc AudioOutputUnitStop;
+ ma_proc AudioUnitAddPropertyListener;
+ ma_proc AudioUnitGetPropertyInfo;
+ ma_proc AudioUnitGetProperty;
+ ma_proc AudioUnitSetProperty;
+ ma_proc AudioUnitInitialize;
+ ma_proc AudioUnitRender;
-#ifdef MA_WIN32
-#include
-#else
-#include /* For malloc(), free(), wcstombs(). */
-#include /* For memset() */
-#include
-#include /* select() (used for ma_sleep()). */
+ /*AudioComponent*/ ma_ptr component;
+ ma_bool32 noAudioSessionDeactivate; /* For tracking whether or not the iOS audio session should be explicitly deactivated. Set from the config in ma_context_init__coreaudio(). */
+ } coreaudio;
#endif
-
-#include /* For fstat(), etc. */
-
-#ifdef MA_EMSCRIPTEN
-#include
+#ifdef MA_SUPPORT_SNDIO
+ struct
+ {
+ ma_handle sndioSO;
+ ma_proc sio_open;
+ ma_proc sio_close;
+ ma_proc sio_setpar;
+ ma_proc sio_getpar;
+ ma_proc sio_getcap;
+ ma_proc sio_start;
+ ma_proc sio_stop;
+ ma_proc sio_read;
+ ma_proc sio_write;
+ ma_proc sio_onmove;
+ ma_proc sio_nfds;
+ ma_proc sio_pollfd;
+ ma_proc sio_revents;
+ ma_proc sio_eof;
+ ma_proc sio_setvol;
+ ma_proc sio_onvol;
+ ma_proc sio_initpar;
+ } sndio;
#endif
-
-#if !defined(MA_64BIT) && !defined(MA_32BIT)
-#ifdef _WIN32
-#ifdef _WIN64
-#define MA_64BIT
-#else
-#define MA_32BIT
+#ifdef MA_SUPPORT_AUDIO4
+ struct
+ {
+ int _unused;
+ } audio4;
#endif
+#ifdef MA_SUPPORT_OSS
+ struct
+ {
+ int versionMajor;
+ int versionMinor;
+ } oss;
#endif
+#ifdef MA_SUPPORT_AAUDIO
+ struct
+ {
+ ma_handle hAAudio; /* libaaudio.so */
+ ma_proc AAudio_createStreamBuilder;
+ ma_proc AAudioStreamBuilder_delete;
+ ma_proc AAudioStreamBuilder_setDeviceId;
+ ma_proc AAudioStreamBuilder_setDirection;
+ ma_proc AAudioStreamBuilder_setSharingMode;
+ ma_proc AAudioStreamBuilder_setFormat;
+ ma_proc AAudioStreamBuilder_setChannelCount;
+ ma_proc AAudioStreamBuilder_setSampleRate;
+ ma_proc AAudioStreamBuilder_setBufferCapacityInFrames;
+ ma_proc AAudioStreamBuilder_setFramesPerDataCallback;
+ ma_proc AAudioStreamBuilder_setDataCallback;
+ ma_proc AAudioStreamBuilder_setErrorCallback;
+ ma_proc AAudioStreamBuilder_setPerformanceMode;
+ ma_proc AAudioStreamBuilder_setUsage;
+ ma_proc AAudioStreamBuilder_setContentType;
+ ma_proc AAudioStreamBuilder_setInputPreset;
+ ma_proc AAudioStreamBuilder_openStream;
+ ma_proc AAudioStream_close;
+ ma_proc AAudioStream_getState;
+ ma_proc AAudioStream_waitForStateChange;
+ ma_proc AAudioStream_getFormat;
+ ma_proc AAudioStream_getChannelCount;
+ ma_proc AAudioStream_getSampleRate;
+ ma_proc AAudioStream_getBufferCapacityInFrames;
+ ma_proc AAudioStream_getFramesPerDataCallback;
+ ma_proc AAudioStream_getFramesPerBurst;
+ ma_proc AAudioStream_requestStart;
+ ma_proc AAudioStream_requestStop;
+ ma_device_job_thread jobThread; /* For processing operations outside of the error callback, specifically device disconnections and rerouting. */
+ } aaudio;
#endif
-
-#if !defined(MA_64BIT) && !defined(MA_32BIT)
-#ifdef __GNUC__
-#ifdef __LP64__
-#define MA_64BIT
-#else
-#define MA_32BIT
+#ifdef MA_SUPPORT_OPENSL
+ struct
+ {
+ ma_handle libOpenSLES;
+ ma_handle SL_IID_ENGINE;
+ ma_handle SL_IID_AUDIOIODEVICECAPABILITIES;
+ ma_handle SL_IID_ANDROIDSIMPLEBUFFERQUEUE;
+ ma_handle SL_IID_RECORD;
+ ma_handle SL_IID_PLAY;
+ ma_handle SL_IID_OUTPUTMIX;
+ ma_handle SL_IID_ANDROIDCONFIGURATION;
+ ma_proc slCreateEngine;
+ } opensl;
#endif
+#ifdef MA_SUPPORT_WEBAUDIO
+ struct
+ {
+ int _unused;
+ } webaudio;
#endif
+#ifdef MA_SUPPORT_NULL
+ struct
+ {
+ int _unused;
+ } null_backend;
#endif
+ };
-#if !defined(MA_64BIT) && !defined(MA_32BIT)
-#include
-#if INTPTR_MAX == INT64_MAX
-#define MA_64BIT
-#else
-#define MA_32BIT
-#endif
-#endif
+ union
+ {
+#ifdef MA_WIN32
+ struct
+ {
+ /*HMODULE*/ ma_handle hOle32DLL;
+ ma_proc CoInitializeEx;
+ ma_proc CoUninitialize;
+ ma_proc CoCreateInstance;
+ ma_proc CoTaskMemFree;
+ ma_proc PropVariantClear;
+ ma_proc StringFromGUID2;
-/* Architecture Detection */
-#if defined(__x86_64__) || defined(_M_X64)
-#define MA_X64
-#elif defined(__i386) || defined(_M_IX86)
-#define MA_X86
-#elif defined(__arm__) || defined(_M_ARM)
-#define MA_ARM
-#endif
+ /*HMODULE*/ ma_handle hUser32DLL;
+ ma_proc GetForegroundWindow;
+ ma_proc GetDesktopWindow;
-/* Cannot currently support AVX-512 if AVX is disabled. */
-#if !defined(MA_NO_AVX512) && defined(MA_NO_AVX2)
-#define MA_NO_AVX512
+ /*HMODULE*/ ma_handle hAdvapi32DLL;
+ ma_proc RegOpenKeyExA;
+ ma_proc RegCloseKey;
+ ma_proc RegQueryValueExA;
+ } win32;
+#endif
+#ifdef MA_POSIX
+ struct
+ {
+ ma_handle pthreadSO;
+ ma_proc pthread_create;
+ ma_proc pthread_join;
+ ma_proc pthread_mutex_init;
+ ma_proc pthread_mutex_destroy;
+ ma_proc pthread_mutex_lock;
+ ma_proc pthread_mutex_unlock;
+ ma_proc pthread_cond_init;
+ ma_proc pthread_cond_destroy;
+ ma_proc pthread_cond_wait;
+ ma_proc pthread_cond_signal;
+ ma_proc pthread_attr_init;
+ ma_proc pthread_attr_destroy;
+ ma_proc pthread_attr_setschedpolicy;
+ ma_proc pthread_attr_getschedparam;
+ ma_proc pthread_attr_setschedparam;
+ } posix;
#endif
+ int _unused;
+ };
+};
-/* Intrinsics Support */
-#if defined(MA_X64) || defined(MA_X86)
- #if defined(_MSC_VER) && !defined(__clang__)
- /* MSVC. */
- #if _MSC_VER >= 1400 && !defined(MA_NO_SSE2) /* 2005 */
- #define MA_SUPPORT_SSE2
- #endif
- /*#if _MSC_VER >= 1600 && !defined(MA_NO_AVX)*/ /* 2010 */
- /* #define MA_SUPPORT_AVX*/
- /*#endif*/
- #if _MSC_VER >= 1700 && !defined(MA_NO_AVX2) /* 2012 */
- #define MA_SUPPORT_AVX2
- #endif
- #if _MSC_VER >= 1910 && !defined(MA_NO_AVX512) /* 2017 */
- #define MA_SUPPORT_AVX512
- #endif
- #else
- /* Assume GNUC-style. */
- #if defined(__SSE2__) && !defined(MA_NO_SSE2)
- #define MA_SUPPORT_SSE2
- #endif
- /*#if defined(__AVX__) && !defined(MA_NO_AVX)*/
- /* #define MA_SUPPORT_AVX*/
- /*#endif*/
- #if defined(__AVX2__) && !defined(MA_NO_AVX2)
- #define MA_SUPPORT_AVX2
- #endif
- #if defined(__AVX512F__) && !defined(MA_NO_AVX512)
- #define MA_SUPPORT_AVX512
- #endif
- #endif
-
- /* If at this point we still haven't determined compiler support for the intrinsics just fall back to __has_include. */
- #if !defined(__GNUC__) && !defined(__clang__) && defined(__has_include)
- #if !defined(MA_SUPPORT_SSE2) && !defined(MA_NO_SSE2) && __has_include()
- #define MA_SUPPORT_SSE2
- #endif
- /*#if !defined(MA_SUPPORT_AVX) && !defined(MA_NO_AVX) && __has_include()*/
- /* #define MA_SUPPORT_AVX*/
- /*#endif*/
- #if !defined(MA_SUPPORT_AVX2) && !defined(MA_NO_AVX2) && __has_include()
- #define MA_SUPPORT_AVX2
- #endif
- #if !defined(MA_SUPPORT_AVX512) && !defined(MA_NO_AVX512) && __has_include()
- #define MA_SUPPORT_AVX512
- #endif
- #endif
+struct ma_device
+{
+ ma_context* pContext;
+ ma_device_type type;
+ ma_uint32 sampleRate;
+ MA_ATOMIC(4, ma_device_state) state; /* The state of the device is variable and can change at any time on any thread. Must be used atomically. */
+ ma_device_data_proc onData; /* Set once at initialization time and should not be changed after. */
+ ma_device_notification_proc onNotification; /* Set once at initialization time and should not be changed after. */
+ ma_stop_proc onStop; /* DEPRECATED. Use the notification callback instead. Set once at initialization time and should not be changed after. */
+ void* pUserData; /* Application defined data. */
+ ma_mutex startStopLock;
+ ma_event wakeupEvent;
+ ma_event startEvent;
+ ma_event stopEvent;
+ ma_thread thread;
+ ma_result workResult; /* This is set by the worker thread after it's finished doing a job. */
+ ma_bool8 isOwnerOfContext; /* When set to true, uninitializing the device will also uninitialize the context. Set to true when NULL is passed into ma_device_init(). */
+ ma_bool8 noPreSilencedOutputBuffer;
+ ma_bool8 noClip;
+ ma_bool8 noDisableDenormals;
+ ma_bool8 noFixedSizedCallback;
+ MA_ATOMIC(4, float) masterVolumeFactor; /* Linear 0..1. Can be read and written simultaneously by different threads. Must be used atomically. */
+ ma_duplex_rb duplexRB; /* Intermediary buffer for duplex device on asynchronous backends. */
+ struct
+ {
+ ma_resample_algorithm algorithm;
+ ma_resampling_backend_vtable* pBackendVTable;
+ void* pBackendUserData;
+ struct
+ {
+ ma_uint32 lpfOrder;
+ } linear;
+ } resampling;
+ struct
+ {
+ ma_device_id* pID; /* Set to NULL if using default ID, otherwise set to the address of "id". */
+ ma_device_id id; /* If using an explicit device, will be set to a copy of the ID used for initialization. Otherwise cleared to 0. */
+ char name[MA_MAX_DEVICE_NAME_LENGTH + 1]; /* Maybe temporary. Likely to be replaced with a query API. */
+ ma_share_mode shareMode; /* Set to whatever was passed in when the device was initialized. */
+ ma_format format;
+ ma_uint32 channels;
+ ma_channel channelMap[MA_MAX_CHANNELS];
+ ma_format internalFormat;
+ ma_uint32 internalChannels;
+ ma_uint32 internalSampleRate;
+ ma_channel internalChannelMap[MA_MAX_CHANNELS];
+ ma_uint32 internalPeriodSizeInFrames;
+ ma_uint32 internalPeriods;
+ ma_channel_mix_mode channelMixMode;
+ ma_data_converter converter;
+ void* pIntermediaryBuffer; /* For implementing fixed sized buffer callbacks. Will be null if using variable sized callbacks. */
+ ma_uint32 intermediaryBufferCap;
+ ma_uint32 intermediaryBufferLen; /* How many valid frames are sitting in the intermediary buffer. */
+ void* pInputCache; /* In external format. Can be null. */
+ ma_uint64 inputCacheCap;
+ ma_uint64 inputCacheConsumed;
+ ma_uint64 inputCacheRemaining;
+ } playback;
+ struct
+ {
+ ma_device_id* pID; /* Set to NULL if using default ID, otherwise set to the address of "id". */
+ ma_device_id id; /* If using an explicit device, will be set to a copy of the ID used for initialization. Otherwise cleared to 0. */
+ char name[MA_MAX_DEVICE_NAME_LENGTH + 1]; /* Maybe temporary. Likely to be replaced with a query API. */
+ ma_share_mode shareMode; /* Set to whatever was passed in when the device was initialized. */
+ ma_format format;
+ ma_uint32 channels;
+ ma_channel channelMap[MA_MAX_CHANNELS];
+ ma_format internalFormat;
+ ma_uint32 internalChannels;
+ ma_uint32 internalSampleRate;
+ ma_channel internalChannelMap[MA_MAX_CHANNELS];
+ ma_uint32 internalPeriodSizeInFrames;
+ ma_uint32 internalPeriods;
+ ma_channel_mix_mode channelMixMode;
+ ma_data_converter converter;
+ void* pIntermediaryBuffer; /* For implementing fixed sized buffer callbacks. Will be null if using variable sized callbacks. */
+ ma_uint32 intermediaryBufferCap;
+ ma_uint32 intermediaryBufferLen; /* How many valid frames are sitting in the intermediary buffer. */
+ } capture;
- #if defined(MA_SUPPORT_AVX512)
- #include /* Not a mistake. Intentionally including instead of because otherwise the compiler will complain. */
- #elif defined(MA_SUPPORT_AVX2) || defined(MA_SUPPORT_AVX)
- #include
- #elif defined(MA_SUPPORT_SSE2)
- #include
- #endif
+ union
+ {
+#ifdef MA_SUPPORT_WASAPI
+ struct
+ {
+ /*IAudioClient**/ ma_ptr pAudioClientPlayback;
+ /*IAudioClient**/ ma_ptr pAudioClientCapture;
+ /*IAudioRenderClient**/ ma_ptr pRenderClient;
+ /*IAudioCaptureClient**/ ma_ptr pCaptureClient;
+ /*IMMDeviceEnumerator**/ ma_ptr pDeviceEnumerator; /* Used for IMMNotificationClient notifications. Required for detecting default device changes. */
+ ma_IMMNotificationClient notificationClient;
+ /*HANDLE*/ ma_handle hEventPlayback; /* Auto reset. Initialized to signaled. */
+ /*HANDLE*/ ma_handle hEventCapture; /* Auto reset. Initialized to unsignaled. */
+ ma_uint32 actualBufferSizeInFramesPlayback; /* Value from GetBufferSize(). internalPeriodSizeInFrames is not set to the _actual_ buffer size when low-latency shared mode is being used due to the way the IAudioClient3 API works. */
+ ma_uint32 actualBufferSizeInFramesCapture;
+ ma_uint32 originalPeriodSizeInFrames;
+ ma_uint32 originalPeriodSizeInMilliseconds;
+ ma_uint32 originalPeriods;
+ ma_performance_profile originalPerformanceProfile;
+ ma_uint32 periodSizeInFramesPlayback;
+ ma_uint32 periodSizeInFramesCapture;
+ void* pMappedBufferCapture;
+ ma_uint32 mappedBufferCaptureCap;
+ ma_uint32 mappedBufferCaptureLen;
+ void* pMappedBufferPlayback;
+ ma_uint32 mappedBufferPlaybackCap;
+ ma_uint32 mappedBufferPlaybackLen;
+ MA_ATOMIC(4, ma_bool32) isStartedCapture; /* Can be read and written simultaneously across different threads. Must be used atomically, and must be 32-bit. */
+ MA_ATOMIC(4, ma_bool32) isStartedPlayback; /* Can be read and written simultaneously across different threads. Must be used atomically, and must be 32-bit. */
+ ma_bool8 noAutoConvertSRC; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM. */
+ ma_bool8 noDefaultQualitySRC; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY. */
+ ma_bool8 noHardwareOffloading;
+ ma_bool8 allowCaptureAutoStreamRouting;
+ ma_bool8 allowPlaybackAutoStreamRouting;
+ ma_bool8 isDetachedPlayback;
+ ma_bool8 isDetachedCapture;
+ } wasapi;
+#endif
+#ifdef MA_SUPPORT_DSOUND
+ struct
+ {
+ /*LPDIRECTSOUND*/ ma_ptr pPlayback;
+ /*LPDIRECTSOUNDBUFFER*/ ma_ptr pPlaybackPrimaryBuffer;
+ /*LPDIRECTSOUNDBUFFER*/ ma_ptr pPlaybackBuffer;
+ /*LPDIRECTSOUNDCAPTURE*/ ma_ptr pCapture;
+ /*LPDIRECTSOUNDCAPTUREBUFFER*/ ma_ptr pCaptureBuffer;
+ } dsound;
+#endif
+#ifdef MA_SUPPORT_WINMM
+ struct
+ {
+ /*HWAVEOUT*/ ma_handle hDevicePlayback;
+ /*HWAVEIN*/ ma_handle hDeviceCapture;
+ /*HANDLE*/ ma_handle hEventPlayback;
+ /*HANDLE*/ ma_handle hEventCapture;
+ ma_uint32 fragmentSizeInFrames;
+ ma_uint32 iNextHeaderPlayback; /* [0,periods). Used as an index into pWAVEHDRPlayback. */
+ ma_uint32 iNextHeaderCapture; /* [0,periods). Used as an index into pWAVEHDRCapture. */
+ ma_uint32 headerFramesConsumedPlayback; /* The number of PCM frames consumed in the buffer in pWAVEHEADER[iNextHeader]. */
+ ma_uint32 headerFramesConsumedCapture; /* ^^^ */
+ /*WAVEHDR**/ ma_uint8* pWAVEHDRPlayback; /* One instantiation for each period. */
+ /*WAVEHDR**/ ma_uint8* pWAVEHDRCapture; /* One instantiation for each period. */
+ ma_uint8* pIntermediaryBufferPlayback;
+ ma_uint8* pIntermediaryBufferCapture;
+ ma_uint8* _pHeapData; /* Used internally and is used for the heap allocated data for the intermediary buffer and the WAVEHDR structures. */
+ } winmm;
+#endif
+#ifdef MA_SUPPORT_ALSA
+ struct
+ {
+ /*snd_pcm_t**/ ma_ptr pPCMPlayback;
+ /*snd_pcm_t**/ ma_ptr pPCMCapture;
+ /*struct pollfd**/ void* pPollDescriptorsPlayback;
+ /*struct pollfd**/ void* pPollDescriptorsCapture;
+ int pollDescriptorCountPlayback;
+ int pollDescriptorCountCapture;
+ int wakeupfdPlayback; /* eventfd for waking up from poll() when the playback device is stopped. */
+ int wakeupfdCapture; /* eventfd for waking up from poll() when the capture device is stopped. */
+ ma_bool8 isUsingMMapPlayback;
+ ma_bool8 isUsingMMapCapture;
+ } alsa;
+#endif
+#ifdef MA_SUPPORT_PULSEAUDIO
+ struct
+ {
+ /*pa_mainloop**/ ma_ptr pMainLoop;
+ /*pa_context**/ ma_ptr pPulseContext;
+ /*pa_stream**/ ma_ptr pStreamPlayback;
+ /*pa_stream**/ ma_ptr pStreamCapture;
+ } pulse;
+#endif
+#ifdef MA_SUPPORT_JACK
+ struct
+ {
+ /*jack_client_t**/ ma_ptr pClient;
+ /*jack_port_t**/ ma_ptr* ppPortsPlayback;
+ /*jack_port_t**/ ma_ptr* ppPortsCapture;
+ float* pIntermediaryBufferPlayback; /* Typed as a float because JACK is always floating point. */
+ float* pIntermediaryBufferCapture;
+ } jack;
+#endif
+#ifdef MA_SUPPORT_COREAUDIO
+ struct
+ {
+ ma_uint32 deviceObjectIDPlayback;
+ ma_uint32 deviceObjectIDCapture;
+ /*AudioUnit*/ ma_ptr audioUnitPlayback;
+ /*AudioUnit*/ ma_ptr audioUnitCapture;
+ /*AudioBufferList**/ ma_ptr pAudioBufferList; /* Only used for input devices. */
+ ma_uint32 audioBufferCapInFrames; /* Only used for input devices. The capacity in frames of each buffer in pAudioBufferList. */
+ ma_event stopEvent;
+ ma_uint32 originalPeriodSizeInFrames;
+ ma_uint32 originalPeriodSizeInMilliseconds;
+ ma_uint32 originalPeriods;
+ ma_performance_profile originalPerformanceProfile;
+ ma_bool32 isDefaultPlaybackDevice;
+ ma_bool32 isDefaultCaptureDevice;
+ ma_bool32 isSwitchingPlaybackDevice; /* <-- Set to true when the default device has changed and miniaudio is in the process of switching. */
+ ma_bool32 isSwitchingCaptureDevice; /* <-- Set to true when the default device has changed and miniaudio is in the process of switching. */
+ void* pNotificationHandler; /* Only used on mobile platforms. Obj-C object for handling route changes. */
+ } coreaudio;
+#endif
+#ifdef MA_SUPPORT_SNDIO
+ struct
+ {
+ ma_ptr handlePlayback;
+ ma_ptr handleCapture;
+ ma_bool32 isStartedPlayback;
+ ma_bool32 isStartedCapture;
+ } sndio;
+#endif
+#ifdef MA_SUPPORT_AUDIO4
+ struct
+ {
+ int fdPlayback;
+ int fdCapture;
+ } audio4;
+#endif
+#ifdef MA_SUPPORT_OSS
+ struct
+ {
+ int fdPlayback;
+ int fdCapture;
+ } oss;
+#endif
+#ifdef MA_SUPPORT_AAUDIO
+ struct
+ {
+ /*AAudioStream**/ ma_ptr pStreamPlayback;
+ /*AAudioStream**/ ma_ptr pStreamCapture;
+ ma_aaudio_usage usage;
+ ma_aaudio_content_type contentType;
+ ma_aaudio_input_preset inputPreset;
+ ma_bool32 noAutoStartAfterReroute;
+ } aaudio;
+#endif
+#ifdef MA_SUPPORT_OPENSL
+ struct
+ {
+ /*SLObjectItf*/ ma_ptr pOutputMixObj;
+ /*SLOutputMixItf*/ ma_ptr pOutputMix;
+ /*SLObjectItf*/ ma_ptr pAudioPlayerObj;
+ /*SLPlayItf*/ ma_ptr pAudioPlayer;
+ /*SLObjectItf*/ ma_ptr pAudioRecorderObj;
+ /*SLRecordItf*/ ma_ptr pAudioRecorder;
+ /*SLAndroidSimpleBufferQueueItf*/ ma_ptr pBufferQueuePlayback;
+ /*SLAndroidSimpleBufferQueueItf*/ ma_ptr pBufferQueueCapture;
+ ma_bool32 isDrainingCapture;
+ ma_bool32 isDrainingPlayback;
+ ma_uint32 currentBufferIndexPlayback;
+ ma_uint32 currentBufferIndexCapture;
+ ma_uint8* pBufferPlayback; /* This is malloc()'d and is used for storing audio data. Typed as ma_uint8 for easy offsetting. */
+ ma_uint8* pBufferCapture;
+ } opensl;
+#endif
+#ifdef MA_SUPPORT_WEBAUDIO
+ struct
+ {
+ int indexPlayback; /* We use a factory on the JavaScript side to manage devices and use an index for JS/C interop. */
+ int indexCapture;
+ } webaudio;
+#endif
+#ifdef MA_SUPPORT_NULL
+ struct
+ {
+ ma_thread deviceThread;
+ ma_event operationEvent;
+ ma_event operationCompletionEvent;
+ ma_semaphore operationSemaphore;
+ ma_uint32 operation;
+ ma_result operationResult;
+ ma_timer timer;
+ double priorRunTime;
+ ma_uint32 currentPeriodFramesRemainingPlayback;
+ ma_uint32 currentPeriodFramesRemainingCapture;
+ ma_uint64 lastProcessedFramePlayback;
+ ma_uint64 lastProcessedFrameCapture;
+ MA_ATOMIC(4, ma_bool32) isStarted; /* Read and written by multiple threads. Must be used atomically, and must be 32-bit for compiler compatibility. */
+ } null_device;
+#endif
+ };
+};
+#if defined(_MSC_VER) && !defined(__clang__)
+ #pragma warning(pop)
+#elif defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8)))
+ #pragma GCC diagnostic pop /* For ISO C99 doesn't support unnamed structs/unions [-Wpedantic] */
#endif
-#if defined(MA_ARM)
- #if !defined(MA_NO_NEON) && (defined(__ARM_NEON) || defined(__aarch64__) || defined(_M_ARM64))
- #define MA_SUPPORT_NEON
- #endif
+/*
+Initializes a `ma_context_config` object.
- /* Fall back to looking for the #include file. */
- #if !defined(__GNUC__) && !defined(__clang__) && defined(__has_include)
- #if !defined(MA_SUPPORT_NEON) && !defined(MA_NO_NEON) && __has_include()
- #define MA_SUPPORT_NEON
- #endif
- #endif
- #if defined(MA_SUPPORT_NEON)
- #include
- #endif
-#endif
+Return Value
+------------
+A `ma_context_config` initialized to defaults.
-/* Begin globally disabled warnings. */
-#if defined(_MSC_VER)
- #pragma warning(push)
- #pragma warning(disable:4752) /* found Intel(R) Advanced Vector Extensions; consider using /arch:AVX */
-#endif
-#if defined(MA_X64) || defined(MA_X86)
- #if defined(_MSC_VER) && !defined(__clang__)
- #if _MSC_VER >= 1400
- #include
- static MA_INLINE void ma_cpuid(int info[4], int fid)
- {
- __cpuid(info, fid);
- }
- #else
- #define MA_NO_CPUID
- #endif
+Remarks
+-------
+You must always use this to initialize the default state of the `ma_context_config` object. Not using this will result in your program breaking when miniaudio
+is updated and new members are added to `ma_context_config`. It also sets logical defaults.
- #if _MSC_VER >= 1600 && (defined(_MSC_FULL_VER) && _MSC_FULL_VER >= 160040219)
- static MA_INLINE unsigned __int64 ma_xgetbv(int reg)
- {
- return _xgetbv(reg);
- }
- #else
- #define MA_NO_XGETBV
- #endif
- #elif (defined(__GNUC__) || defined(__clang__)) && !defined(MA_ANDROID)
- static MA_INLINE void ma_cpuid(int info[4], int fid)
- {
- /*
- It looks like the -fPIC option uses the ebx register which GCC complains about. We can work around this by just using a different register, the
- specific register of which I'm letting the compiler decide on. The "k" prefix is used to specify a 32-bit register. The {...} syntax is for
- supporting different assembly dialects.
+You can override members of the returned object by changing it's members directly.
- What's basically happening is that we're saving and restoring the ebx register manually.
- */
- #if defined(DRFLAC_X86) && defined(__PIC__)
- __asm__ __volatile__ (
- "xchg{l} {%%}ebx, %k1;"
- "cpuid;"
- "xchg{l} {%%}ebx, %k1;"
- : "=a"(info[0]), "=&r"(info[1]), "=c"(info[2]), "=d"(info[3]) : "a"(fid), "c"(0)
- );
- #else
- __asm__ __volatile__ (
- "cpuid" : "=a"(info[0]), "=b"(info[1]), "=c"(info[2]), "=d"(info[3]) : "a"(fid), "c"(0)
- );
- #endif
- }
- static MA_INLINE ma_uint64 ma_xgetbv(int reg)
- {
- unsigned int hi;
- unsigned int lo;
+See Also
+--------
+ma_context_init()
+*/
+MA_API ma_context_config ma_context_config_init(void);
- __asm__ __volatile__ (
- "xgetbv" : "=a"(lo), "=d"(hi) : "c"(reg)
- );
+/*
+Initializes a context.
- return ((ma_uint64)hi << 32) | (ma_uint64)lo;
- }
- #else
- #define MA_NO_CPUID
- #define MA_NO_XGETBV
- #endif
-#else
- #define MA_NO_CPUID
- #define MA_NO_XGETBV
-#endif
+The context is used for selecting and initializing an appropriate backend and to represent the backend at a more global level than that of an individual
+device. There is one context to many devices, and a device is created from a context. A context is required to enumerate devices.
-static MA_INLINE ma_bool32 ma_has_sse2(void)
-{
-#if defined(MA_SUPPORT_SSE2)
- #if (defined(MA_X64) || defined(MA_X86)) && !defined(MA_NO_SSE2)
- #if defined(MA_X64)
- return MA_TRUE; /* 64-bit targets always support SSE2. */
- #elif (defined(_M_IX86_FP) && _M_IX86_FP == 2) || defined(__SSE2__)
- return MA_TRUE; /* If the compiler is allowed to freely generate SSE2 code we can assume support. */
- #else
- #if defined(MA_NO_CPUID)
- return MA_FALSE;
- #else
- int info[4];
- ma_cpuid(info, 1);
- return (info[3] & (1 << 26)) != 0;
- #endif
- #endif
- #else
- return MA_FALSE; /* SSE2 is only supported on x86 and x64 architectures. */
- #endif
-#else
- return MA_FALSE; /* No compiler support. */
-#endif
-}
-#if 0
-static MA_INLINE ma_bool32 ma_has_avx()
-{
-#if defined(MA_SUPPORT_AVX)
- #if (defined(MA_X64) || defined(MA_X86)) && !defined(MA_NO_AVX)
- #if defined(_AVX_) || defined(__AVX__)
- return MA_TRUE; /* If the compiler is allowed to freely generate AVX code we can assume support. */
- #else
- /* AVX requires both CPU and OS support. */
- #if defined(MA_NO_CPUID) || defined(MA_NO_XGETBV)
- return MA_FALSE;
- #else
- int info[4];
- ma_cpuid(info, 1);
- if (((info[2] & (1 << 27)) != 0) && ((info[2] & (1 << 28)) != 0)) {
- ma_uint64 xrc = ma_xgetbv(0);
- if ((xrc & 0x06) == 0x06) {
- return MA_TRUE;
- } else {
- return MA_FALSE;
- }
- } else {
- return MA_FALSE;
- }
- #endif
- #endif
- #else
- return MA_FALSE; /* AVX is only supported on x86 and x64 architectures. */
- #endif
-#else
- return MA_FALSE; /* No compiler support. */
-#endif
-}
-#endif
+Parameters
+----------
+backends (in, optional)
+ A list of backends to try initializing, in priority order. Can be NULL, in which case it uses default priority order.
-static MA_INLINE ma_bool32 ma_has_avx2(void)
-{
-#if defined(MA_SUPPORT_AVX2)
- #if (defined(MA_X64) || defined(MA_X86)) && !defined(MA_NO_AVX2)
- #if defined(_AVX2_) || defined(__AVX2__)
- return MA_TRUE; /* If the compiler is allowed to freely generate AVX2 code we can assume support. */
- #else
- /* AVX2 requires both CPU and OS support. */
- #if defined(MA_NO_CPUID) || defined(MA_NO_XGETBV)
- return MA_FALSE;
- #else
- int info1[4];
- int info7[4];
- ma_cpuid(info1, 1);
- ma_cpuid(info7, 7);
- if (((info1[2] & (1 << 27)) != 0) && ((info7[1] & (1 << 5)) != 0)) {
- ma_uint64 xrc = ma_xgetbv(0);
- if ((xrc & 0x06) == 0x06) {
- return MA_TRUE;
- } else {
- return MA_FALSE;
- }
- } else {
- return MA_FALSE;
- }
- #endif
- #endif
- #else
- return MA_FALSE; /* AVX2 is only supported on x86 and x64 architectures. */
- #endif
-#else
- return MA_FALSE; /* No compiler support. */
-#endif
-}
+backendCount (in, optional)
+ The number of items in `backend`. Ignored if `backend` is NULL.
-static MA_INLINE ma_bool32 ma_has_avx512f(void)
-{
-#if defined(MA_SUPPORT_AVX512)
- #if (defined(MA_X64) || defined(MA_X86)) && !defined(MA_NO_AVX512)
- #if defined(__AVX512F__)
- return MA_TRUE; /* If the compiler is allowed to freely generate AVX-512F code we can assume support. */
- #else
- /* AVX-512 requires both CPU and OS support. */
- #if defined(MA_NO_CPUID) || defined(MA_NO_XGETBV)
- return MA_FALSE;
- #else
- int info1[4];
- int info7[4];
- ma_cpuid(info1, 1);
- ma_cpuid(info7, 7);
- if (((info1[2] & (1 << 27)) != 0) && ((info7[1] & (1 << 16)) != 0)) {
- ma_uint64 xrc = ma_xgetbv(0);
- if ((xrc & 0xE6) == 0xE6) {
- return MA_TRUE;
- } else {
- return MA_FALSE;
- }
- } else {
- return MA_FALSE;
- }
- #endif
- #endif
- #else
- return MA_FALSE; /* AVX-512F is only supported on x86 and x64 architectures. */
- #endif
-#else
- return MA_FALSE; /* No compiler support. */
-#endif
-}
+pConfig (in, optional)
+ The context configuration.
-static MA_INLINE ma_bool32 ma_has_neon(void)
-{
-#if defined(MA_SUPPORT_NEON)
- #if defined(MA_ARM) && !defined(MA_NO_NEON)
- #if (defined(__ARM_NEON) || defined(__aarch64__) || defined(_M_ARM64))
- return MA_TRUE; /* If the compiler is allowed to freely generate NEON code we can assume support. */
- #else
- /* TODO: Runtime check. */
- return MA_FALSE;
- #endif
- #else
- return MA_FALSE; /* NEON is only supported on ARM architectures. */
- #endif
-#else
- return MA_FALSE; /* No compiler support. */
-#endif
-}
+pContext (in)
+ A pointer to the context object being initialized.
-#define MA_SIMD_NONE 0
-#define MA_SIMD_SSE2 1
-#define MA_SIMD_AVX2 2
-#define MA_SIMD_NEON 3
-#ifndef MA_PREFERRED_SIMD
- # if defined(MA_SUPPORT_SSE2) && defined(MA_PREFER_SSE2)
- #define MA_PREFERRED_SIMD MA_SIMD_SSE2
- #elif defined(MA_SUPPORT_AVX2) && defined(MA_PREFER_AVX2)
- #define MA_PREFERRED_SIMD MA_SIMD_AVX2
- #elif defined(MA_SUPPORT_NEON) && defined(MA_PREFER_NEON)
- #define MA_PREFERRED_SIMD MA_SIMD_NEON
- #else
- #define MA_PREFERRED_SIMD MA_SIMD_NONE
- #endif
-#endif
+Return Value
+------------
+MA_SUCCESS if successful; any other error code otherwise.
-#if defined(__has_builtin)
- #define MA_COMPILER_HAS_BUILTIN(x) __has_builtin(x)
-#else
- #define MA_COMPILER_HAS_BUILTIN(x) 0
-#endif
-
-#ifndef MA_ASSUME
- #if MA_COMPILER_HAS_BUILTIN(__builtin_assume)
- #define MA_ASSUME(x) __builtin_assume(x)
- #elif MA_COMPILER_HAS_BUILTIN(__builtin_unreachable)
- #define MA_ASSUME(x) do { if (!(x)) __builtin_unreachable(); } while (0)
- #elif defined(_MSC_VER)
- #define MA_ASSUME(x) __assume(x)
- #else
- #define MA_ASSUME(x) while(0)
- #endif
-#endif
-#ifndef MA_RESTRICT
- #if defined(__clang__) || defined(__GNUC__) || defined(_MSC_VER)
- #define MA_RESTRICT __restrict
- #else
- #define MA_RESTRICT
- #endif
-#endif
+Thread Safety
+-------------
+Unsafe. Do not call this function across multiple threads as some backends read and write to global state.
-#if defined(_MSC_VER) && _MSC_VER >= 1400
- #define MA_HAS_BYTESWAP16_INTRINSIC
- #define MA_HAS_BYTESWAP32_INTRINSIC
- #define MA_HAS_BYTESWAP64_INTRINSIC
-#elif defined(__clang__)
- #if MA_COMPILER_HAS_BUILTIN(__builtin_bswap16)
- #define MA_HAS_BYTESWAP16_INTRINSIC
- #endif
- #if MA_COMPILER_HAS_BUILTIN(__builtin_bswap32)
- #define MA_HAS_BYTESWAP32_INTRINSIC
- #endif
- #if MA_COMPILER_HAS_BUILTIN(__builtin_bswap64)
- #define MA_HAS_BYTESWAP64_INTRINSIC
- #endif
-#elif defined(__GNUC__)
- #if ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 3))
- #define MA_HAS_BYTESWAP32_INTRINSIC
- #define MA_HAS_BYTESWAP64_INTRINSIC
- #endif
- #if ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8))
- #define MA_HAS_BYTESWAP16_INTRINSIC
- #endif
-#endif
+Remarks
+-------
+When `backends` is NULL, the default priority order will be used. Below is a list of backends in priority order:
-static MA_INLINE ma_bool32 ma_is_little_endian(void)
-{
-#if defined(MA_X86) || defined(MA_X64)
- return MA_TRUE;
-#else
- int n = 1;
- return (*(char*)&n) == 1;
-#endif
-}
+ |-------------|-----------------------|--------------------------------------------------------|
+ | Name | Enum Name | Supported Operating Systems |
+ |-------------|-----------------------|--------------------------------------------------------|
+ | WASAPI | ma_backend_wasapi | Windows Vista+ |
+ | DirectSound | ma_backend_dsound | Windows XP+ |
+ | WinMM | ma_backend_winmm | Windows XP+ (may work on older versions, but untested) |
+ | Core Audio | ma_backend_coreaudio | macOS, iOS |
+ | ALSA | ma_backend_alsa | Linux |
+ | PulseAudio | ma_backend_pulseaudio | Cross Platform (disabled on Windows, BSD and Android) |
+ | JACK | ma_backend_jack | Cross Platform (disabled on BSD and Android) |
+ | sndio | ma_backend_sndio | OpenBSD |
+ | audio(4) | ma_backend_audio4 | NetBSD, OpenBSD |
+ | OSS | ma_backend_oss | FreeBSD |
+ | AAudio | ma_backend_aaudio | Android 8+ |
+ | OpenSL|ES | ma_backend_opensl | Android (API level 16+) |
+ | Web Audio | ma_backend_webaudio | Web (via Emscripten) |
+ | Null | ma_backend_null | Cross Platform (not used on Web) |
+ |-------------|-----------------------|--------------------------------------------------------|
-static MA_INLINE ma_bool32 ma_is_big_endian(void)
-{
- return !ma_is_little_endian();
-}
+The context can be configured via the `pConfig` argument. The config object is initialized with `ma_context_config_init()`. Individual configuration settings
+can then be set directly on the structure. Below are the members of the `ma_context_config` object.
+ pLog
+ A pointer to the `ma_log` to post log messages to. Can be NULL if the application does not
+ require logging. See the `ma_log` API for details on how to use the logging system.
-static MA_INLINE ma_uint32 ma_swap_endian_uint32(ma_uint32 n)
-{
-#ifdef MA_HAS_BYTESWAP32_INTRINSIC
- #if defined(_MSC_VER)
- return _byteswap_ulong(n);
- #elif defined(__GNUC__) || defined(__clang__)
- #if defined(MA_ARM) && (defined(__ARM_ARCH) && __ARM_ARCH >= 6) && !defined(MA_64BIT) /* <-- 64-bit inline assembly has not been tested, so disabling for now. */
- /* Inline assembly optimized implementation for ARM. In my testing, GCC does not generate optimized code with __builtin_bswap32(). */
- ma_uint32 r;
- __asm__ __volatile__ (
- #if defined(MA_64BIT)
- "rev %w[out], %w[in]" : [out]"=r"(r) : [in]"r"(n) /* <-- This is untested. If someone in the community could test this, that would be appreciated! */
- #else
- "rev %[out], %[in]" : [out]"=r"(r) : [in]"r"(n)
- #endif
- );
- return r;
- #else
- return __builtin_bswap32(n);
- #endif
- #else
- #error "This compiler does not support the byte swap intrinsic."
- #endif
-#else
- return ((n & 0xFF000000) >> 24) |
- ((n & 0x00FF0000) >> 8) |
- ((n & 0x0000FF00) << 8) |
- ((n & 0x000000FF) << 24);
-#endif
-}
+ threadPriority
+ The desired priority to use for the audio thread. Allowable values include the following:
+ |--------------------------------------|
+ | Thread Priority |
+ |--------------------------------------|
+ | ma_thread_priority_idle |
+ | ma_thread_priority_lowest |
+ | ma_thread_priority_low |
+ | ma_thread_priority_normal |
+ | ma_thread_priority_high |
+ | ma_thread_priority_highest (default) |
+ | ma_thread_priority_realtime |
+ | ma_thread_priority_default |
+ |--------------------------------------|
-#if !defined(MA_EMSCRIPTEN)
-#ifdef MA_WIN32
-static void ma_sleep__win32(ma_uint32 milliseconds)
-{
- Sleep((DWORD)milliseconds);
-}
-#endif
-#ifdef MA_POSIX
-static void ma_sleep__posix(ma_uint32 milliseconds)
-{
-#ifdef MA_EMSCRIPTEN
- (void)milliseconds;
- MA_ASSERT(MA_FALSE); /* The Emscripten build should never sleep. */
-#else
- #if _POSIX_C_SOURCE >= 199309L
- struct timespec ts;
- ts.tv_sec = milliseconds / 1000;
- ts.tv_nsec = milliseconds % 1000 * 1000000;
- nanosleep(&ts, NULL);
- #else
- struct timeval tv;
- tv.tv_sec = milliseconds / 1000;
- tv.tv_usec = milliseconds % 1000 * 1000;
- select(0, NULL, NULL, NULL, &tv);
- #endif
-#endif
-}
-#endif
+ threadStackSize
+ The desired size of the stack for the audio thread. Defaults to the operating system's default.
-static void ma_sleep(ma_uint32 milliseconds)
-{
-#ifdef MA_WIN32
- ma_sleep__win32(milliseconds);
-#endif
-#ifdef MA_POSIX
- ma_sleep__posix(milliseconds);
-#endif
-}
-#endif
+ pUserData
+ A pointer to application-defined data. This can be accessed from the context object directly such as `context.pUserData`.
-static MA_INLINE void ma_yield()
-{
-#if defined(__i386) || defined(_M_IX86) || defined(__x86_64__) || defined(_M_X64)
- /* x86/x64 */
- #if (defined(_MSC_VER) || defined(__WATCOMC__) || defined(__DMC__)) && !defined(__clang__)
- #if _MSC_VER >= 1400
- _mm_pause();
- #else
- #if defined(__DMC__)
- /* Digital Mars does not recognize the PAUSE opcode. Fall back to NOP. */
- __asm nop;
- #else
- __asm pause;
- #endif
- #endif
- #else
- __asm__ __volatile__ ("pause");
- #endif
-#elif (defined(__arm__) && defined(__ARM_ARCH) && __ARM_ARCH >= 7) || (defined(_M_ARM) && _M_ARM >= 7) || defined(__ARM_ARCH_6K__) || defined(__ARM_ARCH_6T2__)
- /* ARM */
- #if defined(_MSC_VER)
- /* Apparently there is a __yield() intrinsic that's compatible with ARM, but I cannot find documentation for it nor can I find where it's declared. */
- __yield();
- #else
- __asm__ __volatile__ ("yield"); /* ARMv6K/ARMv6T2 and above. */
- #endif
-#else
- /* Unknown or unsupported architecture. No-op. */
-#endif
-}
+ allocationCallbacks
+ Structure containing custom allocation callbacks. Leaving this at defaults will cause it to use MA_MALLOC, MA_REALLOC and MA_FREE. These allocation
+ callbacks will be used for anything tied to the context, including devices.
+ alsa.useVerboseDeviceEnumeration
+ ALSA will typically enumerate many different devices which can be intrusive and not user-friendly. To combat this, miniaudio will enumerate only unique
+ card/device pairs by default. The problem with this is that you lose a bit of flexibility and control. Setting alsa.useVerboseDeviceEnumeration makes
+ it so the ALSA backend includes all devices. Defaults to false.
+ pulse.pApplicationName
+ PulseAudio only. The application name to use when initializing the PulseAudio context with `pa_context_new()`.
-#ifndef MA_COINIT_VALUE
-#define MA_COINIT_VALUE 0 /* 0 = COINIT_MULTITHREADED */
-#endif
+ pulse.pServerName
+ PulseAudio only. The name of the server to connect to with `pa_context_connect()`.
+ pulse.tryAutoSpawn
+ PulseAudio only. Whether or not to try automatically starting the PulseAudio daemon. Defaults to false. If you set this to true, keep in mind that
+ miniaudio uses a trial and error method to find the most appropriate backend, and this will result in the PulseAudio daemon starting which may be
+ intrusive for the end user.
-#ifndef MA_FLT_MAX
- #ifdef FLT_MAX
- #define MA_FLT_MAX FLT_MAX
- #else
- #define MA_FLT_MAX 3.402823466e+38F
- #endif
-#endif
+ coreaudio.sessionCategory
+ iOS only. The session category to use for the shared AudioSession instance. Below is a list of allowable values and their Core Audio equivalents.
+ |-----------------------------------------|-------------------------------------|
+ | miniaudio Token | Core Audio Token |
+ |-----------------------------------------|-------------------------------------|
+ | ma_ios_session_category_ambient | AVAudioSessionCategoryAmbient |
+ | ma_ios_session_category_solo_ambient | AVAudioSessionCategorySoloAmbient |
+ | ma_ios_session_category_playback | AVAudioSessionCategoryPlayback |
+ | ma_ios_session_category_record | AVAudioSessionCategoryRecord |
+ | ma_ios_session_category_play_and_record | AVAudioSessionCategoryPlayAndRecord |
+ | ma_ios_session_category_multi_route | AVAudioSessionCategoryMultiRoute |
+ | ma_ios_session_category_none | AVAudioSessionCategoryAmbient |
+ | ma_ios_session_category_default | AVAudioSessionCategoryAmbient |
+ |-----------------------------------------|-------------------------------------|
-#ifndef MA_PI
-#define MA_PI 3.14159265358979323846264f
-#endif
-#ifndef MA_PI_D
-#define MA_PI_D 3.14159265358979323846264
-#endif
-#ifndef MA_TAU
-#define MA_TAU 6.28318530717958647693f
-#endif
-#ifndef MA_TAU_D
-#define MA_TAU_D 6.28318530717958647693
-#endif
+ coreaudio.sessionCategoryOptions
+ iOS only. Session category options to use with the shared AudioSession instance. Below is a list of allowable values and their Core Audio equivalents.
+ |---------------------------------------------------------------------------|------------------------------------------------------------------|
+ | miniaudio Token | Core Audio Token |
+ |---------------------------------------------------------------------------|------------------------------------------------------------------|
+ | ma_ios_session_category_option_mix_with_others | AVAudioSessionCategoryOptionMixWithOthers |
+ | ma_ios_session_category_option_duck_others | AVAudioSessionCategoryOptionDuckOthers |
+ | ma_ios_session_category_option_allow_bluetooth | AVAudioSessionCategoryOptionAllowBluetooth |
+ | ma_ios_session_category_option_default_to_speaker | AVAudioSessionCategoryOptionDefaultToSpeaker |
+ | ma_ios_session_category_option_interrupt_spoken_audio_and_mix_with_others | AVAudioSessionCategoryOptionInterruptSpokenAudioAndMixWithOthers |
+ | ma_ios_session_category_option_allow_bluetooth_a2dp | AVAudioSessionCategoryOptionAllowBluetoothA2DP |
+ | ma_ios_session_category_option_allow_air_play | AVAudioSessionCategoryOptionAllowAirPlay |
+ |---------------------------------------------------------------------------|------------------------------------------------------------------|
-/* The default format when ma_format_unknown (0) is requested when initializing a device. */
-#ifndef MA_DEFAULT_FORMAT
-#define MA_DEFAULT_FORMAT ma_format_f32
-#endif
+ coreaudio.noAudioSessionActivate
+ iOS only. When set to true, does not perform an explicit [[AVAudioSession sharedInstace] setActive:true] on initialization.
-/* The default channel count to use when 0 is used when initializing a device. */
-#ifndef MA_DEFAULT_CHANNELS
-#define MA_DEFAULT_CHANNELS 2
-#endif
+ coreaudio.noAudioSessionDeactivate
+ iOS only. When set to true, does not perform an explicit [[AVAudioSession sharedInstace] setActive:false] on uninitialization.
-/* The default sample rate to use when 0 is used when initializing a device. */
-#ifndef MA_DEFAULT_SAMPLE_RATE
-#define MA_DEFAULT_SAMPLE_RATE 48000
-#endif
+ jack.pClientName
+ The name of the client to pass to `jack_client_open()`.
-/* Default periods when none is specified in ma_device_init(). More periods means more work on the CPU. */
-#ifndef MA_DEFAULT_PERIODS
-#define MA_DEFAULT_PERIODS 3
-#endif
+ jack.tryStartServer
+ Whether or not to try auto-starting the JACK server. Defaults to false.
-/* The default period size in milliseconds for low latency mode. */
-#ifndef MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY
-#define MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY 10
-#endif
-/* The default buffer size in milliseconds for conservative mode. */
-#ifndef MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE
-#define MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE 100
-#endif
+It is recommended that only a single context is active at any given time because it's a bulky data structure which performs run-time linking for the
+relevant backends every time it's initialized.
-/* The default LPF filter order for linear resampling. Note that this is clamped to MA_MAX_FILTER_ORDER. */
-#ifndef MA_DEFAULT_RESAMPLER_LPF_ORDER
- #if MA_MAX_FILTER_ORDER >= 4
- #define MA_DEFAULT_RESAMPLER_LPF_ORDER 4
- #else
- #define MA_DEFAULT_RESAMPLER_LPF_ORDER MA_MAX_FILTER_ORDER
- #endif
-#endif
+The location of the context cannot change throughout it's lifetime. Consider allocating the `ma_context` object with `malloc()` if this is an issue. The
+reason for this is that a pointer to the context is stored in the `ma_device` structure.
-#if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
- #pragma GCC diagnostic push
- #pragma GCC diagnostic ignored "-Wunused-variable"
-#endif
+Example 1 - Default Initialization
+----------------------------------
+The example below shows how to initialize the context using the default configuration.
-/* Standard sample rates, in order of priority. */
-static ma_uint32 g_maStandardSampleRatePriorities[] = {
- (ma_uint32)ma_standard_sample_rate_48000,
- (ma_uint32)ma_standard_sample_rate_44100,
+```c
+ma_context context;
+ma_result result = ma_context_init(NULL, 0, NULL, &context);
+if (result != MA_SUCCESS) {
+ // Error.
+}
+```
- (ma_uint32)ma_standard_sample_rate_32000,
- (ma_uint32)ma_standard_sample_rate_24000,
- (ma_uint32)ma_standard_sample_rate_22050,
- (ma_uint32)ma_standard_sample_rate_88200,
- (ma_uint32)ma_standard_sample_rate_96000,
- (ma_uint32)ma_standard_sample_rate_176400,
- (ma_uint32)ma_standard_sample_rate_192000,
+Example 2 - Custom Configuration
+--------------------------------
+The example below shows how to initialize the context using custom backend priorities and a custom configuration. In this hypothetical example, the program
+wants to prioritize ALSA over PulseAudio on Linux. They also want to avoid using the WinMM backend on Windows because it's latency is too high. They also
+want an error to be returned if no valid backend is available which they achieve by excluding the Null backend.
- (ma_uint32)ma_standard_sample_rate_16000,
- (ma_uint32)ma_standard_sample_rate_11025,
- (ma_uint32)ma_standard_sample_rate_8000,
+For the configuration, the program wants to capture any log messages so they can, for example, route it to a log file and user interface.
- (ma_uint32)ma_standard_sample_rate_352800,
- (ma_uint32)ma_standard_sample_rate_384000
+```c
+ma_backend backends[] = {
+ ma_backend_alsa,
+ ma_backend_pulseaudio,
+ ma_backend_wasapi,
+ ma_backend_dsound
};
-static MA_INLINE ma_bool32 ma_is_standard_sample_rate(ma_uint32 sampleRate)
-{
- ma_uint32 iSampleRate;
+ma_log log;
+ma_log_init(&log);
+ma_log_register_callback(&log, ma_log_callback_init(my_log_callbac, pMyLogUserData));
- for (iSampleRate = 0; iSampleRate < sizeof(g_maStandardSampleRatePriorities) / sizeof(g_maStandardSampleRatePriorities[0]); iSampleRate += 1) {
- if (g_maStandardSampleRatePriorities[iSampleRate] == sampleRate) {
- return MA_TRUE;
- }
- }
+ma_context_config config = ma_context_config_init();
+config.pLog = &log; // Specify a custom log object in the config so any logs that are posted from ma_context_init() are captured.
- /* Getting here means the sample rate is not supported. */
- return MA_FALSE;
+ma_context context;
+ma_result result = ma_context_init(backends, sizeof(backends)/sizeof(backends[0]), &config, &context);
+if (result != MA_SUCCESS) {
+ // Error.
+ if (result == MA_NO_BACKEND) {
+ // Couldn't find an appropriate backend.
+ }
}
+// You could also attach a log callback post-initialization:
+ma_log_register_callback(ma_context_get_log(&context), ma_log_callback_init(my_log_callback, pMyLogUserData));
+```
-static ma_format g_maFormatPriorities[] = {
- ma_format_s16, /* Most common */
- ma_format_f32,
- /*ma_format_s24_32,*/ /* Clean alignment */
- ma_format_s32,
+See Also
+--------
+ma_context_config_init()
+ma_context_uninit()
+*/
+MA_API ma_result ma_context_init(const ma_backend backends[], ma_uint32 backendCount, const ma_context_config* pConfig, ma_context* pContext);
- ma_format_s24, /* Unclean alignment */
+/*
+Uninitializes a context.
- ma_format_u8 /* Low quality */
-};
-#if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
- #pragma GCC diagnostic pop
-#endif
+Return Value
+------------
+MA_SUCCESS if successful; any other error code otherwise.
-MA_API void ma_version(ma_uint32* pMajor, ma_uint32* pMinor, ma_uint32* pRevision)
-{
- if (pMajor) {
- *pMajor = MA_VERSION_MAJOR;
- }
- if (pMinor) {
- *pMinor = MA_VERSION_MINOR;
- }
+Thread Safety
+-------------
+Unsafe. Do not call this function across multiple threads as some backends read and write to global state.
- if (pRevision) {
- *pRevision = MA_VERSION_REVISION;
- }
-}
-MA_API const char* ma_version_string(void)
-{
- return MA_VERSION_STRING;
-}
+Remarks
+-------
+Results are undefined if you call this while any device created by this context is still active.
-/******************************************************************************
+See Also
+--------
+ma_context_init()
+*/
+MA_API ma_result ma_context_uninit(ma_context* pContext);
-Standard Library Stuff
+/*
+Retrieves the size of the ma_context object.
-******************************************************************************/
-#ifndef MA_MALLOC
-#ifdef MA_WIN32
-#define MA_MALLOC(sz) HeapAlloc(GetProcessHeap(), 0, (sz))
-#else
-#define MA_MALLOC(sz) malloc((sz))
-#endif
-#endif
+This is mainly for the purpose of bindings to know how much memory to allocate.
+*/
+MA_API size_t ma_context_sizeof(void);
-#ifndef MA_REALLOC
-#ifdef MA_WIN32
-#define MA_REALLOC(p, sz) (((sz) > 0) ? ((p) ? HeapReAlloc(GetProcessHeap(), 0, (p), (sz)) : HeapAlloc(GetProcessHeap(), 0, (sz))) : ((VOID*)(size_t)(HeapFree(GetProcessHeap(), 0, (p)) & 0)))
-#else
-#define MA_REALLOC(p, sz) realloc((p), (sz))
-#endif
-#endif
+/*
+Retrieves a pointer to the log object associated with this context.
-#ifndef MA_FREE
-#ifdef MA_WIN32
-#define MA_FREE(p) HeapFree(GetProcessHeap(), 0, (p))
-#else
-#define MA_FREE(p) free((p))
-#endif
-#endif
-#ifndef MA_ZERO_MEMORY
-#ifdef MA_WIN32
-#define MA_ZERO_MEMORY(p, sz) ZeroMemory((p), (sz))
-#else
-#define MA_ZERO_MEMORY(p, sz) memset((p), 0, (sz))
-#endif
-#endif
+Remarks
+-------
+Pass the returned pointer to `ma_log_post()`, `ma_log_postv()` or `ma_log_postf()` to post a log
+message.
-#ifndef MA_COPY_MEMORY
-#ifdef MA_WIN32
-#define MA_COPY_MEMORY(dst, src, sz) CopyMemory((dst), (src), (sz))
-#else
-#define MA_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz))
-#endif
-#endif
+You can attach your own logging callback to the log with `ma_log_register_callback()`
-#ifndef MA_MOVE_MEMORY
-#ifdef MA_WIN32
-#define MA_MOVE_MEMORY(dst, src, sz) MoveMemory((dst), (src), (sz))
-#else
-#define MA_MOVE_MEMORY(dst, src, sz) memmove((dst), (src), (sz))
-#endif
-#endif
-#ifndef MA_ASSERT
-#ifdef MA_WIN32
-#define MA_ASSERT(condition) assert(condition)
-#else
-#define MA_ASSERT(condition) assert(condition)
-#endif
-#endif
+Return Value
+------------
+A pointer to the `ma_log` object that the context uses to post log messages. If some error occurs,
+NULL will be returned.
+*/
+MA_API ma_log* ma_context_get_log(ma_context* pContext);
-#define MA_ZERO_OBJECT(p) MA_ZERO_MEMORY((p), sizeof(*(p)))
+/*
+Enumerates over every device (both playback and capture).
-#define ma_countof(x) (sizeof(x) / sizeof(x[0]))
-#define ma_max(x, y) (((x) > (y)) ? (x) : (y))
-#define ma_min(x, y) (((x) < (y)) ? (x) : (y))
-#define ma_abs(x) (((x) > 0) ? (x) : -(x))
-#define ma_clamp(x, lo, hi) (ma_max(lo, ma_min(x, hi)))
-#define ma_offset_ptr(p, offset) (((ma_uint8*)(p)) + (offset))
-#define ma_align(x, a) ((x + (a-1)) & ~(a-1))
-#define ma_align_64(x) ma_align(x, 8)
+This is a lower-level enumeration function to the easier to use `ma_context_get_devices()`. Use `ma_context_enumerate_devices()` if you would rather not incur
+an internal heap allocation, or it simply suits your code better.
-#define ma_buffer_frame_capacity(buffer, channels, format) (sizeof(buffer) / ma_get_bytes_per_sample(format) / (channels))
+Note that this only retrieves the ID and name/description of the device. The reason for only retrieving basic information is that it would otherwise require
+opening the backend device in order to probe it for more detailed information which can be inefficient. Consider using `ma_context_get_device_info()` for this,
+but don't call it from within the enumeration callback.
-static MA_INLINE double ma_sind(double x)
-{
- /* TODO: Implement custom sin(x). */
- return sin(x);
-}
+Returning false from the callback will stop enumeration. Returning true will continue enumeration.
-static MA_INLINE double ma_expd(double x)
-{
- /* TODO: Implement custom exp(x). */
- return exp(x);
-}
-static MA_INLINE double ma_logd(double x)
-{
- /* TODO: Implement custom log(x). */
- return log(x);
-}
+Parameters
+----------
+pContext (in)
+ A pointer to the context performing the enumeration.
-static MA_INLINE double ma_powd(double x, double y)
-{
- /* TODO: Implement custom pow(x, y). */
- return pow(x, y);
-}
+callback (in)
+ The callback to fire for each enumerated device.
-static MA_INLINE double ma_sqrtd(double x)
-{
- /* TODO: Implement custom sqrt(x). */
- return sqrt(x);
-}
+pUserData (in)
+ A pointer to application-defined data passed to the callback.
-static MA_INLINE double ma_cosd(double x)
-{
- return ma_sind((MA_PI_D*0.5) - x);
-}
+Return Value
+------------
+MA_SUCCESS if successful; any other error code otherwise.
-static MA_INLINE double ma_log10d(double x)
-{
- return ma_logd(x) * 0.43429448190325182765;
-}
-static MA_INLINE float ma_powf(float x, float y)
-{
- return (float)ma_powd((double)x, (double)y);
-}
+Thread Safety
+-------------
+Safe. This is guarded using a simple mutex lock.
-static MA_INLINE float ma_log10f(float x)
-{
- return (float)ma_log10d((double)x);
-}
+Remarks
+-------
+Do _not_ assume the first enumerated device of a given type is the default device.
-static MA_INLINE double ma_degrees_to_radians(double degrees)
-{
- return degrees * 0.01745329252;
-}
+Some backends and platforms may only support default playback and capture devices.
-static MA_INLINE double ma_radians_to_degrees(double radians)
-{
- return radians * 57.295779512896;
-}
+In general, you should not do anything complicated from within the callback. In particular, do not try initializing a device from within the callback. Also,
+do not try to call `ma_context_get_device_info()` from within the callback.
-static MA_INLINE float ma_degrees_to_radians_f(float degrees)
+Consider using `ma_context_get_devices()` for a simpler and safer API, albeit at the expense of an internal heap allocation.
+
+
+Example 1 - Simple Enumeration
+------------------------------
+ma_bool32 ma_device_enum_callback(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pInfo, void* pUserData)
{
- return degrees * 0.01745329252f;
+ printf("Device Name: %s\n", pInfo->name);
+ return MA_TRUE;
}
-static MA_INLINE float ma_radians_to_degrees_f(float radians)
-{
- return radians * 57.295779512896f;
+ma_result result = ma_context_enumerate_devices(&context, my_device_enum_callback, pMyUserData);
+if (result != MA_SUCCESS) {
+ // Error.
}
+See Also
+--------
+ma_context_get_devices()
+*/
+MA_API ma_result ma_context_enumerate_devices(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData);
+
/*
-Return Values:
- 0: Success
- 22: EINVAL
- 34: ERANGE
+Retrieves basic information about every active playback and/or capture device.
-Not using symbolic constants for errors because I want to avoid #including errno.h
-*/
-MA_API int ma_strcpy_s(char* dst, size_t dstSizeInBytes, const char* src)
-{
- size_t i;
+This function will allocate memory internally for the device lists and return a pointer to them through the `ppPlaybackDeviceInfos` and `ppCaptureDeviceInfos`
+parameters. If you do not want to incur the overhead of these allocations consider using `ma_context_enumerate_devices()` which will instead use a callback.
- if (dst == 0) {
- return 22;
- }
- if (dstSizeInBytes == 0) {
- return 34;
- }
- if (src == 0) {
- dst[0] = '\0';
- return 22;
- }
- for (i = 0; i < dstSizeInBytes && src[i] != '\0'; ++i) {
- dst[i] = src[i];
- }
+Parameters
+----------
+pContext (in)
+ A pointer to the context performing the enumeration.
- if (i < dstSizeInBytes) {
- dst[i] = '\0';
- return 0;
- }
+ppPlaybackDeviceInfos (out)
+ A pointer to a pointer that will receive the address of a buffer containing the list of `ma_device_info` structures for playback devices.
- dst[0] = '\0';
- return 34;
-}
+pPlaybackDeviceCount (out)
+ A pointer to an unsigned integer that will receive the number of playback devices.
-MA_API int ma_wcscpy_s(wchar_t* dst, size_t dstCap, const wchar_t* src)
-{
- size_t i;
+ppCaptureDeviceInfos (out)
+ A pointer to a pointer that will receive the address of a buffer containing the list of `ma_device_info` structures for capture devices.
- if (dst == 0) {
- return 22;
- }
- if (dstCap == 0) {
- return 34;
- }
- if (src == 0) {
- dst[0] = '\0';
- return 22;
- }
+pCaptureDeviceCount (out)
+ A pointer to an unsigned integer that will receive the number of capture devices.
- for (i = 0; i < dstCap && src[i] != '\0'; ++i) {
- dst[i] = src[i];
- }
- if (i < dstCap) {
- dst[i] = '\0';
- return 0;
- }
+Return Value
+------------
+MA_SUCCESS if successful; any other error code otherwise.
- dst[0] = '\0';
- return 34;
-}
+Thread Safety
+-------------
+Unsafe. Since each call to this function invalidates the pointers from the previous call, you should not be calling this simultaneously across multiple
+threads. Instead, you need to make a copy of the returned data with your own higher level synchronization.
-MA_API int ma_strncpy_s(char* dst, size_t dstSizeInBytes, const char* src, size_t count)
-{
- size_t maxcount;
- size_t i;
- if (dst == 0) {
- return 22;
- }
- if (dstSizeInBytes == 0) {
- return 34;
- }
- if (src == 0) {
- dst[0] = '\0';
- return 22;
- }
+Remarks
+-------
+It is _not_ safe to assume the first device in the list is the default device.
- maxcount = count;
- if (count == ((size_t)-1) || count >= dstSizeInBytes) { /* -1 = _TRUNCATE */
- maxcount = dstSizeInBytes - 1;
- }
+You can pass in NULL for the playback or capture lists in which case they'll be ignored.
- for (i = 0; i < maxcount && src[i] != '\0'; ++i) {
- dst[i] = src[i];
- }
+The returned pointers will become invalid upon the next call this this function, or when the context is uninitialized. Do not free the returned pointers.
- if (src[i] == '\0' || i == count || count == ((size_t)-1)) {
- dst[i] = '\0';
- return 0;
- }
- dst[0] = '\0';
- return 34;
-}
+See Also
+--------
+ma_context_get_devices()
+*/
+MA_API ma_result ma_context_get_devices(ma_context* pContext, ma_device_info** ppPlaybackDeviceInfos, ma_uint32* pPlaybackDeviceCount, ma_device_info** ppCaptureDeviceInfos, ma_uint32* pCaptureDeviceCount);
-MA_API int ma_strcat_s(char* dst, size_t dstSizeInBytes, const char* src)
-{
- char* dstorig;
+/*
+Retrieves information about a device of the given type, with the specified ID and share mode.
- if (dst == 0) {
- return 22;
- }
- if (dstSizeInBytes == 0) {
- return 34;
- }
- if (src == 0) {
- dst[0] = '\0';
- return 22;
- }
- dstorig = dst;
+Parameters
+----------
+pContext (in)
+ A pointer to the context performing the query.
- while (dstSizeInBytes > 0 && dst[0] != '\0') {
- dst += 1;
- dstSizeInBytes -= 1;
- }
+deviceType (in)
+ The type of the device being queried. Must be either `ma_device_type_playback` or `ma_device_type_capture`.
- if (dstSizeInBytes == 0) {
- return 22; /* Unterminated. */
- }
+pDeviceID (in)
+ The ID of the device being queried.
+pDeviceInfo (out)
+ A pointer to the `ma_device_info` structure that will receive the device information.
- while (dstSizeInBytes > 0 && src[0] != '\0') {
- *dst++ = *src++;
- dstSizeInBytes -= 1;
- }
- if (dstSizeInBytes > 0) {
- dst[0] = '\0';
- } else {
- dstorig[0] = '\0';
- return 34;
- }
+Return Value
+------------
+MA_SUCCESS if successful; any other error code otherwise.
- return 0;
-}
-MA_API int ma_strncat_s(char* dst, size_t dstSizeInBytes, const char* src, size_t count)
-{
- char* dstorig;
+Thread Safety
+-------------
+Safe. This is guarded using a simple mutex lock.
- if (dst == 0) {
- return 22;
- }
- if (dstSizeInBytes == 0) {
- return 34;
- }
- if (src == 0) {
- return 22;
- }
- dstorig = dst;
+Remarks
+-------
+Do _not_ call this from within the `ma_context_enumerate_devices()` callback.
- while (dstSizeInBytes > 0 && dst[0] != '\0') {
- dst += 1;
- dstSizeInBytes -= 1;
- }
+It's possible for a device to have different information and capabilities depending on whether or not it's opened in shared or exclusive mode. For example, in
+shared mode, WASAPI always uses floating point samples for mixing, but in exclusive mode it can be anything. Therefore, this function allows you to specify
+which share mode you want information for. Note that not all backends and devices support shared or exclusive mode, in which case this function will fail if
+the requested share mode is unsupported.
- if (dstSizeInBytes == 0) {
- return 22; /* Unterminated. */
- }
+This leaves pDeviceInfo unmodified in the result of an error.
+*/
+MA_API ma_result ma_context_get_device_info(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo);
+/*
+Determines if the given context supports loopback mode.
- if (count == ((size_t)-1)) { /* _TRUNCATE */
- count = dstSizeInBytes - 1;
- }
- while (dstSizeInBytes > 0 && src[0] != '\0' && count > 0) {
- *dst++ = *src++;
- dstSizeInBytes -= 1;
- count -= 1;
- }
+Parameters
+----------
+pContext (in)
+ A pointer to the context getting queried.
- if (dstSizeInBytes > 0) {
- dst[0] = '\0';
- } else {
- dstorig[0] = '\0';
- return 34;
- }
- return 0;
-}
+Return Value
+------------
+MA_TRUE if the context supports loopback mode; MA_FALSE otherwise.
+*/
+MA_API ma_bool32 ma_context_is_loopback_supported(ma_context* pContext);
-MA_API int ma_itoa_s(int value, char* dst, size_t dstSizeInBytes, int radix)
-{
- int sign;
- unsigned int valueU;
- char* dstEnd;
- if (dst == NULL || dstSizeInBytes == 0) {
- return 22;
- }
- if (radix < 2 || radix > 36) {
- dst[0] = '\0';
- return 22;
- }
- sign = (value < 0 && radix == 10) ? -1 : 1; /* The negative sign is only used when the base is 10. */
+/*
+Initializes a device config with default settings.
- if (value < 0) {
- valueU = -value;
- } else {
- valueU = value;
- }
- dstEnd = dst;
- do
- {
- int remainder = valueU % radix;
- if (remainder > 9) {
- *dstEnd = (char)((remainder - 10) + 'a');
- } else {
- *dstEnd = (char)(remainder + '0');
- }
+Parameters
+----------
+deviceType (in)
+ The type of the device this config is being initialized for. This must set to one of the following:
- dstEnd += 1;
- dstSizeInBytes -= 1;
- valueU /= radix;
- } while (dstSizeInBytes > 0 && valueU > 0);
+ |-------------------------|
+ | Device Type |
+ |-------------------------|
+ | ma_device_type_playback |
+ | ma_device_type_capture |
+ | ma_device_type_duplex |
+ | ma_device_type_loopback |
+ |-------------------------|
- if (dstSizeInBytes == 0) {
- dst[0] = '\0';
- return 22; /* Ran out of room in the output buffer. */
- }
- if (sign < 0) {
- *dstEnd++ = '-';
- dstSizeInBytes -= 1;
- }
+Return Value
+------------
+A new device config object with default settings. You will typically want to adjust the config after this function returns. See remarks.
- if (dstSizeInBytes == 0) {
- dst[0] = '\0';
- return 22; /* Ran out of room in the output buffer. */
- }
- *dstEnd = '\0';
+Thread Safety
+-------------
+Safe.
- /* At this point the string will be reversed. */
- dstEnd -= 1;
- while (dst < dstEnd) {
- char temp = *dst;
- *dst = *dstEnd;
- *dstEnd = temp;
+Callback Safety
+---------------
+Safe, but don't try initializing a device in a callback.
- dst += 1;
- dstEnd -= 1;
- }
- return 0;
-}
+Remarks
+-------
+The returned config will be initialized to defaults. You will normally want to customize a few variables before initializing the device. See Example 1 for a
+typical configuration which sets the sample format, channel count, sample rate, data callback and user data. These are usually things you will want to change
+before initializing the device.
-MA_API int ma_strcmp(const char* str1, const char* str2)
-{
- if (str1 == str2) return 0;
+See `ma_device_init()` for details on specific configuration options.
- /* These checks differ from the standard implementation. It's not important, but I prefer it just for sanity. */
- if (str1 == NULL) return -1;
- if (str2 == NULL) return 1;
- for (;;) {
- if (str1[0] == '\0') {
- break;
- }
- if (str1[0] != str2[0]) {
- break;
- }
+Example 1 - Simple Configuration
+--------------------------------
+The example below is what a program will typically want to configure for each device at a minimum. Notice how `ma_device_config_init()` is called first, and
+then the returned object is modified directly. This is important because it ensures that your program continues to work as new configuration options are added
+to the `ma_device_config` structure.
- str1 += 1;
- str2 += 1;
- }
+```c
+ma_device_config config = ma_device_config_init(ma_device_type_playback);
+config.playback.format = ma_format_f32;
+config.playback.channels = 2;
+config.sampleRate = 48000;
+config.dataCallback = ma_data_callback;
+config.pUserData = pMyUserData;
+```
- return ((unsigned char*)str1)[0] - ((unsigned char*)str2)[0];
-}
-MA_API int ma_strappend(char* dst, size_t dstSize, const char* srcA, const char* srcB)
-{
- int result;
+See Also
+--------
+ma_device_init()
+ma_device_init_ex()
+*/
+MA_API ma_device_config ma_device_config_init(ma_device_type deviceType);
- result = ma_strncpy_s(dst, dstSize, srcA, (size_t)-1);
- if (result != 0) {
- return result;
- }
- result = ma_strncat_s(dst, dstSize, srcB, (size_t)-1);
- if (result != 0) {
- return result;
- }
+/*
+Initializes a device.
- return result;
-}
+A device represents a physical audio device. The idea is you send or receive audio data from the device to either play it back through a speaker, or capture it
+from a microphone. Whether or not you should send or receive data from the device (or both) depends on the type of device you are initializing which can be
+playback, capture, full-duplex or loopback. (Note that loopback mode is only supported on select backends.) Sending and receiving audio data to and from the
+device is done via a callback which is fired by miniaudio at periodic time intervals.
-MA_API char* ma_copy_string(const char* src, const ma_allocation_callbacks* pAllocationCallbacks)
-{
- size_t sz = strlen(src)+1;
- char* dst = (char*)ma_malloc(sz, pAllocationCallbacks);
- if (dst == NULL) {
- return NULL;
- }
+The frequency at which data is delivered to and from a device depends on the size of it's period. The size of the period can be defined in terms of PCM frames
+or milliseconds, whichever is more convenient. Generally speaking, the smaller the period, the lower the latency at the expense of higher CPU usage and
+increased risk of glitching due to the more frequent and granular data deliver intervals. The size of a period will depend on your requirements, but
+miniaudio's defaults should work fine for most scenarios. If you're building a game you should leave this fairly small, whereas if you're building a simple
+media player you can make it larger. Note that the period size you request is actually just a hint - miniaudio will tell the backend what you want, but the
+backend is ultimately responsible for what it gives you. You cannot assume you will get exactly what you ask for.
- ma_strcpy_s(dst, sz, src);
+When delivering data to and from a device you need to make sure it's in the correct format which you can set through the device configuration. You just set the
+format that you want to use and miniaudio will perform all of the necessary conversion for you internally. When delivering data to and from the callback you
+can assume the format is the same as what you requested when you initialized the device. See Remarks for more details on miniaudio's data conversion pipeline.
- return dst;
-}
-MA_API wchar_t* ma_copy_string_w(const wchar_t* src, const ma_allocation_callbacks* pAllocationCallbacks)
-{
- size_t sz = wcslen(src)+1;
- wchar_t* dst = (wchar_t*)ma_malloc(sz * sizeof(*dst), pAllocationCallbacks);
- if (dst == NULL) {
- return NULL;
- }
+Parameters
+----------
+pContext (in, optional)
+ A pointer to the context that owns the device. This can be null, in which case it creates a default context internally.
- ma_wcscpy_s(dst, sz, src);
+pConfig (in)
+ A pointer to the device configuration. Cannot be null. See remarks for details.
- return dst;
-}
+pDevice (out)
+ A pointer to the device object being initialized.
-#include
-static ma_result ma_result_from_errno(int e)
-{
- switch (e)
- {
- case 0: return MA_SUCCESS;
- #ifdef EPERM
- case EPERM: return MA_INVALID_OPERATION;
- #endif
- #ifdef ENOENT
- case ENOENT: return MA_DOES_NOT_EXIST;
- #endif
- #ifdef ESRCH
- case ESRCH: return MA_DOES_NOT_EXIST;
- #endif
- #ifdef EINTR
- case EINTR: return MA_INTERRUPT;
- #endif
- #ifdef EIO
- case EIO: return MA_IO_ERROR;
- #endif
- #ifdef ENXIO
- case ENXIO: return MA_DOES_NOT_EXIST;
- #endif
- #ifdef E2BIG
- case E2BIG: return MA_INVALID_ARGS;
- #endif
- #ifdef ENOEXEC
- case ENOEXEC: return MA_INVALID_FILE;
- #endif
- #ifdef EBADF
- case EBADF: return MA_INVALID_FILE;
- #endif
- #ifdef ECHILD
- case ECHILD: return MA_ERROR;
- #endif
- #ifdef EAGAIN
- case EAGAIN: return MA_UNAVAILABLE;
- #endif
- #ifdef ENOMEM
- case ENOMEM: return MA_OUT_OF_MEMORY;
- #endif
- #ifdef EACCES
- case EACCES: return MA_ACCESS_DENIED;
- #endif
- #ifdef EFAULT
- case EFAULT: return MA_BAD_ADDRESS;
- #endif
- #ifdef ENOTBLK
- case ENOTBLK: return MA_ERROR;
- #endif
- #ifdef EBUSY
- case EBUSY: return MA_BUSY;
- #endif
- #ifdef EEXIST
- case EEXIST: return MA_ALREADY_EXISTS;
- #endif
- #ifdef EXDEV
- case EXDEV: return MA_ERROR;
- #endif
- #ifdef ENODEV
- case ENODEV: return MA_DOES_NOT_EXIST;
- #endif
- #ifdef ENOTDIR
- case ENOTDIR: return MA_NOT_DIRECTORY;
- #endif
- #ifdef EISDIR
- case EISDIR: return MA_IS_DIRECTORY;
- #endif
- #ifdef EINVAL
- case EINVAL: return MA_INVALID_ARGS;
- #endif
- #ifdef ENFILE
- case ENFILE: return MA_TOO_MANY_OPEN_FILES;
- #endif
- #ifdef EMFILE
- case EMFILE: return MA_TOO_MANY_OPEN_FILES;
- #endif
- #ifdef ENOTTY
- case ENOTTY: return MA_INVALID_OPERATION;
- #endif
- #ifdef ETXTBSY
- case ETXTBSY: return MA_BUSY;
- #endif
- #ifdef EFBIG
- case EFBIG: return MA_TOO_BIG;
- #endif
- #ifdef ENOSPC
- case ENOSPC: return MA_NO_SPACE;
- #endif
- #ifdef ESPIPE
- case ESPIPE: return MA_BAD_SEEK;
- #endif
- #ifdef EROFS
- case EROFS: return MA_ACCESS_DENIED;
- #endif
- #ifdef EMLINK
- case EMLINK: return MA_TOO_MANY_LINKS;
- #endif
- #ifdef EPIPE
- case EPIPE: return MA_BAD_PIPE;
- #endif
- #ifdef EDOM
- case EDOM: return MA_OUT_OF_RANGE;
- #endif
- #ifdef ERANGE
- case ERANGE: return MA_OUT_OF_RANGE;
- #endif
- #ifdef EDEADLK
- case EDEADLK: return MA_DEADLOCK;
- #endif
- #ifdef ENAMETOOLONG
- case ENAMETOOLONG: return MA_PATH_TOO_LONG;
- #endif
- #ifdef ENOLCK
- case ENOLCK: return MA_ERROR;
- #endif
- #ifdef ENOSYS
- case ENOSYS: return MA_NOT_IMPLEMENTED;
- #endif
- #ifdef ENOTEMPTY
- case ENOTEMPTY: return MA_DIRECTORY_NOT_EMPTY;
- #endif
- #ifdef ELOOP
- case ELOOP: return MA_TOO_MANY_LINKS;
- #endif
- #ifdef ENOMSG
- case ENOMSG: return MA_NO_MESSAGE;
- #endif
- #ifdef EIDRM
- case EIDRM: return MA_ERROR;
- #endif
- #ifdef ECHRNG
- case ECHRNG: return MA_ERROR;
- #endif
- #ifdef EL2NSYNC
- case EL2NSYNC: return MA_ERROR;
- #endif
- #ifdef EL3HLT
- case EL3HLT: return MA_ERROR;
- #endif
- #ifdef EL3RST
- case EL3RST: return MA_ERROR;
- #endif
- #ifdef ELNRNG
- case ELNRNG: return MA_OUT_OF_RANGE;
- #endif
- #ifdef EUNATCH
- case EUNATCH: return MA_ERROR;
- #endif
- #ifdef ENOCSI
- case ENOCSI: return MA_ERROR;
- #endif
- #ifdef EL2HLT
- case EL2HLT: return MA_ERROR;
- #endif
- #ifdef EBADE
- case EBADE: return MA_ERROR;
- #endif
- #ifdef EBADR
- case EBADR: return MA_ERROR;
- #endif
- #ifdef EXFULL
- case EXFULL: return MA_ERROR;
- #endif
- #ifdef ENOANO
- case ENOANO: return MA_ERROR;
- #endif
- #ifdef EBADRQC
- case EBADRQC: return MA_ERROR;
- #endif
- #ifdef EBADSLT
- case EBADSLT: return MA_ERROR;
- #endif
- #ifdef EBFONT
- case EBFONT: return MA_INVALID_FILE;
- #endif
- #ifdef ENOSTR
- case ENOSTR: return MA_ERROR;
- #endif
- #ifdef ENODATA
- case ENODATA: return MA_NO_DATA_AVAILABLE;
- #endif
- #ifdef ETIME
- case ETIME: return MA_TIMEOUT;
- #endif
- #ifdef ENOSR
- case ENOSR: return MA_NO_DATA_AVAILABLE;
- #endif
- #ifdef ENONET
- case ENONET: return MA_NO_NETWORK;
- #endif
- #ifdef ENOPKG
- case ENOPKG: return MA_ERROR;
- #endif
- #ifdef EREMOTE
- case EREMOTE: return MA_ERROR;
- #endif
- #ifdef ENOLINK
- case ENOLINK: return MA_ERROR;
- #endif
- #ifdef EADV
- case EADV: return MA_ERROR;
- #endif
- #ifdef ESRMNT
- case ESRMNT: return MA_ERROR;
- #endif
- #ifdef ECOMM
- case ECOMM: return MA_ERROR;
- #endif
- #ifdef EPROTO
- case EPROTO: return MA_ERROR;
- #endif
- #ifdef EMULTIHOP
- case EMULTIHOP: return MA_ERROR;
- #endif
- #ifdef EDOTDOT
- case EDOTDOT: return MA_ERROR;
- #endif
- #ifdef EBADMSG
- case EBADMSG: return MA_BAD_MESSAGE;
- #endif
- #ifdef EOVERFLOW
- case EOVERFLOW: return MA_TOO_BIG;
- #endif
- #ifdef ENOTUNIQ
- case ENOTUNIQ: return MA_NOT_UNIQUE;
- #endif
- #ifdef EBADFD
- case EBADFD: return MA_ERROR;
- #endif
- #ifdef EREMCHG
- case EREMCHG: return MA_ERROR;
- #endif
- #ifdef ELIBACC
- case ELIBACC: return MA_ACCESS_DENIED;
- #endif
- #ifdef ELIBBAD
- case ELIBBAD: return MA_INVALID_FILE;
- #endif
- #ifdef ELIBSCN
- case ELIBSCN: return MA_INVALID_FILE;
- #endif
- #ifdef ELIBMAX
- case ELIBMAX: return MA_ERROR;
- #endif
- #ifdef ELIBEXEC
- case ELIBEXEC: return MA_ERROR;
- #endif
- #ifdef EILSEQ
- case EILSEQ: return MA_INVALID_DATA;
- #endif
- #ifdef ERESTART
- case ERESTART: return MA_ERROR;
- #endif
- #ifdef ESTRPIPE
- case ESTRPIPE: return MA_ERROR;
- #endif
- #ifdef EUSERS
- case EUSERS: return MA_ERROR;
- #endif
- #ifdef ENOTSOCK
- case ENOTSOCK: return MA_NOT_SOCKET;
- #endif
- #ifdef EDESTADDRREQ
- case EDESTADDRREQ: return MA_NO_ADDRESS;
- #endif
- #ifdef EMSGSIZE
- case EMSGSIZE: return MA_TOO_BIG;
- #endif
- #ifdef EPROTOTYPE
- case EPROTOTYPE: return MA_BAD_PROTOCOL;
- #endif
- #ifdef ENOPROTOOPT
- case ENOPROTOOPT: return MA_PROTOCOL_UNAVAILABLE;
- #endif
- #ifdef EPROTONOSUPPORT
- case EPROTONOSUPPORT: return MA_PROTOCOL_NOT_SUPPORTED;
- #endif
- #ifdef ESOCKTNOSUPPORT
- case ESOCKTNOSUPPORT: return MA_SOCKET_NOT_SUPPORTED;
- #endif
- #ifdef EOPNOTSUPP
- case EOPNOTSUPP: return MA_INVALID_OPERATION;
- #endif
- #ifdef EPFNOSUPPORT
- case EPFNOSUPPORT: return MA_PROTOCOL_FAMILY_NOT_SUPPORTED;
- #endif
- #ifdef EAFNOSUPPORT
- case EAFNOSUPPORT: return MA_ADDRESS_FAMILY_NOT_SUPPORTED;
- #endif
- #ifdef EADDRINUSE
- case EADDRINUSE: return MA_ALREADY_IN_USE;
- #endif
- #ifdef EADDRNOTAVAIL
- case EADDRNOTAVAIL: return MA_ERROR;
- #endif
- #ifdef ENETDOWN
- case ENETDOWN: return MA_NO_NETWORK;
- #endif
- #ifdef ENETUNREACH
- case ENETUNREACH: return MA_NO_NETWORK;
- #endif
- #ifdef ENETRESET
- case ENETRESET: return MA_NO_NETWORK;
- #endif
- #ifdef ECONNABORTED
- case ECONNABORTED: return MA_NO_NETWORK;
- #endif
- #ifdef ECONNRESET
- case ECONNRESET: return MA_CONNECTION_RESET;
- #endif
- #ifdef ENOBUFS
- case ENOBUFS: return MA_NO_SPACE;
- #endif
- #ifdef EISCONN
- case EISCONN: return MA_ALREADY_CONNECTED;
- #endif
- #ifdef ENOTCONN
- case ENOTCONN: return MA_NOT_CONNECTED;
- #endif
- #ifdef ESHUTDOWN
- case ESHUTDOWN: return MA_ERROR;
- #endif
- #ifdef ETOOMANYREFS
- case ETOOMANYREFS: return MA_ERROR;
- #endif
- #ifdef ETIMEDOUT
- case ETIMEDOUT: return MA_TIMEOUT;
- #endif
- #ifdef ECONNREFUSED
- case ECONNREFUSED: return MA_CONNECTION_REFUSED;
- #endif
- #ifdef EHOSTDOWN
- case EHOSTDOWN: return MA_NO_HOST;
- #endif
- #ifdef EHOSTUNREACH
- case EHOSTUNREACH: return MA_NO_HOST;
- #endif
- #ifdef EALREADY
- case EALREADY: return MA_IN_PROGRESS;
- #endif
- #ifdef EINPROGRESS
- case EINPROGRESS: return MA_IN_PROGRESS;
- #endif
- #ifdef ESTALE
- case ESTALE: return MA_INVALID_FILE;
- #endif
- #ifdef EUCLEAN
- case EUCLEAN: return MA_ERROR;
- #endif
- #ifdef ENOTNAM
- case ENOTNAM: return MA_ERROR;
- #endif
- #ifdef ENAVAIL
- case ENAVAIL: return MA_ERROR;
- #endif
- #ifdef EISNAM
- case EISNAM: return MA_ERROR;
- #endif
- #ifdef EREMOTEIO
- case EREMOTEIO: return MA_IO_ERROR;
- #endif
- #ifdef EDQUOT
- case EDQUOT: return MA_NO_SPACE;
- #endif
- #ifdef ENOMEDIUM
- case ENOMEDIUM: return MA_DOES_NOT_EXIST;
- #endif
- #ifdef EMEDIUMTYPE
- case EMEDIUMTYPE: return MA_ERROR;
- #endif
- #ifdef ECANCELED
- case ECANCELED: return MA_CANCELLED;
- #endif
- #ifdef ENOKEY
- case ENOKEY: return MA_ERROR;
- #endif
- #ifdef EKEYEXPIRED
- case EKEYEXPIRED: return MA_ERROR;
- #endif
- #ifdef EKEYREVOKED
- case EKEYREVOKED: return MA_ERROR;
- #endif
- #ifdef EKEYREJECTED
- case EKEYREJECTED: return MA_ERROR;
- #endif
- #ifdef EOWNERDEAD
- case EOWNERDEAD: return MA_ERROR;
- #endif
- #ifdef ENOTRECOVERABLE
- case ENOTRECOVERABLE: return MA_ERROR;
- #endif
- #ifdef ERFKILL
- case ERFKILL: return MA_ERROR;
- #endif
- #ifdef EHWPOISON
- case EHWPOISON: return MA_ERROR;
- #endif
- default: return MA_ERROR;
- }
-}
+Return Value
+------------
+MA_SUCCESS if successful; any other error code otherwise.
-MA_API ma_result ma_fopen(FILE** ppFile, const char* pFilePath, const char* pOpenMode)
-{
-#if defined(_MSC_VER) && _MSC_VER >= 1400
- errno_t err;
-#endif
- if (ppFile != NULL) {
- *ppFile = NULL; /* Safety. */
- }
+Thread Safety
+-------------
+Unsafe. It is not safe to call this function simultaneously for different devices because some backends depend on and mutate global state. The same applies to
+calling this at the same time as `ma_device_uninit()`.
- if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) {
- return MA_INVALID_ARGS;
- }
-#if defined(_MSC_VER) && _MSC_VER >= 1400
- err = fopen_s(ppFile, pFilePath, pOpenMode);
- if (err != 0) {
- return ma_result_from_errno(err);
- }
-#else
-#if defined(_WIN32) || defined(__APPLE__)
- *ppFile = fopen(pFilePath, pOpenMode);
-#else
- #if defined(_FILE_OFFSET_BITS) && _FILE_OFFSET_BITS == 64 && defined(_LARGEFILE64_SOURCE)
- *ppFile = fopen64(pFilePath, pOpenMode);
- #else
- *ppFile = fopen(pFilePath, pOpenMode);
- #endif
-#endif
- if (*ppFile == NULL) {
- ma_result result = ma_result_from_errno(errno);
- if (result == MA_SUCCESS) {
- result = MA_ERROR; /* Just a safety check to make sure we never ever return success when pFile == NULL. */
- }
+Callback Safety
+---------------
+Unsafe. It is not safe to call this inside any callback.
- return result;
- }
-#endif
- return MA_SUCCESS;
-}
+Remarks
+-------
+Setting `pContext` to NULL will result in miniaudio creating a default context internally and is equivalent to passing in a context initialized like so:
+ ```c
+ ma_context_init(NULL, 0, NULL, &context);
+ ```
+Do not set `pContext` to NULL if you are needing to open multiple devices. You can, however, use NULL when initializing the first device, and then use
+device.pContext for the initialization of other devices.
-/*
-_wfopen() isn't always available in all compilation environments.
+The device can be configured via the `pConfig` argument. The config object is initialized with `ma_device_config_init()`. Individual configuration settings can
+then be set directly on the structure. Below are the members of the `ma_device_config` object.
- * Windows only.
- * MSVC seems to support it universally as far back as VC6 from what I can tell (haven't checked further back).
- * MinGW-64 (both 32- and 64-bit) seems to support it.
- * MinGW wraps it in !defined(__STRICT_ANSI__).
- * OpenWatcom wraps it in !defined(_NO_EXT_KEYS).
+ deviceType
+ Must be `ma_device_type_playback`, `ma_device_type_capture`, `ma_device_type_duplex` of `ma_device_type_loopback`.
-This can be reviewed as compatibility issues arise. The preference is to use _wfopen_s() and _wfopen() as opposed to the wcsrtombs()
-fallback, so if you notice your compiler not detecting this properly I'm happy to look at adding support.
-*/
-#if defined(_WIN32)
- #if defined(_MSC_VER) || defined(__MINGW64__) || (!defined(__STRICT_ANSI__) && !defined(_NO_EXT_KEYS))
- #define MA_HAS_WFOPEN
- #endif
-#endif
+ sampleRate
+ The sample rate, in hertz. The most common sample rates are 48000 and 44100. Setting this to 0 will use the device's native sample rate.
-MA_API ma_result ma_wfopen(FILE** ppFile, const wchar_t* pFilePath, const wchar_t* pOpenMode, const ma_allocation_callbacks* pAllocationCallbacks)
-{
- if (ppFile != NULL) {
- *ppFile = NULL; /* Safety. */
- }
+ periodSizeInFrames
+ The desired size of a period in PCM frames. If this is 0, `periodSizeInMilliseconds` will be used instead. If both are 0 the default buffer size will
+ be used depending on the selected performance profile. This value affects latency. See below for details.
- if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) {
- return MA_INVALID_ARGS;
- }
+ periodSizeInMilliseconds
+ The desired size of a period in milliseconds. If this is 0, `periodSizeInFrames` will be used instead. If both are 0 the default buffer size will be
+ used depending on the selected performance profile. The value affects latency. See below for details.
-#if defined(MA_HAS_WFOPEN)
- {
- /* Use _wfopen() on Windows. */
- #if defined(_MSC_VER) && _MSC_VER >= 1400
- errno_t err = _wfopen_s(ppFile, pFilePath, pOpenMode);
- if (err != 0) {
- return ma_result_from_errno(err);
- }
- #else
- *ppFile = _wfopen(pFilePath, pOpenMode);
- if (*ppFile == NULL) {
- return ma_result_from_errno(errno);
- }
- #endif
- (void)pAllocationCallbacks;
- }
-#else
- /*
- Use fopen() on anything other than Windows. Requires a conversion. This is annoying because fopen() is locale specific. The only real way I can
- think of to do this is with wcsrtombs(). Note that wcstombs() is apparently not thread-safe because it uses a static global mbstate_t object for
- maintaining state. I've checked this with -std=c89 and it works, but if somebody get's a compiler error I'll look into improving compatibility.
- */
- {
- mbstate_t mbs;
- size_t lenMB;
- const wchar_t* pFilePathTemp = pFilePath;
- char* pFilePathMB = NULL;
- char pOpenModeMB[32] = {0};
+ periods
+ The number of periods making up the device's entire buffer. The total buffer size is `periodSizeInFrames` or `periodSizeInMilliseconds` multiplied by
+ this value. This is just a hint as backends will be the ones who ultimately decide how your periods will be configured.
- /* Get the length first. */
- MA_ZERO_OBJECT(&mbs);
- lenMB = wcsrtombs(NULL, &pFilePathTemp, 0, &mbs);
- if (lenMB == (size_t)-1) {
- return ma_result_from_errno(errno);
- }
+ performanceProfile
+ A hint to miniaudio as to the performance requirements of your program. Can be either `ma_performance_profile_low_latency` (default) or
+ `ma_performance_profile_conservative`. This mainly affects the size of default buffers and can usually be left at it's default value.
- pFilePathMB = (char*)ma_malloc(lenMB + 1, pAllocationCallbacks);
- if (pFilePathMB == NULL) {
- return MA_OUT_OF_MEMORY;
- }
+ noPreSilencedOutputBuffer
+ When set to true, the contents of the output buffer passed into the data callback will be left undefined. When set to false (default), the contents of
+ the output buffer will be cleared the zero. You can use this to avoid the overhead of zeroing out the buffer if you can guarantee that your data
+ callback will write to every sample in the output buffer, or if you are doing your own clearing.
- pFilePathTemp = pFilePath;
- MA_ZERO_OBJECT(&mbs);
- wcsrtombs(pFilePathMB, &pFilePathTemp, lenMB + 1, &mbs);
+ noClip
+ When set to true, the contents of the output buffer passed into the data callback will be clipped after returning. When set to false (default), the
+ contents of the output buffer are left alone after returning and it will be left up to the backend itself to decide whether or not the clip. This only
+ applies when the playback sample format is f32.
- /* The open mode should always consist of ASCII characters so we should be able to do a trivial conversion. */
- {
- size_t i = 0;
- for (;;) {
- if (pOpenMode[i] == 0) {
- pOpenModeMB[i] = '\0';
- break;
- }
+ noDisableDenormals
+ By default, miniaudio will disable denormals when the data callback is called. Setting this to true will prevent the disabling of denormals.
- pOpenModeMB[i] = (char)pOpenMode[i];
- i += 1;
- }
- }
+ noFixedSizedCallback
+ Allows miniaudio to fire the data callback with any frame count. When this is set to true, the data callback will be fired with a consistent frame
+ count as specified by `periodSizeInFrames` or `periodSizeInMilliseconds`. When set to false, miniaudio will fire the callback with whatever the
+ backend requests, which could be anything.
- *ppFile = fopen(pFilePathMB, pOpenModeMB);
+ dataCallback
+ The callback to fire whenever data is ready to be delivered to or from the device.
- ma_free(pFilePathMB, pAllocationCallbacks);
- }
+ notificationCallback
+ The callback to fire when something has changed with the device, such as whether or not it has been started or stopped.
- if (*ppFile == NULL) {
- return MA_ERROR;
- }
-#endif
+ pUserData
+ The user data pointer to use with the device. You can access this directly from the device object like `device.pUserData`.
- return MA_SUCCESS;
-}
+ resampling.algorithm
+ The resampling algorithm to use when miniaudio needs to perform resampling between the rate specified by `sampleRate` and the device's native rate. The
+ default value is `ma_resample_algorithm_linear`, and the quality can be configured with `resampling.linear.lpfOrder`.
+ resampling.pBackendVTable
+ A pointer to an optional vtable that can be used for plugging in a custom resampler.
+ resampling.pBackendUserData
+ A pointer that will passed to callbacks in pBackendVTable.
-static MA_INLINE void ma_copy_memory_64(void* dst, const void* src, ma_uint64 sizeInBytes)
-{
-#if 0xFFFFFFFFFFFFFFFF <= MA_SIZE_MAX
- MA_COPY_MEMORY(dst, src, (size_t)sizeInBytes);
-#else
- while (sizeInBytes > 0) {
- ma_uint64 bytesToCopyNow = sizeInBytes;
- if (bytesToCopyNow > MA_SIZE_MAX) {
- bytesToCopyNow = MA_SIZE_MAX;
- }
+ resampling.linear.lpfOrder
+ The linear resampler applies a low-pass filter as part of it's procesing for anti-aliasing. This setting controls the order of the filter. The higher
+ the value, the better the quality, in general. Setting this to 0 will disable low-pass filtering altogether. The maximum value is
+ `MA_MAX_FILTER_ORDER`. The default value is `min(4, MA_MAX_FILTER_ORDER)`.
- MA_COPY_MEMORY(dst, src, (size_t)bytesToCopyNow); /* Safe cast to size_t. */
+ playback.pDeviceID
+ A pointer to a `ma_device_id` structure containing the ID of the playback device to initialize. Setting this NULL (default) will use the system's
+ default playback device. Retrieve the device ID from the `ma_device_info` structure, which can be retrieved using device enumeration.
- sizeInBytes -= bytesToCopyNow;
- dst = ( void*)(( ma_uint8*)dst + bytesToCopyNow);
- src = (const void*)((const ma_uint8*)src + bytesToCopyNow);
- }
-#endif
-}
+ playback.format
+ The sample format to use for playback. When set to `ma_format_unknown` the device's native format will be used. This can be retrieved after
+ initialization from the device object directly with `device.playback.format`.
-static MA_INLINE void ma_zero_memory_64(void* dst, ma_uint64 sizeInBytes)
-{
-#if 0xFFFFFFFFFFFFFFFF <= MA_SIZE_MAX
- MA_ZERO_MEMORY(dst, (size_t)sizeInBytes);
-#else
- while (sizeInBytes > 0) {
- ma_uint64 bytesToZeroNow = sizeInBytes;
- if (bytesToZeroNow > MA_SIZE_MAX) {
- bytesToZeroNow = MA_SIZE_MAX;
- }
+ playback.channels
+ The number of channels to use for playback. When set to 0 the device's native channel count will be used. This can be retrieved after initialization
+ from the device object directly with `device.playback.channels`.
- MA_ZERO_MEMORY(dst, (size_t)bytesToZeroNow); /* Safe cast to size_t. */
+ playback.pChannelMap
+ The channel map to use for playback. When left empty, the device's native channel map will be used. This can be retrieved after initialization from the
+ device object direct with `device.playback.pChannelMap`. When set, the buffer should contain `channels` items.
- sizeInBytes -= bytesToZeroNow;
- dst = (void*)((ma_uint8*)dst + bytesToZeroNow);
- }
-#endif
-}
+ playback.shareMode
+ The preferred share mode to use for playback. Can be either `ma_share_mode_shared` (default) or `ma_share_mode_exclusive`. Note that if you specify
+ exclusive mode, but it's not supported by the backend, initialization will fail. You can then fall back to shared mode if desired by changing this to
+ ma_share_mode_shared and reinitializing.
+ capture.pDeviceID
+ A pointer to a `ma_device_id` structure containing the ID of the capture device to initialize. Setting this NULL (default) will use the system's
+ default capture device. Retrieve the device ID from the `ma_device_info` structure, which can be retrieved using device enumeration.
-/* Thanks to good old Bit Twiddling Hacks for this one: http://graphics.stanford.edu/~seander/bithacks.html#RoundUpPowerOf2 */
-static MA_INLINE unsigned int ma_next_power_of_2(unsigned int x)
-{
- x--;
- x |= x >> 1;
- x |= x >> 2;
- x |= x >> 4;
- x |= x >> 8;
- x |= x >> 16;
- x++;
+ capture.format
+ The sample format to use for capture. When set to `ma_format_unknown` the device's native format will be used. This can be retrieved after
+ initialization from the device object directly with `device.capture.format`.
- return x;
-}
+ capture.channels
+ The number of channels to use for capture. When set to 0 the device's native channel count will be used. This can be retrieved after initialization
+ from the device object directly with `device.capture.channels`.
-static MA_INLINE unsigned int ma_prev_power_of_2(unsigned int x)
-{
- return ma_next_power_of_2(x) >> 1;
-}
+ capture.pChannelMap
+ The channel map to use for capture. When left empty, the device's native channel map will be used. This can be retrieved after initialization from the
+ device object direct with `device.capture.pChannelMap`. When set, the buffer should contain `channels` items.
-static MA_INLINE unsigned int ma_round_to_power_of_2(unsigned int x)
-{
- unsigned int prev = ma_prev_power_of_2(x);
- unsigned int next = ma_next_power_of_2(x);
- if ((next - x) > (x - prev)) {
- return prev;
- } else {
- return next;
- }
-}
+ capture.shareMode
+ The preferred share mode to use for capture. Can be either `ma_share_mode_shared` (default) or `ma_share_mode_exclusive`. Note that if you specify
+ exclusive mode, but it's not supported by the backend, initialization will fail. You can then fall back to shared mode if desired by changing this to
+ ma_share_mode_shared and reinitializing.
-static MA_INLINE unsigned int ma_count_set_bits(unsigned int x)
-{
- unsigned int count = 0;
- while (x != 0) {
- if (x & 1) {
- count += 1;
- }
+ wasapi.noAutoConvertSRC
+ WASAPI only. When set to true, disables WASAPI's automatic resampling and forces the use of miniaudio's resampler. Defaults to false.
- x = x >> 1;
- }
+ wasapi.noDefaultQualitySRC
+ WASAPI only. Only used when `wasapi.noAutoConvertSRC` is set to false. When set to true, disables the use of `AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY`.
+ You should usually leave this set to false, which is the default.
- return count;
-}
+ wasapi.noAutoStreamRouting
+ WASAPI only. When set to true, disables automatic stream routing on the WASAPI backend. Defaults to false.
+ wasapi.noHardwareOffloading
+ WASAPI only. When set to true, disables the use of WASAPI's hardware offloading feature. Defaults to false.
+ alsa.noMMap
+ ALSA only. When set to true, disables MMap mode. Defaults to false.
-/**************************************************************************************************************************************************************
+ alsa.noAutoFormat
+ ALSA only. When set to true, disables ALSA's automatic format conversion by including the SND_PCM_NO_AUTO_FORMAT flag. Defaults to false.
-Allocation Callbacks
+ alsa.noAutoChannels
+ ALSA only. When set to true, disables ALSA's automatic channel conversion by including the SND_PCM_NO_AUTO_CHANNELS flag. Defaults to false.
-**************************************************************************************************************************************************************/
-static void* ma__malloc_default(size_t sz, void* pUserData)
-{
- (void)pUserData;
- return MA_MALLOC(sz);
-}
+ alsa.noAutoResample
+ ALSA only. When set to true, disables ALSA's automatic resampling by including the SND_PCM_NO_AUTO_RESAMPLE flag. Defaults to false.
-static void* ma__realloc_default(void* p, size_t sz, void* pUserData)
-{
- (void)pUserData;
- return MA_REALLOC(p, sz);
-}
+ pulse.pStreamNamePlayback
+ PulseAudio only. Sets the stream name for playback.
-static void ma__free_default(void* p, void* pUserData)
-{
- (void)pUserData;
- MA_FREE(p);
-}
+ pulse.pStreamNameCapture
+ PulseAudio only. Sets the stream name for capture.
+ coreaudio.allowNominalSampleRateChange
+ Core Audio only. Desktop only. When enabled, allows the sample rate of the device to be changed at the operating system level. This
+ is disabled by default in order to prevent intrusive changes to the user's system. This is useful if you want to use a sample rate
+ that is known to be natively supported by the hardware thereby avoiding the cost of resampling. When set to true, miniaudio will
+ find the closest match between the sample rate requested in the device config and the sample rates natively supported by the
+ hardware. When set to false, the sample rate currently set by the operating system will always be used.
-static void* ma__malloc_from_callbacks(size_t sz, const ma_allocation_callbacks* pAllocationCallbacks)
-{
- if (pAllocationCallbacks == NULL) {
- return NULL;
- }
+ opensl.streamType
+ OpenSL only. Explicitly sets the stream type. If left unset (`ma_opensl_stream_type_default`), the
+ stream type will be left unset. Think of this as the type of audio you're playing.
- if (pAllocationCallbacks->onMalloc != NULL) {
- return pAllocationCallbacks->onMalloc(sz, pAllocationCallbacks->pUserData);
- }
+ opensl.recordingPreset
+ OpenSL only. Explicitly sets the type of recording your program will be doing. When left
+ unset, the recording preset will be left unchanged.
- /* Try using realloc(). */
- if (pAllocationCallbacks->onRealloc != NULL) {
- return pAllocationCallbacks->onRealloc(NULL, sz, pAllocationCallbacks->pUserData);
- }
+ aaudio.usage
+ AAudio only. Explicitly sets the nature of the audio the program will be consuming. When
+ left unset, the usage will be left unchanged.
- return NULL;
-}
+ aaudio.contentType
+ AAudio only. Sets the content type. When left unset, the content type will be left unchanged.
-static void* ma__realloc_from_callbacks(void* p, size_t szNew, size_t szOld, const ma_allocation_callbacks* pAllocationCallbacks)
-{
- if (pAllocationCallbacks == NULL) {
- return NULL;
- }
+ aaudio.inputPreset
+ AAudio only. Explicitly sets the type of recording your program will be doing. When left
+ unset, the input preset will be left unchanged.
- if (pAllocationCallbacks->onRealloc != NULL) {
- return pAllocationCallbacks->onRealloc(p, szNew, pAllocationCallbacks->pUserData);
- }
+ aaudio.noAutoStartAfterReroute
+ AAudio only. Controls whether or not the device should be automatically restarted after a
+ stream reroute. When set to false (default) the device will be restarted automatically;
+ otherwise the device will be stopped.
- /* Try emulating realloc() in terms of malloc()/free(). */
- if (pAllocationCallbacks->onMalloc != NULL && pAllocationCallbacks->onFree != NULL) {
- void* p2;
- p2 = pAllocationCallbacks->onMalloc(szNew, pAllocationCallbacks->pUserData);
- if (p2 == NULL) {
- return NULL;
- }
+Once initialized, the device's config is immutable. If you need to change the config you will need to initialize a new device.
- if (p != NULL) {
- MA_COPY_MEMORY(p2, p, szOld);
- pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData);
- }
+After initializing the device it will be in a stopped state. To start it, use `ma_device_start()`.
- return p2;
- }
+If both `periodSizeInFrames` and `periodSizeInMilliseconds` are set to zero, it will default to `MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY` or
+`MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE`, depending on whether or not `performanceProfile` is set to `ma_performance_profile_low_latency` or
+`ma_performance_profile_conservative`.
- return NULL;
-}
+If you request exclusive mode and the backend does not support it an error will be returned. For robustness, you may want to first try initializing the device
+in exclusive mode, and then fall back to shared mode if required. Alternatively you can just request shared mode (the default if you leave it unset in the
+config) which is the most reliable option. Some backends do not have a practical way of choosing whether or not the device should be exclusive or not (ALSA,
+for example) in which case it just acts as a hint. Unless you have special requirements you should try avoiding exclusive mode as it's intrusive to the user.
+Starting with Windows 10, miniaudio will use low-latency shared mode where possible which may make exclusive mode unnecessary.
-static MA_INLINE void* ma__calloc_from_callbacks(size_t sz, const ma_allocation_callbacks* pAllocationCallbacks)
-{
- void* p = ma__malloc_from_callbacks(sz, pAllocationCallbacks);
- if (p != NULL) {
- MA_ZERO_MEMORY(p, sz);
- }
+When sending or receiving data to/from a device, miniaudio will internally perform a format conversion to convert between the format specified by the config
+and the format used internally by the backend. If you pass in 0 for the sample format, channel count, sample rate _and_ channel map, data transmission will run
+on an optimized pass-through fast path. You can retrieve the format, channel count and sample rate by inspecting the `playback/capture.format`,
+`playback/capture.channels` and `sampleRate` members of the device object.
- return p;
-}
+When compiling for UWP you must ensure you call this function on the main UI thread because the operating system may need to present the user with a message
+asking for permissions. Please refer to the official documentation for ActivateAudioInterfaceAsync() for more information.
-static void ma__free_from_callbacks(void* p, const ma_allocation_callbacks* pAllocationCallbacks)
-{
- if (p == NULL || pAllocationCallbacks == NULL) {
- return;
- }
+ALSA Specific: When initializing the default device, requesting shared mode will try using the "dmix" device for playback and the "dsnoop" device for capture.
+If these fail it will try falling back to the "hw" device.
- if (pAllocationCallbacks->onFree != NULL) {
- pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData);
- }
-}
-static ma_allocation_callbacks ma_allocation_callbacks_init_default(void)
-{
- ma_allocation_callbacks callbacks;
- callbacks.pUserData = NULL;
- callbacks.onMalloc = ma__malloc_default;
- callbacks.onRealloc = ma__realloc_default;
- callbacks.onFree = ma__free_default;
+Example 1 - Simple Initialization
+---------------------------------
+This example shows how to initialize a simple playback device using a standard configuration. If you are just needing to do simple playback from the default
+playback device this is usually all you need.
- return callbacks;
+```c
+ma_device_config config = ma_device_config_init(ma_device_type_playback);
+config.playback.format = ma_format_f32;
+config.playback.channels = 2;
+config.sampleRate = 48000;
+config.dataCallback = ma_data_callback;
+config.pMyUserData = pMyUserData;
+
+ma_device device;
+ma_result result = ma_device_init(NULL, &config, &device);
+if (result != MA_SUCCESS) {
+ // Error
}
+```
-static ma_result ma_allocation_callbacks_init_copy(ma_allocation_callbacks* pDst, const ma_allocation_callbacks* pSrc)
-{
- if (pDst == NULL) {
- return MA_INVALID_ARGS;
- }
- if (pSrc == NULL) {
- *pDst = ma_allocation_callbacks_init_default();
- } else {
- if (pSrc->pUserData == NULL && pSrc->onFree == NULL && pSrc->onMalloc == NULL && pSrc->onRealloc == NULL) {
- *pDst = ma_allocation_callbacks_init_default();
- } else {
- if (pSrc->onFree == NULL || (pSrc->onMalloc == NULL && pSrc->onRealloc == NULL)) {
- return MA_INVALID_ARGS; /* Invalid allocation callbacks. */
- } else {
- *pDst = *pSrc;
- }
- }
- }
+Example 2 - Advanced Initialization
+-----------------------------------
+This example shows how you might do some more advanced initialization. In this hypothetical example we want to control the latency by setting the buffer size
+and period count. We also want to allow the user to be able to choose which device to output from which means we need a context so we can perform device
+enumeration.
- return MA_SUCCESS;
+```c
+ma_context context;
+ma_result result = ma_context_init(NULL, 0, NULL, &context);
+if (result != MA_SUCCESS) {
+ // Error
}
+ma_device_info* pPlaybackDeviceInfos;
+ma_uint32 playbackDeviceCount;
+result = ma_context_get_devices(&context, &pPlaybackDeviceInfos, &playbackDeviceCount, NULL, NULL);
+if (result != MA_SUCCESS) {
+ // Error
+}
+// ... choose a device from pPlaybackDeviceInfos ...
+ma_device_config config = ma_device_config_init(ma_device_type_playback);
+config.playback.pDeviceID = pMyChosenDeviceID; // <-- Get this from the `id` member of one of the `ma_device_info` objects returned by ma_context_get_devices().
+config.playback.format = ma_format_f32;
+config.playback.channels = 2;
+config.sampleRate = 48000;
+config.dataCallback = ma_data_callback;
+config.pUserData = pMyUserData;
+config.periodSizeInMilliseconds = 10;
+config.periods = 3;
-/**************************************************************************************************************************************************************
-
-Logging
-
-**************************************************************************************************************************************************************/
-MA_API const char* ma_log_level_to_string(ma_uint32 logLevel)
-{
- switch (logLevel)
- {
- case MA_LOG_LEVEL_DEBUG: return "DEBUG";
- case MA_LOG_LEVEL_INFO: return "INFO";
- case MA_LOG_LEVEL_WARNING: return "WARNING";
- case MA_LOG_LEVEL_ERROR: return "ERROR";
- default: return "ERROR";
- }
+ma_device device;
+result = ma_device_init(&context, &config, &device);
+if (result != MA_SUCCESS) {
+ // Error
}
+```
-#if defined(MA_DEBUG_OUTPUT)
-/* Customize this to use a specific tag in __android_log_print() for debug output messages. */
-#ifndef MA_ANDROID_LOG_TAG
-#define MA_ANDROID_LOG_TAG "miniaudio"
-#endif
+See Also
+--------
+ma_device_config_init()
+ma_device_uninit()
+ma_device_start()
+ma_context_init()
+ma_context_get_devices()
+ma_context_enumerate_devices()
+*/
+MA_API ma_result ma_device_init(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice);
-void ma_log_callback_debug(void* pUserData, ma_uint32 level, const char* pMessage)
-{
- (void)pUserData;
+/*
+Initializes a device without a context, with extra parameters for controlling the configuration of the internal self-managed context.
- /* Special handling for some platforms. */
- #if defined(MA_ANDROID)
- {
- /* Android. */
- __android_log_print(ANDROID_LOG_DEBUG, MA_ANDROID_LOG_TAG, "%s: %s", ma_log_level_to_string(level), pMessage);
- }
- #else
- {
- /* Everything else. */
- printf("%s: %s", ma_log_level_to_string(level), pMessage);
- }
- #endif
-}
-#endif
+This is the same as `ma_device_init()`, only instead of a context being passed in, the parameters from `ma_context_init()` are passed in instead. This function
+allows you to configure the internally created context.
-MA_API ma_log_callback ma_log_callback_init(ma_log_callback_proc onLog, void* pUserData)
-{
- ma_log_callback callback;
- MA_ZERO_OBJECT(&callback);
- callback.onLog = onLog;
- callback.pUserData = pUserData;
+Parameters
+----------
+backends (in, optional)
+ A list of backends to try initializing, in priority order. Can be NULL, in which case it uses default priority order.
- return callback;
-}
+backendCount (in, optional)
+ The number of items in `backend`. Ignored if `backend` is NULL.
+pContextConfig (in, optional)
+ The context configuration.
-MA_API ma_result ma_log_init(const ma_allocation_callbacks* pAllocationCallbacks, ma_log* pLog)
-{
- if (pLog == NULL) {
- return MA_INVALID_ARGS;
- }
+pConfig (in)
+ A pointer to the device configuration. Cannot be null. See remarks for details.
- MA_ZERO_OBJECT(pLog);
- ma_allocation_callbacks_init_copy(&pLog->allocationCallbacks, pAllocationCallbacks);
+pDevice (out)
+ A pointer to the device object being initialized.
- /* We need a mutex for thread safety. */
- #ifndef MA_NO_THREADING
- {
- ma_result result = ma_mutex_init(&pLog->lock);
- if (result != MA_SUCCESS) {
- return result;
- }
- }
- #endif
-
- /* If we're using debug output, enable it. */
- #if defined(MA_DEBUG_OUTPUT)
- {
- ma_log_register_callback(pLog, ma_log_callback_init(ma_log_callback_debug, NULL)); /* Doesn't really matter if this fails. */
- }
- #endif
- return MA_SUCCESS;
-}
+Return Value
+------------
+MA_SUCCESS if successful; any other error code otherwise.
-MA_API void ma_log_uninit(ma_log* pLog)
-{
- if (pLog == NULL) {
- return;
- }
-#ifndef MA_NO_THREADING
- ma_mutex_uninit(&pLog->lock);
-#endif
-}
+Thread Safety
+-------------
+Unsafe. It is not safe to call this function simultaneously for different devices because some backends depend on and mutate global state. The same applies to
+calling this at the same time as `ma_device_uninit()`.
-static void ma_log_lock(ma_log* pLog)
-{
-#ifndef MA_NO_THREADING
- ma_mutex_lock(&pLog->lock);
-#else
- (void)pLog;
-#endif
-}
-static void ma_log_unlock(ma_log* pLog)
-{
-#ifndef MA_NO_THREADING
- ma_mutex_unlock(&pLog->lock);
-#else
- (void)pLog;
-#endif
-}
+Callback Safety
+---------------
+Unsafe. It is not safe to call this inside any callback.
-MA_API ma_result ma_log_register_callback(ma_log* pLog, ma_log_callback callback)
-{
- ma_result result = MA_SUCCESS;
- if (pLog == NULL || callback.onLog == NULL) {
- return MA_INVALID_ARGS;
- }
+Remarks
+-------
+You only need to use this function if you want to configure the context differently to it's defaults. You should never use this function if you want to manage
+your own context.
- ma_log_lock(pLog);
- {
- if (pLog->callbackCount == ma_countof(pLog->callbacks)) {
- result = MA_OUT_OF_MEMORY; /* Reached the maximum allowed log callbacks. */
- } else {
- pLog->callbacks[pLog->callbackCount] = callback;
- pLog->callbackCount += 1;
- }
- }
- ma_log_unlock(pLog);
+See the documentation for `ma_context_init()` for information on the different context configuration options.
- return result;
-}
-MA_API ma_result ma_log_unregister_callback(ma_log* pLog, ma_log_callback callback)
-{
- if (pLog == NULL) {
- return MA_INVALID_ARGS;
- }
+See Also
+--------
+ma_device_init()
+ma_device_uninit()
+ma_device_config_init()
+ma_context_init()
+*/
+MA_API ma_result ma_device_init_ex(const ma_backend backends[], ma_uint32 backendCount, const ma_context_config* pContextConfig, const ma_device_config* pConfig, ma_device* pDevice);
- ma_log_lock(pLog);
- {
- ma_uint32 iLog;
- for (iLog = 0; iLog < pLog->callbackCount; ) {
- if (pLog->callbacks[iLog].onLog == callback.onLog) {
- /* Found. Move everything down a slot. */
- ma_uint32 jLog;
- for (jLog = iLog; jLog < pLog->callbackCount-1; jLog += 1) {
- pLog->callbacks[jLog] = pLog->callbacks[jLog + 1];
- }
+/*
+Uninitializes a device.
- pLog->callbackCount -= 1;
- } else {
- /* Not found. */
- iLog += 1;
- }
- }
- }
- ma_log_unlock(pLog);
+This will explicitly stop the device. You do not need to call `ma_device_stop()` beforehand, but it's harmless if you do.
- return MA_SUCCESS;
-}
-MA_API ma_result ma_log_post(ma_log* pLog, ma_uint32 level, const char* pMessage)
-{
- if (pLog == NULL || pMessage == NULL) {
- return MA_INVALID_ARGS;
- }
+Parameters
+----------
+pDevice (in)
+ A pointer to the device to stop.
- /* If it's a debug log, ignore it unless MA_DEBUG_OUTPUT is enabled. */
- #if !defined(MA_DEBUG_OUTPUT)
- {
- if (level == MA_LOG_LEVEL_DEBUG) {
- return MA_INVALID_ARGS; /* Don't post debug messages if debug output is disabled. */
- }
- }
- #endif
- ma_log_lock(pLog);
- {
- ma_uint32 iLog;
- for (iLog = 0; iLog < pLog->callbackCount; iLog += 1) {
- if (pLog->callbacks[iLog].onLog) {
- pLog->callbacks[iLog].onLog(pLog->callbacks[iLog].pUserData, level, pMessage);
- }
- }
- }
- ma_log_unlock(pLog);
+Return Value
+------------
+Nothing
- return MA_SUCCESS;
-}
+
+Thread Safety
+-------------
+Unsafe. As soon as this API is called the device should be considered undefined.
+
+
+Callback Safety
+---------------
+Unsafe. It is not safe to call this inside any callback. Doing this will result in a deadlock.
+
+
+See Also
+--------
+ma_device_init()
+ma_device_stop()
+*/
+MA_API void ma_device_uninit(ma_device* pDevice);
/*
-We need to emulate _vscprintf() for the VC6 build. This can be more efficient, but since it's only VC6, and it's just a
-logging function, I'm happy to keep this simple. In the VC6 build we can implement this in terms of _vsnprintf().
+Retrieves a pointer to the context that owns the given device.
*/
-#if defined(_MSC_VER) && _MSC_VER < 1900
-static int ma_vscprintf(const ma_allocation_callbacks* pAllocationCallbacks, const char* format, va_list args)
-{
-#if _MSC_VER > 1200
- return _vscprintf(format, args);
-#else
- int result;
- char* pTempBuffer = NULL;
- size_t tempBufferCap = 1024;
+MA_API ma_context* ma_device_get_context(ma_device* pDevice);
- if (format == NULL) {
- errno = EINVAL;
- return -1;
- }
+/*
+Helper function for retrieving the log object associated with the context that owns this device.
+*/
+MA_API ma_log* ma_device_get_log(ma_device* pDevice);
- for (;;) {
- char* pNewTempBuffer = (char*)ma_realloc(pTempBuffer, tempBufferCap, pAllocationCallbacks);
- if (pNewTempBuffer == NULL) {
- ma_free(pTempBuffer, pAllocationCallbacks);
- errno = ENOMEM;
- return -1; /* Out of memory. */
- }
- pTempBuffer = pNewTempBuffer;
+/*
+Retrieves information about the device.
- result = _vsnprintf(pTempBuffer, tempBufferCap, format, args);
- ma_free(pTempBuffer, NULL);
- if (result != -1) {
- break; /* Got it. */
- }
+Parameters
+----------
+pDevice (in)
+ A pointer to the device whose information is being retrieved.
- /* Buffer wasn't big enough. Ideally it'd be nice to use an error code to know the reason for sure, but this is reliable enough. */
- tempBufferCap *= 2;
- }
+type (in)
+ The device type. This parameter is required for duplex devices. When retrieving device
+ information, you are doing so for an individual playback or capture device.
- return result;
-#endif
-}
-#endif
+pDeviceInfo (out)
+ A pointer to the `ma_device_info` that will receive the device information.
-MA_API ma_result ma_log_postv(ma_log* pLog, ma_uint32 level, const char* pFormat, va_list args)
-{
- if (pLog == NULL || pFormat == NULL) {
- return MA_INVALID_ARGS;
- }
- /*
- If it's a debug log, ignore it unless MA_DEBUG_OUTPUT is enabled. Do this before generating the
- formatted message string so that we don't waste time only to have ma_log_post() reject it.
- */
- #if !defined(MA_DEBUG_OUTPUT)
- {
- if (level == MA_LOG_LEVEL_DEBUG) {
- return MA_INVALID_ARGS; /* Don't post debug messages if debug output is disabled. */
- }
- }
- #endif
+Return Value
+------------
+MA_SUCCESS if successful; any other error code otherwise.
- #if (defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L) || ((!defined(_MSC_VER) || _MSC_VER >= 1900) && !defined(__STRICT_ANSI__) && !defined(_NO_EXT_KEYS))
- {
- ma_result result;
- int length;
- char pFormattedMessageStack[1024];
- char* pFormattedMessageHeap = NULL;
- /* First try formatting into our fixed sized stack allocated buffer. If this is too small we'll fallback to a heap allocation. */
- length = vsnprintf(pFormattedMessageStack, sizeof(pFormattedMessageStack), pFormat, args);
- if (length < 0) {
- return MA_INVALID_OPERATION; /* An error occured when trying to convert the buffer. */
- }
+Thread Safety
+-------------
+Unsafe. This should be considered unsafe because it may be calling into the backend which may or
+may not be safe.
- if ((size_t)length < sizeof(pFormattedMessageStack)) {
- /* The string was written to the stack. */
- result = ma_log_post(pLog, level, pFormattedMessageStack);
- } else {
- /* The stack buffer was too small, try the heap. */
- pFormattedMessageHeap = (char*)ma_malloc(length + 1, &pLog->allocationCallbacks);
- if (pFormattedMessageHeap == NULL) {
- return MA_OUT_OF_MEMORY;
- }
- length = vsnprintf(pFormattedMessageHeap, length + 1, pFormat, args);
- if (length < 0) {
- ma_free(pFormattedMessageHeap, &pLog->allocationCallbacks);
- return MA_INVALID_OPERATION;
- }
+Callback Safety
+---------------
+Unsafe. You should avoid calling this in the data callback because it may call into the backend
+which may or may not be safe.
+*/
+MA_API ma_result ma_device_get_info(ma_device* pDevice, ma_device_type type, ma_device_info* pDeviceInfo);
- result = ma_log_post(pLog, level, pFormattedMessageHeap);
- ma_free(pFormattedMessageHeap, &pLog->allocationCallbacks);
- }
- return result;
- }
- #else
- {
- /*
- Without snprintf() we need to first measure the string and then heap allocate it. I'm only aware of Visual Studio having support for this without snprintf(), so we'll
- need to restrict this branch to Visual Studio. For other compilers we need to just not support formatted logging because I don't want the security risk of overflowing
- a fixed sized stack allocated buffer.
- */
- #if defined(_MSC_VER) && _MSC_VER >= 1200 /* 1200 = VC6 */
- {
- ma_result result;
- int formattedLen;
- char* pFormattedMessage = NULL;
- va_list args2;
+/*
+Retrieves the name of the device.
- #if _MSC_VER >= 1800
- {
- va_copy(args2, args);
- }
- #else
- {
- args2 = args;
- }
- #endif
- formattedLen = ma_vscprintf(&pLog->allocationCallbacks, pFormat, args2);
- va_end(args2);
+Parameters
+----------
+pDevice (in)
+ A pointer to the device whose information is being retrieved.
- if (formattedLen <= 0) {
- return MA_INVALID_OPERATION;
- }
+type (in)
+ The device type. This parameter is required for duplex devices. When retrieving device
+ information, you are doing so for an individual playback or capture device.
- pFormattedMessage = (char*)ma_malloc(formattedLen + 1, &pLog->allocationCallbacks);
- if (pFormattedMessage == NULL) {
- return MA_OUT_OF_MEMORY;
- }
+pName (out)
+ A pointer to the buffer that will receive the name.
- /* We'll get errors on newer versions of Visual Studio if we try to use vsprintf(). */
- #if _MSC_VER >= 1400 /* 1400 = Visual Studio 2005 */
- {
- vsprintf_s(pFormattedMessage, formattedLen + 1, pFormat, args);
- }
- #else
- {
- vsprintf(pFormattedMessage, pFormat, args);
- }
- #endif
+nameCap (in)
+ The capacity of the output buffer, including space for the null terminator.
- result = ma_log_post(pLog, level, pFormattedMessage);
- ma_free(pFormattedMessage, &pLog->allocationCallbacks);
+pLengthNotIncludingNullTerminator (out, optional)
+ A pointer to the variable that will receive the length of the name, not including the null
+ terminator.
- return result;
- }
- #else
- {
- /* Can't do anything because we don't have a safe way of to emulate vsnprintf() without a manual solution. */
- (void)level;
- (void)args;
- return MA_INVALID_OPERATION;
- }
- #endif
- }
- #endif
-}
+Return Value
+------------
+MA_SUCCESS if successful; any other error code otherwise.
-MA_API ma_result ma_log_postf(ma_log* pLog, ma_uint32 level, const char* pFormat, ...)
-{
- ma_result result;
- va_list args;
- if (pLog == NULL || pFormat == NULL) {
- return MA_INVALID_ARGS;
- }
+Thread Safety
+-------------
+Unsafe. This should be considered unsafe because it may be calling into the backend which may or
+may not be safe.
- /*
- If it's a debug log, ignore it unless MA_DEBUG_OUTPUT is enabled. Do this before generating the
- formatted message string so that we don't waste time only to have ma_log_post() reject it.
- */
- #if !defined(MA_DEBUG_OUTPUT)
- {
- if (level == MA_LOG_LEVEL_DEBUG) {
- return MA_INVALID_ARGS; /* Don't post debug messages if debug output is disabled. */
- }
- }
- #endif
- va_start(args, pFormat);
- {
- result = ma_log_postv(pLog, level, pFormat, args);
- }
- va_end(args);
+Callback Safety
+---------------
+Unsafe. You should avoid calling this in the data callback because it may call into the backend
+which may or may not be safe.
- return result;
-}
+Remarks
+-------
+If the name does not fully fit into the output buffer, it'll be truncated. You can pass in NULL to
+`pName` if you want to first get the length of the name for the purpose of memory allocation of the
+output buffer. Allocating a buffer of size `MA_MAX_DEVICE_NAME_LENGTH + 1` should be enough for
+most cases and will avoid the need for the inefficiency of calling this function twice.
+This is implemented in terms of `ma_device_get_info()`.
+*/
+MA_API ma_result ma_device_get_name(ma_device* pDevice, ma_device_type type, char* pName, size_t nameCap, size_t* pLengthNotIncludingNullTerminator);
-/* Clamps an f32 sample to -1..1 */
-static MA_INLINE float ma_clip_f32(float x)
-{
- if (x < -1) return -1;
- if (x > +1) return +1;
- return x;
-}
+/*
+Starts the device. For playback devices this begins playback. For capture devices it begins recording.
-static MA_INLINE float ma_mix_f32(float x, float y, float a)
-{
- return x*(1-a) + y*a;
-}
-static MA_INLINE float ma_mix_f32_fast(float x, float y, float a)
-{
- float r0 = (y - x);
- float r1 = r0*a;
- return x + r1;
- /*return x + (y - x)*a;*/
-}
+Use `ma_device_stop()` to stop the device.
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE __m128 ma_mix_f32_fast__sse2(__m128 x, __m128 y, __m128 a)
-{
- return _mm_add_ps(x, _mm_mul_ps(_mm_sub_ps(y, x), a));
-}
-#endif
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE __m256 ma_mix_f32_fast__avx2(__m256 x, __m256 y, __m256 a)
-{
- return _mm256_add_ps(x, _mm256_mul_ps(_mm256_sub_ps(y, x), a));
-}
-#endif
-#if defined(MA_SUPPORT_AVX512)
-static MA_INLINE __m512 ma_mix_f32_fast__avx512(__m512 x, __m512 y, __m512 a)
-{
- return _mm512_add_ps(x, _mm512_mul_ps(_mm512_sub_ps(y, x), a));
-}
-#endif
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE float32x4_t ma_mix_f32_fast__neon(float32x4_t x, float32x4_t y, float32x4_t a)
-{
- return vaddq_f32(x, vmulq_f32(vsubq_f32(y, x), a));
-}
-#endif
+Parameters
+----------
+pDevice (in)
+ A pointer to the device to start.
-static MA_INLINE double ma_mix_f64(double x, double y, double a)
-{
- return x*(1-a) + y*a;
-}
-static MA_INLINE double ma_mix_f64_fast(double x, double y, double a)
-{
- return x + (y - x)*a;
-}
-static MA_INLINE float ma_scale_to_range_f32(float x, float lo, float hi)
-{
- return lo + x*(hi-lo);
-}
+Return Value
+------------
+MA_SUCCESS if successful; any other error code otherwise.
-/*
-Greatest common factor using Euclid's algorithm iteratively.
-*/
-static MA_INLINE ma_uint32 ma_gcf_u32(ma_uint32 a, ma_uint32 b)
-{
- for (;;) {
- if (b == 0) {
- break;
- } else {
- ma_uint32 t = a;
- a = b;
- b = t % a;
- }
- }
+Thread Safety
+-------------
+Safe. It's safe to call this from any thread with the exception of the callback thread.
- return a;
-}
+Callback Safety
+---------------
+Unsafe. It is not safe to call this inside any callback.
-/*
-Random Number Generation
-miniaudio uses the LCG random number generation algorithm. This is good enough for audio.
+Remarks
+-------
+For a playback device, this will retrieve an initial chunk of audio data from the client before returning. The reason for this is to ensure there is valid
+audio data in the buffer, which needs to be done before the device begins playback.
-Note that miniaudio's global LCG implementation uses global state which is _not_ thread-local. When this is called across
-multiple threads, results will be unpredictable. However, it won't crash and results will still be random enough for
-miniaudio's purposes.
+This API waits until the backend device has been started for real by the worker thread. It also waits on a mutex for thread-safety.
+
+Do not call this in any callback.
+
+
+See Also
+--------
+ma_device_stop()
*/
-#ifndef MA_DEFAULT_LCG_SEED
-#define MA_DEFAULT_LCG_SEED 4321
-#endif
+MA_API ma_result ma_device_start(ma_device* pDevice);
-#define MA_LCG_M 2147483647
-#define MA_LCG_A 48271
-#define MA_LCG_C 0
+/*
+Stops the device. For playback devices this stops playback. For capture devices it stops recording.
-static ma_lcg g_maLCG = {MA_DEFAULT_LCG_SEED}; /* Non-zero initial seed. Use ma_seed() to use an explicit seed. */
+Use `ma_device_start()` to start the device again.
-static MA_INLINE void ma_lcg_seed(ma_lcg* pLCG, ma_int32 seed)
-{
- MA_ASSERT(pLCG != NULL);
- pLCG->state = seed;
-}
-static MA_INLINE ma_int32 ma_lcg_rand_s32(ma_lcg* pLCG)
-{
- pLCG->state = (MA_LCG_A * pLCG->state + MA_LCG_C) % MA_LCG_M;
- return pLCG->state;
-}
+Parameters
+----------
+pDevice (in)
+ A pointer to the device to stop.
-static MA_INLINE ma_uint32 ma_lcg_rand_u32(ma_lcg* pLCG)
-{
- return (ma_uint32)ma_lcg_rand_s32(pLCG);
-}
-static MA_INLINE ma_int16 ma_lcg_rand_s16(ma_lcg* pLCG)
-{
- return (ma_int16)(ma_lcg_rand_s32(pLCG) & 0xFFFF);
-}
+Return Value
+------------
+MA_SUCCESS if successful; any other error code otherwise.
-static MA_INLINE double ma_lcg_rand_f64(ma_lcg* pLCG)
-{
- return ma_lcg_rand_s32(pLCG) / (double)0x7FFFFFFF;
-}
-static MA_INLINE float ma_lcg_rand_f32(ma_lcg* pLCG)
-{
- return (float)ma_lcg_rand_f64(pLCG);
-}
+Thread Safety
+-------------
+Safe. It's safe to call this from any thread with the exception of the callback thread.
-static MA_INLINE float ma_lcg_rand_range_f32(ma_lcg* pLCG, float lo, float hi)
-{
- return ma_scale_to_range_f32(ma_lcg_rand_f32(pLCG), lo, hi);
-}
-static MA_INLINE ma_int32 ma_lcg_rand_range_s32(ma_lcg* pLCG, ma_int32 lo, ma_int32 hi)
-{
- if (lo == hi) {
- return lo;
- }
+Callback Safety
+---------------
+Unsafe. It is not safe to call this inside any callback. Doing this will result in a deadlock.
- return lo + ma_lcg_rand_u32(pLCG) / (0xFFFFFFFF / (hi - lo + 1) + 1);
-}
+Remarks
+-------
+This API needs to wait on the worker thread to stop the backend device properly before returning. It also waits on a mutex for thread-safety. In addition, some
+backends need to wait for the device to finish playback/recording of the current fragment which can take some time (usually proportionate to the buffer size
+that was specified at initialization time).
+Backends are required to either pause the stream in-place or drain the buffer if pausing is not possible. The reason for this is that stopping the device and
+the resuming it with ma_device_start() (which you might do when your program loses focus) may result in a situation where those samples are never output to the
+speakers or received from the microphone which can in turn result in de-syncs.
-static MA_INLINE void ma_seed(ma_int32 seed)
-{
- ma_lcg_seed(&g_maLCG, seed);
-}
+Do not call this in any callback.
-static MA_INLINE ma_int32 ma_rand_s32(void)
-{
- return ma_lcg_rand_s32(&g_maLCG);
-}
+This will be called implicitly by `ma_device_uninit()`.
-static MA_INLINE ma_uint32 ma_rand_u32(void)
-{
- return ma_lcg_rand_u32(&g_maLCG);
-}
-static MA_INLINE double ma_rand_f64(void)
-{
- return ma_lcg_rand_f64(&g_maLCG);
-}
+See Also
+--------
+ma_device_start()
+*/
+MA_API ma_result ma_device_stop(ma_device* pDevice);
-static MA_INLINE float ma_rand_f32(void)
-{
- return ma_lcg_rand_f32(&g_maLCG);
-}
+/*
+Determines whether or not the device is started.
-static MA_INLINE float ma_rand_range_f32(float lo, float hi)
-{
- return ma_lcg_rand_range_f32(&g_maLCG, lo, hi);
-}
-static MA_INLINE ma_int32 ma_rand_range_s32(ma_int32 lo, ma_int32 hi)
-{
- return ma_lcg_rand_range_s32(&g_maLCG, lo, hi);
-}
+Parameters
+----------
+pDevice (in)
+ A pointer to the device whose start state is being retrieved.
-static MA_INLINE float ma_dither_f32_rectangle(float ditherMin, float ditherMax)
-{
- return ma_rand_range_f32(ditherMin, ditherMax);
-}
+Return Value
+------------
+True if the device is started, false otherwise.
-static MA_INLINE float ma_dither_f32_triangle(float ditherMin, float ditherMax)
-{
- float a = ma_rand_range_f32(ditherMin, 0);
- float b = ma_rand_range_f32(0, ditherMax);
- return a + b;
-}
-static MA_INLINE float ma_dither_f32(ma_dither_mode ditherMode, float ditherMin, float ditherMax)
-{
- if (ditherMode == ma_dither_mode_rectangle) {
- return ma_dither_f32_rectangle(ditherMin, ditherMax);
- }
- if (ditherMode == ma_dither_mode_triangle) {
- return ma_dither_f32_triangle(ditherMin, ditherMax);
- }
+Thread Safety
+-------------
+Safe. If another thread calls `ma_device_start()` or `ma_device_stop()` at this same time as this function is called, there's a very small chance the return
+value will be out of sync.
- return 0;
-}
-static MA_INLINE ma_int32 ma_dither_s32(ma_dither_mode ditherMode, ma_int32 ditherMin, ma_int32 ditherMax)
-{
- if (ditherMode == ma_dither_mode_rectangle) {
- ma_int32 a = ma_rand_range_s32(ditherMin, ditherMax);
- return a;
- }
- if (ditherMode == ma_dither_mode_triangle) {
- ma_int32 a = ma_rand_range_s32(ditherMin, 0);
- ma_int32 b = ma_rand_range_s32(0, ditherMax);
- return a + b;
- }
+Callback Safety
+---------------
+Safe. This is implemented as a simple accessor.
- return 0;
-}
+See Also
+--------
+ma_device_start()
+ma_device_stop()
+*/
+MA_API ma_bool32 ma_device_is_started(const ma_device* pDevice);
-/**************************************************************************************************************************************************************
-Atomics
+/*
+Retrieves the state of the device.
-**************************************************************************************************************************************************************/
-/* c89atomic.h begin */
-#ifndef c89atomic_h
-#define c89atomic_h
-#if defined(__cplusplus)
-extern "C" {
-#endif
-typedef signed char c89atomic_int8;
-typedef unsigned char c89atomic_uint8;
-typedef signed short c89atomic_int16;
-typedef unsigned short c89atomic_uint16;
-typedef signed int c89atomic_int32;
-typedef unsigned int c89atomic_uint32;
-#if defined(_MSC_VER)
- typedef signed __int64 c89atomic_int64;
- typedef unsigned __int64 c89atomic_uint64;
-#else
- #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
- #pragma GCC diagnostic push
- #pragma GCC diagnostic ignored "-Wlong-long"
- #if defined(__clang__)
- #pragma GCC diagnostic ignored "-Wc++11-long-long"
- #endif
- #endif
- typedef signed long long c89atomic_int64;
- typedef unsigned long long c89atomic_uint64;
- #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
- #pragma GCC diagnostic pop
- #endif
-#endif
-typedef int c89atomic_memory_order;
-typedef unsigned char c89atomic_bool;
+
+Parameters
+----------
+pDevice (in)
+ A pointer to the device whose state is being retrieved.
+
+
+Return Value
+------------
+The current state of the device. The return value will be one of the following:
+
+ +-------------------------------+------------------------------------------------------------------------------+
+ | ma_device_state_uninitialized | Will only be returned if the device is in the middle of initialization. |
+ +-------------------------------+------------------------------------------------------------------------------+
+ | ma_device_state_stopped | The device is stopped. The initial state of the device after initialization. |
+ +-------------------------------+------------------------------------------------------------------------------+
+ | ma_device_state_started | The device started and requesting and/or delivering audio data. |
+ +-------------------------------+------------------------------------------------------------------------------+
+ | ma_device_state_starting | The device is in the process of starting. |
+ +-------------------------------+------------------------------------------------------------------------------+
+ | ma_device_state_stopping | The device is in the process of stopping. |
+ +-------------------------------+------------------------------------------------------------------------------+
+
+
+Thread Safety
+-------------
+Safe. This is implemented as a simple accessor. Note that if the device is started or stopped at the same time as this function is called,
+there's a possibility the return value could be out of sync. See remarks.
+
+
+Callback Safety
+---------------
+Safe. This is implemented as a simple accessor.
+
+
+Remarks
+-------
+The general flow of a devices state goes like this:
+
+ ```
+ ma_device_init() -> ma_device_state_uninitialized -> ma_device_state_stopped
+ ma_device_start() -> ma_device_state_starting -> ma_device_state_started
+ ma_device_stop() -> ma_device_state_stopping -> ma_device_state_stopped
+ ```
+
+When the state of the device is changed with `ma_device_start()` or `ma_device_stop()` at this same time as this function is called, the
+value returned by this function could potentially be out of sync. If this is significant to your program you need to implement your own
+synchronization.
+*/
+MA_API ma_device_state ma_device_get_state(const ma_device* pDevice);
+
+
+/*
+Performs post backend initialization routines for setting up internal data conversion.
+
+This should be called whenever the backend is initialized. The only time this should be called from
+outside of miniaudio is if you're implementing a custom backend, and you would only do it if you
+are reinitializing the backend due to rerouting or reinitializing for some reason.
+
+
+Parameters
+----------
+pDevice [in]
+ A pointer to the device.
+
+deviceType [in]
+ The type of the device that was just reinitialized.
+
+pPlaybackDescriptor [in]
+ The descriptor of the playback device containing the internal data format and buffer sizes.
+
+pPlaybackDescriptor [in]
+ The descriptor of the capture device containing the internal data format and buffer sizes.
+
+
+Return Value
+------------
+MA_SUCCESS if successful; any other error otherwise.
+
+
+Thread Safety
+-------------
+Unsafe. This will be reinitializing internal data converters which may be in use by another thread.
+
+
+Callback Safety
+---------------
+Unsafe. This will be reinitializing internal data converters which may be in use by the callback.
+
+
+Remarks
+-------
+For a duplex device, you can call this for only one side of the system. This is why the deviceType
+is specified as a parameter rather than deriving it from the device.
+
+You do not need to call this manually unless you are doing a custom backend, in which case you need
+only do it if you're manually performing rerouting or reinitialization.
+*/
+MA_API ma_result ma_device_post_init(ma_device* pDevice, ma_device_type deviceType, const ma_device_descriptor* pPlaybackDescriptor, const ma_device_descriptor* pCaptureDescriptor);
+
+
+/*
+Sets the master volume factor for the device.
+
+The volume factor must be between 0 (silence) and 1 (full volume). Use `ma_device_set_master_volume_db()` to use decibel notation, where 0 is full volume and
+values less than 0 decreases the volume.
+
+
+Parameters
+----------
+pDevice (in)
+ A pointer to the device whose volume is being set.
+
+volume (in)
+ The new volume factor. Must be >= 0.
+
+
+Return Value
+------------
+MA_SUCCESS if the volume was set successfully.
+MA_INVALID_ARGS if pDevice is NULL.
+MA_INVALID_ARGS if volume is negative.
+
+
+Thread Safety
+-------------
+Safe. This just sets a local member of the device object.
+
+
+Callback Safety
+---------------
+Safe. If you set the volume in the data callback, that data written to the output buffer will have the new volume applied.
+
+
+Remarks
+-------
+This applies the volume factor across all channels.
+
+This does not change the operating system's volume. It only affects the volume for the given `ma_device` object's audio stream.
+
+
+See Also
+--------
+ma_device_get_master_volume()
+ma_device_set_master_volume_db()
+ma_device_get_master_volume_db()
+*/
+MA_API ma_result ma_device_set_master_volume(ma_device* pDevice, float volume);
+
+/*
+Retrieves the master volume factor for the device.
+
+
+Parameters
+----------
+pDevice (in)
+ A pointer to the device whose volume factor is being retrieved.
+
+pVolume (in)
+ A pointer to the variable that will receive the volume factor. The returned value will be in the range of [0, 1].
+
+
+Return Value
+------------
+MA_SUCCESS if successful.
+MA_INVALID_ARGS if pDevice is NULL.
+MA_INVALID_ARGS if pVolume is NULL.
+
+
+Thread Safety
+-------------
+Safe. This just a simple member retrieval.
+
+
+Callback Safety
+---------------
+Safe.
+
+
+Remarks
+-------
+If an error occurs, `*pVolume` will be set to 0.
+
+
+See Also
+--------
+ma_device_set_master_volume()
+ma_device_set_master_volume_gain_db()
+ma_device_get_master_volume_gain_db()
+*/
+MA_API ma_result ma_device_get_master_volume(ma_device* pDevice, float* pVolume);
+
+/*
+Sets the master volume for the device as gain in decibels.
+
+A gain of 0 is full volume, whereas a gain of < 0 will decrease the volume.
+
+
+Parameters
+----------
+pDevice (in)
+ A pointer to the device whose gain is being set.
+
+gainDB (in)
+ The new volume as gain in decibels. Must be less than or equal to 0, where 0 is full volume and anything less than 0 decreases the volume.
+
+
+Return Value
+------------
+MA_SUCCESS if the volume was set successfully.
+MA_INVALID_ARGS if pDevice is NULL.
+MA_INVALID_ARGS if the gain is > 0.
+
+
+Thread Safety
+-------------
+Safe. This just sets a local member of the device object.
+
+
+Callback Safety
+---------------
+Safe. If you set the volume in the data callback, that data written to the output buffer will have the new volume applied.
+
+
+Remarks
+-------
+This applies the gain across all channels.
+
+This does not change the operating system's volume. It only affects the volume for the given `ma_device` object's audio stream.
+
+
+See Also
+--------
+ma_device_get_master_volume_gain_db()
+ma_device_set_master_volume()
+ma_device_get_master_volume()
+*/
+MA_API ma_result ma_device_set_master_volume_db(ma_device* pDevice, float gainDB);
+
+/*
+Retrieves the master gain in decibels.
+
+
+Parameters
+----------
+pDevice (in)
+ A pointer to the device whose gain is being retrieved.
+
+pGainDB (in)
+ A pointer to the variable that will receive the gain in decibels. The returned value will be <= 0.
+
+
+Return Value
+------------
+MA_SUCCESS if successful.
+MA_INVALID_ARGS if pDevice is NULL.
+MA_INVALID_ARGS if pGainDB is NULL.
+
+
+Thread Safety
+-------------
+Safe. This just a simple member retrieval.
+
+
+Callback Safety
+---------------
+Safe.
+
+
+Remarks
+-------
+If an error occurs, `*pGainDB` will be set to 0.
+
+
+See Also
+--------
+ma_device_set_master_volume_db()
+ma_device_set_master_volume()
+ma_device_get_master_volume()
+*/
+MA_API ma_result ma_device_get_master_volume_db(ma_device* pDevice, float* pGainDB);
+
+
+/*
+Called from the data callback of asynchronous backends to allow miniaudio to process the data and fire the miniaudio data callback.
+
+
+Parameters
+----------
+pDevice (in)
+ A pointer to device whose processing the data callback.
+
+pOutput (out)
+ A pointer to the buffer that will receive the output PCM frame data. On a playback device this must not be NULL. On a duplex device
+ this can be NULL, in which case pInput must not be NULL.
+
+pInput (in)
+ A pointer to the buffer containing input PCM frame data. On a capture device this must not be NULL. On a duplex device this can be
+ NULL, in which case `pOutput` must not be NULL.
+
+frameCount (in)
+ The number of frames being processed.
+
+
+Return Value
+------------
+MA_SUCCESS if successful; any other result code otherwise.
+
+
+Thread Safety
+-------------
+This function should only ever be called from the internal data callback of the backend. It is safe to call this simultaneously between a
+playback and capture device in duplex setups.
+
+
+Callback Safety
+---------------
+Do not call this from the miniaudio data callback. It should only ever be called from the internal data callback of the backend.
+
+
+Remarks
+-------
+If both `pOutput` and `pInput` are NULL, and error will be returned. In duplex scenarios, both `pOutput` and `pInput` can be non-NULL, in
+which case `pInput` will be processed first, followed by `pOutput`.
+
+If you are implementing a custom backend, and that backend uses a callback for data delivery, you'll need to call this from inside that
+callback.
+*/
+MA_API ma_result ma_device_handle_backend_data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount);
+
+
+/*
+Calculates an appropriate buffer size from a descriptor, native sample rate and performance profile.
+
+This function is used by backends for helping determine an appropriately sized buffer to use with
+the device depending on the values of `periodSizeInFrames` and `periodSizeInMilliseconds` in the
+`pDescriptor` object. Since buffer size calculations based on time depends on the sample rate, a
+best guess at the device's native sample rate is also required which is where `nativeSampleRate`
+comes in. In addition, the performance profile is also needed for cases where both the period size
+in frames and milliseconds are both zero.
+
+
+Parameters
+----------
+pDescriptor (in)
+ A pointer to device descriptor whose `periodSizeInFrames` and `periodSizeInMilliseconds` members
+ will be used for the calculation of the buffer size.
+
+nativeSampleRate (in)
+ The device's native sample rate. This is only ever used when the `periodSizeInFrames` member of
+ `pDescriptor` is zero. In this case, `periodSizeInMilliseconds` will be used instead, in which
+ case a sample rate is required to convert to a size in frames.
+
+performanceProfile (in)
+ When both the `periodSizeInFrames` and `periodSizeInMilliseconds` members of `pDescriptor` are
+ zero, miniaudio will fall back to a buffer size based on the performance profile. The profile
+ to use for this calculation is determine by this parameter.
+
+
+Return Value
+------------
+The calculated buffer size in frames.
+
+
+Thread Safety
+-------------
+This is safe so long as nothing modifies `pDescriptor` at the same time. However, this function
+should only ever be called from within the backend's device initialization routine and therefore
+shouldn't have any multithreading concerns.
+
+
+Callback Safety
+---------------
+This is safe to call within the data callback, but there is no reason to ever do this.
+
+
+Remarks
+-------
+If `nativeSampleRate` is zero, this function will fall back to `pDescriptor->sampleRate`. If that
+is also zero, `MA_DEFAULT_SAMPLE_RATE` will be used instead.
+*/
+MA_API ma_uint32 ma_calculate_buffer_size_in_frames_from_descriptor(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile);
+
+
+
+/*
+Retrieves a friendly name for a backend.
+*/
+MA_API const char* ma_get_backend_name(ma_backend backend);
+
+/*
+Determines whether or not the given backend is available by the compilation environment.
+*/
+MA_API ma_bool32 ma_is_backend_enabled(ma_backend backend);
+
+/*
+Retrieves compile-time enabled backends.
+
+
+Parameters
+----------
+pBackends (out, optional)
+ A pointer to the buffer that will receive the enabled backends. Set to NULL to retrieve the backend count. Setting
+ the capacity of the buffer to `MA_BUFFER_COUNT` will guarantee it's large enough for all backends.
+
+backendCap (in)
+ The capacity of the `pBackends` buffer.
+
+pBackendCount (out)
+ A pointer to the variable that will receive the enabled backend count.
+
+
+Return Value
+------------
+MA_SUCCESS if successful.
+MA_INVALID_ARGS if `pBackendCount` is NULL.
+MA_NO_SPACE if the capacity of `pBackends` is not large enough.
+
+If `MA_NO_SPACE` is returned, the `pBackends` buffer will be filled with `*pBackendCount` values.
+
+
+Thread Safety
+-------------
+Safe.
+
+
+Callback Safety
+---------------
+Safe.
+
+
+Remarks
+-------
+If you want to retrieve the number of backends so you can determine the capacity of `pBackends` buffer, you can call
+this function with `pBackends` set to NULL.
+
+This will also enumerate the null backend. If you don't want to include this you need to check for `ma_backend_null`
+when you enumerate over the returned backends and handle it appropriately. Alternatively, you can disable it at
+compile time with `MA_NO_NULL`.
+
+The returned backends are determined based on compile time settings, not the platform it's currently running on. For
+example, PulseAudio will be returned if it was enabled at compile time, even when the user doesn't actually have
+PulseAudio installed.
+
+
+Example 1
+---------
+The example below retrieves the enabled backend count using a fixed sized buffer allocated on the stack. The buffer is
+given a capacity of `MA_BACKEND_COUNT` which will guarantee it'll be large enough to store all available backends.
+Since `MA_BACKEND_COUNT` is always a relatively small value, this should be suitable for most scenarios.
+
+```
+ma_backend enabledBackends[MA_BACKEND_COUNT];
+size_t enabledBackendCount;
+
+result = ma_get_enabled_backends(enabledBackends, MA_BACKEND_COUNT, &enabledBackendCount);
+if (result != MA_SUCCESS) {
+ // Failed to retrieve enabled backends. Should never happen in this example since all inputs are valid.
+}
+```
+
+
+See Also
+--------
+ma_is_backend_enabled()
+*/
+MA_API ma_result ma_get_enabled_backends(ma_backend* pBackends, size_t backendCap, size_t* pBackendCount);
+
+/*
+Determines whether or not loopback mode is support by a backend.
+*/
+MA_API ma_bool32 ma_is_loopback_supported(ma_backend backend);
+
+#endif /* MA_NO_DEVICE_IO */
+
+
+
+/************************************************************************************************************************************************************
+
+Utiltities
+
+************************************************************************************************************************************************************/
+
+/*
+Calculates a buffer size in milliseconds from the specified number of frames and sample rate.
+*/
+MA_API ma_uint32 ma_calculate_buffer_size_in_milliseconds_from_frames(ma_uint32 bufferSizeInFrames, ma_uint32 sampleRate);
+
+/*
+Calculates a buffer size in frames from the specified number of milliseconds and sample rate.
+*/
+MA_API ma_uint32 ma_calculate_buffer_size_in_frames_from_milliseconds(ma_uint32 bufferSizeInMilliseconds, ma_uint32 sampleRate);
+
+/*
+Copies PCM frames from one buffer to another.
+*/
+MA_API void ma_copy_pcm_frames(void* dst, const void* src, ma_uint64 frameCount, ma_format format, ma_uint32 channels);
+
+/*
+Copies silent frames into the given buffer.
+
+Remarks
+-------
+For all formats except `ma_format_u8`, the output buffer will be filled with 0. For `ma_format_u8` it will be filled with 128. The reason for this is that it
+makes more sense for the purpose of mixing to initialize it to the center point.
+*/
+MA_API void ma_silence_pcm_frames(void* p, ma_uint64 frameCount, ma_format format, ma_uint32 channels);
+
+
+/*
+Offsets a pointer by the specified number of PCM frames.
+*/
+MA_API void* ma_offset_pcm_frames_ptr(void* p, ma_uint64 offsetInFrames, ma_format format, ma_uint32 channels);
+MA_API const void* ma_offset_pcm_frames_const_ptr(const void* p, ma_uint64 offsetInFrames, ma_format format, ma_uint32 channels);
+static MA_INLINE float* ma_offset_pcm_frames_ptr_f32(float* p, ma_uint64 offsetInFrames, ma_uint32 channels) { return (float*)ma_offset_pcm_frames_ptr((void*)p, offsetInFrames, ma_format_f32, channels); }
+static MA_INLINE const float* ma_offset_pcm_frames_const_ptr_f32(const float* p, ma_uint64 offsetInFrames, ma_uint32 channels) { return (const float*)ma_offset_pcm_frames_const_ptr((const void*)p, offsetInFrames, ma_format_f32, channels); }
+
+
+/*
+Clips samples.
+*/
+MA_API void ma_clip_samples_u8(ma_uint8* pDst, const ma_int16* pSrc, ma_uint64 count);
+MA_API void ma_clip_samples_s16(ma_int16* pDst, const ma_int32* pSrc, ma_uint64 count);
+MA_API void ma_clip_samples_s24(ma_uint8* pDst, const ma_int64* pSrc, ma_uint64 count);
+MA_API void ma_clip_samples_s32(ma_int32* pDst, const ma_int64* pSrc, ma_uint64 count);
+MA_API void ma_clip_samples_f32(float* pDst, const float* pSrc, ma_uint64 count);
+MA_API void ma_clip_pcm_frames(void* pDst, const void* pSrc, ma_uint64 frameCount, ma_format format, ma_uint32 channels);
+
+/*
+Helper for applying a volume factor to samples.
+
+Note that the source and destination buffers can be the same, in which case it'll perform the operation in-place.
+*/
+MA_API void ma_copy_and_apply_volume_factor_u8(ma_uint8* pSamplesOut, const ma_uint8* pSamplesIn, ma_uint64 sampleCount, float factor);
+MA_API void ma_copy_and_apply_volume_factor_s16(ma_int16* pSamplesOut, const ma_int16* pSamplesIn, ma_uint64 sampleCount, float factor);
+MA_API void ma_copy_and_apply_volume_factor_s24(void* pSamplesOut, const void* pSamplesIn, ma_uint64 sampleCount, float factor);
+MA_API void ma_copy_and_apply_volume_factor_s32(ma_int32* pSamplesOut, const ma_int32* pSamplesIn, ma_uint64 sampleCount, float factor);
+MA_API void ma_copy_and_apply_volume_factor_f32(float* pSamplesOut, const float* pSamplesIn, ma_uint64 sampleCount, float factor);
+
+MA_API void ma_apply_volume_factor_u8(ma_uint8* pSamples, ma_uint64 sampleCount, float factor);
+MA_API void ma_apply_volume_factor_s16(ma_int16* pSamples, ma_uint64 sampleCount, float factor);
+MA_API void ma_apply_volume_factor_s24(void* pSamples, ma_uint64 sampleCount, float factor);
+MA_API void ma_apply_volume_factor_s32(ma_int32* pSamples, ma_uint64 sampleCount, float factor);
+MA_API void ma_apply_volume_factor_f32(float* pSamples, ma_uint64 sampleCount, float factor);
+
+MA_API void ma_copy_and_apply_volume_factor_pcm_frames_u8(ma_uint8* pFramesOut, const ma_uint8* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor);
+MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s16(ma_int16* pFramesOut, const ma_int16* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor);
+MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s24(void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor);
+MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s32(ma_int32* pFramesOut, const ma_int32* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor);
+MA_API void ma_copy_and_apply_volume_factor_pcm_frames_f32(float* pFramesOut, const float* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor);
+MA_API void ma_copy_and_apply_volume_factor_pcm_frames(void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount, ma_format format, ma_uint32 channels, float factor);
+
+MA_API void ma_apply_volume_factor_pcm_frames_u8(ma_uint8* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor);
+MA_API void ma_apply_volume_factor_pcm_frames_s16(ma_int16* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor);
+MA_API void ma_apply_volume_factor_pcm_frames_s24(void* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor);
+MA_API void ma_apply_volume_factor_pcm_frames_s32(ma_int32* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor);
+MA_API void ma_apply_volume_factor_pcm_frames_f32(float* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor);
+MA_API void ma_apply_volume_factor_pcm_frames(void* pFrames, ma_uint64 frameCount, ma_format format, ma_uint32 channels, float factor);
+
+MA_API void ma_copy_and_apply_volume_factor_per_channel_f32(float* pFramesOut, const float* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float* pChannelGains);
+
+
+MA_API void ma_copy_and_apply_volume_and_clip_samples_u8(ma_uint8* pDst, const ma_int16* pSrc, ma_uint64 count, float volume);
+MA_API void ma_copy_and_apply_volume_and_clip_samples_s16(ma_int16* pDst, const ma_int32* pSrc, ma_uint64 count, float volume);
+MA_API void ma_copy_and_apply_volume_and_clip_samples_s24(ma_uint8* pDst, const ma_int64* pSrc, ma_uint64 count, float volume);
+MA_API void ma_copy_and_apply_volume_and_clip_samples_s32(ma_int32* pDst, const ma_int64* pSrc, ma_uint64 count, float volume);
+MA_API void ma_copy_and_apply_volume_and_clip_samples_f32(float* pDst, const float* pSrc, ma_uint64 count, float volume);
+MA_API void ma_copy_and_apply_volume_and_clip_pcm_frames(void* pDst, const void* pSrc, ma_uint64 frameCount, ma_format format, ma_uint32 channels, float volume);
+
+
+/*
+Helper for converting a linear factor to gain in decibels.
+*/
+MA_API float ma_volume_linear_to_db(float factor);
+
+/*
+Helper for converting gain in decibels to a linear factor.
+*/
+MA_API float ma_volume_db_to_linear(float gain);
+
+
+
+
+/**************************************************************************************************
+
+Data Source
+
+**************************************************************************************************/
+typedef void ma_data_source;
+
+#define MA_DATA_SOURCE_SELF_MANAGED_RANGE_AND_LOOP_POINT 0x00000001
+
+typedef struct
+{
+ ma_result (* onRead)(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
+ ma_result (* onSeek)(ma_data_source* pDataSource, ma_uint64 frameIndex);
+ ma_result (* onGetDataFormat)(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
+ ma_result (* onGetCursor)(ma_data_source* pDataSource, ma_uint64* pCursor);
+ ma_result (* onGetLength)(ma_data_source* pDataSource, ma_uint64* pLength);
+ ma_result (* onSetLooping)(ma_data_source* pDataSource, ma_bool32 isLooping);
+ ma_uint32 flags;
+} ma_data_source_vtable;
+
+typedef ma_data_source* (* ma_data_source_get_next_proc)(ma_data_source* pDataSource);
+
+typedef struct
+{
+ const ma_data_source_vtable* vtable;
+} ma_data_source_config;
+
+MA_API ma_data_source_config ma_data_source_config_init(void);
+
+
+typedef struct
+{
+ const ma_data_source_vtable* vtable;
+ ma_uint64 rangeBegInFrames;
+ ma_uint64 rangeEndInFrames; /* Set to -1 for unranged (default). */
+ ma_uint64 loopBegInFrames; /* Relative to rangeBegInFrames. */
+ ma_uint64 loopEndInFrames; /* Relative to rangeBegInFrames. Set to -1 for the end of the range. */
+ ma_data_source* pCurrent; /* When non-NULL, the data source being initialized will act as a proxy and will route all operations to pCurrent. Used in conjunction with pNext/onGetNext for seamless chaining. */
+ ma_data_source* pNext; /* When set to NULL, onGetNext will be used. */
+ ma_data_source_get_next_proc onGetNext; /* Will be used when pNext is NULL. If both are NULL, no next will be used. */
+ MA_ATOMIC(4, ma_bool32) isLooping;
+} ma_data_source_base;
+
+MA_API ma_result ma_data_source_init(const ma_data_source_config* pConfig, ma_data_source* pDataSource);
+MA_API void ma_data_source_uninit(ma_data_source* pDataSource);
+MA_API ma_result ma_data_source_read_pcm_frames(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead); /* Must support pFramesOut = NULL in which case a forward seek should be performed. */
+MA_API ma_result ma_data_source_seek_pcm_frames(ma_data_source* pDataSource, ma_uint64 frameCount, ma_uint64* pFramesSeeked); /* Can only seek forward. Equivalent to ma_data_source_read_pcm_frames(pDataSource, NULL, frameCount, &framesRead); */
+MA_API ma_result ma_data_source_seek_to_pcm_frame(ma_data_source* pDataSource, ma_uint64 frameIndex);
+MA_API ma_result ma_data_source_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
+MA_API ma_result ma_data_source_get_cursor_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pCursor);
+MA_API ma_result ma_data_source_get_length_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pLength); /* Returns MA_NOT_IMPLEMENTED if the length is unknown or cannot be determined. Decoders can return this. */
+MA_API ma_result ma_data_source_get_cursor_in_seconds(ma_data_source* pDataSource, float* pCursor);
+MA_API ma_result ma_data_source_get_length_in_seconds(ma_data_source* pDataSource, float* pLength);
+MA_API ma_result ma_data_source_set_looping(ma_data_source* pDataSource, ma_bool32 isLooping);
+MA_API ma_bool32 ma_data_source_is_looping(ma_data_source* pDataSource);
+MA_API ma_result ma_data_source_set_range_in_pcm_frames(ma_data_source* pDataSource, ma_uint64 rangeBegInFrames, ma_uint64 rangeEndInFrames);
+MA_API void ma_data_source_get_range_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pRangeBegInFrames, ma_uint64* pRangeEndInFrames);
+MA_API ma_result ma_data_source_set_loop_point_in_pcm_frames(ma_data_source* pDataSource, ma_uint64 loopBegInFrames, ma_uint64 loopEndInFrames);
+MA_API void ma_data_source_get_loop_point_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pLoopBegInFrames, ma_uint64* pLoopEndInFrames);
+MA_API ma_result ma_data_source_set_current(ma_data_source* pDataSource, ma_data_source* pCurrentDataSource);
+MA_API ma_data_source* ma_data_source_get_current(ma_data_source* pDataSource);
+MA_API ma_result ma_data_source_set_next(ma_data_source* pDataSource, ma_data_source* pNextDataSource);
+MA_API ma_data_source* ma_data_source_get_next(ma_data_source* pDataSource);
+MA_API ma_result ma_data_source_set_next_callback(ma_data_source* pDataSource, ma_data_source_get_next_proc onGetNext);
+MA_API ma_data_source_get_next_proc ma_data_source_get_next_callback(ma_data_source* pDataSource);
+
+
+typedef struct
+{
+ ma_data_source_base ds;
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint64 cursor;
+ ma_uint64 sizeInFrames;
+ const void* pData;
+} ma_audio_buffer_ref;
+
+MA_API ma_result ma_audio_buffer_ref_init(ma_format format, ma_uint32 channels, const void* pData, ma_uint64 sizeInFrames, ma_audio_buffer_ref* pAudioBufferRef);
+MA_API void ma_audio_buffer_ref_uninit(ma_audio_buffer_ref* pAudioBufferRef);
+MA_API ma_result ma_audio_buffer_ref_set_data(ma_audio_buffer_ref* pAudioBufferRef, const void* pData, ma_uint64 sizeInFrames);
+MA_API ma_uint64 ma_audio_buffer_ref_read_pcm_frames(ma_audio_buffer_ref* pAudioBufferRef, void* pFramesOut, ma_uint64 frameCount, ma_bool32 loop);
+MA_API ma_result ma_audio_buffer_ref_seek_to_pcm_frame(ma_audio_buffer_ref* pAudioBufferRef, ma_uint64 frameIndex);
+MA_API ma_result ma_audio_buffer_ref_map(ma_audio_buffer_ref* pAudioBufferRef, void** ppFramesOut, ma_uint64* pFrameCount);
+MA_API ma_result ma_audio_buffer_ref_unmap(ma_audio_buffer_ref* pAudioBufferRef, ma_uint64 frameCount); /* Returns MA_AT_END if the end has been reached. This should be considered successful. */
+MA_API ma_bool32 ma_audio_buffer_ref_at_end(const ma_audio_buffer_ref* pAudioBufferRef);
+MA_API ma_result ma_audio_buffer_ref_get_cursor_in_pcm_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pCursor);
+MA_API ma_result ma_audio_buffer_ref_get_length_in_pcm_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pLength);
+MA_API ma_result ma_audio_buffer_ref_get_available_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pAvailableFrames);
+
+
+
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint64 sizeInFrames;
+ const void* pData; /* If set to NULL, will allocate a block of memory for you. */
+ ma_allocation_callbacks allocationCallbacks;
+} ma_audio_buffer_config;
+
+MA_API ma_audio_buffer_config ma_audio_buffer_config_init(ma_format format, ma_uint32 channels, ma_uint64 sizeInFrames, const void* pData, const ma_allocation_callbacks* pAllocationCallbacks);
+
+typedef struct
+{
+ ma_audio_buffer_ref ref;
+ ma_allocation_callbacks allocationCallbacks;
+ ma_bool32 ownsData; /* Used to control whether or not miniaudio owns the data buffer. If set to true, pData will be freed in ma_audio_buffer_uninit(). */
+ ma_uint8 _pExtraData[1]; /* For allocating a buffer with the memory located directly after the other memory of the structure. */
+} ma_audio_buffer;
+
+MA_API ma_result ma_audio_buffer_init(const ma_audio_buffer_config* pConfig, ma_audio_buffer* pAudioBuffer);
+MA_API ma_result ma_audio_buffer_init_copy(const ma_audio_buffer_config* pConfig, ma_audio_buffer* pAudioBuffer);
+MA_API ma_result ma_audio_buffer_alloc_and_init(const ma_audio_buffer_config* pConfig, ma_audio_buffer** ppAudioBuffer); /* Always copies the data. Doesn't make sense to use this otherwise. Use ma_audio_buffer_uninit_and_free() to uninit. */
+MA_API void ma_audio_buffer_uninit(ma_audio_buffer* pAudioBuffer);
+MA_API void ma_audio_buffer_uninit_and_free(ma_audio_buffer* pAudioBuffer);
+MA_API ma_uint64 ma_audio_buffer_read_pcm_frames(ma_audio_buffer* pAudioBuffer, void* pFramesOut, ma_uint64 frameCount, ma_bool32 loop);
+MA_API ma_result ma_audio_buffer_seek_to_pcm_frame(ma_audio_buffer* pAudioBuffer, ma_uint64 frameIndex);
+MA_API ma_result ma_audio_buffer_map(ma_audio_buffer* pAudioBuffer, void** ppFramesOut, ma_uint64* pFrameCount);
+MA_API ma_result ma_audio_buffer_unmap(ma_audio_buffer* pAudioBuffer, ma_uint64 frameCount); /* Returns MA_AT_END if the end has been reached. This should be considered successful. */
+MA_API ma_bool32 ma_audio_buffer_at_end(const ma_audio_buffer* pAudioBuffer);
+MA_API ma_result ma_audio_buffer_get_cursor_in_pcm_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pCursor);
+MA_API ma_result ma_audio_buffer_get_length_in_pcm_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pLength);
+MA_API ma_result ma_audio_buffer_get_available_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pAvailableFrames);
+
+
+/*
+Paged Audio Buffer
+==================
+A paged audio buffer is made up of a linked list of pages. It's expandable, but not shrinkable. It
+can be used for cases where audio data is streamed in asynchronously while allowing data to be read
+at the same time.
+
+This is lock-free, but not 100% thread safe. You can append a page and read from the buffer across
+simultaneously across different threads, however only one thread at a time can append, and only one
+thread at a time can read and seek.
+*/
+typedef struct ma_paged_audio_buffer_page ma_paged_audio_buffer_page;
+struct ma_paged_audio_buffer_page
+{
+ MA_ATOMIC(MA_SIZEOF_PTR, ma_paged_audio_buffer_page*) pNext;
+ ma_uint64 sizeInFrames;
+ ma_uint8 pAudioData[1];
+};
+
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ ma_paged_audio_buffer_page head; /* Dummy head for the lock-free algorithm. Always has a size of 0. */
+ MA_ATOMIC(MA_SIZEOF_PTR, ma_paged_audio_buffer_page*) pTail; /* Never null. Initially set to &head. */
+} ma_paged_audio_buffer_data;
+
+MA_API ma_result ma_paged_audio_buffer_data_init(ma_format format, ma_uint32 channels, ma_paged_audio_buffer_data* pData);
+MA_API void ma_paged_audio_buffer_data_uninit(ma_paged_audio_buffer_data* pData, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_paged_audio_buffer_page* ma_paged_audio_buffer_data_get_head(ma_paged_audio_buffer_data* pData);
+MA_API ma_paged_audio_buffer_page* ma_paged_audio_buffer_data_get_tail(ma_paged_audio_buffer_data* pData);
+MA_API ma_result ma_paged_audio_buffer_data_get_length_in_pcm_frames(ma_paged_audio_buffer_data* pData, ma_uint64* pLength);
+MA_API ma_result ma_paged_audio_buffer_data_allocate_page(ma_paged_audio_buffer_data* pData, ma_uint64 pageSizeInFrames, const void* pInitialData, const ma_allocation_callbacks* pAllocationCallbacks, ma_paged_audio_buffer_page** ppPage);
+MA_API ma_result ma_paged_audio_buffer_data_free_page(ma_paged_audio_buffer_data* pData, ma_paged_audio_buffer_page* pPage, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_paged_audio_buffer_data_append_page(ma_paged_audio_buffer_data* pData, ma_paged_audio_buffer_page* pPage);
+MA_API ma_result ma_paged_audio_buffer_data_allocate_and_append_page(ma_paged_audio_buffer_data* pData, ma_uint32 pageSizeInFrames, const void* pInitialData, const ma_allocation_callbacks* pAllocationCallbacks);
+
+
+typedef struct
+{
+ ma_paged_audio_buffer_data* pData; /* Must not be null. */
+} ma_paged_audio_buffer_config;
+
+MA_API ma_paged_audio_buffer_config ma_paged_audio_buffer_config_init(ma_paged_audio_buffer_data* pData);
+
+
+typedef struct
+{
+ ma_data_source_base ds;
+ ma_paged_audio_buffer_data* pData; /* Audio data is read from here. Cannot be null. */
+ ma_paged_audio_buffer_page* pCurrent;
+ ma_uint64 relativeCursor; /* Relative to the current page. */
+ ma_uint64 absoluteCursor;
+} ma_paged_audio_buffer;
+
+MA_API ma_result ma_paged_audio_buffer_init(const ma_paged_audio_buffer_config* pConfig, ma_paged_audio_buffer* pPagedAudioBuffer);
+MA_API void ma_paged_audio_buffer_uninit(ma_paged_audio_buffer* pPagedAudioBuffer);
+MA_API ma_result ma_paged_audio_buffer_read_pcm_frames(ma_paged_audio_buffer* pPagedAudioBuffer, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead); /* Returns MA_AT_END if no more pages available. */
+MA_API ma_result ma_paged_audio_buffer_seek_to_pcm_frame(ma_paged_audio_buffer* pPagedAudioBuffer, ma_uint64 frameIndex);
+MA_API ma_result ma_paged_audio_buffer_get_cursor_in_pcm_frames(ma_paged_audio_buffer* pPagedAudioBuffer, ma_uint64* pCursor);
+MA_API ma_result ma_paged_audio_buffer_get_length_in_pcm_frames(ma_paged_audio_buffer* pPagedAudioBuffer, ma_uint64* pLength);
+
+
+
+/************************************************************************************************************************************************************
+
+VFS
+===
+
+The VFS object (virtual file system) is what's used to customize file access. This is useful in cases where stdio FILE* based APIs may not be entirely
+appropriate for a given situation.
+
+************************************************************************************************************************************************************/
+typedef void ma_vfs;
+typedef ma_handle ma_vfs_file;
+
+typedef enum
+{
+ MA_OPEN_MODE_READ = 0x00000001,
+ MA_OPEN_MODE_WRITE = 0x00000002
+} ma_open_mode_flags;
+
+typedef enum
+{
+ ma_seek_origin_start,
+ ma_seek_origin_current,
+ ma_seek_origin_end /* Not used by decoders. */
+} ma_seek_origin;
+
+typedef struct
+{
+ ma_uint64 sizeInBytes;
+} ma_file_info;
+
+typedef struct
+{
+ ma_result (* onOpen) (ma_vfs* pVFS, const char* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile);
+ ma_result (* onOpenW)(ma_vfs* pVFS, const wchar_t* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile);
+ ma_result (* onClose)(ma_vfs* pVFS, ma_vfs_file file);
+ ma_result (* onRead) (ma_vfs* pVFS, ma_vfs_file file, void* pDst, size_t sizeInBytes, size_t* pBytesRead);
+ ma_result (* onWrite)(ma_vfs* pVFS, ma_vfs_file file, const void* pSrc, size_t sizeInBytes, size_t* pBytesWritten);
+ ma_result (* onSeek) (ma_vfs* pVFS, ma_vfs_file file, ma_int64 offset, ma_seek_origin origin);
+ ma_result (* onTell) (ma_vfs* pVFS, ma_vfs_file file, ma_int64* pCursor);
+ ma_result (* onInfo) (ma_vfs* pVFS, ma_vfs_file file, ma_file_info* pInfo);
+} ma_vfs_callbacks;
+
+MA_API ma_result ma_vfs_open(ma_vfs* pVFS, const char* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile);
+MA_API ma_result ma_vfs_open_w(ma_vfs* pVFS, const wchar_t* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile);
+MA_API ma_result ma_vfs_close(ma_vfs* pVFS, ma_vfs_file file);
+MA_API ma_result ma_vfs_read(ma_vfs* pVFS, ma_vfs_file file, void* pDst, size_t sizeInBytes, size_t* pBytesRead);
+MA_API ma_result ma_vfs_write(ma_vfs* pVFS, ma_vfs_file file, const void* pSrc, size_t sizeInBytes, size_t* pBytesWritten);
+MA_API ma_result ma_vfs_seek(ma_vfs* pVFS, ma_vfs_file file, ma_int64 offset, ma_seek_origin origin);
+MA_API ma_result ma_vfs_tell(ma_vfs* pVFS, ma_vfs_file file, ma_int64* pCursor);
+MA_API ma_result ma_vfs_info(ma_vfs* pVFS, ma_vfs_file file, ma_file_info* pInfo);
+MA_API ma_result ma_vfs_open_and_read_file(ma_vfs* pVFS, const char* pFilePath, void** ppData, size_t* pSize, const ma_allocation_callbacks* pAllocationCallbacks);
+
+typedef struct
+{
+ ma_vfs_callbacks cb;
+ ma_allocation_callbacks allocationCallbacks; /* Only used for the wchar_t version of open() on non-Windows platforms. */
+} ma_default_vfs;
+
+MA_API ma_result ma_default_vfs_init(ma_default_vfs* pVFS, const ma_allocation_callbacks* pAllocationCallbacks);
+
+
+
+typedef ma_result (* ma_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead, size_t* pBytesRead);
+typedef ma_result (* ma_seek_proc)(void* pUserData, ma_int64 offset, ma_seek_origin origin);
+typedef ma_result (* ma_tell_proc)(void* pUserData, ma_int64* pCursor);
+
+
+
+#if !defined(MA_NO_DECODING) || !defined(MA_NO_ENCODING)
+typedef enum
+{
+ ma_encoding_format_unknown = 0,
+ ma_encoding_format_wav,
+ ma_encoding_format_flac,
+ ma_encoding_format_mp3,
+ ma_encoding_format_vorbis
+} ma_encoding_format;
+#endif
+
+/************************************************************************************************************************************************************
+
+Decoding
+========
+
+Decoders are independent of the main device API. Decoding APIs can be called freely inside the device's data callback, but they are not thread safe unless
+you do your own synchronization.
+
+************************************************************************************************************************************************************/
+#ifndef MA_NO_DECODING
+typedef struct ma_decoder ma_decoder;
+
+
+typedef struct
+{
+ ma_format preferredFormat;
+ ma_uint32 seekPointCount; /* Set to > 0 to generate a seektable if the decoding backend supports it. */
+} ma_decoding_backend_config;
+
+MA_API ma_decoding_backend_config ma_decoding_backend_config_init(ma_format preferredFormat, ma_uint32 seekPointCount);
+
+
+typedef struct
+{
+ ma_result (* onInit )(void* pUserData, ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend);
+ ma_result (* onInitFile )(void* pUserData, const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend); /* Optional. */
+ ma_result (* onInitFileW )(void* pUserData, const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend); /* Optional. */
+ ma_result (* onInitMemory)(void* pUserData, const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend); /* Optional. */
+ void (* onUninit )(void* pUserData, ma_data_source* pBackend, const ma_allocation_callbacks* pAllocationCallbacks);
+} ma_decoding_backend_vtable;
+
+
+typedef ma_result (* ma_decoder_read_proc)(ma_decoder* pDecoder, void* pBufferOut, size_t bytesToRead, size_t* pBytesRead); /* Returns the number of bytes read. */
+typedef ma_result (* ma_decoder_seek_proc)(ma_decoder* pDecoder, ma_int64 byteOffset, ma_seek_origin origin);
+typedef ma_result (* ma_decoder_tell_proc)(ma_decoder* pDecoder, ma_int64* pCursor);
+
+typedef struct
+{
+ ma_format format; /* Set to 0 or ma_format_unknown to use the stream's internal format. */
+ ma_uint32 channels; /* Set to 0 to use the stream's internal channels. */
+ ma_uint32 sampleRate; /* Set to 0 to use the stream's internal sample rate. */
+ ma_channel* pChannelMap;
+ ma_channel_mix_mode channelMixMode;
+ ma_dither_mode ditherMode;
+ ma_resampler_config resampling;
+ ma_allocation_callbacks allocationCallbacks;
+ ma_encoding_format encodingFormat;
+ ma_uint32 seekPointCount; /* When set to > 0, specifies the number of seek points to use for the generation of a seek table. Not all decoding backends support this. */
+ ma_decoding_backend_vtable** ppCustomBackendVTables;
+ ma_uint32 customBackendCount;
+ void* pCustomBackendUserData;
+} ma_decoder_config;
+
+struct ma_decoder
+{
+ ma_data_source_base ds;
+ ma_data_source* pBackend; /* The decoding backend we'll be pulling data from. */
+ const ma_decoding_backend_vtable* pBackendVTable; /* The vtable for the decoding backend. This needs to be stored so we can access the onUninit() callback. */
+ void* pBackendUserData;
+ ma_decoder_read_proc onRead;
+ ma_decoder_seek_proc onSeek;
+ ma_decoder_tell_proc onTell;
+ void* pUserData;
+ ma_uint64 readPointerInPCMFrames; /* In output sample rate. Used for keeping track of how many frames are available for decoding. */
+ ma_format outputFormat;
+ ma_uint32 outputChannels;
+ ma_uint32 outputSampleRate;
+ ma_data_converter converter; /* Data conversion is achieved by running frames through this. */
+ void* pInputCache; /* In input format. Can be null if it's not needed. */
+ ma_uint64 inputCacheCap; /* The capacity of the input cache. */
+ ma_uint64 inputCacheConsumed; /* The number of frames that have been consumed in the cache. Used for determining the next valid frame. */
+ ma_uint64 inputCacheRemaining; /* The number of valid frames remaining in the cahce. */
+ ma_allocation_callbacks allocationCallbacks;
+ union
+ {
+ struct
+ {
+ ma_vfs* pVFS;
+ ma_vfs_file file;
+ } vfs;
+ struct
+ {
+ const ma_uint8* pData;
+ size_t dataSize;
+ size_t currentReadPos;
+ } memory; /* Only used for decoders that were opened against a block of memory. */
+ } data;
+};
+
+MA_API ma_decoder_config ma_decoder_config_init(ma_format outputFormat, ma_uint32 outputChannels, ma_uint32 outputSampleRate);
+MA_API ma_decoder_config ma_decoder_config_init_default(void);
+
+MA_API ma_result ma_decoder_init(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
+MA_API ma_result ma_decoder_init_memory(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
+MA_API ma_result ma_decoder_init_vfs(ma_vfs* pVFS, const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
+MA_API ma_result ma_decoder_init_vfs_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
+MA_API ma_result ma_decoder_init_file(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
+MA_API ma_result ma_decoder_init_file_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
+
+/*
+Uninitializes a decoder.
+*/
+MA_API ma_result ma_decoder_uninit(ma_decoder* pDecoder);
+
+/*
+Reads PCM frames from the given decoder.
+
+This is not thread safe without your own synchronization.
+*/
+MA_API ma_result ma_decoder_read_pcm_frames(ma_decoder* pDecoder, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
+
+/*
+Seeks to a PCM frame based on it's absolute index.
+
+This is not thread safe without your own synchronization.
+*/
+MA_API ma_result ma_decoder_seek_to_pcm_frame(ma_decoder* pDecoder, ma_uint64 frameIndex);
+
+/*
+Retrieves the decoder's output data format.
+*/
+MA_API ma_result ma_decoder_get_data_format(ma_decoder* pDecoder, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
+
+/*
+Retrieves the current position of the read cursor in PCM frames.
+*/
+MA_API ma_result ma_decoder_get_cursor_in_pcm_frames(ma_decoder* pDecoder, ma_uint64* pCursor);
+
+/*
+Retrieves the length of the decoder in PCM frames.
+
+Do not call this on streams of an undefined length, such as internet radio.
+
+If the length is unknown or an error occurs, 0 will be returned.
+
+This will always return 0 for Vorbis decoders. This is due to a limitation with stb_vorbis in push mode which is what miniaudio
+uses internally.
+
+For MP3's, this will decode the entire file. Do not call this in time critical scenarios.
+
+This function is not thread safe without your own synchronization.
+*/
+MA_API ma_result ma_decoder_get_length_in_pcm_frames(ma_decoder* pDecoder, ma_uint64* pLength);
+
+/*
+Retrieves the number of frames that can be read before reaching the end.
+
+This calls `ma_decoder_get_length_in_pcm_frames()` so you need to be aware of the rules for that function, in
+particular ensuring you do not call it on streams of an undefined length, such as internet radio.
+
+If the total length of the decoder cannot be retrieved, such as with Vorbis decoders, `MA_NOT_IMPLEMENTED` will be
+returned.
+*/
+MA_API ma_result ma_decoder_get_available_frames(ma_decoder* pDecoder, ma_uint64* pAvailableFrames);
+
+/*
+Helper for opening and decoding a file into a heap allocated block of memory. Free the returned pointer with ma_free(). On input,
+pConfig should be set to what you want. On output it will be set to what you got.
+*/
+MA_API ma_result ma_decode_from_vfs(ma_vfs* pVFS, const char* pFilePath, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut);
+MA_API ma_result ma_decode_file(const char* pFilePath, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut);
+MA_API ma_result ma_decode_memory(const void* pData, size_t dataSize, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut);
+
+#endif /* MA_NO_DECODING */
+
+
+/************************************************************************************************************************************************************
+
+Encoding
+========
+
+Encoders do not perform any format conversion for you. If your target format does not support the format, and error will be returned.
+
+************************************************************************************************************************************************************/
+#ifndef MA_NO_ENCODING
+typedef struct ma_encoder ma_encoder;
+
+typedef ma_result (* ma_encoder_write_proc) (ma_encoder* pEncoder, const void* pBufferIn, size_t bytesToWrite, size_t* pBytesWritten);
+typedef ma_result (* ma_encoder_seek_proc) (ma_encoder* pEncoder, ma_int64 offset, ma_seek_origin origin);
+typedef ma_result (* ma_encoder_init_proc) (ma_encoder* pEncoder);
+typedef void (* ma_encoder_uninit_proc) (ma_encoder* pEncoder);
+typedef ma_result (* ma_encoder_write_pcm_frames_proc)(ma_encoder* pEncoder, const void* pFramesIn, ma_uint64 frameCount, ma_uint64* pFramesWritten);
+
+typedef struct
+{
+ ma_encoding_format encodingFormat;
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ ma_allocation_callbacks allocationCallbacks;
+} ma_encoder_config;
+
+MA_API ma_encoder_config ma_encoder_config_init(ma_encoding_format encodingFormat, ma_format format, ma_uint32 channels, ma_uint32 sampleRate);
+
+struct ma_encoder
+{
+ ma_encoder_config config;
+ ma_encoder_write_proc onWrite;
+ ma_encoder_seek_proc onSeek;
+ ma_encoder_init_proc onInit;
+ ma_encoder_uninit_proc onUninit;
+ ma_encoder_write_pcm_frames_proc onWritePCMFrames;
+ void* pUserData;
+ void* pInternalEncoder; /* <-- The drwav/drflac/stb_vorbis/etc. objects. */
+ union
+ {
+ struct
+ {
+ ma_vfs* pVFS;
+ ma_vfs_file file;
+ } vfs;
+ } data;
+};
+
+MA_API ma_result ma_encoder_init(ma_encoder_write_proc onWrite, ma_encoder_seek_proc onSeek, void* pUserData, const ma_encoder_config* pConfig, ma_encoder* pEncoder);
+MA_API ma_result ma_encoder_init_vfs(ma_vfs* pVFS, const char* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder);
+MA_API ma_result ma_encoder_init_vfs_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder);
+MA_API ma_result ma_encoder_init_file(const char* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder);
+MA_API ma_result ma_encoder_init_file_w(const wchar_t* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder);
+MA_API void ma_encoder_uninit(ma_encoder* pEncoder);
+MA_API ma_result ma_encoder_write_pcm_frames(ma_encoder* pEncoder, const void* pFramesIn, ma_uint64 frameCount, ma_uint64* pFramesWritten);
+
+#endif /* MA_NO_ENCODING */
+
+
+/************************************************************************************************************************************************************
+
+Generation
+
+************************************************************************************************************************************************************/
+#ifndef MA_NO_GENERATION
+typedef enum
+{
+ ma_waveform_type_sine,
+ ma_waveform_type_square,
+ ma_waveform_type_triangle,
+ ma_waveform_type_sawtooth
+} ma_waveform_type;
+
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ ma_waveform_type type;
+ double amplitude;
+ double frequency;
+} ma_waveform_config;
+
+MA_API ma_waveform_config ma_waveform_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_waveform_type type, double amplitude, double frequency);
+
+typedef struct
+{
+ ma_data_source_base ds;
+ ma_waveform_config config;
+ double advance;
+ double time;
+} ma_waveform;
+
+MA_API ma_result ma_waveform_init(const ma_waveform_config* pConfig, ma_waveform* pWaveform);
+MA_API void ma_waveform_uninit(ma_waveform* pWaveform);
+MA_API ma_result ma_waveform_read_pcm_frames(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
+MA_API ma_result ma_waveform_seek_to_pcm_frame(ma_waveform* pWaveform, ma_uint64 frameIndex);
+MA_API ma_result ma_waveform_set_amplitude(ma_waveform* pWaveform, double amplitude);
+MA_API ma_result ma_waveform_set_frequency(ma_waveform* pWaveform, double frequency);
+MA_API ma_result ma_waveform_set_type(ma_waveform* pWaveform, ma_waveform_type type);
+MA_API ma_result ma_waveform_set_sample_rate(ma_waveform* pWaveform, ma_uint32 sampleRate);
+
+typedef enum
+{
+ ma_noise_type_white,
+ ma_noise_type_pink,
+ ma_noise_type_brownian
+} ma_noise_type;
+
+
+typedef struct
+{
+ ma_format format;
+ ma_uint32 channels;
+ ma_noise_type type;
+ ma_int32 seed;
+ double amplitude;
+ ma_bool32 duplicateChannels;
+} ma_noise_config;
+
+MA_API ma_noise_config ma_noise_config_init(ma_format format, ma_uint32 channels, ma_noise_type type, ma_int32 seed, double amplitude);
+
+typedef struct
+{
+ ma_data_source_vtable ds;
+ ma_noise_config config;
+ ma_lcg lcg;
+ union
+ {
+ struct
+ {
+ double** bin;
+ double* accumulation;
+ ma_uint32* counter;
+ } pink;
+ struct
+ {
+ double* accumulation;
+ } brownian;
+ } state;
+
+ /* Memory management. */
+ void* _pHeap;
+ ma_bool32 _ownsHeap;
+} ma_noise;
+
+MA_API ma_result ma_noise_get_heap_size(const ma_noise_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_noise_init_preallocated(const ma_noise_config* pConfig, void* pHeap, ma_noise* pNoise);
+MA_API ma_result ma_noise_init(const ma_noise_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_noise* pNoise);
+MA_API void ma_noise_uninit(ma_noise* pNoise, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_noise_read_pcm_frames(ma_noise* pNoise, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
+MA_API ma_result ma_noise_set_amplitude(ma_noise* pNoise, double amplitude);
+MA_API ma_result ma_noise_set_seed(ma_noise* pNoise, ma_int32 seed);
+MA_API ma_result ma_noise_set_type(ma_noise* pNoise, ma_noise_type type);
+
+#endif /* MA_NO_GENERATION */
+
+
+
+/************************************************************************************************************************************************************
+
+Resource Manager
+
+************************************************************************************************************************************************************/
+/* The resource manager cannot be enabled if there is no decoder. */
+#if !defined(MA_NO_RESOURCE_MANAGER) && defined(MA_NO_DECODING)
+#define MA_NO_RESOURCE_MANAGER
+#endif
+
+#ifndef MA_NO_RESOURCE_MANAGER
+typedef struct ma_resource_manager ma_resource_manager;
+typedef struct ma_resource_manager_data_buffer_node ma_resource_manager_data_buffer_node;
+typedef struct ma_resource_manager_data_buffer ma_resource_manager_data_buffer;
+typedef struct ma_resource_manager_data_stream ma_resource_manager_data_stream;
+typedef struct ma_resource_manager_data_source ma_resource_manager_data_source;
+
+typedef enum
+{
+ MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM = 0x00000001, /* When set, does not load the entire data source in memory. Disk I/O will happen on job threads. */
+ MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE = 0x00000002, /* Decode data before storing in memory. When set, decoding is done at the resource manager level rather than the mixing thread. Results in faster mixing, but higher memory usage. */
+ MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC = 0x00000004, /* When set, the resource manager will load the data source asynchronously. */
+ MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT = 0x00000008 /* When set, waits for initialization of the underlying data source before returning from ma_resource_manager_data_source_init(). */
+} ma_resource_manager_data_source_flags;
+
+
+/*
+Pipeline notifications used by the resource manager. Made up of both an async notification and a fence, both of which are optional.
+*/
+typedef struct
+{
+ ma_async_notification* pNotification;
+ ma_fence* pFence;
+} ma_resource_manager_pipeline_stage_notification;
+
+typedef struct
+{
+ ma_resource_manager_pipeline_stage_notification init; /* Initialization of the decoder. */
+ ma_resource_manager_pipeline_stage_notification done; /* Decoding fully completed. */
+} ma_resource_manager_pipeline_notifications;
+
+MA_API ma_resource_manager_pipeline_notifications ma_resource_manager_pipeline_notifications_init(void);
+
+
+
+/* BEGIN BACKWARDS COMPATIBILITY */
+/* TODO: Remove this block in version 0.12. */
+#if 1
+#define ma_resource_manager_job ma_job
+#define ma_resource_manager_job_init ma_job_init
+#define MA_JOB_TYPE_RESOURCE_MANAGER_QUEUE_FLAG_NON_BLOCKING MA_JOB_QUEUE_FLAG_NON_BLOCKING
+#define ma_resource_manager_job_queue_config ma_job_queue_config
+#define ma_resource_manager_job_queue_config_init ma_job_queue_config_init
+#define ma_resource_manager_job_queue ma_job_queue
+#define ma_resource_manager_job_queue_get_heap_size ma_job_queue_get_heap_size
+#define ma_resource_manager_job_queue_init_preallocated ma_job_queue_init_preallocated
+#define ma_resource_manager_job_queue_init ma_job_queue_init
+#define ma_resource_manager_job_queue_uninit ma_job_queue_uninit
+#define ma_resource_manager_job_queue_post ma_job_queue_post
+#define ma_resource_manager_job_queue_next ma_job_queue_next
+#endif
+/* END BACKWARDS COMPATIBILITY */
+
+
+
+
+/* Maximum job thread count will be restricted to this, but this may be removed later and replaced with a heap allocation thereby removing any limitation. */
+#ifndef MA_RESOURCE_MANAGER_MAX_JOB_THREAD_COUNT
+#define MA_RESOURCE_MANAGER_MAX_JOB_THREAD_COUNT 64
+#endif
+
+typedef enum
+{
+ /* Indicates ma_resource_manager_next_job() should not block. Only valid when the job thread count is 0. */
+ MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING = 0x00000001,
+
+ /* Disables any kind of multithreading. Implicitly enables MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING. */
+ MA_RESOURCE_MANAGER_FLAG_NO_THREADING = 0x00000002
+} ma_resource_manager_flags;
+
+typedef struct
+{
+ const char* pFilePath;
+ const wchar_t* pFilePathW;
+ const ma_resource_manager_pipeline_notifications* pNotifications;
+ ma_uint64 initialSeekPointInPCMFrames;
+ ma_uint64 rangeBegInPCMFrames;
+ ma_uint64 rangeEndInPCMFrames;
+ ma_uint64 loopPointBegInPCMFrames;
+ ma_uint64 loopPointEndInPCMFrames;
+ ma_bool32 isLooping;
+ ma_uint32 flags;
+} ma_resource_manager_data_source_config;
+
+MA_API ma_resource_manager_data_source_config ma_resource_manager_data_source_config_init(void);
+
+
+typedef enum
+{
+ ma_resource_manager_data_supply_type_unknown = 0, /* Used for determining whether or the data supply has been initialized. */
+ ma_resource_manager_data_supply_type_encoded, /* Data supply is an encoded buffer. Connector is ma_decoder. */
+ ma_resource_manager_data_supply_type_decoded, /* Data supply is a decoded buffer. Connector is ma_audio_buffer. */
+ ma_resource_manager_data_supply_type_decoded_paged /* Data supply is a linked list of decoded buffers. Connector is ma_paged_audio_buffer. */
+} ma_resource_manager_data_supply_type;
+
+typedef struct
+{
+ MA_ATOMIC(4, ma_resource_manager_data_supply_type) type; /* Read and written from different threads so needs to be accessed atomically. */
+ union
+ {
+ struct
+ {
+ const void* pData;
+ size_t sizeInBytes;
+ } encoded;
+ struct
+ {
+ const void* pData;
+ ma_uint64 totalFrameCount;
+ ma_uint64 decodedFrameCount;
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ } decoded;
+ struct
+ {
+ ma_paged_audio_buffer_data data;
+ ma_uint64 decodedFrameCount;
+ ma_uint32 sampleRate;
+ } decodedPaged;
+ } backend;
+} ma_resource_manager_data_supply;
+
+struct ma_resource_manager_data_buffer_node
+{
+ ma_uint32 hashedName32; /* The hashed name. This is the key. */
+ ma_uint32 refCount;
+ MA_ATOMIC(4, ma_result) result; /* Result from asynchronous loading. When loading set to MA_BUSY. When fully loaded set to MA_SUCCESS. When deleting set to MA_UNAVAILABLE. */
+ MA_ATOMIC(4, ma_uint32) executionCounter; /* For allocating execution orders for jobs. */
+ MA_ATOMIC(4, ma_uint32) executionPointer; /* For managing the order of execution for asynchronous jobs relating to this object. Incremented as jobs complete processing. */
+ ma_bool32 isDataOwnedByResourceManager; /* Set to true when the underlying data buffer was allocated the resource manager. Set to false if it is owned by the application (via ma_resource_manager_register_*()). */
+ ma_resource_manager_data_supply data;
+ ma_resource_manager_data_buffer_node* pParent;
+ ma_resource_manager_data_buffer_node* pChildLo;
+ ma_resource_manager_data_buffer_node* pChildHi;
+};
+
+struct ma_resource_manager_data_buffer
+{
+ ma_data_source_base ds; /* Base data source. A data buffer is a data source. */
+ ma_resource_manager* pResourceManager; /* A pointer to the resource manager that owns this buffer. */
+ ma_resource_manager_data_buffer_node* pNode; /* The data node. This is reference counted and is what supplies the data. */
+ ma_uint32 flags; /* The flags that were passed used to initialize the buffer. */
+ MA_ATOMIC(4, ma_uint32) executionCounter; /* For allocating execution orders for jobs. */
+ MA_ATOMIC(4, ma_uint32) executionPointer; /* For managing the order of execution for asynchronous jobs relating to this object. Incremented as jobs complete processing. */
+ ma_uint64 seekTargetInPCMFrames; /* Only updated by the public API. Never written nor read from the job thread. */
+ ma_bool32 seekToCursorOnNextRead; /* On the next read we need to seek to the frame cursor. */
+ MA_ATOMIC(4, ma_result) result; /* Keeps track of a result of decoding. Set to MA_BUSY while the buffer is still loading. Set to MA_SUCCESS when loading is finished successfully. Otherwise set to some other code. */
+ MA_ATOMIC(4, ma_bool32) isLooping; /* Can be read and written by different threads at the same time. Must be used atomically. */
+ ma_bool32 isConnectorInitialized; /* Used for asynchronous loading to ensure we don't try to initialize the connector multiple times while waiting for the node to fully load. */
+ union
+ {
+ ma_decoder decoder; /* Supply type is ma_resource_manager_data_supply_type_encoded */
+ ma_audio_buffer buffer; /* Supply type is ma_resource_manager_data_supply_type_decoded */
+ ma_paged_audio_buffer pagedBuffer; /* Supply type is ma_resource_manager_data_supply_type_decoded_paged */
+ } connector; /* Connects this object to the node's data supply. */
+};
+
+struct ma_resource_manager_data_stream
+{
+ ma_data_source_base ds; /* Base data source. A data stream is a data source. */
+ ma_resource_manager* pResourceManager; /* A pointer to the resource manager that owns this data stream. */
+ ma_uint32 flags; /* The flags that were passed used to initialize the stream. */
+ ma_decoder decoder; /* Used for filling pages with data. This is only ever accessed by the job thread. The public API should never touch this. */
+ ma_bool32 isDecoderInitialized; /* Required for determining whether or not the decoder should be uninitialized in MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_STREAM. */
+ ma_uint64 totalLengthInPCMFrames; /* This is calculated when first loaded by the MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_STREAM. */
+ ma_uint32 relativeCursor; /* The playback cursor, relative to the current page. Only ever accessed by the public API. Never accessed by the job thread. */
+ MA_ATOMIC(8, ma_uint64) absoluteCursor; /* The playback cursor, in absolute position starting from the start of the file. */
+ ma_uint32 currentPageIndex; /* Toggles between 0 and 1. Index 0 is the first half of pPageData. Index 1 is the second half. Only ever accessed by the public API. Never accessed by the job thread. */
+ MA_ATOMIC(4, ma_uint32) executionCounter; /* For allocating execution orders for jobs. */
+ MA_ATOMIC(4, ma_uint32) executionPointer; /* For managing the order of execution for asynchronous jobs relating to this object. Incremented as jobs complete processing. */
+
+ /* Written by the public API, read by the job thread. */
+ MA_ATOMIC(4, ma_bool32) isLooping; /* Whether or not the stream is looping. It's important to set the looping flag at the data stream level for smooth loop transitions. */
+
+ /* Written by the job thread, read by the public API. */
+ void* pPageData; /* Buffer containing the decoded data of each page. Allocated once at initialization time. */
+ MA_ATOMIC(4, ma_uint32) pageFrameCount[2]; /* The number of valid PCM frames in each page. Used to determine the last valid frame. */
+
+ /* Written and read by both the public API and the job thread. These must be atomic. */
+ MA_ATOMIC(4, ma_result) result; /* Result from asynchronous loading. When loading set to MA_BUSY. When initialized set to MA_SUCCESS. When deleting set to MA_UNAVAILABLE. If an error occurs when loading, set to an error code. */
+ MA_ATOMIC(4, ma_bool32) isDecoderAtEnd; /* Whether or not the decoder has reached the end. */
+ MA_ATOMIC(4, ma_bool32) isPageValid[2]; /* Booleans to indicate whether or not a page is valid. Set to false by the public API, set to true by the job thread. Set to false as the pages are consumed, true when they are filled. */
+ MA_ATOMIC(4, ma_bool32) seekCounter; /* When 0, no seeking is being performed. When > 0, a seek is being performed and reading should be delayed with MA_BUSY. */
+};
+
+struct ma_resource_manager_data_source
+{
+ union
+ {
+ ma_resource_manager_data_buffer buffer;
+ ma_resource_manager_data_stream stream;
+ } backend; /* Must be the first item because we need the first item to be the data source callbacks for the buffer or stream. */
+
+ ma_uint32 flags; /* The flags that were passed in to ma_resource_manager_data_source_init(). */
+ MA_ATOMIC(4, ma_uint32) executionCounter; /* For allocating execution orders for jobs. */
+ MA_ATOMIC(4, ma_uint32) executionPointer; /* For managing the order of execution for asynchronous jobs relating to this object. Incremented as jobs complete processing. */
+};
+
+typedef struct
+{
+ ma_allocation_callbacks allocationCallbacks;
+ ma_log* pLog;
+ ma_format decodedFormat; /* The decoded format to use. Set to ma_format_unknown (default) to use the file's native format. */
+ ma_uint32 decodedChannels; /* The decoded channel count to use. Set to 0 (default) to use the file's native channel count. */
+ ma_uint32 decodedSampleRate; /* the decoded sample rate to use. Set to 0 (default) to use the file's native sample rate. */
+ ma_uint32 jobThreadCount; /* Set to 0 if you want to self-manage your job threads. Defaults to 1. */
+ ma_uint32 jobQueueCapacity; /* The maximum number of jobs that can fit in the queue at a time. Defaults to MA_JOB_TYPE_RESOURCE_MANAGER_QUEUE_CAPACITY. Cannot be zero. */
+ ma_uint32 flags;
+ ma_vfs* pVFS; /* Can be NULL in which case defaults will be used. */
+ ma_decoding_backend_vtable** ppCustomDecodingBackendVTables;
+ ma_uint32 customDecodingBackendCount;
+ void* pCustomDecodingBackendUserData;
+} ma_resource_manager_config;
+
+MA_API ma_resource_manager_config ma_resource_manager_config_init(void);
+
+struct ma_resource_manager
+{
+ ma_resource_manager_config config;
+ ma_resource_manager_data_buffer_node* pRootDataBufferNode; /* The root buffer in the binary tree. */
+#ifndef MA_NO_THREADING
+ ma_mutex dataBufferBSTLock; /* For synchronizing access to the data buffer binary tree. */
+ ma_thread jobThreads[MA_RESOURCE_MANAGER_MAX_JOB_THREAD_COUNT]; /* The threads for executing jobs. */
+#endif
+ ma_job_queue jobQueue; /* Multi-consumer, multi-producer job queue for managing jobs for asynchronous decoding and streaming. */
+ ma_default_vfs defaultVFS; /* Only used if a custom VFS is not specified. */
+ ma_log log; /* Only used if no log was specified in the config. */
+};
+
+/* Init. */
+MA_API ma_result ma_resource_manager_init(const ma_resource_manager_config* pConfig, ma_resource_manager* pResourceManager);
+MA_API void ma_resource_manager_uninit(ma_resource_manager* pResourceManager);
+MA_API ma_log* ma_resource_manager_get_log(ma_resource_manager* pResourceManager);
+
+/* Registration. */
+MA_API ma_result ma_resource_manager_register_file(ma_resource_manager* pResourceManager, const char* pFilePath, ma_uint32 flags);
+MA_API ma_result ma_resource_manager_register_file_w(ma_resource_manager* pResourceManager, const wchar_t* pFilePath, ma_uint32 flags);
+MA_API ma_result ma_resource_manager_register_decoded_data(ma_resource_manager* pResourceManager, const char* pName, const void* pData, ma_uint64 frameCount, ma_format format, ma_uint32 channels, ma_uint32 sampleRate); /* Does not copy. Increments the reference count if already exists and returns MA_SUCCESS. */
+MA_API ma_result ma_resource_manager_register_decoded_data_w(ma_resource_manager* pResourceManager, const wchar_t* pName, const void* pData, ma_uint64 frameCount, ma_format format, ma_uint32 channels, ma_uint32 sampleRate);
+MA_API ma_result ma_resource_manager_register_encoded_data(ma_resource_manager* pResourceManager, const char* pName, const void* pData, size_t sizeInBytes); /* Does not copy. Increments the reference count if already exists and returns MA_SUCCESS. */
+MA_API ma_result ma_resource_manager_register_encoded_data_w(ma_resource_manager* pResourceManager, const wchar_t* pName, const void* pData, size_t sizeInBytes);
+MA_API ma_result ma_resource_manager_unregister_file(ma_resource_manager* pResourceManager, const char* pFilePath);
+MA_API ma_result ma_resource_manager_unregister_file_w(ma_resource_manager* pResourceManager, const wchar_t* pFilePath);
+MA_API ma_result ma_resource_manager_unregister_data(ma_resource_manager* pResourceManager, const char* pName);
+MA_API ma_result ma_resource_manager_unregister_data_w(ma_resource_manager* pResourceManager, const wchar_t* pName);
+
+/* Data Buffers. */
+MA_API ma_result ma_resource_manager_data_buffer_init_ex(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source_config* pConfig, ma_resource_manager_data_buffer* pDataBuffer);
+MA_API ma_result ma_resource_manager_data_buffer_init(ma_resource_manager* pResourceManager, const char* pFilePath, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_buffer* pDataBuffer);
+MA_API ma_result ma_resource_manager_data_buffer_init_w(ma_resource_manager* pResourceManager, const wchar_t* pFilePath, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_buffer* pDataBuffer);
+MA_API ma_result ma_resource_manager_data_buffer_init_copy(ma_resource_manager* pResourceManager, const ma_resource_manager_data_buffer* pExistingDataBuffer, ma_resource_manager_data_buffer* pDataBuffer);
+MA_API ma_result ma_resource_manager_data_buffer_uninit(ma_resource_manager_data_buffer* pDataBuffer);
+MA_API ma_result ma_resource_manager_data_buffer_read_pcm_frames(ma_resource_manager_data_buffer* pDataBuffer, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
+MA_API ma_result ma_resource_manager_data_buffer_seek_to_pcm_frame(ma_resource_manager_data_buffer* pDataBuffer, ma_uint64 frameIndex);
+MA_API ma_result ma_resource_manager_data_buffer_get_data_format(ma_resource_manager_data_buffer* pDataBuffer, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
+MA_API ma_result ma_resource_manager_data_buffer_get_cursor_in_pcm_frames(ma_resource_manager_data_buffer* pDataBuffer, ma_uint64* pCursor);
+MA_API ma_result ma_resource_manager_data_buffer_get_length_in_pcm_frames(ma_resource_manager_data_buffer* pDataBuffer, ma_uint64* pLength);
+MA_API ma_result ma_resource_manager_data_buffer_result(const ma_resource_manager_data_buffer* pDataBuffer);
+MA_API ma_result ma_resource_manager_data_buffer_set_looping(ma_resource_manager_data_buffer* pDataBuffer, ma_bool32 isLooping);
+MA_API ma_bool32 ma_resource_manager_data_buffer_is_looping(const ma_resource_manager_data_buffer* pDataBuffer);
+MA_API ma_result ma_resource_manager_data_buffer_get_available_frames(ma_resource_manager_data_buffer* pDataBuffer, ma_uint64* pAvailableFrames);
+
+/* Data Streams. */
+MA_API ma_result ma_resource_manager_data_stream_init_ex(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source_config* pConfig, ma_resource_manager_data_stream* pDataStream);
+MA_API ma_result ma_resource_manager_data_stream_init(ma_resource_manager* pResourceManager, const char* pFilePath, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_stream* pDataStream);
+MA_API ma_result ma_resource_manager_data_stream_init_w(ma_resource_manager* pResourceManager, const wchar_t* pFilePath, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_stream* pDataStream);
+MA_API ma_result ma_resource_manager_data_stream_uninit(ma_resource_manager_data_stream* pDataStream);
+MA_API ma_result ma_resource_manager_data_stream_read_pcm_frames(ma_resource_manager_data_stream* pDataStream, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
+MA_API ma_result ma_resource_manager_data_stream_seek_to_pcm_frame(ma_resource_manager_data_stream* pDataStream, ma_uint64 frameIndex);
+MA_API ma_result ma_resource_manager_data_stream_get_data_format(ma_resource_manager_data_stream* pDataStream, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
+MA_API ma_result ma_resource_manager_data_stream_get_cursor_in_pcm_frames(ma_resource_manager_data_stream* pDataStream, ma_uint64* pCursor);
+MA_API ma_result ma_resource_manager_data_stream_get_length_in_pcm_frames(ma_resource_manager_data_stream* pDataStream, ma_uint64* pLength);
+MA_API ma_result ma_resource_manager_data_stream_result(const ma_resource_manager_data_stream* pDataStream);
+MA_API ma_result ma_resource_manager_data_stream_set_looping(ma_resource_manager_data_stream* pDataStream, ma_bool32 isLooping);
+MA_API ma_bool32 ma_resource_manager_data_stream_is_looping(const ma_resource_manager_data_stream* pDataStream);
+MA_API ma_result ma_resource_manager_data_stream_get_available_frames(ma_resource_manager_data_stream* pDataStream, ma_uint64* pAvailableFrames);
+
+/* Data Sources. */
+MA_API ma_result ma_resource_manager_data_source_init_ex(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source_config* pConfig, ma_resource_manager_data_source* pDataSource);
+MA_API ma_result ma_resource_manager_data_source_init(ma_resource_manager* pResourceManager, const char* pName, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_source* pDataSource);
+MA_API ma_result ma_resource_manager_data_source_init_w(ma_resource_manager* pResourceManager, const wchar_t* pName, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_source* pDataSource);
+MA_API ma_result ma_resource_manager_data_source_init_copy(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source* pExistingDataSource, ma_resource_manager_data_source* pDataSource);
+MA_API ma_result ma_resource_manager_data_source_uninit(ma_resource_manager_data_source* pDataSource);
+MA_API ma_result ma_resource_manager_data_source_read_pcm_frames(ma_resource_manager_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
+MA_API ma_result ma_resource_manager_data_source_seek_to_pcm_frame(ma_resource_manager_data_source* pDataSource, ma_uint64 frameIndex);
+MA_API ma_result ma_resource_manager_data_source_get_data_format(ma_resource_manager_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
+MA_API ma_result ma_resource_manager_data_source_get_cursor_in_pcm_frames(ma_resource_manager_data_source* pDataSource, ma_uint64* pCursor);
+MA_API ma_result ma_resource_manager_data_source_get_length_in_pcm_frames(ma_resource_manager_data_source* pDataSource, ma_uint64* pLength);
+MA_API ma_result ma_resource_manager_data_source_result(const ma_resource_manager_data_source* pDataSource);
+MA_API ma_result ma_resource_manager_data_source_set_looping(ma_resource_manager_data_source* pDataSource, ma_bool32 isLooping);
+MA_API ma_bool32 ma_resource_manager_data_source_is_looping(const ma_resource_manager_data_source* pDataSource);
+MA_API ma_result ma_resource_manager_data_source_get_available_frames(ma_resource_manager_data_source* pDataSource, ma_uint64* pAvailableFrames);
+
+/* Job management. */
+MA_API ma_result ma_resource_manager_post_job(ma_resource_manager* pResourceManager, const ma_job* pJob);
+MA_API ma_result ma_resource_manager_post_job_quit(ma_resource_manager* pResourceManager); /* Helper for posting a quit job. */
+MA_API ma_result ma_resource_manager_next_job(ma_resource_manager* pResourceManager, ma_job* pJob);
+MA_API ma_result ma_resource_manager_process_job(ma_resource_manager* pResourceManager, ma_job* pJob); /* DEPRECATED. Use ma_job_process(). Will be removed in version 0.12. */
+MA_API ma_result ma_resource_manager_process_next_job(ma_resource_manager* pResourceManager); /* Returns MA_CANCELLED if a MA_JOB_TYPE_QUIT job is found. In non-blocking mode, returns MA_NO_DATA_AVAILABLE if no jobs are available. */
+#endif /* MA_NO_RESOURCE_MANAGER */
+
+
+
+/************************************************************************************************************************************************************
+
+Node Graph
+
+************************************************************************************************************************************************************/
+#ifndef MA_NO_NODE_GRAPH
+/* Must never exceed 254. */
+#ifndef MA_MAX_NODE_BUS_COUNT
+#define MA_MAX_NODE_BUS_COUNT 254
+#endif
+
+/* Used internally by miniaudio for memory management. Must never exceed MA_MAX_NODE_BUS_COUNT. */
+#ifndef MA_MAX_NODE_LOCAL_BUS_COUNT
+#define MA_MAX_NODE_LOCAL_BUS_COUNT 2
+#endif
+
+/* Use this when the bus count is determined by the node instance rather than the vtable. */
+#define MA_NODE_BUS_COUNT_UNKNOWN 255
+
+typedef struct ma_node_graph ma_node_graph;
+typedef void ma_node;
+
+
+/* Node flags. */
+typedef enum
+{
+ MA_NODE_FLAG_PASSTHROUGH = 0x00000001,
+ MA_NODE_FLAG_CONTINUOUS_PROCESSING = 0x00000002,
+ MA_NODE_FLAG_ALLOW_NULL_INPUT = 0x00000004,
+ MA_NODE_FLAG_DIFFERENT_PROCESSING_RATES = 0x00000008,
+ MA_NODE_FLAG_SILENT_OUTPUT = 0x00000010
+} ma_node_flags;
+
+
+/* The playback state of a node. Either started or stopped. */
+typedef enum
+{
+ ma_node_state_started = 0,
+ ma_node_state_stopped = 1
+} ma_node_state;
+
+
+typedef struct
+{
+ /*
+ Extended processing callback. This callback is used for effects that process input and output
+ at different rates (i.e. they perform resampling). This is similar to the simple version, only
+ they take two seperate frame counts: one for input, and one for output.
+
+ On input, `pFrameCountOut` is equal to the capacity of the output buffer for each bus, whereas
+ `pFrameCountIn` will be equal to the number of PCM frames in each of the buffers in `ppFramesIn`.
+
+ On output, set `pFrameCountOut` to the number of PCM frames that were actually output and set
+ `pFrameCountIn` to the number of input frames that were consumed.
+ */
+ void (* onProcess)(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut);
+
+ /*
+ A callback for retrieving the number of a input frames that are required to output the
+ specified number of output frames. You would only want to implement this when the node performs
+ resampling. This is optional, even for nodes that perform resampling, but it does offer a
+ small reduction in latency as it allows miniaudio to calculate the exact number of input frames
+ to read at a time instead of having to estimate.
+ */
+ ma_result (* onGetRequiredInputFrameCount)(ma_node* pNode, ma_uint32 outputFrameCount, ma_uint32* pInputFrameCount);
+
+ /*
+ The number of input buses. This is how many sub-buffers will be contained in the `ppFramesIn`
+ parameters of the callbacks above.
+ */
+ ma_uint8 inputBusCount;
+
+ /*
+ The number of output buses. This is how many sub-buffers will be contained in the `ppFramesOut`
+ parameters of the callbacks above.
+ */
+ ma_uint8 outputBusCount;
+
+ /*
+ Flags describing characteristics of the node. This is currently just a placeholder for some
+ ideas for later on.
+ */
+ ma_uint32 flags;
+} ma_node_vtable;
+
+typedef struct
+{
+ const ma_node_vtable* vtable; /* Should never be null. Initialization of the node will fail if so. */
+ ma_node_state initialState; /* Defaults to ma_node_state_started. */
+ ma_uint32 inputBusCount; /* Only used if the vtable specifies an input bus count of `MA_NODE_BUS_COUNT_UNKNOWN`, otherwise must be set to `MA_NODE_BUS_COUNT_UNKNOWN` (default). */
+ ma_uint32 outputBusCount; /* Only used if the vtable specifies an output bus count of `MA_NODE_BUS_COUNT_UNKNOWN`, otherwise be set to `MA_NODE_BUS_COUNT_UNKNOWN` (default). */
+ const ma_uint32* pInputChannels; /* The number of elements are determined by the input bus count as determined by the vtable, or `inputBusCount` if the vtable specifies `MA_NODE_BUS_COUNT_UNKNOWN`. */
+ const ma_uint32* pOutputChannels; /* The number of elements are determined by the output bus count as determined by the vtable, or `outputBusCount` if the vtable specifies `MA_NODE_BUS_COUNT_UNKNOWN`. */
+} ma_node_config;
+
+MA_API ma_node_config ma_node_config_init(void);
+
+
+/*
+A node has multiple output buses. An output bus is attached to an input bus as an item in a linked
+list. Think of the input bus as a linked list, with the output bus being an item in that list.
+*/
+typedef struct ma_node_output_bus ma_node_output_bus;
+struct ma_node_output_bus
+{
+ /* Immutable. */
+ ma_node* pNode; /* The node that owns this output bus. The input node. Will be null for dummy head and tail nodes. */
+ ma_uint8 outputBusIndex; /* The index of the output bus on pNode that this output bus represents. */
+ ma_uint8 channels; /* The number of channels in the audio stream for this bus. */
+
+ /* Mutable via multiple threads. Must be used atomically. The weird ordering here is for packing reasons. */
+ MA_ATOMIC(1, ma_uint8) inputNodeInputBusIndex; /* The index of the input bus on the input. Required for detaching. */
+ MA_ATOMIC(4, ma_uint32) flags; /* Some state flags for tracking the read state of the output buffer. A combination of MA_NODE_OUTPUT_BUS_FLAG_*. */
+ MA_ATOMIC(4, ma_uint32) refCount; /* Reference count for some thread-safety when detaching. */
+ MA_ATOMIC(4, ma_bool32) isAttached; /* This is used to prevent iteration of nodes that are in the middle of being detached. Used for thread safety. */
+ MA_ATOMIC(4, ma_spinlock) lock; /* Unfortunate lock, but significantly simplifies the implementation. Required for thread-safe attaching and detaching. */
+ MA_ATOMIC(4, float) volume; /* Linear. */
+ MA_ATOMIC(MA_SIZEOF_PTR, ma_node_output_bus*) pNext; /* If null, it's the tail node or detached. */
+ MA_ATOMIC(MA_SIZEOF_PTR, ma_node_output_bus*) pPrev; /* If null, it's the head node or detached. */
+ MA_ATOMIC(MA_SIZEOF_PTR, ma_node*) pInputNode; /* The node that this output bus is attached to. Required for detaching. */
+};
+
+/*
+A node has multiple input buses. The output buses of a node are connecting to the input busses of
+another. An input bus is essentially just a linked list of output buses.
+*/
+typedef struct ma_node_input_bus ma_node_input_bus;
+struct ma_node_input_bus
+{
+ /* Mutable via multiple threads. */
+ ma_node_output_bus head; /* Dummy head node for simplifying some lock-free thread-safety stuff. */
+ MA_ATOMIC(4, ma_uint32) nextCounter; /* This is used to determine whether or not the input bus is finding the next node in the list. Used for thread safety when detaching output buses. */
+ MA_ATOMIC(4, ma_spinlock) lock; /* Unfortunate lock, but significantly simplifies the implementation. Required for thread-safe attaching and detaching. */
+
+ /* Set once at startup. */
+ ma_uint8 channels; /* The number of channels in the audio stream for this bus. */
+};
+
+
+typedef struct ma_node_base ma_node_base;
+struct ma_node_base
+{
+ /* These variables are set once at startup. */
+ ma_node_graph* pNodeGraph; /* The graph this node belongs to. */
+ const ma_node_vtable* vtable;
+ float* pCachedData; /* Allocated on the heap. Fixed size. Needs to be stored on the heap because reading from output buses is done in separate function calls. */
+ ma_uint16 cachedDataCapInFramesPerBus; /* The capacity of the input data cache in frames, per bus. */
+
+ /* These variables are read and written only from the audio thread. */
+ ma_uint16 cachedFrameCountOut;
+ ma_uint16 cachedFrameCountIn;
+ ma_uint16 consumedFrameCountIn;
+
+ /* These variables are read and written between different threads. */
+ MA_ATOMIC(4, ma_node_state) state; /* When set to stopped, nothing will be read, regardless of the times in stateTimes. */
+ MA_ATOMIC(8, ma_uint64) stateTimes[2]; /* Indexed by ma_node_state. Specifies the time based on the global clock that a node should be considered to be in the relevant state. */
+ MA_ATOMIC(8, ma_uint64) localTime; /* The node's local clock. This is just a running sum of the number of output frames that have been processed. Can be modified by any thread with `ma_node_set_time()`. */
+ ma_uint32 inputBusCount;
+ ma_uint32 outputBusCount;
+ ma_node_input_bus* pInputBuses;
+ ma_node_output_bus* pOutputBuses;
+
+ /* Memory management. */
+ ma_node_input_bus _inputBuses[MA_MAX_NODE_LOCAL_BUS_COUNT];
+ ma_node_output_bus _outputBuses[MA_MAX_NODE_LOCAL_BUS_COUNT];
+ void* _pHeap; /* A heap allocation for internal use only. pInputBuses and/or pOutputBuses will point to this if the bus count exceeds MA_MAX_NODE_LOCAL_BUS_COUNT. */
+ ma_bool32 _ownsHeap; /* If set to true, the node owns the heap allocation and _pHeap will be freed in ma_node_uninit(). */
+};
+
+MA_API ma_result ma_node_get_heap_size(ma_node_graph* pNodeGraph, const ma_node_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_node_init_preallocated(ma_node_graph* pNodeGraph, const ma_node_config* pConfig, void* pHeap, ma_node* pNode);
+MA_API ma_result ma_node_init(ma_node_graph* pNodeGraph, const ma_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_node* pNode);
+MA_API void ma_node_uninit(ma_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_node_graph* ma_node_get_node_graph(const ma_node* pNode);
+MA_API ma_uint32 ma_node_get_input_bus_count(const ma_node* pNode);
+MA_API ma_uint32 ma_node_get_output_bus_count(const ma_node* pNode);
+MA_API ma_uint32 ma_node_get_input_channels(const ma_node* pNode, ma_uint32 inputBusIndex);
+MA_API ma_uint32 ma_node_get_output_channels(const ma_node* pNode, ma_uint32 outputBusIndex);
+MA_API ma_result ma_node_attach_output_bus(ma_node* pNode, ma_uint32 outputBusIndex, ma_node* pOtherNode, ma_uint32 otherNodeInputBusIndex);
+MA_API ma_result ma_node_detach_output_bus(ma_node* pNode, ma_uint32 outputBusIndex);
+MA_API ma_result ma_node_detach_all_output_buses(ma_node* pNode);
+MA_API ma_result ma_node_set_output_bus_volume(ma_node* pNode, ma_uint32 outputBusIndex, float volume);
+MA_API float ma_node_get_output_bus_volume(const ma_node* pNode, ma_uint32 outputBusIndex);
+MA_API ma_result ma_node_set_state(ma_node* pNode, ma_node_state state);
+MA_API ma_node_state ma_node_get_state(const ma_node* pNode);
+MA_API ma_result ma_node_set_state_time(ma_node* pNode, ma_node_state state, ma_uint64 globalTime);
+MA_API ma_uint64 ma_node_get_state_time(const ma_node* pNode, ma_node_state state);
+MA_API ma_node_state ma_node_get_state_by_time(const ma_node* pNode, ma_uint64 globalTime);
+MA_API ma_node_state ma_node_get_state_by_time_range(const ma_node* pNode, ma_uint64 globalTimeBeg, ma_uint64 globalTimeEnd);
+MA_API ma_uint64 ma_node_get_time(const ma_node* pNode);
+MA_API ma_result ma_node_set_time(ma_node* pNode, ma_uint64 localTime);
+
+
+typedef struct
+{
+ ma_uint32 channels;
+ ma_uint16 nodeCacheCapInFrames;
+} ma_node_graph_config;
+
+MA_API ma_node_graph_config ma_node_graph_config_init(ma_uint32 channels);
+
+
+struct ma_node_graph
+{
+ /* Immutable. */
+ ma_node_base base; /* The node graph itself is a node so it can be connected as an input to different node graph. This has zero inputs and calls ma_node_graph_read_pcm_frames() to generate it's output. */
+ ma_node_base endpoint; /* Special node that all nodes eventually connect to. Data is read from this node in ma_node_graph_read_pcm_frames(). */
+ ma_uint16 nodeCacheCapInFrames;
+
+ /* Read and written by multiple threads. */
+ MA_ATOMIC(4, ma_bool32) isReading;
+};
+
+MA_API ma_result ma_node_graph_init(const ma_node_graph_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_node_graph* pNodeGraph);
+MA_API void ma_node_graph_uninit(ma_node_graph* pNodeGraph, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_node* ma_node_graph_get_endpoint(ma_node_graph* pNodeGraph);
+MA_API ma_result ma_node_graph_read_pcm_frames(ma_node_graph* pNodeGraph, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
+MA_API ma_uint32 ma_node_graph_get_channels(const ma_node_graph* pNodeGraph);
+MA_API ma_uint64 ma_node_graph_get_time(const ma_node_graph* pNodeGraph);
+MA_API ma_result ma_node_graph_set_time(ma_node_graph* pNodeGraph, ma_uint64 globalTime);
+
+
+
+/* Data source node. 0 input buses, 1 output bus. Used for reading from a data source. */
+typedef struct
+{
+ ma_node_config nodeConfig;
+ ma_data_source* pDataSource;
+} ma_data_source_node_config;
+
+MA_API ma_data_source_node_config ma_data_source_node_config_init(ma_data_source* pDataSource);
+
+
+typedef struct
+{
+ ma_node_base base;
+ ma_data_source* pDataSource;
+} ma_data_source_node;
+
+MA_API ma_result ma_data_source_node_init(ma_node_graph* pNodeGraph, const ma_data_source_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source_node* pDataSourceNode);
+MA_API void ma_data_source_node_uninit(ma_data_source_node* pDataSourceNode, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_data_source_node_set_looping(ma_data_source_node* pDataSourceNode, ma_bool32 isLooping);
+MA_API ma_bool32 ma_data_source_node_is_looping(ma_data_source_node* pDataSourceNode);
+
+
+/* Splitter Node. 1 input, 2 outputs. Used for splitting/copying a stream so it can be as input into two separate output nodes. */
+typedef struct
+{
+ ma_node_config nodeConfig;
+ ma_uint32 channels;
+} ma_splitter_node_config;
+
+MA_API ma_splitter_node_config ma_splitter_node_config_init(ma_uint32 channels);
+
+
+typedef struct
+{
+ ma_node_base base;
+} ma_splitter_node;
+
+MA_API ma_result ma_splitter_node_init(ma_node_graph* pNodeGraph, const ma_splitter_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_splitter_node* pSplitterNode);
+MA_API void ma_splitter_node_uninit(ma_splitter_node* pSplitterNode, const ma_allocation_callbacks* pAllocationCallbacks);
+
+
+/*
+Biquad Node
+*/
+typedef struct
+{
+ ma_node_config nodeConfig;
+ ma_biquad_config biquad;
+} ma_biquad_node_config;
+
+MA_API ma_biquad_node_config ma_biquad_node_config_init(ma_uint32 channels, float b0, float b1, float b2, float a0, float a1, float a2);
+
+
+typedef struct
+{
+ ma_node_base baseNode;
+ ma_biquad biquad;
+} ma_biquad_node;
+
+MA_API ma_result ma_biquad_node_init(ma_node_graph* pNodeGraph, const ma_biquad_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_biquad_node* pNode);
+MA_API ma_result ma_biquad_node_reinit(const ma_biquad_config* pConfig, ma_biquad_node* pNode);
+MA_API void ma_biquad_node_uninit(ma_biquad_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks);
+
+
+/*
+Low Pass Filter Node
+*/
+typedef struct
+{
+ ma_node_config nodeConfig;
+ ma_lpf_config lpf;
+} ma_lpf_node_config;
+
+MA_API ma_lpf_node_config ma_lpf_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order);
+
+
+typedef struct
+{
+ ma_node_base baseNode;
+ ma_lpf lpf;
+} ma_lpf_node;
+
+MA_API ma_result ma_lpf_node_init(ma_node_graph* pNodeGraph, const ma_lpf_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_lpf_node* pNode);
+MA_API ma_result ma_lpf_node_reinit(const ma_lpf_config* pConfig, ma_lpf_node* pNode);
+MA_API void ma_lpf_node_uninit(ma_lpf_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks);
+
+
+/*
+High Pass Filter Node
+*/
+typedef struct
+{
+ ma_node_config nodeConfig;
+ ma_hpf_config hpf;
+} ma_hpf_node_config;
+
+MA_API ma_hpf_node_config ma_hpf_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order);
+
+
+typedef struct
+{
+ ma_node_base baseNode;
+ ma_hpf hpf;
+} ma_hpf_node;
+
+MA_API ma_result ma_hpf_node_init(ma_node_graph* pNodeGraph, const ma_hpf_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hpf_node* pNode);
+MA_API ma_result ma_hpf_node_reinit(const ma_hpf_config* pConfig, ma_hpf_node* pNode);
+MA_API void ma_hpf_node_uninit(ma_hpf_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks);
+
+
+/*
+Band Pass Filter Node
+*/
+typedef struct
+{
+ ma_node_config nodeConfig;
+ ma_bpf_config bpf;
+} ma_bpf_node_config;
+
+MA_API ma_bpf_node_config ma_bpf_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order);
+
+
+typedef struct
+{
+ ma_node_base baseNode;
+ ma_bpf bpf;
+} ma_bpf_node;
+
+MA_API ma_result ma_bpf_node_init(ma_node_graph* pNodeGraph, const ma_bpf_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_bpf_node* pNode);
+MA_API ma_result ma_bpf_node_reinit(const ma_bpf_config* pConfig, ma_bpf_node* pNode);
+MA_API void ma_bpf_node_uninit(ma_bpf_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks);
+
+
+/*
+Notching Filter Node
+*/
+typedef struct
+{
+ ma_node_config nodeConfig;
+ ma_notch_config notch;
+} ma_notch_node_config;
+
+MA_API ma_notch_node_config ma_notch_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double q, double frequency);
+
+
+typedef struct
+{
+ ma_node_base baseNode;
+ ma_notch2 notch;
+} ma_notch_node;
+
+MA_API ma_result ma_notch_node_init(ma_node_graph* pNodeGraph, const ma_notch_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_notch_node* pNode);
+MA_API ma_result ma_notch_node_reinit(const ma_notch_config* pConfig, ma_notch_node* pNode);
+MA_API void ma_notch_node_uninit(ma_notch_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks);
+
+
+/*
+Peaking Filter Node
+*/
+typedef struct
+{
+ ma_node_config nodeConfig;
+ ma_peak_config peak;
+} ma_peak_node_config;
+
+MA_API ma_peak_node_config ma_peak_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double q, double frequency);
+
+
+typedef struct
+{
+ ma_node_base baseNode;
+ ma_peak2 peak;
+} ma_peak_node;
+
+MA_API ma_result ma_peak_node_init(ma_node_graph* pNodeGraph, const ma_peak_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_peak_node* pNode);
+MA_API ma_result ma_peak_node_reinit(const ma_peak_config* pConfig, ma_peak_node* pNode);
+MA_API void ma_peak_node_uninit(ma_peak_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks);
+
+
+/*
+Low Shelf Filter Node
+*/
+typedef struct
+{
+ ma_node_config nodeConfig;
+ ma_loshelf_config loshelf;
+} ma_loshelf_node_config;
+
+MA_API ma_loshelf_node_config ma_loshelf_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double q, double frequency);
+
+
+typedef struct
+{
+ ma_node_base baseNode;
+ ma_loshelf2 loshelf;
+} ma_loshelf_node;
+
+MA_API ma_result ma_loshelf_node_init(ma_node_graph* pNodeGraph, const ma_loshelf_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_loshelf_node* pNode);
+MA_API ma_result ma_loshelf_node_reinit(const ma_loshelf_config* pConfig, ma_loshelf_node* pNode);
+MA_API void ma_loshelf_node_uninit(ma_loshelf_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks);
+
+
+/*
+High Shelf Filter Node
+*/
+typedef struct
+{
+ ma_node_config nodeConfig;
+ ma_hishelf_config hishelf;
+} ma_hishelf_node_config;
+
+MA_API ma_hishelf_node_config ma_hishelf_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double q, double frequency);
+
+
+typedef struct
+{
+ ma_node_base baseNode;
+ ma_hishelf2 hishelf;
+} ma_hishelf_node;
+
+MA_API ma_result ma_hishelf_node_init(ma_node_graph* pNodeGraph, const ma_hishelf_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hishelf_node* pNode);
+MA_API ma_result ma_hishelf_node_reinit(const ma_hishelf_config* pConfig, ma_hishelf_node* pNode);
+MA_API void ma_hishelf_node_uninit(ma_hishelf_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks);
+
+
+typedef struct
+{
+ ma_node_config nodeConfig;
+ ma_delay_config delay;
+} ma_delay_node_config;
+
+MA_API ma_delay_node_config ma_delay_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 delayInFrames, float decay);
+
+
+typedef struct
+{
+ ma_node_base baseNode;
+ ma_delay delay;
+} ma_delay_node;
+
+MA_API ma_result ma_delay_node_init(ma_node_graph* pNodeGraph, const ma_delay_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_delay_node* pDelayNode);
+MA_API void ma_delay_node_uninit(ma_delay_node* pDelayNode, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API void ma_delay_node_set_wet(ma_delay_node* pDelayNode, float value);
+MA_API float ma_delay_node_get_wet(const ma_delay_node* pDelayNode);
+MA_API void ma_delay_node_set_dry(ma_delay_node* pDelayNode, float value);
+MA_API float ma_delay_node_get_dry(const ma_delay_node* pDelayNode);
+MA_API void ma_delay_node_set_decay(ma_delay_node* pDelayNode, float value);
+MA_API float ma_delay_node_get_decay(const ma_delay_node* pDelayNode);
+#endif /* MA_NO_NODE_GRAPH */
+
+
+/************************************************************************************************************************************************************
+
+Engine
+
+************************************************************************************************************************************************************/
+#if !defined(MA_NO_ENGINE) && !defined(MA_NO_NODE_GRAPH)
+typedef struct ma_engine ma_engine;
+typedef struct ma_sound ma_sound;
+
+
+/* Sound flags. */
+typedef enum
+{
+ MA_SOUND_FLAG_STREAM = 0x00000001, /* MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM */
+ MA_SOUND_FLAG_DECODE = 0x00000002, /* MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE */
+ MA_SOUND_FLAG_ASYNC = 0x00000004, /* MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC */
+ MA_SOUND_FLAG_WAIT_INIT = 0x00000008, /* MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT */
+ MA_SOUND_FLAG_NO_DEFAULT_ATTACHMENT = 0x00000010, /* Do not attach to the endpoint by default. Useful for when setting up nodes in a complex graph system. */
+ MA_SOUND_FLAG_NO_PITCH = 0x00000020, /* Disable pitch shifting with ma_sound_set_pitch() and ma_sound_group_set_pitch(). This is an optimization. */
+ MA_SOUND_FLAG_NO_SPATIALIZATION = 0x00000040 /* Disable spatialization. */
+} ma_sound_flags;
+
+#ifndef MA_ENGINE_MAX_LISTENERS
+#define MA_ENGINE_MAX_LISTENERS 4
+#endif
+
+#define MA_LISTENER_INDEX_CLOSEST ((ma_uint8)-1)
+
+typedef enum
+{
+ ma_engine_node_type_sound,
+ ma_engine_node_type_group
+} ma_engine_node_type;
+
+typedef struct
+{
+ ma_engine* pEngine;
+ ma_engine_node_type type;
+ ma_uint32 channelsIn;
+ ma_uint32 channelsOut;
+ ma_uint32 sampleRate; /* Only used when the type is set to ma_engine_node_type_sound. */
+ ma_bool8 isPitchDisabled; /* Pitching can be explicitly disable with MA_SOUND_FLAG_NO_PITCH to optimize processing. */
+ ma_bool8 isSpatializationDisabled; /* Spatialization can be explicitly disabled with MA_SOUND_FLAG_NO_SPATIALIZATION. */
+ ma_uint8 pinnedListenerIndex; /* The index of the listener this node should always use for spatialization. If set to MA_LISTENER_INDEX_CLOSEST the engine will use the closest listener. */
+} ma_engine_node_config;
+
+MA_API ma_engine_node_config ma_engine_node_config_init(ma_engine* pEngine, ma_engine_node_type type, ma_uint32 flags);
+
+
+/* Base node object for both ma_sound and ma_sound_group. */
+typedef struct
+{
+ ma_node_base baseNode; /* Must be the first member for compatiblity with the ma_node API. */
+ ma_engine* pEngine; /* A pointer to the engine. Set based on the value from the config. */
+ ma_uint32 sampleRate; /* The sample rate of the input data. For sounds backed by a data source, this will be the data source's sample rate. Otherwise it'll be the engine's sample rate. */
+ ma_fader fader;
+ ma_linear_resampler resampler; /* For pitch shift. */
+ ma_spatializer spatializer;
+ ma_panner panner;
+ MA_ATOMIC(4, float) pitch;
+ float oldPitch; /* For determining whether or not the resampler needs to be updated to reflect the new pitch. The resampler will be updated on the mixing thread. */
+ float oldDopplerPitch; /* For determining whether or not the resampler needs to be updated to take a new doppler pitch into account. */
+ MA_ATOMIC(4, ma_bool32) isPitchDisabled; /* When set to true, pitching will be disabled which will allow the resampler to be bypassed to save some computation. */
+ MA_ATOMIC(4, ma_bool32) isSpatializationDisabled; /* Set to false by default. When set to false, will not have spatialisation applied. */
+ MA_ATOMIC(4, ma_uint32) pinnedListenerIndex; /* The index of the listener this node should always use for spatialization. If set to MA_LISTENER_INDEX_CLOSEST the engine will use the closest listener. */
+
+ /* Memory management. */
+ ma_bool8 _ownsHeap;
+ void* _pHeap;
+} ma_engine_node;
+
+MA_API ma_result ma_engine_node_get_heap_size(const ma_engine_node_config* pConfig, size_t* pHeapSizeInBytes);
+MA_API ma_result ma_engine_node_init_preallocated(const ma_engine_node_config* pConfig, void* pHeap, ma_engine_node* pEngineNode);
+MA_API ma_result ma_engine_node_init(const ma_engine_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_engine_node* pEngineNode);
+MA_API void ma_engine_node_uninit(ma_engine_node* pEngineNode, const ma_allocation_callbacks* pAllocationCallbacks);
+
+
+#define MA_SOUND_SOURCE_CHANNEL_COUNT 0xFFFFFFFF
+
+typedef struct
+{
+ const char* pFilePath; /* Set this to load from the resource manager. */
+ const wchar_t* pFilePathW; /* Set this to load from the resource manager. */
+ ma_data_source* pDataSource; /* Set this to load from an existing data source. */
+ ma_node* pInitialAttachment; /* If set, the sound will be attached to an input of this node. This can be set to a ma_sound. If set to NULL, the sound will be attached directly to the endpoint unless MA_SOUND_FLAG_NO_DEFAULT_ATTACHMENT is set in `flags`. */
+ ma_uint32 initialAttachmentInputBusIndex; /* The index of the input bus of pInitialAttachment to attach the sound to. */
+ ma_uint32 channelsIn; /* Ignored if using a data source as input (the data source's channel count will be used always). Otherwise, setting to 0 will cause the engine's channel count to be used. */
+ ma_uint32 channelsOut; /* Set this to 0 (default) to use the engine's channel count. Set to MA_SOUND_SOURCE_CHANNEL_COUNT to use the data source's channel count (only used if using a data source as input). */
+ ma_uint32 flags; /* A combination of MA_SOUND_FLAG_* flags. */
+ ma_uint64 initialSeekPointInPCMFrames; /* Initializes the sound such that it's seeked to this location by default. */
+ ma_uint64 rangeBegInPCMFrames;
+ ma_uint64 rangeEndInPCMFrames;
+ ma_uint64 loopPointBegInPCMFrames;
+ ma_uint64 loopPointEndInPCMFrames;
+ ma_bool32 isLooping;
+ ma_fence* pDoneFence; /* Released when the resource manager has finished decoding the entire sound. Not used with streams. */
+} ma_sound_config;
+
+MA_API ma_sound_config ma_sound_config_init(void);
+
+struct ma_sound
+{
+ ma_engine_node engineNode; /* Must be the first member for compatibility with the ma_node API. */
+ ma_data_source* pDataSource;
+ ma_uint64 seekTarget; /* The PCM frame index to seek to in the mixing thread. Set to (~(ma_uint64)0) to not perform any seeking. */
+ MA_ATOMIC(4, ma_bool32) atEnd;
+ ma_bool8 ownsDataSource;
+
+ /*
+ We're declaring a resource manager data source object here to save us a malloc when loading a
+ sound via the resource manager, which I *think* will be the most common scenario.
+ */
+#ifndef MA_NO_RESOURCE_MANAGER
+ ma_resource_manager_data_source* pResourceManagerDataSource;
+#endif
+};
+
+/* Structure specifically for sounds played with ma_engine_play_sound(). Making this a separate structure to reduce overhead. */
+typedef struct ma_sound_inlined ma_sound_inlined;
+struct ma_sound_inlined
+{
+ ma_sound sound;
+ ma_sound_inlined* pNext;
+ ma_sound_inlined* pPrev;
+};
+
+/* A sound group is just a sound. */
+typedef ma_sound_config ma_sound_group_config;
+typedef ma_sound ma_sound_group;
+
+MA_API ma_sound_group_config ma_sound_group_config_init(void);
+
+
+typedef struct
+{
+#if !defined(MA_NO_RESOURCE_MANAGER)
+ ma_resource_manager* pResourceManager; /* Can be null in which case a resource manager will be created for you. */
+#endif
+#if !defined(MA_NO_DEVICE_IO)
+ ma_context* pContext;
+ ma_device* pDevice; /* If set, the caller is responsible for calling ma_engine_data_callback() in the device's data callback. */
+ ma_device_id* pPlaybackDeviceID; /* The ID of the playback device to use with the default listener. */
+#endif
+ ma_log* pLog; /* When set to NULL, will use the context's log. */
+ ma_uint32 listenerCount; /* Must be between 1 and MA_ENGINE_MAX_LISTENERS. */
+ ma_uint32 channels; /* The number of channels to use when mixing and spatializing. When set to 0, will use the native channel count of the device. */
+ ma_uint32 sampleRate; /* The sample rate. When set to 0 will use the native channel count of the device. */
+ ma_uint32 periodSizeInFrames; /* If set to something other than 0, updates will always be exactly this size. The underlying device may be a different size, but from the perspective of the mixer that won't matter.*/
+ ma_uint32 periodSizeInMilliseconds; /* Used if periodSizeInFrames is unset. */
+ ma_uint32 gainSmoothTimeInFrames; /* The number of frames to interpolate the gain of spatialized sounds across. If set to 0, will use gainSmoothTimeInMilliseconds. */
+ ma_uint32 gainSmoothTimeInMilliseconds; /* When set to 0, gainSmoothTimeInFrames will be used. If both are set to 0, a default value will be used. */
+ ma_allocation_callbacks allocationCallbacks;
+ ma_bool32 noAutoStart; /* When set to true, requires an explicit call to ma_engine_start(). This is false by default, meaning the engine will be started automatically in ma_engine_init(). */
+ ma_bool32 noDevice; /* When set to true, don't create a default device. ma_engine_read_pcm_frames() can be called manually to read data. */
+ ma_mono_expansion_mode monoExpansionMode; /* Controls how the mono channel should be expanded to other channels when spatialization is disabled on a sound. */
+ ma_vfs* pResourceManagerVFS; /* A pointer to a pre-allocated VFS object to use with the resource manager. This is ignored if pResourceManager is not NULL. */
+} ma_engine_config;
+
+MA_API ma_engine_config ma_engine_config_init(void);
+
+
+struct ma_engine
+{
+ ma_node_graph nodeGraph; /* An engine is a node graph. It should be able to be plugged into any ma_node_graph API (with a cast) which means this must be the first member of this struct. */
+#if !defined(MA_NO_RESOURCE_MANAGER)
+ ma_resource_manager* pResourceManager;
+#endif
+#if !defined(MA_NO_DEVICE_IO)
+ ma_device* pDevice; /* Optionally set via the config, otherwise allocated by the engine in ma_engine_init(). */
+#endif
+ ma_log* pLog;
+ ma_uint32 sampleRate;
+ ma_uint32 listenerCount;
+ ma_spatializer_listener listeners[MA_ENGINE_MAX_LISTENERS];
+ ma_allocation_callbacks allocationCallbacks;
+ ma_bool8 ownsResourceManager;
+ ma_bool8 ownsDevice;
+ ma_spinlock inlinedSoundLock; /* For synchronizing access so the inlined sound list. */
+ ma_sound_inlined* pInlinedSoundHead; /* The first inlined sound. Inlined sounds are tracked in a linked list. */
+ MA_ATOMIC(4, ma_uint32) inlinedSoundCount; /* The total number of allocated inlined sound objects. Used for debugging. */
+ ma_uint32 gainSmoothTimeInFrames; /* The number of frames to interpolate the gain of spatialized sounds across. */
+ ma_mono_expansion_mode monoExpansionMode;
+};
+
+MA_API ma_result ma_engine_init(const ma_engine_config* pConfig, ma_engine* pEngine);
+MA_API void ma_engine_uninit(ma_engine* pEngine);
+MA_API ma_result ma_engine_read_pcm_frames(ma_engine* pEngine, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
+MA_API ma_node_graph* ma_engine_get_node_graph(ma_engine* pEngine);
+#if !defined(MA_NO_RESOURCE_MANAGER)
+MA_API ma_resource_manager* ma_engine_get_resource_manager(ma_engine* pEngine);
+#endif
+MA_API ma_device* ma_engine_get_device(ma_engine* pEngine);
+MA_API ma_log* ma_engine_get_log(ma_engine* pEngine);
+MA_API ma_node* ma_engine_get_endpoint(ma_engine* pEngine);
+MA_API ma_uint64 ma_engine_get_time(const ma_engine* pEngine);
+MA_API ma_result ma_engine_set_time(ma_engine* pEngine, ma_uint64 globalTime);
+MA_API ma_uint32 ma_engine_get_channels(const ma_engine* pEngine);
+MA_API ma_uint32 ma_engine_get_sample_rate(const ma_engine* pEngine);
+
+MA_API ma_result ma_engine_start(ma_engine* pEngine);
+MA_API ma_result ma_engine_stop(ma_engine* pEngine);
+MA_API ma_result ma_engine_set_volume(ma_engine* pEngine, float volume);
+MA_API ma_result ma_engine_set_gain_db(ma_engine* pEngine, float gainDB);
+
+MA_API ma_uint32 ma_engine_get_listener_count(const ma_engine* pEngine);
+MA_API ma_uint32 ma_engine_find_closest_listener(const ma_engine* pEngine, float absolutePosX, float absolutePosY, float absolutePosZ);
+MA_API void ma_engine_listener_set_position(ma_engine* pEngine, ma_uint32 listenerIndex, float x, float y, float z);
+MA_API ma_vec3f ma_engine_listener_get_position(const ma_engine* pEngine, ma_uint32 listenerIndex);
+MA_API void ma_engine_listener_set_direction(ma_engine* pEngine, ma_uint32 listenerIndex, float x, float y, float z);
+MA_API ma_vec3f ma_engine_listener_get_direction(const ma_engine* pEngine, ma_uint32 listenerIndex);
+MA_API void ma_engine_listener_set_velocity(ma_engine* pEngine, ma_uint32 listenerIndex, float x, float y, float z);
+MA_API ma_vec3f ma_engine_listener_get_velocity(const ma_engine* pEngine, ma_uint32 listenerIndex);
+MA_API void ma_engine_listener_set_cone(ma_engine* pEngine, ma_uint32 listenerIndex, float innerAngleInRadians, float outerAngleInRadians, float outerGain);
+MA_API void ma_engine_listener_get_cone(const ma_engine* pEngine, ma_uint32 listenerIndex, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain);
+MA_API void ma_engine_listener_set_world_up(ma_engine* pEngine, ma_uint32 listenerIndex, float x, float y, float z);
+MA_API ma_vec3f ma_engine_listener_get_world_up(const ma_engine* pEngine, ma_uint32 listenerIndex);
+MA_API void ma_engine_listener_set_enabled(ma_engine* pEngine, ma_uint32 listenerIndex, ma_bool32 isEnabled);
+MA_API ma_bool32 ma_engine_listener_is_enabled(const ma_engine* pEngine, ma_uint32 listenerIndex);
+
+#ifndef MA_NO_RESOURCE_MANAGER
+MA_API ma_result ma_engine_play_sound_ex(ma_engine* pEngine, const char* pFilePath, ma_node* pNode, ma_uint32 nodeInputBusIndex);
+MA_API ma_result ma_engine_play_sound(ma_engine* pEngine, const char* pFilePath, ma_sound_group* pGroup); /* Fire and forget. */
+#endif
+
+#ifndef MA_NO_RESOURCE_MANAGER
+MA_API ma_result ma_sound_init_from_file(ma_engine* pEngine, const char* pFilePath, ma_uint32 flags, ma_sound_group* pGroup, ma_fence* pDoneFence, ma_sound* pSound);
+MA_API ma_result ma_sound_init_from_file_w(ma_engine* pEngine, const wchar_t* pFilePath, ma_uint32 flags, ma_sound_group* pGroup, ma_fence* pDoneFence, ma_sound* pSound);
+MA_API ma_result ma_sound_init_copy(ma_engine* pEngine, const ma_sound* pExistingSound, ma_uint32 flags, ma_sound_group* pGroup, ma_sound* pSound);
+#endif
+MA_API ma_result ma_sound_init_from_data_source(ma_engine* pEngine, ma_data_source* pDataSource, ma_uint32 flags, ma_sound_group* pGroup, ma_sound* pSound);
+MA_API ma_result ma_sound_init_ex(ma_engine* pEngine, const ma_sound_config* pConfig, ma_sound* pSound);
+MA_API void ma_sound_uninit(ma_sound* pSound);
+MA_API ma_engine* ma_sound_get_engine(const ma_sound* pSound);
+MA_API ma_data_source* ma_sound_get_data_source(const ma_sound* pSound);
+MA_API ma_result ma_sound_start(ma_sound* pSound);
+MA_API ma_result ma_sound_stop(ma_sound* pSound);
+MA_API void ma_sound_set_volume(ma_sound* pSound, float volume);
+MA_API float ma_sound_get_volume(const ma_sound* pSound);
+MA_API void ma_sound_set_pan(ma_sound* pSound, float pan);
+MA_API float ma_sound_get_pan(const ma_sound* pSound);
+MA_API void ma_sound_set_pan_mode(ma_sound* pSound, ma_pan_mode panMode);
+MA_API ma_pan_mode ma_sound_get_pan_mode(const ma_sound* pSound);
+MA_API void ma_sound_set_pitch(ma_sound* pSound, float pitch);
+MA_API float ma_sound_get_pitch(const ma_sound* pSound);
+MA_API void ma_sound_set_spatialization_enabled(ma_sound* pSound, ma_bool32 enabled);
+MA_API ma_bool32 ma_sound_is_spatialization_enabled(const ma_sound* pSound);
+MA_API void ma_sound_set_pinned_listener_index(ma_sound* pSound, ma_uint32 listenerIndex);
+MA_API ma_uint32 ma_sound_get_pinned_listener_index(const ma_sound* pSound);
+MA_API ma_uint32 ma_sound_get_listener_index(const ma_sound* pSound);
+MA_API ma_vec3f ma_sound_get_direction_to_listener(const ma_sound* pSound);
+MA_API void ma_sound_set_position(ma_sound* pSound, float x, float y, float z);
+MA_API ma_vec3f ma_sound_get_position(const ma_sound* pSound);
+MA_API void ma_sound_set_direction(ma_sound* pSound, float x, float y, float z);
+MA_API ma_vec3f ma_sound_get_direction(const ma_sound* pSound);
+MA_API void ma_sound_set_velocity(ma_sound* pSound, float x, float y, float z);
+MA_API ma_vec3f ma_sound_get_velocity(const ma_sound* pSound);
+MA_API void ma_sound_set_attenuation_model(ma_sound* pSound, ma_attenuation_model attenuationModel);
+MA_API ma_attenuation_model ma_sound_get_attenuation_model(const ma_sound* pSound);
+MA_API void ma_sound_set_positioning(ma_sound* pSound, ma_positioning positioning);
+MA_API ma_positioning ma_sound_get_positioning(const ma_sound* pSound);
+MA_API void ma_sound_set_rolloff(ma_sound* pSound, float rolloff);
+MA_API float ma_sound_get_rolloff(const ma_sound* pSound);
+MA_API void ma_sound_set_min_gain(ma_sound* pSound, float minGain);
+MA_API float ma_sound_get_min_gain(const ma_sound* pSound);
+MA_API void ma_sound_set_max_gain(ma_sound* pSound, float maxGain);
+MA_API float ma_sound_get_max_gain(const ma_sound* pSound);
+MA_API void ma_sound_set_min_distance(ma_sound* pSound, float minDistance);
+MA_API float ma_sound_get_min_distance(const ma_sound* pSound);
+MA_API void ma_sound_set_max_distance(ma_sound* pSound, float maxDistance);
+MA_API float ma_sound_get_max_distance(const ma_sound* pSound);
+MA_API void ma_sound_set_cone(ma_sound* pSound, float innerAngleInRadians, float outerAngleInRadians, float outerGain);
+MA_API void ma_sound_get_cone(const ma_sound* pSound, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain);
+MA_API void ma_sound_set_doppler_factor(ma_sound* pSound, float dopplerFactor);
+MA_API float ma_sound_get_doppler_factor(const ma_sound* pSound);
+MA_API void ma_sound_set_directional_attenuation_factor(ma_sound* pSound, float directionalAttenuationFactor);
+MA_API float ma_sound_get_directional_attenuation_factor(const ma_sound* pSound);
+MA_API void ma_sound_set_fade_in_pcm_frames(ma_sound* pSound, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInFrames);
+MA_API void ma_sound_set_fade_in_milliseconds(ma_sound* pSound, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInMilliseconds);
+MA_API float ma_sound_get_current_fade_volume(ma_sound* pSound);
+MA_API void ma_sound_set_start_time_in_pcm_frames(ma_sound* pSound, ma_uint64 absoluteGlobalTimeInFrames);
+MA_API void ma_sound_set_start_time_in_milliseconds(ma_sound* pSound, ma_uint64 absoluteGlobalTimeInMilliseconds);
+MA_API void ma_sound_set_stop_time_in_pcm_frames(ma_sound* pSound, ma_uint64 absoluteGlobalTimeInFrames);
+MA_API void ma_sound_set_stop_time_in_milliseconds(ma_sound* pSound, ma_uint64 absoluteGlobalTimeInMilliseconds);
+MA_API ma_bool32 ma_sound_is_playing(const ma_sound* pSound);
+MA_API ma_uint64 ma_sound_get_time_in_pcm_frames(const ma_sound* pSound);
+MA_API void ma_sound_set_looping(ma_sound* pSound, ma_bool32 isLooping);
+MA_API ma_bool32 ma_sound_is_looping(const ma_sound* pSound);
+MA_API ma_bool32 ma_sound_at_end(const ma_sound* pSound);
+MA_API ma_result ma_sound_seek_to_pcm_frame(ma_sound* pSound, ma_uint64 frameIndex); /* Just a wrapper around ma_data_source_seek_to_pcm_frame(). */
+MA_API ma_result ma_sound_get_data_format(ma_sound* pSound, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
+MA_API ma_result ma_sound_get_cursor_in_pcm_frames(ma_sound* pSound, ma_uint64* pCursor);
+MA_API ma_result ma_sound_get_length_in_pcm_frames(ma_sound* pSound, ma_uint64* pLength);
+MA_API ma_result ma_sound_get_cursor_in_seconds(ma_sound* pSound, float* pCursor);
+MA_API ma_result ma_sound_get_length_in_seconds(ma_sound* pSound, float* pLength);
+
+MA_API ma_result ma_sound_group_init(ma_engine* pEngine, ma_uint32 flags, ma_sound_group* pParentGroup, ma_sound_group* pGroup);
+MA_API ma_result ma_sound_group_init_ex(ma_engine* pEngine, const ma_sound_group_config* pConfig, ma_sound_group* pGroup);
+MA_API void ma_sound_group_uninit(ma_sound_group* pGroup);
+MA_API ma_engine* ma_sound_group_get_engine(const ma_sound_group* pGroup);
+MA_API ma_result ma_sound_group_start(ma_sound_group* pGroup);
+MA_API ma_result ma_sound_group_stop(ma_sound_group* pGroup);
+MA_API void ma_sound_group_set_volume(ma_sound_group* pGroup, float volume);
+MA_API float ma_sound_group_get_volume(const ma_sound_group* pGroup);
+MA_API void ma_sound_group_set_pan(ma_sound_group* pGroup, float pan);
+MA_API float ma_sound_group_get_pan(const ma_sound_group* pGroup);
+MA_API void ma_sound_group_set_pan_mode(ma_sound_group* pGroup, ma_pan_mode panMode);
+MA_API ma_pan_mode ma_sound_group_get_pan_mode(const ma_sound_group* pGroup);
+MA_API void ma_sound_group_set_pitch(ma_sound_group* pGroup, float pitch);
+MA_API float ma_sound_group_get_pitch(const ma_sound_group* pGroup);
+MA_API void ma_sound_group_set_spatialization_enabled(ma_sound_group* pGroup, ma_bool32 enabled);
+MA_API ma_bool32 ma_sound_group_is_spatialization_enabled(const ma_sound_group* pGroup);
+MA_API void ma_sound_group_set_pinned_listener_index(ma_sound_group* pGroup, ma_uint32 listenerIndex);
+MA_API ma_uint32 ma_sound_group_get_pinned_listener_index(const ma_sound_group* pGroup);
+MA_API ma_uint32 ma_sound_group_get_listener_index(const ma_sound_group* pGroup);
+MA_API ma_vec3f ma_sound_group_get_direction_to_listener(const ma_sound_group* pGroup);
+MA_API void ma_sound_group_set_position(ma_sound_group* pGroup, float x, float y, float z);
+MA_API ma_vec3f ma_sound_group_get_position(const ma_sound_group* pGroup);
+MA_API void ma_sound_group_set_direction(ma_sound_group* pGroup, float x, float y, float z);
+MA_API ma_vec3f ma_sound_group_get_direction(const ma_sound_group* pGroup);
+MA_API void ma_sound_group_set_velocity(ma_sound_group* pGroup, float x, float y, float z);
+MA_API ma_vec3f ma_sound_group_get_velocity(const ma_sound_group* pGroup);
+MA_API void ma_sound_group_set_attenuation_model(ma_sound_group* pGroup, ma_attenuation_model attenuationModel);
+MA_API ma_attenuation_model ma_sound_group_get_attenuation_model(const ma_sound_group* pGroup);
+MA_API void ma_sound_group_set_positioning(ma_sound_group* pGroup, ma_positioning positioning);
+MA_API ma_positioning ma_sound_group_get_positioning(const ma_sound_group* pGroup);
+MA_API void ma_sound_group_set_rolloff(ma_sound_group* pGroup, float rolloff);
+MA_API float ma_sound_group_get_rolloff(const ma_sound_group* pGroup);
+MA_API void ma_sound_group_set_min_gain(ma_sound_group* pGroup, float minGain);
+MA_API float ma_sound_group_get_min_gain(const ma_sound_group* pGroup);
+MA_API void ma_sound_group_set_max_gain(ma_sound_group* pGroup, float maxGain);
+MA_API float ma_sound_group_get_max_gain(const ma_sound_group* pGroup);
+MA_API void ma_sound_group_set_min_distance(ma_sound_group* pGroup, float minDistance);
+MA_API float ma_sound_group_get_min_distance(const ma_sound_group* pGroup);
+MA_API void ma_sound_group_set_max_distance(ma_sound_group* pGroup, float maxDistance);
+MA_API float ma_sound_group_get_max_distance(const ma_sound_group* pGroup);
+MA_API void ma_sound_group_set_cone(ma_sound_group* pGroup, float innerAngleInRadians, float outerAngleInRadians, float outerGain);
+MA_API void ma_sound_group_get_cone(const ma_sound_group* pGroup, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain);
+MA_API void ma_sound_group_set_doppler_factor(ma_sound_group* pGroup, float dopplerFactor);
+MA_API float ma_sound_group_get_doppler_factor(const ma_sound_group* pGroup);
+MA_API void ma_sound_group_set_directional_attenuation_factor(ma_sound_group* pGroup, float directionalAttenuationFactor);
+MA_API float ma_sound_group_get_directional_attenuation_factor(const ma_sound_group* pGroup);
+MA_API void ma_sound_group_set_fade_in_pcm_frames(ma_sound_group* pGroup, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInFrames);
+MA_API void ma_sound_group_set_fade_in_milliseconds(ma_sound_group* pGroup, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInMilliseconds);
+MA_API float ma_sound_group_get_current_fade_volume(ma_sound_group* pGroup);
+MA_API void ma_sound_group_set_start_time_in_pcm_frames(ma_sound_group* pGroup, ma_uint64 absoluteGlobalTimeInFrames);
+MA_API void ma_sound_group_set_start_time_in_milliseconds(ma_sound_group* pGroup, ma_uint64 absoluteGlobalTimeInMilliseconds);
+MA_API void ma_sound_group_set_stop_time_in_pcm_frames(ma_sound_group* pGroup, ma_uint64 absoluteGlobalTimeInFrames);
+MA_API void ma_sound_group_set_stop_time_in_milliseconds(ma_sound_group* pGroup, ma_uint64 absoluteGlobalTimeInMilliseconds);
+MA_API ma_bool32 ma_sound_group_is_playing(const ma_sound_group* pGroup);
+MA_API ma_uint64 ma_sound_group_get_time_in_pcm_frames(const ma_sound_group* pGroup);
+#endif /* MA_NO_ENGINE */
+
+#ifdef __cplusplus
+}
+#endif
+#endif /* miniaudio_h */
+
+
+/*
+This is for preventing greying out of the implementation section.
+*/
+#if defined(Q_CREATOR_RUN) || defined(__INTELLISENSE__) || defined(__CDT_PARSER__)
+#define MINIAUDIO_IMPLEMENTATION
+#endif
+
+/************************************************************************************************************************************************************
+*************************************************************************************************************************************************************
+
+IMPLEMENTATION
+
+*************************************************************************************************************************************************************
+************************************************************************************************************************************************************/
+#if defined(MINIAUDIO_IMPLEMENTATION) || defined(MA_IMPLEMENTATION)
+#ifndef miniaudio_c
+#define miniaudio_c
+
+#include
+#include /* For INT_MAX */
+#include /* sin(), etc. */
+
+#include
+#include
+#if !defined(_MSC_VER) && !defined(__DMC__)
+ #include /* For strcasecmp(). */
+ #include /* For wcslen(), wcsrtombs() */
+#endif
+#ifdef _MSC_VER
+ #include /* For _controlfp_s constants */
+#endif
+
+#ifdef MA_WIN32
+#include
+#else
+#include /* For malloc(), free(), wcstombs(). */
+#include /* For memset() */
+#include
+#include /* select() (used for ma_sleep()). */
+#include
+#endif
+
+#include /* For fstat(), etc. */
+
+#ifdef MA_EMSCRIPTEN
+#include
+#endif
+
+#if !defined(MA_64BIT) && !defined(MA_32BIT)
+#ifdef _WIN32
+#ifdef _WIN64
+#define MA_64BIT
+#else
+#define MA_32BIT
+#endif
+#endif
+#endif
+
+#if !defined(MA_64BIT) && !defined(MA_32BIT)
+#ifdef __GNUC__
+#ifdef __LP64__
+#define MA_64BIT
+#else
+#define MA_32BIT
+#endif
+#endif
+#endif
+
+#if !defined(MA_64BIT) && !defined(MA_32BIT)
+#include
+#if INTPTR_MAX == INT64_MAX
+#define MA_64BIT
+#else
+#define MA_32BIT
+#endif
+#endif
+
+/* Architecture Detection */
+#if defined(__x86_64__) || defined(_M_X64)
+#define MA_X64
+#elif defined(__i386) || defined(_M_IX86)
+#define MA_X86
+#elif defined(__arm__) || defined(_M_ARM) || defined(__arm64) || defined(__arm64__) || defined(__aarch64__) || defined(_M_ARM64)
+#define MA_ARM
+#endif
+
+/* Intrinsics Support */
+#if defined(MA_X64) || defined(MA_X86)
+ #if defined(_MSC_VER) && !defined(__clang__)
+ /* MSVC. */
+ #if _MSC_VER >= 1400 && !defined(MA_NO_SSE2) /* 2005 */
+ #define MA_SUPPORT_SSE2
+ #endif
+ /*#if _MSC_VER >= 1600 && !defined(MA_NO_AVX)*/ /* 2010 */
+ /* #define MA_SUPPORT_AVX*/
+ /*#endif*/
+ #if _MSC_VER >= 1700 && !defined(MA_NO_AVX2) /* 2012 */
+ #define MA_SUPPORT_AVX2
+ #endif
+ #else
+ /* Assume GNUC-style. */
+ #if defined(__SSE2__) && !defined(MA_NO_SSE2)
+ #define MA_SUPPORT_SSE2
+ #endif
+ /*#if defined(__AVX__) && !defined(MA_NO_AVX)*/
+ /* #define MA_SUPPORT_AVX*/
+ /*#endif*/
+ #if defined(__AVX2__) && !defined(MA_NO_AVX2)
+ #define MA_SUPPORT_AVX2
+ #endif
+ #endif
+
+ /* If at this point we still haven't determined compiler support for the intrinsics just fall back to __has_include. */
+ #if !defined(__GNUC__) && !defined(__clang__) && defined(__has_include)
+ #if !defined(MA_SUPPORT_SSE2) && !defined(MA_NO_SSE2) && __has_include()
+ #define MA_SUPPORT_SSE2
+ #endif
+ /*#if !defined(MA_SUPPORT_AVX) && !defined(MA_NO_AVX) && __has_include()*/
+ /* #define MA_SUPPORT_AVX*/
+ /*#endif*/
+ #if !defined(MA_SUPPORT_AVX2) && !defined(MA_NO_AVX2) && __has_include()
+ #define MA_SUPPORT_AVX2
+ #endif
+ #endif
+
+ #if defined(MA_SUPPORT_AVX2) || defined(MA_SUPPORT_AVX)
+ #include
+ #elif defined(MA_SUPPORT_SSE2)
+ #include
+ #endif
+#endif
+
+#if defined(MA_ARM)
+ #if !defined(MA_NO_NEON) && (defined(__ARM_NEON) || defined(__aarch64__) || defined(_M_ARM64))
+ #define MA_SUPPORT_NEON
+ #include
+ #endif
+#endif
+
+/* Begin globally disabled warnings. */
+#if defined(_MSC_VER)
+ #pragma warning(push)
+ #pragma warning(disable:4752) /* found Intel(R) Advanced Vector Extensions; consider using /arch:AVX */
+ #pragma warning(disable:4049) /* compiler limit : terminating line number emission */
+#endif
+
+#if defined(MA_X64) || defined(MA_X86)
+ #if defined(_MSC_VER) && !defined(__clang__)
+ #if _MSC_VER >= 1400
+ #include
+ static MA_INLINE void ma_cpuid(int info[4], int fid)
+ {
+ __cpuid(info, fid);
+ }
+ #else
+ #define MA_NO_CPUID
+ #endif
+
+ #if _MSC_VER >= 1600 && (defined(_MSC_FULL_VER) && _MSC_FULL_VER >= 160040219)
+ static MA_INLINE unsigned __int64 ma_xgetbv(int reg)
+ {
+ return _xgetbv(reg);
+ }
+ #else
+ #define MA_NO_XGETBV
+ #endif
+ #elif (defined(__GNUC__) || defined(__clang__)) && !defined(MA_ANDROID)
+ static MA_INLINE void ma_cpuid(int info[4], int fid)
+ {
+ /*
+ It looks like the -fPIC option uses the ebx register which GCC complains about. We can work around this by just using a different register, the
+ specific register of which I'm letting the compiler decide on. The "k" prefix is used to specify a 32-bit register. The {...} syntax is for
+ supporting different assembly dialects.
+
+ What's basically happening is that we're saving and restoring the ebx register manually.
+ */
+ #if defined(DRFLAC_X86) && defined(__PIC__)
+ __asm__ __volatile__ (
+ "xchg{l} {%%}ebx, %k1;"
+ "cpuid;"
+ "xchg{l} {%%}ebx, %k1;"
+ : "=a"(info[0]), "=&r"(info[1]), "=c"(info[2]), "=d"(info[3]) : "a"(fid), "c"(0)
+ );
+ #else
+ __asm__ __volatile__ (
+ "cpuid" : "=a"(info[0]), "=b"(info[1]), "=c"(info[2]), "=d"(info[3]) : "a"(fid), "c"(0)
+ );
+ #endif
+ }
+
+ static MA_INLINE ma_uint64 ma_xgetbv(int reg)
+ {
+ unsigned int hi;
+ unsigned int lo;
+
+ __asm__ __volatile__ (
+ "xgetbv" : "=a"(lo), "=d"(hi) : "c"(reg)
+ );
+
+ return ((ma_uint64)hi << 32) | (ma_uint64)lo;
+ }
+ #else
+ #define MA_NO_CPUID
+ #define MA_NO_XGETBV
+ #endif
+#else
+ #define MA_NO_CPUID
+ #define MA_NO_XGETBV
+#endif
+
+static MA_INLINE ma_bool32 ma_has_sse2(void)
+{
+#if defined(MA_SUPPORT_SSE2)
+ #if (defined(MA_X64) || defined(MA_X86)) && !defined(MA_NO_SSE2)
+ #if defined(MA_X64)
+ return MA_TRUE; /* 64-bit targets always support SSE2. */
+ #elif (defined(_M_IX86_FP) && _M_IX86_FP == 2) || defined(__SSE2__)
+ return MA_TRUE; /* If the compiler is allowed to freely generate SSE2 code we can assume support. */
+ #else
+ #if defined(MA_NO_CPUID)
+ return MA_FALSE;
+ #else
+ int info[4];
+ ma_cpuid(info, 1);
+ return (info[3] & (1 << 26)) != 0;
+ #endif
+ #endif
+ #else
+ return MA_FALSE; /* SSE2 is only supported on x86 and x64 architectures. */
+ #endif
+#else
+ return MA_FALSE; /* No compiler support. */
+#endif
+}
+
+#if 0
+static MA_INLINE ma_bool32 ma_has_avx()
+{
+#if defined(MA_SUPPORT_AVX)
+ #if (defined(MA_X64) || defined(MA_X86)) && !defined(MA_NO_AVX)
+ #if defined(_AVX_) || defined(__AVX__)
+ return MA_TRUE; /* If the compiler is allowed to freely generate AVX code we can assume support. */
+ #else
+ /* AVX requires both CPU and OS support. */
+ #if defined(MA_NO_CPUID) || defined(MA_NO_XGETBV)
+ return MA_FALSE;
+ #else
+ int info[4];
+ ma_cpuid(info, 1);
+ if (((info[2] & (1 << 27)) != 0) && ((info[2] & (1 << 28)) != 0)) {
+ ma_uint64 xrc = ma_xgetbv(0);
+ if ((xrc & 0x06) == 0x06) {
+ return MA_TRUE;
+ } else {
+ return MA_FALSE;
+ }
+ } else {
+ return MA_FALSE;
+ }
+ #endif
+ #endif
+ #else
+ return MA_FALSE; /* AVX is only supported on x86 and x64 architectures. */
+ #endif
+#else
+ return MA_FALSE; /* No compiler support. */
+#endif
+}
+#endif
+
+static MA_INLINE ma_bool32 ma_has_avx2(void)
+{
+#if defined(MA_SUPPORT_AVX2)
+ #if (defined(MA_X64) || defined(MA_X86)) && !defined(MA_NO_AVX2)
+ #if defined(_AVX2_) || defined(__AVX2__)
+ return MA_TRUE; /* If the compiler is allowed to freely generate AVX2 code we can assume support. */
+ #else
+ /* AVX2 requires both CPU and OS support. */
+ #if defined(MA_NO_CPUID) || defined(MA_NO_XGETBV)
+ return MA_FALSE;
+ #else
+ int info1[4];
+ int info7[4];
+ ma_cpuid(info1, 1);
+ ma_cpuid(info7, 7);
+ if (((info1[2] & (1 << 27)) != 0) && ((info7[1] & (1 << 5)) != 0)) {
+ ma_uint64 xrc = ma_xgetbv(0);
+ if ((xrc & 0x06) == 0x06) {
+ return MA_TRUE;
+ } else {
+ return MA_FALSE;
+ }
+ } else {
+ return MA_FALSE;
+ }
+ #endif
+ #endif
+ #else
+ return MA_FALSE; /* AVX2 is only supported on x86 and x64 architectures. */
+ #endif
+#else
+ return MA_FALSE; /* No compiler support. */
+#endif
+}
+
+static MA_INLINE ma_bool32 ma_has_neon(void)
+{
+#if defined(MA_SUPPORT_NEON)
+ #if defined(MA_ARM) && !defined(MA_NO_NEON)
+ #if (defined(__ARM_NEON) || defined(__aarch64__) || defined(_M_ARM64))
+ return MA_TRUE; /* If the compiler is allowed to freely generate NEON code we can assume support. */
+ #else
+ /* TODO: Runtime check. */
+ return MA_FALSE;
+ #endif
+ #else
+ return MA_FALSE; /* NEON is only supported on ARM architectures. */
+ #endif
+#else
+ return MA_FALSE; /* No compiler support. */
+#endif
+}
+
+#define MA_SIMD_NONE 0
+#define MA_SIMD_SSE2 1
+#define MA_SIMD_AVX2 2
+#define MA_SIMD_NEON 3
+
+#ifndef MA_PREFERRED_SIMD
+ # if defined(MA_SUPPORT_SSE2) && defined(MA_PREFER_SSE2)
+ #define MA_PREFERRED_SIMD MA_SIMD_SSE2
+ #elif defined(MA_SUPPORT_AVX2) && defined(MA_PREFER_AVX2)
+ #define MA_PREFERRED_SIMD MA_SIMD_AVX2
+ #elif defined(MA_SUPPORT_NEON) && defined(MA_PREFER_NEON)
+ #define MA_PREFERRED_SIMD MA_SIMD_NEON
+ #else
+ #define MA_PREFERRED_SIMD MA_SIMD_NONE
+ #endif
+#endif
+
+#if defined(__has_builtin)
+ #define MA_COMPILER_HAS_BUILTIN(x) __has_builtin(x)
+#else
+ #define MA_COMPILER_HAS_BUILTIN(x) 0
+#endif
+
+#ifndef MA_ASSUME
+ #if MA_COMPILER_HAS_BUILTIN(__builtin_assume)
+ #define MA_ASSUME(x) __builtin_assume(x)
+ #elif MA_COMPILER_HAS_BUILTIN(__builtin_unreachable)
+ #define MA_ASSUME(x) do { if (!(x)) __builtin_unreachable(); } while (0)
+ #elif defined(_MSC_VER)
+ #define MA_ASSUME(x) __assume(x)
+ #else
+ #define MA_ASSUME(x) (void)(x)
+ #endif
+#endif
+
+#ifndef MA_RESTRICT
+ #if defined(__clang__) || defined(__GNUC__) || defined(_MSC_VER)
+ #define MA_RESTRICT __restrict
+ #else
+ #define MA_RESTRICT
+ #endif
+#endif
+
+#if defined(_MSC_VER) && _MSC_VER >= 1400
+ #define MA_HAS_BYTESWAP16_INTRINSIC
+ #define MA_HAS_BYTESWAP32_INTRINSIC
+ #define MA_HAS_BYTESWAP64_INTRINSIC
+#elif defined(__clang__)
+ #if MA_COMPILER_HAS_BUILTIN(__builtin_bswap16)
+ #define MA_HAS_BYTESWAP16_INTRINSIC
+ #endif
+ #if MA_COMPILER_HAS_BUILTIN(__builtin_bswap32)
+ #define MA_HAS_BYTESWAP32_INTRINSIC
+ #endif
+ #if MA_COMPILER_HAS_BUILTIN(__builtin_bswap64)
+ #define MA_HAS_BYTESWAP64_INTRINSIC
+ #endif
+#elif defined(__GNUC__)
+ #if ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 3))
+ #define MA_HAS_BYTESWAP32_INTRINSIC
+ #define MA_HAS_BYTESWAP64_INTRINSIC
+ #endif
+ #if ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8))
+ #define MA_HAS_BYTESWAP16_INTRINSIC
+ #endif
+#endif
+
+
+static MA_INLINE ma_bool32 ma_is_little_endian(void)
+{
+#if defined(MA_X86) || defined(MA_X64)
+ return MA_TRUE;
+#else
+ int n = 1;
+ return (*(char*)&n) == 1;
+#endif
+}
+
+static MA_INLINE ma_bool32 ma_is_big_endian(void)
+{
+ return !ma_is_little_endian();
+}
+
+
+static MA_INLINE ma_uint32 ma_swap_endian_uint32(ma_uint32 n)
+{
+#ifdef MA_HAS_BYTESWAP32_INTRINSIC
+ #if defined(_MSC_VER)
+ return _byteswap_ulong(n);
+ #elif defined(__GNUC__) || defined(__clang__)
+ #if defined(MA_ARM) && (defined(__ARM_ARCH) && __ARM_ARCH >= 6) && !defined(MA_64BIT) /* <-- 64-bit inline assembly has not been tested, so disabling for now. */
+ /* Inline assembly optimized implementation for ARM. In my testing, GCC does not generate optimized code with __builtin_bswap32(). */
+ ma_uint32 r;
+ __asm__ __volatile__ (
+ #if defined(MA_64BIT)
+ "rev %w[out], %w[in]" : [out]"=r"(r) : [in]"r"(n) /* <-- This is untested. If someone in the community could test this, that would be appreciated! */
+ #else
+ "rev %[out], %[in]" : [out]"=r"(r) : [in]"r"(n)
+ #endif
+ );
+ return r;
+ #else
+ return __builtin_bswap32(n);
+ #endif
+ #else
+ #error "This compiler does not support the byte swap intrinsic."
+ #endif
+#else
+ return ((n & 0xFF000000) >> 24) |
+ ((n & 0x00FF0000) >> 8) |
+ ((n & 0x0000FF00) << 8) |
+ ((n & 0x000000FF) << 24);
+#endif
+}
+
+
+#if !defined(MA_EMSCRIPTEN)
+#ifdef MA_WIN32
+static void ma_sleep__win32(ma_uint32 milliseconds)
+{
+ Sleep((DWORD)milliseconds);
+}
+#endif
+#ifdef MA_POSIX
+static void ma_sleep__posix(ma_uint32 milliseconds)
+{
+#ifdef MA_EMSCRIPTEN
+ (void)milliseconds;
+ MA_ASSERT(MA_FALSE); /* The Emscripten build should never sleep. */
+#else
+ #if defined(_POSIX_C_SOURCE) && _POSIX_C_SOURCE >= 199309L
+ struct timespec ts;
+ ts.tv_sec = milliseconds / 1000;
+ ts.tv_nsec = milliseconds % 1000 * 1000000;
+ nanosleep(&ts, NULL);
+ #else
+ struct timeval tv;
+ tv.tv_sec = milliseconds / 1000;
+ tv.tv_usec = milliseconds % 1000 * 1000;
+ select(0, NULL, NULL, NULL, &tv);
+ #endif
+#endif
+}
+#endif
+
+static MA_INLINE void ma_sleep(ma_uint32 milliseconds)
+{
+#ifdef MA_WIN32
+ ma_sleep__win32(milliseconds);
+#endif
+#ifdef MA_POSIX
+ ma_sleep__posix(milliseconds);
+#endif
+}
+#endif
+
+static MA_INLINE void ma_yield()
+{
+#if defined(__i386) || defined(_M_IX86) || defined(__x86_64__) || defined(_M_X64)
+ /* x86/x64 */
+ #if (defined(_MSC_VER) || defined(__WATCOMC__) || defined(__DMC__)) && !defined(__clang__)
+ #if _MSC_VER >= 1400
+ _mm_pause();
+ #else
+ #if defined(__DMC__)
+ /* Digital Mars does not recognize the PAUSE opcode. Fall back to NOP. */
+ __asm nop;
+ #else
+ __asm pause;
+ #endif
+ #endif
+ #else
+ __asm__ __volatile__ ("pause");
+ #endif
+#elif (defined(__arm__) && defined(__ARM_ARCH) && __ARM_ARCH >= 7) || defined(_M_ARM64) || (defined(_M_ARM) && _M_ARM >= 7) || defined(__ARM_ARCH_6K__) || defined(__ARM_ARCH_6T2__)
+ /* ARM */
+ #if defined(_MSC_VER)
+ /* Apparently there is a __yield() intrinsic that's compatible with ARM, but I cannot find documentation for it nor can I find where it's declared. */
+ __yield();
+ #else
+ __asm__ __volatile__ ("yield"); /* ARMv6K/ARMv6T2 and above. */
+ #endif
+#else
+ /* Unknown or unsupported architecture. No-op. */
+#endif
+}
+
+
+#define MA_MM_DENORMALS_ZERO_MASK 0x0040
+#define MA_MM_FLUSH_ZERO_MASK 0x8000
+
+static MA_INLINE unsigned int ma_disable_denormals()
+{
+ unsigned int prevState;
+
+ #if defined(_MSC_VER)
+ {
+ /*
+ Older versions of Visual Studio don't support the "safe" versions of _controlfp_s(). I don't
+ know which version of Visual Studio first added support for _controlfp_s(), but I do know
+ that VC6 lacks support. _MSC_VER = 1200 is VC6, but if you get compilation errors on older
+ versions of Visual Studio, let me know and I'll make the necessary adjustment.
+ */
+ #if _MSC_VER <= 1200
+ {
+ prevState = _statusfp();
+ _controlfp(prevState | _DN_FLUSH, _MCW_DN);
+ }
+ #else
+ {
+ unsigned int unused;
+ _controlfp_s(&prevState, 0, 0);
+ _controlfp_s(&unused, prevState | _DN_FLUSH, _MCW_DN);
+ }
+ #endif
+ }
+ #elif defined(MA_X86) || defined(MA_X64)
+ {
+ #if defined(__SSE2__) && !(defined(__TINYC__) || defined(__WATCOMC__)) /* <-- Add compilers that lack support for _mm_getcsr() and _mm_setcsr() to this list. */
+ {
+ prevState = _mm_getcsr();
+ _mm_setcsr(prevState | MA_MM_DENORMALS_ZERO_MASK | MA_MM_FLUSH_ZERO_MASK);
+ }
+ #else
+ {
+ /* x88/64, but no support for _mm_getcsr()/_mm_setcsr(). May need to fall back to inlined assembly here. */
+ prevState = 0;
+ }
+ #endif
+ }
+ #else
+ {
+ /* Unknown or unsupported architecture. No-op. */
+ prevState = 0;
+ }
+ #endif
+
+ return prevState;
+}
+
+static MA_INLINE void ma_restore_denormals(unsigned int prevState)
+{
+ #if defined(_MSC_VER)
+ {
+ /* Older versions of Visual Studio do not support _controlfp_s(). See ma_disable_denormals(). */
+ #if _MSC_VER <= 1200
+ {
+ _controlfp(prevState, _MCW_DN);
+ }
+ #else
+ {
+ unsigned int unused;
+ _controlfp_s(&unused, prevState, _MCW_DN);
+ }
+ #endif
+ }
+ #elif defined(MA_X86) || defined(MA_X64)
+ {
+ #if defined(__SSE2__) && !(defined(__TINYC__) || defined(__WATCOMC__)) /* <-- Add compilers that lack support for _mm_getcsr() and _mm_setcsr() to this list. */
+ {
+ _mm_setcsr(prevState);
+ }
+ #else
+ {
+ /* x88/64, but no support for _mm_getcsr()/_mm_setcsr(). May need to fall back to inlined assembly here. */
+ (void)prevState;
+ }
+ #endif
+ }
+ #else
+ {
+ /* Unknown or unsupported architecture. No-op. */
+ (void)prevState;
+ }
+ #endif
+}
+
+
+
+#ifndef MA_COINIT_VALUE
+#define MA_COINIT_VALUE 0 /* 0 = COINIT_MULTITHREADED */
+#endif
+
+
+#ifndef MA_FLT_MAX
+ #ifdef FLT_MAX
+ #define MA_FLT_MAX FLT_MAX
+ #else
+ #define MA_FLT_MAX 3.402823466e+38F
+ #endif
+#endif
+
+
+#ifndef MA_PI
+#define MA_PI 3.14159265358979323846264f
+#endif
+#ifndef MA_PI_D
+#define MA_PI_D 3.14159265358979323846264
+#endif
+#ifndef MA_TAU
+#define MA_TAU 6.28318530717958647693f
+#endif
+#ifndef MA_TAU_D
+#define MA_TAU_D 6.28318530717958647693
+#endif
+
+
+/* The default format when ma_format_unknown (0) is requested when initializing a device. */
+#ifndef MA_DEFAULT_FORMAT
+#define MA_DEFAULT_FORMAT ma_format_f32
+#endif
+
+/* The default channel count to use when 0 is used when initializing a device. */
+#ifndef MA_DEFAULT_CHANNELS
+#define MA_DEFAULT_CHANNELS 2
+#endif
+
+/* The default sample rate to use when 0 is used when initializing a device. */
+#ifndef MA_DEFAULT_SAMPLE_RATE
+#define MA_DEFAULT_SAMPLE_RATE 48000
+#endif
+
+/* Default periods when none is specified in ma_device_init(). More periods means more work on the CPU. */
+#ifndef MA_DEFAULT_PERIODS
+#define MA_DEFAULT_PERIODS 3
+#endif
+
+/* The default period size in milliseconds for low latency mode. */
+#ifndef MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY
+#define MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY 10
+#endif
+
+/* The default buffer size in milliseconds for conservative mode. */
+#ifndef MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE
+#define MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE 100
+#endif
+
+/* The default LPF filter order for linear resampling. Note that this is clamped to MA_MAX_FILTER_ORDER. */
+#ifndef MA_DEFAULT_RESAMPLER_LPF_ORDER
+ #if MA_MAX_FILTER_ORDER >= 4
+ #define MA_DEFAULT_RESAMPLER_LPF_ORDER 4
+ #else
+ #define MA_DEFAULT_RESAMPLER_LPF_ORDER MA_MAX_FILTER_ORDER
+ #endif
+#endif
+
+
+#if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
+ #pragma GCC diagnostic push
+ #pragma GCC diagnostic ignored "-Wunused-variable"
+#endif
+
+/* Standard sample rates, in order of priority. */
+static ma_uint32 g_maStandardSampleRatePriorities[] = {
+ (ma_uint32)ma_standard_sample_rate_48000,
+ (ma_uint32)ma_standard_sample_rate_44100,
+
+ (ma_uint32)ma_standard_sample_rate_32000,
+ (ma_uint32)ma_standard_sample_rate_24000,
+ (ma_uint32)ma_standard_sample_rate_22050,
+
+ (ma_uint32)ma_standard_sample_rate_88200,
+ (ma_uint32)ma_standard_sample_rate_96000,
+ (ma_uint32)ma_standard_sample_rate_176400,
+ (ma_uint32)ma_standard_sample_rate_192000,
+
+ (ma_uint32)ma_standard_sample_rate_16000,
+ (ma_uint32)ma_standard_sample_rate_11025,
+ (ma_uint32)ma_standard_sample_rate_8000,
+
+ (ma_uint32)ma_standard_sample_rate_352800,
+ (ma_uint32)ma_standard_sample_rate_384000
+};
+
+static MA_INLINE ma_bool32 ma_is_standard_sample_rate(ma_uint32 sampleRate)
+{
+ ma_uint32 iSampleRate;
+
+ for (iSampleRate = 0; iSampleRate < sizeof(g_maStandardSampleRatePriorities) / sizeof(g_maStandardSampleRatePriorities[0]); iSampleRate += 1) {
+ if (g_maStandardSampleRatePriorities[iSampleRate] == sampleRate) {
+ return MA_TRUE;
+ }
+ }
+
+ /* Getting here means the sample rate is not supported. */
+ return MA_FALSE;
+}
+
+
+static ma_format g_maFormatPriorities[] = {
+ ma_format_s16, /* Most common */
+ ma_format_f32,
+
+ /*ma_format_s24_32,*/ /* Clean alignment */
+ ma_format_s32,
+
+ ma_format_s24, /* Unclean alignment */
+
+ ma_format_u8 /* Low quality */
+};
+#if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
+ #pragma GCC diagnostic pop
+#endif
+
+
+MA_API void ma_version(ma_uint32* pMajor, ma_uint32* pMinor, ma_uint32* pRevision)
+{
+ if (pMajor) {
+ *pMajor = MA_VERSION_MAJOR;
+ }
+
+ if (pMinor) {
+ *pMinor = MA_VERSION_MINOR;
+ }
+
+ if (pRevision) {
+ *pRevision = MA_VERSION_REVISION;
+ }
+}
+
+MA_API const char* ma_version_string(void)
+{
+ return MA_VERSION_STRING;
+}
+
+
+/******************************************************************************
+
+Standard Library Stuff
+
+******************************************************************************/
+#ifndef MA_MALLOC
+#ifdef MA_WIN32
+#define MA_MALLOC(sz) HeapAlloc(GetProcessHeap(), 0, (sz))
+#else
+#define MA_MALLOC(sz) malloc((sz))
+#endif
+#endif
+
+#ifndef MA_REALLOC
+#ifdef MA_WIN32
+#define MA_REALLOC(p, sz) (((sz) > 0) ? ((p) ? HeapReAlloc(GetProcessHeap(), 0, (p), (sz)) : HeapAlloc(GetProcessHeap(), 0, (sz))) : ((VOID*)(size_t)(HeapFree(GetProcessHeap(), 0, (p)) & 0)))
+#else
+#define MA_REALLOC(p, sz) realloc((p), (sz))
+#endif
+#endif
+
+#ifndef MA_FREE
+#ifdef MA_WIN32
+#define MA_FREE(p) HeapFree(GetProcessHeap(), 0, (p))
+#else
+#define MA_FREE(p) free((p))
+#endif
+#endif
+
+#ifndef MA_ZERO_MEMORY
+#ifdef MA_WIN32
+#define MA_ZERO_MEMORY(p, sz) ZeroMemory((p), (sz))
+#else
+#define MA_ZERO_MEMORY(p, sz) memset((p), 0, (sz))
+#endif
+#endif
+
+#ifndef MA_COPY_MEMORY
+#ifdef MA_WIN32
+#define MA_COPY_MEMORY(dst, src, sz) CopyMemory((dst), (src), (sz))
+#else
+#define MA_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz))
+#endif
+#endif
+
+#ifndef MA_MOVE_MEMORY
+#ifdef MA_WIN32
+#define MA_MOVE_MEMORY(dst, src, sz) MoveMemory((dst), (src), (sz))
+#else
+#define MA_MOVE_MEMORY(dst, src, sz) memmove((dst), (src), (sz))
+#endif
+#endif
+
+#ifndef MA_ASSERT
+#ifdef MA_WIN32
+#define MA_ASSERT(condition) assert(condition)
+#else
+#define MA_ASSERT(condition) assert(condition)
+#endif
+#endif
+
+#define MA_ZERO_OBJECT(p) MA_ZERO_MEMORY((p), sizeof(*(p)))
+
+#define ma_countof(x) (sizeof(x) / sizeof(x[0]))
+#define ma_max(x, y) (((x) > (y)) ? (x) : (y))
+#define ma_min(x, y) (((x) < (y)) ? (x) : (y))
+#define ma_abs(x) (((x) > 0) ? (x) : -(x))
+#define ma_clamp(x, lo, hi) (ma_max(lo, ma_min(x, hi)))
+#define ma_offset_ptr(p, offset) (((ma_uint8*)(p)) + (offset))
+#define ma_align(x, a) ((x + (a-1)) & ~(a-1))
+#define ma_align_64(x) ma_align(x, 8)
+
+#define ma_buffer_frame_capacity(buffer, channels, format) (sizeof(buffer) / ma_get_bytes_per_sample(format) / (channels))
+
+static MA_INLINE double ma_sind(double x)
+{
+ /* TODO: Implement custom sin(x). */
+ return sin(x);
+}
+
+static MA_INLINE double ma_expd(double x)
+{
+ /* TODO: Implement custom exp(x). */
+ return exp(x);
+}
+
+static MA_INLINE double ma_logd(double x)
+{
+ /* TODO: Implement custom log(x). */
+ return log(x);
+}
+
+static MA_INLINE double ma_powd(double x, double y)
+{
+ /* TODO: Implement custom pow(x, y). */
+ return pow(x, y);
+}
+
+static MA_INLINE double ma_sqrtd(double x)
+{
+ /* TODO: Implement custom sqrt(x). */
+ return sqrt(x);
+}
+
+
+static MA_INLINE float ma_sinf(float x)
+{
+ return (float)ma_sind((float)x);
+}
+
+static MA_INLINE double ma_cosd(double x)
+{
+ return ma_sind((MA_PI_D*0.5) - x);
+}
+
+static MA_INLINE float ma_cosf(float x)
+{
+ return (float)ma_cosd((float)x);
+}
+
+static MA_INLINE double ma_log10d(double x)
+{
+ return ma_logd(x) * 0.43429448190325182765;
+}
+
+static MA_INLINE float ma_powf(float x, float y)
+{
+ return (float)ma_powd((double)x, (double)y);
+}
+
+static MA_INLINE float ma_log10f(float x)
+{
+ return (float)ma_log10d((double)x);
+}
+
+
+static MA_INLINE double ma_degrees_to_radians(double degrees)
+{
+ return degrees * 0.01745329252;
+}
+
+static MA_INLINE double ma_radians_to_degrees(double radians)
+{
+ return radians * 57.295779512896;
+}
+
+static MA_INLINE float ma_degrees_to_radians_f(float degrees)
+{
+ return degrees * 0.01745329252f;
+}
+
+static MA_INLINE float ma_radians_to_degrees_f(float radians)
+{
+ return radians * 57.295779512896f;
+}
+
+
+/*
+Return Values:
+ 0: Success
+ 22: EINVAL
+ 34: ERANGE
+
+Not using symbolic constants for errors because I want to avoid #including errno.h
+*/
+MA_API int ma_strcpy_s(char* dst, size_t dstSizeInBytes, const char* src)
+{
+ size_t i;
+
+ if (dst == 0) {
+ return 22;
+ }
+ if (dstSizeInBytes == 0) {
+ return 34;
+ }
+ if (src == 0) {
+ dst[0] = '\0';
+ return 22;
+ }
+
+ for (i = 0; i < dstSizeInBytes && src[i] != '\0'; ++i) {
+ dst[i] = src[i];
+ }
+
+ if (i < dstSizeInBytes) {
+ dst[i] = '\0';
+ return 0;
+ }
+
+ dst[0] = '\0';
+ return 34;
+}
+
+MA_API int ma_wcscpy_s(wchar_t* dst, size_t dstCap, const wchar_t* src)
+{
+ size_t i;
+
+ if (dst == 0) {
+ return 22;
+ }
+ if (dstCap == 0) {
+ return 34;
+ }
+ if (src == 0) {
+ dst[0] = '\0';
+ return 22;
+ }
+
+ for (i = 0; i < dstCap && src[i] != '\0'; ++i) {
+ dst[i] = src[i];
+ }
+
+ if (i < dstCap) {
+ dst[i] = '\0';
+ return 0;
+ }
+
+ dst[0] = '\0';
+ return 34;
+}
+
+
+MA_API int ma_strncpy_s(char* dst, size_t dstSizeInBytes, const char* src, size_t count)
+{
+ size_t maxcount;
+ size_t i;
+
+ if (dst == 0) {
+ return 22;
+ }
+ if (dstSizeInBytes == 0) {
+ return 34;
+ }
+ if (src == 0) {
+ dst[0] = '\0';
+ return 22;
+ }
+
+ maxcount = count;
+ if (count == ((size_t)-1) || count >= dstSizeInBytes) { /* -1 = _TRUNCATE */
+ maxcount = dstSizeInBytes - 1;
+ }
+
+ for (i = 0; i < maxcount && src[i] != '\0'; ++i) {
+ dst[i] = src[i];
+ }
+
+ if (src[i] == '\0' || i == count || count == ((size_t)-1)) {
+ dst[i] = '\0';
+ return 0;
+ }
+
+ dst[0] = '\0';
+ return 34;
+}
+
+MA_API int ma_strcat_s(char* dst, size_t dstSizeInBytes, const char* src)
+{
+ char* dstorig;
+
+ if (dst == 0) {
+ return 22;
+ }
+ if (dstSizeInBytes == 0) {
+ return 34;
+ }
+ if (src == 0) {
+ dst[0] = '\0';
+ return 22;
+ }
+
+ dstorig = dst;
+
+ while (dstSizeInBytes > 0 && dst[0] != '\0') {
+ dst += 1;
+ dstSizeInBytes -= 1;
+ }
+
+ if (dstSizeInBytes == 0) {
+ return 22; /* Unterminated. */
+ }
+
+
+ while (dstSizeInBytes > 0 && src[0] != '\0') {
+ *dst++ = *src++;
+ dstSizeInBytes -= 1;
+ }
+
+ if (dstSizeInBytes > 0) {
+ dst[0] = '\0';
+ } else {
+ dstorig[0] = '\0';
+ return 34;
+ }
+
+ return 0;
+}
+
+MA_API int ma_strncat_s(char* dst, size_t dstSizeInBytes, const char* src, size_t count)
+{
+ char* dstorig;
+
+ if (dst == 0) {
+ return 22;
+ }
+ if (dstSizeInBytes == 0) {
+ return 34;
+ }
+ if (src == 0) {
+ return 22;
+ }
+
+ dstorig = dst;
+
+ while (dstSizeInBytes > 0 && dst[0] != '\0') {
+ dst += 1;
+ dstSizeInBytes -= 1;
+ }
+
+ if (dstSizeInBytes == 0) {
+ return 22; /* Unterminated. */
+ }
+
+
+ if (count == ((size_t)-1)) { /* _TRUNCATE */
+ count = dstSizeInBytes - 1;
+ }
+
+ while (dstSizeInBytes > 0 && src[0] != '\0' && count > 0) {
+ *dst++ = *src++;
+ dstSizeInBytes -= 1;
+ count -= 1;
+ }
+
+ if (dstSizeInBytes > 0) {
+ dst[0] = '\0';
+ } else {
+ dstorig[0] = '\0';
+ return 34;
+ }
+
+ return 0;
+}
+
+MA_API int ma_itoa_s(int value, char* dst, size_t dstSizeInBytes, int radix)
+{
+ int sign;
+ unsigned int valueU;
+ char* dstEnd;
+
+ if (dst == NULL || dstSizeInBytes == 0) {
+ return 22;
+ }
+ if (radix < 2 || radix > 36) {
+ dst[0] = '\0';
+ return 22;
+ }
+
+ sign = (value < 0 && radix == 10) ? -1 : 1; /* The negative sign is only used when the base is 10. */
+
+ if (value < 0) {
+ valueU = -value;
+ } else {
+ valueU = value;
+ }
+
+ dstEnd = dst;
+ do
+ {
+ int remainder = valueU % radix;
+ if (remainder > 9) {
+ *dstEnd = (char)((remainder - 10) + 'a');
+ } else {
+ *dstEnd = (char)(remainder + '0');
+ }
+
+ dstEnd += 1;
+ dstSizeInBytes -= 1;
+ valueU /= radix;
+ } while (dstSizeInBytes > 0 && valueU > 0);
+
+ if (dstSizeInBytes == 0) {
+ dst[0] = '\0';
+ return 22; /* Ran out of room in the output buffer. */
+ }
+
+ if (sign < 0) {
+ *dstEnd++ = '-';
+ dstSizeInBytes -= 1;
+ }
+
+ if (dstSizeInBytes == 0) {
+ dst[0] = '\0';
+ return 22; /* Ran out of room in the output buffer. */
+ }
+
+ *dstEnd = '\0';
+
+
+ /* At this point the string will be reversed. */
+ dstEnd -= 1;
+ while (dst < dstEnd) {
+ char temp = *dst;
+ *dst = *dstEnd;
+ *dstEnd = temp;
+
+ dst += 1;
+ dstEnd -= 1;
+ }
+
+ return 0;
+}
+
+MA_API int ma_strcmp(const char* str1, const char* str2)
+{
+ if (str1 == str2) return 0;
+
+ /* These checks differ from the standard implementation. It's not important, but I prefer it just for sanity. */
+ if (str1 == NULL) return -1;
+ if (str2 == NULL) return 1;
+
+ for (;;) {
+ if (str1[0] == '\0') {
+ break;
+ }
+ if (str1[0] != str2[0]) {
+ break;
+ }
+
+ str1 += 1;
+ str2 += 1;
+ }
+
+ return ((unsigned char*)str1)[0] - ((unsigned char*)str2)[0];
+}
+
+MA_API int ma_strappend(char* dst, size_t dstSize, const char* srcA, const char* srcB)
+{
+ int result;
+
+ result = ma_strncpy_s(dst, dstSize, srcA, (size_t)-1);
+ if (result != 0) {
+ return result;
+ }
+
+ result = ma_strncat_s(dst, dstSize, srcB, (size_t)-1);
+ if (result != 0) {
+ return result;
+ }
+
+ return result;
+}
+
+MA_API char* ma_copy_string(const char* src, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (src == NULL) {
+ return NULL;
+ }
+
+ size_t sz = strlen(src)+1;
+ char* dst = (char*)ma_malloc(sz, pAllocationCallbacks);
+ if (dst == NULL) {
+ return NULL;
+ }
+
+ ma_strcpy_s(dst, sz, src);
+
+ return dst;
+}
+
+MA_API wchar_t* ma_copy_string_w(const wchar_t* src, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ size_t sz = wcslen(src)+1;
+ wchar_t* dst = (wchar_t*)ma_malloc(sz * sizeof(*dst), pAllocationCallbacks);
+ if (dst == NULL) {
+ return NULL;
+ }
+
+ ma_wcscpy_s(dst, sz, src);
+
+ return dst;
+}
+
+
+#include
+static ma_result ma_result_from_errno(int e)
+{
+ switch (e)
+ {
+ case 0: return MA_SUCCESS;
+ #ifdef EPERM
+ case EPERM: return MA_INVALID_OPERATION;
+ #endif
+ #ifdef ENOENT
+ case ENOENT: return MA_DOES_NOT_EXIST;
+ #endif
+ #ifdef ESRCH
+ case ESRCH: return MA_DOES_NOT_EXIST;
+ #endif
+ #ifdef EINTR
+ case EINTR: return MA_INTERRUPT;
+ #endif
+ #ifdef EIO
+ case EIO: return MA_IO_ERROR;
+ #endif
+ #ifdef ENXIO
+ case ENXIO: return MA_DOES_NOT_EXIST;
+ #endif
+ #ifdef E2BIG
+ case E2BIG: return MA_INVALID_ARGS;
+ #endif
+ #ifdef ENOEXEC
+ case ENOEXEC: return MA_INVALID_FILE;
+ #endif
+ #ifdef EBADF
+ case EBADF: return MA_INVALID_FILE;
+ #endif
+ #ifdef ECHILD
+ case ECHILD: return MA_ERROR;
+ #endif
+ #ifdef EAGAIN
+ case EAGAIN: return MA_UNAVAILABLE;
+ #endif
+ #ifdef ENOMEM
+ case ENOMEM: return MA_OUT_OF_MEMORY;
+ #endif
+ #ifdef EACCES
+ case EACCES: return MA_ACCESS_DENIED;
+ #endif
+ #ifdef EFAULT
+ case EFAULT: return MA_BAD_ADDRESS;
+ #endif
+ #ifdef ENOTBLK
+ case ENOTBLK: return MA_ERROR;
+ #endif
+ #ifdef EBUSY
+ case EBUSY: return MA_BUSY;
+ #endif
+ #ifdef EEXIST
+ case EEXIST: return MA_ALREADY_EXISTS;
+ #endif
+ #ifdef EXDEV
+ case EXDEV: return MA_ERROR;
+ #endif
+ #ifdef ENODEV
+ case ENODEV: return MA_DOES_NOT_EXIST;
+ #endif
+ #ifdef ENOTDIR
+ case ENOTDIR: return MA_NOT_DIRECTORY;
+ #endif
+ #ifdef EISDIR
+ case EISDIR: return MA_IS_DIRECTORY;
+ #endif
+ #ifdef EINVAL
+ case EINVAL: return MA_INVALID_ARGS;
+ #endif
+ #ifdef ENFILE
+ case ENFILE: return MA_TOO_MANY_OPEN_FILES;
+ #endif
+ #ifdef EMFILE
+ case EMFILE: return MA_TOO_MANY_OPEN_FILES;
+ #endif
+ #ifdef ENOTTY
+ case ENOTTY: return MA_INVALID_OPERATION;
+ #endif
+ #ifdef ETXTBSY
+ case ETXTBSY: return MA_BUSY;
+ #endif
+ #ifdef EFBIG
+ case EFBIG: return MA_TOO_BIG;
+ #endif
+ #ifdef ENOSPC
+ case ENOSPC: return MA_NO_SPACE;
+ #endif
+ #ifdef ESPIPE
+ case ESPIPE: return MA_BAD_SEEK;
+ #endif
+ #ifdef EROFS
+ case EROFS: return MA_ACCESS_DENIED;
+ #endif
+ #ifdef EMLINK
+ case EMLINK: return MA_TOO_MANY_LINKS;
+ #endif
+ #ifdef EPIPE
+ case EPIPE: return MA_BAD_PIPE;
+ #endif
+ #ifdef EDOM
+ case EDOM: return MA_OUT_OF_RANGE;
+ #endif
+ #ifdef ERANGE
+ case ERANGE: return MA_OUT_OF_RANGE;
+ #endif
+ #ifdef EDEADLK
+ case EDEADLK: return MA_DEADLOCK;
+ #endif
+ #ifdef ENAMETOOLONG
+ case ENAMETOOLONG: return MA_PATH_TOO_LONG;
+ #endif
+ #ifdef ENOLCK
+ case ENOLCK: return MA_ERROR;
+ #endif
+ #ifdef ENOSYS
+ case ENOSYS: return MA_NOT_IMPLEMENTED;
+ #endif
+ #ifdef ENOTEMPTY
+ case ENOTEMPTY: return MA_DIRECTORY_NOT_EMPTY;
+ #endif
+ #ifdef ELOOP
+ case ELOOP: return MA_TOO_MANY_LINKS;
+ #endif
+ #ifdef ENOMSG
+ case ENOMSG: return MA_NO_MESSAGE;
+ #endif
+ #ifdef EIDRM
+ case EIDRM: return MA_ERROR;
+ #endif
+ #ifdef ECHRNG
+ case ECHRNG: return MA_ERROR;
+ #endif
+ #ifdef EL2NSYNC
+ case EL2NSYNC: return MA_ERROR;
+ #endif
+ #ifdef EL3HLT
+ case EL3HLT: return MA_ERROR;
+ #endif
+ #ifdef EL3RST
+ case EL3RST: return MA_ERROR;
+ #endif
+ #ifdef ELNRNG
+ case ELNRNG: return MA_OUT_OF_RANGE;
+ #endif
+ #ifdef EUNATCH
+ case EUNATCH: return MA_ERROR;
+ #endif
+ #ifdef ENOCSI
+ case ENOCSI: return MA_ERROR;
+ #endif
+ #ifdef EL2HLT
+ case EL2HLT: return MA_ERROR;
+ #endif
+ #ifdef EBADE
+ case EBADE: return MA_ERROR;
+ #endif
+ #ifdef EBADR
+ case EBADR: return MA_ERROR;
+ #endif
+ #ifdef EXFULL
+ case EXFULL: return MA_ERROR;
+ #endif
+ #ifdef ENOANO
+ case ENOANO: return MA_ERROR;
+ #endif
+ #ifdef EBADRQC
+ case EBADRQC: return MA_ERROR;
+ #endif
+ #ifdef EBADSLT
+ case EBADSLT: return MA_ERROR;
+ #endif
+ #ifdef EBFONT
+ case EBFONT: return MA_INVALID_FILE;
+ #endif
+ #ifdef ENOSTR
+ case ENOSTR: return MA_ERROR;
+ #endif
+ #ifdef ENODATA
+ case ENODATA: return MA_NO_DATA_AVAILABLE;
+ #endif
+ #ifdef ETIME
+ case ETIME: return MA_TIMEOUT;
+ #endif
+ #ifdef ENOSR
+ case ENOSR: return MA_NO_DATA_AVAILABLE;
+ #endif
+ #ifdef ENONET
+ case ENONET: return MA_NO_NETWORK;
+ #endif
+ #ifdef ENOPKG
+ case ENOPKG: return MA_ERROR;
+ #endif
+ #ifdef EREMOTE
+ case EREMOTE: return MA_ERROR;
+ #endif
+ #ifdef ENOLINK
+ case ENOLINK: return MA_ERROR;
+ #endif
+ #ifdef EADV
+ case EADV: return MA_ERROR;
+ #endif
+ #ifdef ESRMNT
+ case ESRMNT: return MA_ERROR;
+ #endif
+ #ifdef ECOMM
+ case ECOMM: return MA_ERROR;
+ #endif
+ #ifdef EPROTO
+ case EPROTO: return MA_ERROR;
+ #endif
+ #ifdef EMULTIHOP
+ case EMULTIHOP: return MA_ERROR;
+ #endif
+ #ifdef EDOTDOT
+ case EDOTDOT: return MA_ERROR;
+ #endif
+ #ifdef EBADMSG
+ case EBADMSG: return MA_BAD_MESSAGE;
+ #endif
+ #ifdef EOVERFLOW
+ case EOVERFLOW: return MA_TOO_BIG;
+ #endif
+ #ifdef ENOTUNIQ
+ case ENOTUNIQ: return MA_NOT_UNIQUE;
+ #endif
+ #ifdef EBADFD
+ case EBADFD: return MA_ERROR;
+ #endif
+ #ifdef EREMCHG
+ case EREMCHG: return MA_ERROR;
+ #endif
+ #ifdef ELIBACC
+ case ELIBACC: return MA_ACCESS_DENIED;
+ #endif
+ #ifdef ELIBBAD
+ case ELIBBAD: return MA_INVALID_FILE;
+ #endif
+ #ifdef ELIBSCN
+ case ELIBSCN: return MA_INVALID_FILE;
+ #endif
+ #ifdef ELIBMAX
+ case ELIBMAX: return MA_ERROR;
+ #endif
+ #ifdef ELIBEXEC
+ case ELIBEXEC: return MA_ERROR;
+ #endif
+ #ifdef EILSEQ
+ case EILSEQ: return MA_INVALID_DATA;
+ #endif
+ #ifdef ERESTART
+ case ERESTART: return MA_ERROR;
+ #endif
+ #ifdef ESTRPIPE
+ case ESTRPIPE: return MA_ERROR;
+ #endif
+ #ifdef EUSERS
+ case EUSERS: return MA_ERROR;
+ #endif
+ #ifdef ENOTSOCK
+ case ENOTSOCK: return MA_NOT_SOCKET;
+ #endif
+ #ifdef EDESTADDRREQ
+ case EDESTADDRREQ: return MA_NO_ADDRESS;
+ #endif
+ #ifdef EMSGSIZE
+ case EMSGSIZE: return MA_TOO_BIG;
+ #endif
+ #ifdef EPROTOTYPE
+ case EPROTOTYPE: return MA_BAD_PROTOCOL;
+ #endif
+ #ifdef ENOPROTOOPT
+ case ENOPROTOOPT: return MA_PROTOCOL_UNAVAILABLE;
+ #endif
+ #ifdef EPROTONOSUPPORT
+ case EPROTONOSUPPORT: return MA_PROTOCOL_NOT_SUPPORTED;
+ #endif
+ #ifdef ESOCKTNOSUPPORT
+ case ESOCKTNOSUPPORT: return MA_SOCKET_NOT_SUPPORTED;
+ #endif
+ #ifdef EOPNOTSUPP
+ case EOPNOTSUPP: return MA_INVALID_OPERATION;
+ #endif
+ #ifdef EPFNOSUPPORT
+ case EPFNOSUPPORT: return MA_PROTOCOL_FAMILY_NOT_SUPPORTED;
+ #endif
+ #ifdef EAFNOSUPPORT
+ case EAFNOSUPPORT: return MA_ADDRESS_FAMILY_NOT_SUPPORTED;
+ #endif
+ #ifdef EADDRINUSE
+ case EADDRINUSE: return MA_ALREADY_IN_USE;
+ #endif
+ #ifdef EADDRNOTAVAIL
+ case EADDRNOTAVAIL: return MA_ERROR;
+ #endif
+ #ifdef ENETDOWN
+ case ENETDOWN: return MA_NO_NETWORK;
+ #endif
+ #ifdef ENETUNREACH
+ case ENETUNREACH: return MA_NO_NETWORK;
+ #endif
+ #ifdef ENETRESET
+ case ENETRESET: return MA_NO_NETWORK;
+ #endif
+ #ifdef ECONNABORTED
+ case ECONNABORTED: return MA_NO_NETWORK;
+ #endif
+ #ifdef ECONNRESET
+ case ECONNRESET: return MA_CONNECTION_RESET;
+ #endif
+ #ifdef ENOBUFS
+ case ENOBUFS: return MA_NO_SPACE;
+ #endif
+ #ifdef EISCONN
+ case EISCONN: return MA_ALREADY_CONNECTED;
+ #endif
+ #ifdef ENOTCONN
+ case ENOTCONN: return MA_NOT_CONNECTED;
+ #endif
+ #ifdef ESHUTDOWN
+ case ESHUTDOWN: return MA_ERROR;
+ #endif
+ #ifdef ETOOMANYREFS
+ case ETOOMANYREFS: return MA_ERROR;
+ #endif
+ #ifdef ETIMEDOUT
+ case ETIMEDOUT: return MA_TIMEOUT;
+ #endif
+ #ifdef ECONNREFUSED
+ case ECONNREFUSED: return MA_CONNECTION_REFUSED;
+ #endif
+ #ifdef EHOSTDOWN
+ case EHOSTDOWN: return MA_NO_HOST;
+ #endif
+ #ifdef EHOSTUNREACH
+ case EHOSTUNREACH: return MA_NO_HOST;
+ #endif
+ #ifdef EALREADY
+ case EALREADY: return MA_IN_PROGRESS;
+ #endif
+ #ifdef EINPROGRESS
+ case EINPROGRESS: return MA_IN_PROGRESS;
+ #endif
+ #ifdef ESTALE
+ case ESTALE: return MA_INVALID_FILE;
+ #endif
+ #ifdef EUCLEAN
+ case EUCLEAN: return MA_ERROR;
+ #endif
+ #ifdef ENOTNAM
+ case ENOTNAM: return MA_ERROR;
+ #endif
+ #ifdef ENAVAIL
+ case ENAVAIL: return MA_ERROR;
+ #endif
+ #ifdef EISNAM
+ case EISNAM: return MA_ERROR;
+ #endif
+ #ifdef EREMOTEIO
+ case EREMOTEIO: return MA_IO_ERROR;
+ #endif
+ #ifdef EDQUOT
+ case EDQUOT: return MA_NO_SPACE;
+ #endif
+ #ifdef ENOMEDIUM
+ case ENOMEDIUM: return MA_DOES_NOT_EXIST;
+ #endif
+ #ifdef EMEDIUMTYPE
+ case EMEDIUMTYPE: return MA_ERROR;
+ #endif
+ #ifdef ECANCELED
+ case ECANCELED: return MA_CANCELLED;
+ #endif
+ #ifdef ENOKEY
+ case ENOKEY: return MA_ERROR;
+ #endif
+ #ifdef EKEYEXPIRED
+ case EKEYEXPIRED: return MA_ERROR;
+ #endif
+ #ifdef EKEYREVOKED
+ case EKEYREVOKED: return MA_ERROR;
+ #endif
+ #ifdef EKEYREJECTED
+ case EKEYREJECTED: return MA_ERROR;
+ #endif
+ #ifdef EOWNERDEAD
+ case EOWNERDEAD: return MA_ERROR;
+ #endif
+ #ifdef ENOTRECOVERABLE
+ case ENOTRECOVERABLE: return MA_ERROR;
+ #endif
+ #ifdef ERFKILL
+ case ERFKILL: return MA_ERROR;
+ #endif
+ #ifdef EHWPOISON
+ case EHWPOISON: return MA_ERROR;
+ #endif
+ default: return MA_ERROR;
+ }
+}
+
+MA_API ma_result ma_fopen(FILE** ppFile, const char* pFilePath, const char* pOpenMode)
+{
+#if defined(_MSC_VER) && _MSC_VER >= 1400
+ errno_t err;
+#endif
+
+ if (ppFile != NULL) {
+ *ppFile = NULL; /* Safety. */
+ }
+
+ if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+#if defined(_MSC_VER) && _MSC_VER >= 1400
+ err = fopen_s(ppFile, pFilePath, pOpenMode);
+ if (err != 0) {
+ return ma_result_from_errno(err);
+ }
+#else
+#if defined(_WIN32) || defined(__APPLE__)
+ *ppFile = fopen(pFilePath, pOpenMode);
+#else
+ #if defined(_FILE_OFFSET_BITS) && _FILE_OFFSET_BITS == 64 && defined(_LARGEFILE64_SOURCE)
+ *ppFile = fopen64(pFilePath, pOpenMode);
+ #else
+ *ppFile = fopen(pFilePath, pOpenMode);
+ #endif
+#endif
+ if (*ppFile == NULL) {
+ ma_result result = ma_result_from_errno(errno);
+ if (result == MA_SUCCESS) {
+ result = MA_ERROR; /* Just a safety check to make sure we never ever return success when pFile == NULL. */
+ }
+
+ return result;
+ }
+#endif
+
+ return MA_SUCCESS;
+}
+
+
+
+/*
+_wfopen() isn't always available in all compilation environments.
+
+ * Windows only.
+ * MSVC seems to support it universally as far back as VC6 from what I can tell (haven't checked further back).
+ * MinGW-64 (both 32- and 64-bit) seems to support it.
+ * MinGW wraps it in !defined(__STRICT_ANSI__).
+ * OpenWatcom wraps it in !defined(_NO_EXT_KEYS).
+
+This can be reviewed as compatibility issues arise. The preference is to use _wfopen_s() and _wfopen() as opposed to the wcsrtombs()
+fallback, so if you notice your compiler not detecting this properly I'm happy to look at adding support.
+*/
+#if defined(_WIN32)
+ #if defined(_MSC_VER) || defined(__MINGW64__) || (!defined(__STRICT_ANSI__) && !defined(_NO_EXT_KEYS))
+ #define MA_HAS_WFOPEN
+ #endif
+#endif
+
+MA_API ma_result ma_wfopen(FILE** ppFile, const wchar_t* pFilePath, const wchar_t* pOpenMode, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (ppFile != NULL) {
+ *ppFile = NULL; /* Safety. */
+ }
+
+ if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+#if defined(MA_HAS_WFOPEN)
+ {
+ /* Use _wfopen() on Windows. */
+ #if defined(_MSC_VER) && _MSC_VER >= 1400
+ errno_t err = _wfopen_s(ppFile, pFilePath, pOpenMode);
+ if (err != 0) {
+ return ma_result_from_errno(err);
+ }
+ #else
+ *ppFile = _wfopen(pFilePath, pOpenMode);
+ if (*ppFile == NULL) {
+ return ma_result_from_errno(errno);
+ }
+ #endif
+ (void)pAllocationCallbacks;
+ }
+#else
+ /*
+ Use fopen() on anything other than Windows. Requires a conversion. This is annoying because fopen() is locale specific. The only real way I can
+ think of to do this is with wcsrtombs(). Note that wcstombs() is apparently not thread-safe because it uses a static global mbstate_t object for
+ maintaining state. I've checked this with -std=c89 and it works, but if somebody get's a compiler error I'll look into improving compatibility.
+ */
+ {
+ mbstate_t mbs;
+ size_t lenMB;
+ const wchar_t* pFilePathTemp = pFilePath;
+ char* pFilePathMB = NULL;
+ char pOpenModeMB[32] = {0};
+
+ /* Get the length first. */
+ MA_ZERO_OBJECT(&mbs);
+ lenMB = wcsrtombs(NULL, &pFilePathTemp, 0, &mbs);
+ if (lenMB == (size_t)-1) {
+ return ma_result_from_errno(errno);
+ }
+
+ pFilePathMB = (char*)ma_malloc(lenMB + 1, pAllocationCallbacks);
+ if (pFilePathMB == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+
+ pFilePathTemp = pFilePath;
+ MA_ZERO_OBJECT(&mbs);
+ wcsrtombs(pFilePathMB, &pFilePathTemp, lenMB + 1, &mbs);
+
+ /* The open mode should always consist of ASCII characters so we should be able to do a trivial conversion. */
+ {
+ size_t i = 0;
+ for (;;) {
+ if (pOpenMode[i] == 0) {
+ pOpenModeMB[i] = '\0';
+ break;
+ }
+
+ pOpenModeMB[i] = (char)pOpenMode[i];
+ i += 1;
+ }
+ }
+
+ *ppFile = fopen(pFilePathMB, pOpenModeMB);
+
+ ma_free(pFilePathMB, pAllocationCallbacks);
+ }
+
+ if (*ppFile == NULL) {
+ return MA_ERROR;
+ }
+#endif
+
+ return MA_SUCCESS;
+}
+
+
+
+static MA_INLINE void ma_copy_memory_64(void* dst, const void* src, ma_uint64 sizeInBytes)
+{
+#if 0xFFFFFFFFFFFFFFFF <= MA_SIZE_MAX
+ MA_COPY_MEMORY(dst, src, (size_t)sizeInBytes);
+#else
+ while (sizeInBytes > 0) {
+ ma_uint64 bytesToCopyNow = sizeInBytes;
+ if (bytesToCopyNow > MA_SIZE_MAX) {
+ bytesToCopyNow = MA_SIZE_MAX;
+ }
+
+ MA_COPY_MEMORY(dst, src, (size_t)bytesToCopyNow); /* Safe cast to size_t. */
+
+ sizeInBytes -= bytesToCopyNow;
+ dst = ( void*)(( ma_uint8*)dst + bytesToCopyNow);
+ src = (const void*)((const ma_uint8*)src + bytesToCopyNow);
+ }
+#endif
+}
+
+static MA_INLINE void ma_zero_memory_64(void* dst, ma_uint64 sizeInBytes)
+{
+#if 0xFFFFFFFFFFFFFFFF <= MA_SIZE_MAX
+ MA_ZERO_MEMORY(dst, (size_t)sizeInBytes);
+#else
+ while (sizeInBytes > 0) {
+ ma_uint64 bytesToZeroNow = sizeInBytes;
+ if (bytesToZeroNow > MA_SIZE_MAX) {
+ bytesToZeroNow = MA_SIZE_MAX;
+ }
+
+ MA_ZERO_MEMORY(dst, (size_t)bytesToZeroNow); /* Safe cast to size_t. */
+
+ sizeInBytes -= bytesToZeroNow;
+ dst = (void*)((ma_uint8*)dst + bytesToZeroNow);
+ }
+#endif
+}
+
+
+/* Thanks to good old Bit Twiddling Hacks for this one: http://graphics.stanford.edu/~seander/bithacks.html#RoundUpPowerOf2 */
+static MA_INLINE unsigned int ma_next_power_of_2(unsigned int x)
+{
+ x--;
+ x |= x >> 1;
+ x |= x >> 2;
+ x |= x >> 4;
+ x |= x >> 8;
+ x |= x >> 16;
+ x++;
+
+ return x;
+}
+
+static MA_INLINE unsigned int ma_prev_power_of_2(unsigned int x)
+{
+ return ma_next_power_of_2(x) >> 1;
+}
+
+static MA_INLINE unsigned int ma_round_to_power_of_2(unsigned int x)
+{
+ unsigned int prev = ma_prev_power_of_2(x);
+ unsigned int next = ma_next_power_of_2(x);
+ if ((next - x) > (x - prev)) {
+ return prev;
+ } else {
+ return next;
+ }
+}
+
+static MA_INLINE unsigned int ma_count_set_bits(unsigned int x)
+{
+ unsigned int count = 0;
+ while (x != 0) {
+ if (x & 1) {
+ count += 1;
+ }
+
+ x = x >> 1;
+ }
+
+ return count;
+}
+
+
+
+/**************************************************************************************************************************************************************
+
+Allocation Callbacks
+
+**************************************************************************************************************************************************************/
+static void* ma__malloc_default(size_t sz, void* pUserData)
+{
+ (void)pUserData;
+ return MA_MALLOC(sz);
+}
+
+static void* ma__realloc_default(void* p, size_t sz, void* pUserData)
+{
+ (void)pUserData;
+ return MA_REALLOC(p, sz);
+}
+
+static void ma__free_default(void* p, void* pUserData)
+{
+ (void)pUserData;
+ MA_FREE(p);
+}
+
+static ma_allocation_callbacks ma_allocation_callbacks_init_default(void)
+{
+ ma_allocation_callbacks callbacks;
+ callbacks.pUserData = NULL;
+ callbacks.onMalloc = ma__malloc_default;
+ callbacks.onRealloc = ma__realloc_default;
+ callbacks.onFree = ma__free_default;
+
+ return callbacks;
+}
+
+static ma_result ma_allocation_callbacks_init_copy(ma_allocation_callbacks* pDst, const ma_allocation_callbacks* pSrc)
+{
+ if (pDst == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (pSrc == NULL) {
+ *pDst = ma_allocation_callbacks_init_default();
+ } else {
+ if (pSrc->pUserData == NULL && pSrc->onFree == NULL && pSrc->onMalloc == NULL && pSrc->onRealloc == NULL) {
+ *pDst = ma_allocation_callbacks_init_default();
+ } else {
+ if (pSrc->onFree == NULL || (pSrc->onMalloc == NULL && pSrc->onRealloc == NULL)) {
+ return MA_INVALID_ARGS; /* Invalid allocation callbacks. */
+ } else {
+ *pDst = *pSrc;
+ }
+ }
+ }
+
+ return MA_SUCCESS;
+}
+
+
+
+
+/**************************************************************************************************************************************************************
+
+Logging
+
+**************************************************************************************************************************************************************/
+MA_API const char* ma_log_level_to_string(ma_uint32 logLevel)
+{
+ switch (logLevel)
+ {
+ case MA_LOG_LEVEL_DEBUG: return "DEBUG";
+ case MA_LOG_LEVEL_INFO: return "INFO";
+ case MA_LOG_LEVEL_WARNING: return "WARNING";
+ case MA_LOG_LEVEL_ERROR: return "ERROR";
+ default: return "ERROR";
+ }
+}
+
+#if defined(MA_DEBUG_OUTPUT)
+
+/* Customize this to use a specific tag in __android_log_print() for debug output messages. */
+#ifndef MA_ANDROID_LOG_TAG
+#define MA_ANDROID_LOG_TAG "miniaudio"
+#endif
+
+void ma_log_callback_debug(void* pUserData, ma_uint32 level, const char* pMessage)
+{
+ (void)pUserData;
+
+ /* Special handling for some platforms. */
+ #if defined(MA_ANDROID)
+ {
+ /* Android. */
+ __android_log_print(ANDROID_LOG_DEBUG, MA_ANDROID_LOG_TAG, "%s: %s", ma_log_level_to_string(level), pMessage);
+ }
+ #else
+ {
+ /* Everything else. */
+ printf("%s: %s", ma_log_level_to_string(level), pMessage);
+ }
+ #endif
+}
+#endif
+
+MA_API ma_log_callback ma_log_callback_init(ma_log_callback_proc onLog, void* pUserData)
+{
+ ma_log_callback callback;
+
+ MA_ZERO_OBJECT(&callback);
+ callback.onLog = onLog;
+ callback.pUserData = pUserData;
+
+ return callback;
+}
+
+
+MA_API ma_result ma_log_init(const ma_allocation_callbacks* pAllocationCallbacks, ma_log* pLog)
+{
+ if (pLog == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ MA_ZERO_OBJECT(pLog);
+ ma_allocation_callbacks_init_copy(&pLog->allocationCallbacks, pAllocationCallbacks);
+
+ /* We need a mutex for thread safety. */
+ #ifndef MA_NO_THREADING
+ {
+ ma_result result = ma_mutex_init(&pLog->lock);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ }
+ #endif
+
+ /* If we're using debug output, enable it. */
+ #if defined(MA_DEBUG_OUTPUT)
+ {
+ ma_log_register_callback(pLog, ma_log_callback_init(ma_log_callback_debug, NULL)); /* Doesn't really matter if this fails. */
+ }
+ #endif
+
+ return MA_SUCCESS;
+}
+
+MA_API void ma_log_uninit(ma_log* pLog)
+{
+ if (pLog == NULL) {
+ return;
+ }
+
+#ifndef MA_NO_THREADING
+ ma_mutex_uninit(&pLog->lock);
+#endif
+}
+
+static void ma_log_lock(ma_log* pLog)
+{
+#ifndef MA_NO_THREADING
+ ma_mutex_lock(&pLog->lock);
+#else
+ (void)pLog;
+#endif
+}
+
+static void ma_log_unlock(ma_log* pLog)
+{
+#ifndef MA_NO_THREADING
+ ma_mutex_unlock(&pLog->lock);
+#else
+ (void)pLog;
+#endif
+}
+
+MA_API ma_result ma_log_register_callback(ma_log* pLog, ma_log_callback callback)
+{
+ ma_result result = MA_SUCCESS;
+
+ if (pLog == NULL || callback.onLog == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ ma_log_lock(pLog);
+ {
+ if (pLog->callbackCount == ma_countof(pLog->callbacks)) {
+ result = MA_OUT_OF_MEMORY; /* Reached the maximum allowed log callbacks. */
+ } else {
+ pLog->callbacks[pLog->callbackCount] = callback;
+ pLog->callbackCount += 1;
+ }
+ }
+ ma_log_unlock(pLog);
+
+ return result;
+}
+
+MA_API ma_result ma_log_unregister_callback(ma_log* pLog, ma_log_callback callback)
+{
+ if (pLog == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ ma_log_lock(pLog);
+ {
+ ma_uint32 iLog;
+ for (iLog = 0; iLog < pLog->callbackCount; ) {
+ if (pLog->callbacks[iLog].onLog == callback.onLog) {
+ /* Found. Move everything down a slot. */
+ ma_uint32 jLog;
+ for (jLog = iLog; jLog < pLog->callbackCount-1; jLog += 1) {
+ pLog->callbacks[jLog] = pLog->callbacks[jLog + 1];
+ }
+
+ pLog->callbackCount -= 1;
+ } else {
+ /* Not found. */
+ iLog += 1;
+ }
+ }
+ }
+ ma_log_unlock(pLog);
+
+ return MA_SUCCESS;
+}
+
+MA_API ma_result ma_log_post(ma_log* pLog, ma_uint32 level, const char* pMessage)
+{
+ if (pLog == NULL || pMessage == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ ma_log_lock(pLog);
+ {
+ ma_uint32 iLog;
+ for (iLog = 0; iLog < pLog->callbackCount; iLog += 1) {
+ if (pLog->callbacks[iLog].onLog) {
+ pLog->callbacks[iLog].onLog(pLog->callbacks[iLog].pUserData, level, pMessage);
+ }
+ }
+ }
+ ma_log_unlock(pLog);
+
+ return MA_SUCCESS;
+}
+
+
+/*
+We need to emulate _vscprintf() for the VC6 build. This can be more efficient, but since it's only VC6, and it's just a
+logging function, I'm happy to keep this simple. In the VC6 build we can implement this in terms of _vsnprintf().
+*/
+#if defined(_MSC_VER) && _MSC_VER < 1900
+static int ma_vscprintf(const ma_allocation_callbacks* pAllocationCallbacks, const char* format, va_list args)
+{
+#if _MSC_VER > 1200
+ return _vscprintf(format, args);
+#else
+ int result;
+ char* pTempBuffer = NULL;
+ size_t tempBufferCap = 1024;
+
+ if (format == NULL) {
+ errno = EINVAL;
+ return -1;
+ }
+
+ for (;;) {
+ char* pNewTempBuffer = (char*)ma_realloc(pTempBuffer, tempBufferCap, pAllocationCallbacks);
+ if (pNewTempBuffer == NULL) {
+ ma_free(pTempBuffer, pAllocationCallbacks);
+ errno = ENOMEM;
+ return -1; /* Out of memory. */
+ }
+
+ pTempBuffer = pNewTempBuffer;
+
+ result = _vsnprintf(pTempBuffer, tempBufferCap, format, args);
+ ma_free(pTempBuffer, NULL);
+
+ if (result != -1) {
+ break; /* Got it. */
+ }
+
+ /* Buffer wasn't big enough. Ideally it'd be nice to use an error code to know the reason for sure, but this is reliable enough. */
+ tempBufferCap *= 2;
+ }
+
+ return result;
+#endif
+}
+#endif
+
+MA_API ma_result ma_log_postv(ma_log* pLog, ma_uint32 level, const char* pFormat, va_list args)
+{
+ if (pLog == NULL || pFormat == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ #if (defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L) || ((!defined(_MSC_VER) || _MSC_VER >= 1900) && !defined(__STRICT_ANSI__) && !defined(_NO_EXT_KEYS))
+ {
+ ma_result result;
+ int length;
+ char pFormattedMessageStack[1024];
+ char* pFormattedMessageHeap = NULL;
+
+ /* First try formatting into our fixed sized stack allocated buffer. If this is too small we'll fallback to a heap allocation. */
+ length = vsnprintf(pFormattedMessageStack, sizeof(pFormattedMessageStack), pFormat, args);
+ if (length < 0) {
+ return MA_INVALID_OPERATION; /* An error occured when trying to convert the buffer. */
+ }
+
+ if ((size_t)length < sizeof(pFormattedMessageStack)) {
+ /* The string was written to the stack. */
+ result = ma_log_post(pLog, level, pFormattedMessageStack);
+ } else {
+ /* The stack buffer was too small, try the heap. */
+ pFormattedMessageHeap = (char*)ma_malloc(length + 1, &pLog->allocationCallbacks);
+ if (pFormattedMessageHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+
+ length = vsnprintf(pFormattedMessageHeap, length + 1, pFormat, args);
+ if (length < 0) {
+ ma_free(pFormattedMessageHeap, &pLog->allocationCallbacks);
+ return MA_INVALID_OPERATION;
+ }
+
+ result = ma_log_post(pLog, level, pFormattedMessageHeap);
+ ma_free(pFormattedMessageHeap, &pLog->allocationCallbacks);
+ }
+
+ return result;
+ }
+ #else
+ {
+ /*
+ Without snprintf() we need to first measure the string and then heap allocate it. I'm only aware of Visual Studio having support for this without snprintf(), so we'll
+ need to restrict this branch to Visual Studio. For other compilers we need to just not support formatted logging because I don't want the security risk of overflowing
+ a fixed sized stack allocated buffer.
+ */
+ #if defined(_MSC_VER) && _MSC_VER >= 1200 /* 1200 = VC6 */
+ {
+ ma_result result;
+ int formattedLen;
+ char* pFormattedMessage = NULL;
+ va_list args2;
+
+ #if _MSC_VER >= 1800
+ {
+ va_copy(args2, args);
+ }
+ #else
+ {
+ args2 = args;
+ }
+ #endif
+
+ formattedLen = ma_vscprintf(&pLog->allocationCallbacks, pFormat, args2);
+ va_end(args2);
+
+ if (formattedLen <= 0) {
+ return MA_INVALID_OPERATION;
+ }
+
+ pFormattedMessage = (char*)ma_malloc(formattedLen + 1, &pLog->allocationCallbacks);
+ if (pFormattedMessage == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+
+ /* We'll get errors on newer versions of Visual Studio if we try to use vsprintf(). */
+ #if _MSC_VER >= 1400 /* 1400 = Visual Studio 2005 */
+ {
+ vsprintf_s(pFormattedMessage, formattedLen + 1, pFormat, args);
+ }
+ #else
+ {
+ vsprintf(pFormattedMessage, pFormat, args);
+ }
+ #endif
+
+ result = ma_log_post(pLog, level, pFormattedMessage);
+ ma_free(pFormattedMessage, &pLog->allocationCallbacks);
+
+ return result;
+ }
+ #else
+ {
+ /* Can't do anything because we don't have a safe way of to emulate vsnprintf() without a manual solution. */
+ (void)level;
+ (void)args;
+
+ return MA_INVALID_OPERATION;
+ }
+ #endif
+ }
+ #endif
+}
+
+MA_API ma_result ma_log_postf(ma_log* pLog, ma_uint32 level, const char* pFormat, ...)
+{
+ ma_result result;
+ va_list args;
+
+ if (pLog == NULL || pFormat == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ va_start(args, pFormat);
+ {
+ result = ma_log_postv(pLog, level, pFormat, args);
+ }
+ va_end(args);
+
+ return result;
+}
+
+
+
+static MA_INLINE ma_uint8 ma_clip_u8(ma_int32 x)
+{
+ return (ma_uint8)(ma_clamp(x, -128, 127) + 128);
+}
+
+static MA_INLINE ma_int16 ma_clip_s16(ma_int32 x)
+{
+ return (ma_int16)ma_clamp(x, -32768, 32767);
+}
+
+static MA_INLINE ma_int64 ma_clip_s24(ma_int64 x)
+{
+ return (ma_int64)ma_clamp(x, -8388608, 8388607);
+}
+
+static MA_INLINE ma_int32 ma_clip_s32(ma_int64 x)
+{
+ /* This dance is to silence warnings with -std=c89. A good compiler should be able to optimize this away. */
+ ma_int64 clipMin;
+ ma_int64 clipMax;
+ clipMin = -((ma_int64)2147483647 + 1);
+ clipMax = (ma_int64)2147483647;
+
+ return (ma_int32)ma_clamp(x, clipMin, clipMax);
+}
+
+static MA_INLINE float ma_clip_f32(float x)
+{
+ if (x < -1) return -1;
+ if (x > +1) return +1;
+ return x;
+}
+
+
+static MA_INLINE float ma_mix_f32(float x, float y, float a)
+{
+ return x*(1-a) + y*a;
+}
+static MA_INLINE float ma_mix_f32_fast(float x, float y, float a)
+{
+ float r0 = (y - x);
+ float r1 = r0*a;
+ return x + r1;
+ /*return x + (y - x)*a;*/
+}
+
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE __m128 ma_mix_f32_fast__sse2(__m128 x, __m128 y, __m128 a)
+{
+ return _mm_add_ps(x, _mm_mul_ps(_mm_sub_ps(y, x), a));
+}
+#endif
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE __m256 ma_mix_f32_fast__avx2(__m256 x, __m256 y, __m256 a)
+{
+ return _mm256_add_ps(x, _mm256_mul_ps(_mm256_sub_ps(y, x), a));
+}
+#endif
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE float32x4_t ma_mix_f32_fast__neon(float32x4_t x, float32x4_t y, float32x4_t a)
+{
+ return vaddq_f32(x, vmulq_f32(vsubq_f32(y, x), a));
+}
+#endif
+
+
+static MA_INLINE double ma_mix_f64(double x, double y, double a)
+{
+ return x*(1-a) + y*a;
+}
+static MA_INLINE double ma_mix_f64_fast(double x, double y, double a)
+{
+ return x + (y - x)*a;
+}
+
+static MA_INLINE float ma_scale_to_range_f32(float x, float lo, float hi)
+{
+ return lo + x*(hi-lo);
+}
+
+
+/*
+Greatest common factor using Euclid's algorithm iteratively.
+*/
+static MA_INLINE ma_uint32 ma_gcf_u32(ma_uint32 a, ma_uint32 b)
+{
+ for (;;) {
+ if (b == 0) {
+ break;
+ } else {
+ ma_uint32 t = a;
+ a = b;
+ b = t % a;
+ }
+ }
+
+ return a;
+}
+
+
+static ma_uint32 ma_ffs_32(ma_uint32 x)
+{
+ ma_uint32 i;
+
+ /* Just a naive implementation just to get things working for now. Will optimize this later. */
+ for (i = 0; i < 32; i += 1) {
+ if ((x & (1 << i)) != 0) {
+ return i;
+ }
+ }
+
+ return i;
+}
+
+static MA_INLINE ma_int16 ma_float_to_fixed_16(float x)
+{
+ return (ma_int16)(x * (1 << 8));
+}
+
+
+
+/*
+Random Number Generation
+
+miniaudio uses the LCG random number generation algorithm. This is good enough for audio.
+
+Note that miniaudio's global LCG implementation uses global state which is _not_ thread-local. When this is called across
+multiple threads, results will be unpredictable. However, it won't crash and results will still be random enough for
+miniaudio's purposes.
+*/
+#ifndef MA_DEFAULT_LCG_SEED
+#define MA_DEFAULT_LCG_SEED 4321
+#endif
+
+#define MA_LCG_M 2147483647
+#define MA_LCG_A 48271
+#define MA_LCG_C 0
+
+static ma_lcg g_maLCG = {MA_DEFAULT_LCG_SEED}; /* Non-zero initial seed. Use ma_seed() to use an explicit seed. */
+
+static MA_INLINE void ma_lcg_seed(ma_lcg* pLCG, ma_int32 seed)
+{
+ MA_ASSERT(pLCG != NULL);
+ pLCG->state = seed;
+}
+
+static MA_INLINE ma_int32 ma_lcg_rand_s32(ma_lcg* pLCG)
+{
+ pLCG->state = (MA_LCG_A * pLCG->state + MA_LCG_C) % MA_LCG_M;
+ return pLCG->state;
+}
+
+static MA_INLINE ma_uint32 ma_lcg_rand_u32(ma_lcg* pLCG)
+{
+ return (ma_uint32)ma_lcg_rand_s32(pLCG);
+}
+
+static MA_INLINE ma_int16 ma_lcg_rand_s16(ma_lcg* pLCG)
+{
+ return (ma_int16)(ma_lcg_rand_s32(pLCG) & 0xFFFF);
+}
+
+static MA_INLINE double ma_lcg_rand_f64(ma_lcg* pLCG)
+{
+ return ma_lcg_rand_s32(pLCG) / (double)0x7FFFFFFF;
+}
+
+static MA_INLINE float ma_lcg_rand_f32(ma_lcg* pLCG)
+{
+ return (float)ma_lcg_rand_f64(pLCG);
+}
+
+static MA_INLINE float ma_lcg_rand_range_f32(ma_lcg* pLCG, float lo, float hi)
+{
+ return ma_scale_to_range_f32(ma_lcg_rand_f32(pLCG), lo, hi);
+}
+
+static MA_INLINE ma_int32 ma_lcg_rand_range_s32(ma_lcg* pLCG, ma_int32 lo, ma_int32 hi)
+{
+ if (lo == hi) {
+ return lo;
+ }
+
+ return lo + ma_lcg_rand_u32(pLCG) / (0xFFFFFFFF / (hi - lo + 1) + 1);
+}
+
+
+
+static MA_INLINE void ma_seed(ma_int32 seed)
+{
+ ma_lcg_seed(&g_maLCG, seed);
+}
+
+static MA_INLINE ma_int32 ma_rand_s32(void)
+{
+ return ma_lcg_rand_s32(&g_maLCG);
+}
+
+static MA_INLINE ma_uint32 ma_rand_u32(void)
+{
+ return ma_lcg_rand_u32(&g_maLCG);
+}
+
+static MA_INLINE double ma_rand_f64(void)
+{
+ return ma_lcg_rand_f64(&g_maLCG);
+}
+
+static MA_INLINE float ma_rand_f32(void)
+{
+ return ma_lcg_rand_f32(&g_maLCG);
+}
+
+static MA_INLINE float ma_rand_range_f32(float lo, float hi)
+{
+ return ma_lcg_rand_range_f32(&g_maLCG, lo, hi);
+}
+
+static MA_INLINE ma_int32 ma_rand_range_s32(ma_int32 lo, ma_int32 hi)
+{
+ return ma_lcg_rand_range_s32(&g_maLCG, lo, hi);
+}
+
+
+static MA_INLINE float ma_dither_f32_rectangle(float ditherMin, float ditherMax)
+{
+ return ma_rand_range_f32(ditherMin, ditherMax);
+}
+
+static MA_INLINE float ma_dither_f32_triangle(float ditherMin, float ditherMax)
+{
+ float a = ma_rand_range_f32(ditherMin, 0);
+ float b = ma_rand_range_f32(0, ditherMax);
+ return a + b;
+}
+
+static MA_INLINE float ma_dither_f32(ma_dither_mode ditherMode, float ditherMin, float ditherMax)
+{
+ if (ditherMode == ma_dither_mode_rectangle) {
+ return ma_dither_f32_rectangle(ditherMin, ditherMax);
+ }
+ if (ditherMode == ma_dither_mode_triangle) {
+ return ma_dither_f32_triangle(ditherMin, ditherMax);
+ }
+
+ return 0;
+}
+
+static MA_INLINE ma_int32 ma_dither_s32(ma_dither_mode ditherMode, ma_int32 ditherMin, ma_int32 ditherMax)
+{
+ if (ditherMode == ma_dither_mode_rectangle) {
+ ma_int32 a = ma_rand_range_s32(ditherMin, ditherMax);
+ return a;
+ }
+ if (ditherMode == ma_dither_mode_triangle) {
+ ma_int32 a = ma_rand_range_s32(ditherMin, 0);
+ ma_int32 b = ma_rand_range_s32(0, ditherMax);
+ return a + b;
+ }
+
+ return 0;
+}
+
+
+/**************************************************************************************************************************************************************
+
+Atomics
+
+**************************************************************************************************************************************************************/
+/* c89atomic.h begin */
+#ifndef c89atomic_h
+#define c89atomic_h
+#if defined(__cplusplus)
+extern "C" {
+#endif
+typedef signed char c89atomic_int8;
+typedef unsigned char c89atomic_uint8;
+typedef signed short c89atomic_int16;
+typedef unsigned short c89atomic_uint16;
+typedef signed int c89atomic_int32;
+typedef unsigned int c89atomic_uint32;
+#if defined(_MSC_VER) && !defined(__clang__)
+ typedef signed __int64 c89atomic_int64;
+ typedef unsigned __int64 c89atomic_uint64;
+#else
+ #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
+ #pragma GCC diagnostic push
+ #pragma GCC diagnostic ignored "-Wlong-long"
+ #if defined(__clang__)
+ #pragma GCC diagnostic ignored "-Wc++11-long-long"
+ #endif
+ #endif
+ typedef signed long long c89atomic_int64;
+ typedef unsigned long long c89atomic_uint64;
+ #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
+ #pragma GCC diagnostic pop
+ #endif
+#endif
+typedef int c89atomic_memory_order;
+typedef unsigned char c89atomic_bool;
+#if !defined(C89ATOMIC_64BIT) && !defined(C89ATOMIC_32BIT)
+#ifdef _WIN32
+#ifdef _WIN64
+#define C89ATOMIC_64BIT
+#else
+#define C89ATOMIC_32BIT
+#endif
+#endif
+#endif
+#if !defined(C89ATOMIC_64BIT) && !defined(C89ATOMIC_32BIT)
+#ifdef __GNUC__
+#ifdef __LP64__
+#define C89ATOMIC_64BIT
+#else
+#define C89ATOMIC_32BIT
+#endif
+#endif
+#endif
#if !defined(C89ATOMIC_64BIT) && !defined(C89ATOMIC_32BIT)
+#include
+#if INTPTR_MAX == INT64_MAX
+#define C89ATOMIC_64BIT
+#else
+#define C89ATOMIC_32BIT
+#endif
+#endif
+#if defined(__x86_64__) || defined(_M_X64)
+#define C89ATOMIC_X64
+#elif defined(__i386) || defined(_M_IX86)
+#define C89ATOMIC_X86
+#elif defined(__arm__) || defined(_M_ARM) || defined(__arm64) || defined(__arm64__) || defined(__aarch64__) || defined(_M_ARM64)
+#define C89ATOMIC_ARM
+#endif
+#if defined(_MSC_VER)
+ #define C89ATOMIC_INLINE __forceinline
+#elif defined(__GNUC__)
+ #if defined(__STRICT_ANSI__)
+ #define C89ATOMIC_INLINE __inline__ __attribute__((always_inline))
+ #else
+ #define C89ATOMIC_INLINE inline __attribute__((always_inline))
+ #endif
+#elif defined(__WATCOMC__) || defined(__DMC__)
+ #define C89ATOMIC_INLINE __inline
+#else
+ #define C89ATOMIC_INLINE
+#endif
+#define C89ATOMIC_HAS_8
+#define C89ATOMIC_HAS_16
+#define C89ATOMIC_HAS_32
+#define C89ATOMIC_HAS_64
+#if (defined(_MSC_VER) ) || defined(__WATCOMC__) || defined(__DMC__)
+ #define c89atomic_memory_order_relaxed 0
+ #define c89atomic_memory_order_consume 1
+ #define c89atomic_memory_order_acquire 2
+ #define c89atomic_memory_order_release 3
+ #define c89atomic_memory_order_acq_rel 4
+ #define c89atomic_memory_order_seq_cst 5
+ #if _MSC_VER < 1600 && defined(C89ATOMIC_X86)
+ #define C89ATOMIC_MSVC_USE_INLINED_ASSEMBLY
+ #endif
+ #if _MSC_VER < 1600
+ #undef C89ATOMIC_HAS_8
+ #undef C89ATOMIC_HAS_16
+ #endif
+ #if !defined(C89ATOMIC_MSVC_USE_INLINED_ASSEMBLY)
+ #include
+ #endif
+ #if defined(C89ATOMIC_MSVC_USE_INLINED_ASSEMBLY)
+ #if defined(C89ATOMIC_HAS_8)
+ static C89ATOMIC_INLINE c89atomic_uint8 __stdcall c89atomic_compare_and_swap_8(volatile c89atomic_uint8* dst, c89atomic_uint8 expected, c89atomic_uint8 desired)
+ {
+ c89atomic_uint8 result = 0;
+ __asm {
+ mov ecx, dst
+ mov al, expected
+ mov dl, desired
+ lock cmpxchg [ecx], dl
+ mov result, al
+ }
+ return result;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_16)
+ static C89ATOMIC_INLINE c89atomic_uint16 __stdcall c89atomic_compare_and_swap_16(volatile c89atomic_uint16* dst, c89atomic_uint16 expected, c89atomic_uint16 desired)
+ {
+ c89atomic_uint16 result = 0;
+ __asm {
+ mov ecx, dst
+ mov ax, expected
+ mov dx, desired
+ lock cmpxchg [ecx], dx
+ mov result, ax
+ }
+ return result;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_32)
+ static C89ATOMIC_INLINE c89atomic_uint32 __stdcall c89atomic_compare_and_swap_32(volatile c89atomic_uint32* dst, c89atomic_uint32 expected, c89atomic_uint32 desired)
+ {
+ c89atomic_uint32 result = 0;
+ __asm {
+ mov ecx, dst
+ mov eax, expected
+ mov edx, desired
+ lock cmpxchg [ecx], edx
+ mov result, eax
+ }
+ return result;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_64)
+ static C89ATOMIC_INLINE c89atomic_uint64 __stdcall c89atomic_compare_and_swap_64(volatile c89atomic_uint64* dst, c89atomic_uint64 expected, c89atomic_uint64 desired)
+ {
+ c89atomic_uint32 resultEAX = 0;
+ c89atomic_uint32 resultEDX = 0;
+ __asm {
+ mov esi, dst
+ mov eax, dword ptr expected
+ mov edx, dword ptr expected + 4
+ mov ebx, dword ptr desired
+ mov ecx, dword ptr desired + 4
+ lock cmpxchg8b qword ptr [esi]
+ mov resultEAX, eax
+ mov resultEDX, edx
+ }
+ return ((c89atomic_uint64)resultEDX << 32) | resultEAX;
+ }
+ #endif
+ #else
+ #if defined(C89ATOMIC_HAS_8)
+ #define c89atomic_compare_and_swap_8( dst, expected, desired) (c89atomic_uint8 )_InterlockedCompareExchange8((volatile char*)dst, (char)desired, (char)expected)
+ #endif
+ #if defined(C89ATOMIC_HAS_16)
+ #define c89atomic_compare_and_swap_16(dst, expected, desired) (c89atomic_uint16)_InterlockedCompareExchange16((volatile short*)dst, (short)desired, (short)expected)
+ #endif
+ #if defined(C89ATOMIC_HAS_32)
+ #define c89atomic_compare_and_swap_32(dst, expected, desired) (c89atomic_uint32)_InterlockedCompareExchange((volatile long*)dst, (long)desired, (long)expected)
+ #endif
+ #if defined(C89ATOMIC_HAS_64)
+ #define c89atomic_compare_and_swap_64(dst, expected, desired) (c89atomic_uint64)_InterlockedCompareExchange64((volatile c89atomic_int64*)dst, (c89atomic_int64)desired, (c89atomic_int64)expected)
+ #endif
+ #endif
+ #if defined(C89ATOMIC_MSVC_USE_INLINED_ASSEMBLY)
+ #if defined(C89ATOMIC_HAS_8)
+ static C89ATOMIC_INLINE c89atomic_uint8 __stdcall c89atomic_exchange_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint8 result = 0;
+ (void)order;
+ __asm {
+ mov ecx, dst
+ mov al, src
+ lock xchg [ecx], al
+ mov result, al
+ }
+ return result;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_16)
+ static C89ATOMIC_INLINE c89atomic_uint16 __stdcall c89atomic_exchange_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint16 result = 0;
+ (void)order;
+ __asm {
+ mov ecx, dst
+ mov ax, src
+ lock xchg [ecx], ax
+ mov result, ax
+ }
+ return result;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_32)
+ static C89ATOMIC_INLINE c89atomic_uint32 __stdcall c89atomic_exchange_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint32 result = 0;
+ (void)order;
+ __asm {
+ mov ecx, dst
+ mov eax, src
+ lock xchg [ecx], eax
+ mov result, eax
+ }
+ return result;
+ }
+ #endif
+ #else
+ #if defined(C89ATOMIC_HAS_8)
+ static C89ATOMIC_INLINE c89atomic_uint8 __stdcall c89atomic_exchange_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return (c89atomic_uint8)_InterlockedExchange8((volatile char*)dst, (char)src);
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_16)
+ static C89ATOMIC_INLINE c89atomic_uint16 __stdcall c89atomic_exchange_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return (c89atomic_uint16)_InterlockedExchange16((volatile short*)dst, (short)src);
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_32)
+ static C89ATOMIC_INLINE c89atomic_uint32 __stdcall c89atomic_exchange_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return (c89atomic_uint32)_InterlockedExchange((volatile long*)dst, (long)src);
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_64) && defined(C89ATOMIC_64BIT)
+ static C89ATOMIC_INLINE c89atomic_uint64 __stdcall c89atomic_exchange_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return (c89atomic_uint64)_InterlockedExchange64((volatile long long*)dst, (long long)src);
+ }
+ #else
+ #endif
+ #endif
+ #if defined(C89ATOMIC_HAS_64) && !defined(C89ATOMIC_64BIT)
+ static C89ATOMIC_INLINE c89atomic_uint64 __stdcall c89atomic_exchange_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint64 oldValue;
+ do {
+ oldValue = *dst;
+ } while (c89atomic_compare_and_swap_64(dst, oldValue, src) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ #endif
+ #if defined(C89ATOMIC_MSVC_USE_INLINED_ASSEMBLY)
+ #if defined(C89ATOMIC_HAS_8)
+ static C89ATOMIC_INLINE c89atomic_uint8 __stdcall c89atomic_fetch_add_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint8 result = 0;
+ (void)order;
+ __asm {
+ mov ecx, dst
+ mov al, src
+ lock xadd [ecx], al
+ mov result, al
+ }
+ return result;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_16)
+ static C89ATOMIC_INLINE c89atomic_uint16 __stdcall c89atomic_fetch_add_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint16 result = 0;
+ (void)order;
+ __asm {
+ mov ecx, dst
+ mov ax, src
+ lock xadd [ecx], ax
+ mov result, ax
+ }
+ return result;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_32)
+ static C89ATOMIC_INLINE c89atomic_uint32 __stdcall c89atomic_fetch_add_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint32 result = 0;
+ (void)order;
+ __asm {
+ mov ecx, dst
+ mov eax, src
+ lock xadd [ecx], eax
+ mov result, eax
+ }
+ return result;
+ }
+ #endif
+ #else
+ #if defined(C89ATOMIC_HAS_8)
+ static C89ATOMIC_INLINE c89atomic_uint8 __stdcall c89atomic_fetch_add_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return (c89atomic_uint8)_InterlockedExchangeAdd8((volatile char*)dst, (char)src);
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_16)
+ static C89ATOMIC_INLINE c89atomic_uint16 __stdcall c89atomic_fetch_add_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return (c89atomic_uint16)_InterlockedExchangeAdd16((volatile short*)dst, (short)src);
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_32)
+ static C89ATOMIC_INLINE c89atomic_uint32 __stdcall c89atomic_fetch_add_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return (c89atomic_uint32)_InterlockedExchangeAdd((volatile long*)dst, (long)src);
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_64) && defined(C89ATOMIC_64BIT)
+ static C89ATOMIC_INLINE c89atomic_uint64 __stdcall c89atomic_fetch_add_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return (c89atomic_uint64)_InterlockedExchangeAdd64((volatile long long*)dst, (long long)src);
+ }
+ #else
+ #endif
+ #endif
+ #if defined(C89ATOMIC_HAS_64) && !defined(C89ATOMIC_64BIT)
+ static C89ATOMIC_INLINE c89atomic_uint64 __stdcall c89atomic_fetch_add_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint64 oldValue;
+ c89atomic_uint64 newValue;
+ do {
+ oldValue = *dst;
+ newValue = oldValue + src;
+ } while (c89atomic_compare_and_swap_64(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ #endif
+ #if defined(C89ATOMIC_MSVC_USE_INLINED_ASSEMBLY)
+ static C89ATOMIC_INLINE void __stdcall c89atomic_thread_fence(c89atomic_memory_order order)
+ {
+ (void)order;
+ __asm {
+ lock add [esp], 0
+ }
+ }
+ #else
+ #if defined(C89ATOMIC_X64)
+ #define c89atomic_thread_fence(order) __faststorefence(), (void)order
+ #else
+ static C89ATOMIC_INLINE void c89atomic_thread_fence(c89atomic_memory_order order)
+ {
+ volatile c89atomic_uint32 barrier = 0;
+ c89atomic_fetch_add_explicit_32(&barrier, 0, order);
+ }
+ #endif
+ #endif
+ #define c89atomic_compiler_fence() c89atomic_thread_fence(c89atomic_memory_order_seq_cst)
+ #define c89atomic_signal_fence(order) c89atomic_thread_fence(order)
+ #if defined(C89ATOMIC_HAS_8)
+ static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_load_explicit_8(volatile const c89atomic_uint8* ptr, c89atomic_memory_order order)
+ {
+ (void)order;
+ return c89atomic_compare_and_swap_8((volatile c89atomic_uint8*)ptr, 0, 0);
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_16)
+ static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_load_explicit_16(volatile const c89atomic_uint16* ptr, c89atomic_memory_order order)
+ {
+ (void)order;
+ return c89atomic_compare_and_swap_16((volatile c89atomic_uint16*)ptr, 0, 0);
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_32)
+ static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_load_explicit_32(volatile const c89atomic_uint32* ptr, c89atomic_memory_order order)
+ {
+ (void)order;
+ return c89atomic_compare_and_swap_32((volatile c89atomic_uint32*)ptr, 0, 0);
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_64)
+ static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_load_explicit_64(volatile const c89atomic_uint64* ptr, c89atomic_memory_order order)
+ {
+ (void)order;
+ return c89atomic_compare_and_swap_64((volatile c89atomic_uint64*)ptr, 0, 0);
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_8)
+ #define c89atomic_store_explicit_8( dst, src, order) (void)c89atomic_exchange_explicit_8 (dst, src, order)
+ #endif
+ #if defined(C89ATOMIC_HAS_16)
+ #define c89atomic_store_explicit_16(dst, src, order) (void)c89atomic_exchange_explicit_16(dst, src, order)
+ #endif
+ #if defined(C89ATOMIC_HAS_32)
+ #define c89atomic_store_explicit_32(dst, src, order) (void)c89atomic_exchange_explicit_32(dst, src, order)
+ #endif
+ #if defined(C89ATOMIC_HAS_64)
+ #define c89atomic_store_explicit_64(dst, src, order) (void)c89atomic_exchange_explicit_64(dst, src, order)
+ #endif
+ #if defined(C89ATOMIC_HAS_8)
+ static C89ATOMIC_INLINE c89atomic_uint8 __stdcall c89atomic_fetch_sub_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint8 oldValue;
+ c89atomic_uint8 newValue;
+ do {
+ oldValue = *dst;
+ newValue = (c89atomic_uint8)(oldValue - src);
+ } while (c89atomic_compare_and_swap_8(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_16)
+ static C89ATOMIC_INLINE c89atomic_uint16 __stdcall c89atomic_fetch_sub_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint16 oldValue;
+ c89atomic_uint16 newValue;
+ do {
+ oldValue = *dst;
+ newValue = (c89atomic_uint16)(oldValue - src);
+ } while (c89atomic_compare_and_swap_16(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_32)
+ static C89ATOMIC_INLINE c89atomic_uint32 __stdcall c89atomic_fetch_sub_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint32 oldValue;
+ c89atomic_uint32 newValue;
+ do {
+ oldValue = *dst;
+ newValue = oldValue - src;
+ } while (c89atomic_compare_and_swap_32(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_64)
+ static C89ATOMIC_INLINE c89atomic_uint64 __stdcall c89atomic_fetch_sub_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint64 oldValue;
+ c89atomic_uint64 newValue;
+ do {
+ oldValue = *dst;
+ newValue = oldValue - src;
+ } while (c89atomic_compare_and_swap_64(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_8)
+ static C89ATOMIC_INLINE c89atomic_uint8 __stdcall c89atomic_fetch_and_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint8 oldValue;
+ c89atomic_uint8 newValue;
+ do {
+ oldValue = *dst;
+ newValue = (c89atomic_uint8)(oldValue & src);
+ } while (c89atomic_compare_and_swap_8(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_16)
+ static C89ATOMIC_INLINE c89atomic_uint16 __stdcall c89atomic_fetch_and_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint16 oldValue;
+ c89atomic_uint16 newValue;
+ do {
+ oldValue = *dst;
+ newValue = (c89atomic_uint16)(oldValue & src);
+ } while (c89atomic_compare_and_swap_16(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_32)
+ static C89ATOMIC_INLINE c89atomic_uint32 __stdcall c89atomic_fetch_and_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint32 oldValue;
+ c89atomic_uint32 newValue;
+ do {
+ oldValue = *dst;
+ newValue = oldValue & src;
+ } while (c89atomic_compare_and_swap_32(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_64)
+ static C89ATOMIC_INLINE c89atomic_uint64 __stdcall c89atomic_fetch_and_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint64 oldValue;
+ c89atomic_uint64 newValue;
+ do {
+ oldValue = *dst;
+ newValue = oldValue & src;
+ } while (c89atomic_compare_and_swap_64(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_8)
+ static C89ATOMIC_INLINE c89atomic_uint8 __stdcall c89atomic_fetch_xor_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint8 oldValue;
+ c89atomic_uint8 newValue;
+ do {
+ oldValue = *dst;
+ newValue = (c89atomic_uint8)(oldValue ^ src);
+ } while (c89atomic_compare_and_swap_8(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_16)
+ static C89ATOMIC_INLINE c89atomic_uint16 __stdcall c89atomic_fetch_xor_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint16 oldValue;
+ c89atomic_uint16 newValue;
+ do {
+ oldValue = *dst;
+ newValue = (c89atomic_uint16)(oldValue ^ src);
+ } while (c89atomic_compare_and_swap_16(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_32)
+ static C89ATOMIC_INLINE c89atomic_uint32 __stdcall c89atomic_fetch_xor_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint32 oldValue;
+ c89atomic_uint32 newValue;
+ do {
+ oldValue = *dst;
+ newValue = oldValue ^ src;
+ } while (c89atomic_compare_and_swap_32(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_64)
+ static C89ATOMIC_INLINE c89atomic_uint64 __stdcall c89atomic_fetch_xor_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint64 oldValue;
+ c89atomic_uint64 newValue;
+ do {
+ oldValue = *dst;
+ newValue = oldValue ^ src;
+ } while (c89atomic_compare_and_swap_64(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_8)
+ static C89ATOMIC_INLINE c89atomic_uint8 __stdcall c89atomic_fetch_or_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint8 oldValue;
+ c89atomic_uint8 newValue;
+ do {
+ oldValue = *dst;
+ newValue = (c89atomic_uint8)(oldValue | src);
+ } while (c89atomic_compare_and_swap_8(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_16)
+ static C89ATOMIC_INLINE c89atomic_uint16 __stdcall c89atomic_fetch_or_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint16 oldValue;
+ c89atomic_uint16 newValue;
+ do {
+ oldValue = *dst;
+ newValue = (c89atomic_uint16)(oldValue | src);
+ } while (c89atomic_compare_and_swap_16(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_32)
+ static C89ATOMIC_INLINE c89atomic_uint32 __stdcall c89atomic_fetch_or_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint32 oldValue;
+ c89atomic_uint32 newValue;
+ do {
+ oldValue = *dst;
+ newValue = oldValue | src;
+ } while (c89atomic_compare_and_swap_32(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_64)
+ static C89ATOMIC_INLINE c89atomic_uint64 __stdcall c89atomic_fetch_or_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint64 oldValue;
+ c89atomic_uint64 newValue;
+ do {
+ oldValue = *dst;
+ newValue = oldValue | src;
+ } while (c89atomic_compare_and_swap_64(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_8)
+ #define c89atomic_test_and_set_explicit_8( dst, order) c89atomic_exchange_explicit_8 (dst, 1, order)
+ #endif
+ #if defined(C89ATOMIC_HAS_16)
+ #define c89atomic_test_and_set_explicit_16(dst, order) c89atomic_exchange_explicit_16(dst, 1, order)
+ #endif
+ #if defined(C89ATOMIC_HAS_32)
+ #define c89atomic_test_and_set_explicit_32(dst, order) c89atomic_exchange_explicit_32(dst, 1, order)
+ #endif
+ #if defined(C89ATOMIC_HAS_64)
+ #define c89atomic_test_and_set_explicit_64(dst, order) c89atomic_exchange_explicit_64(dst, 1, order)
+ #endif
+ #if defined(C89ATOMIC_HAS_8)
+ #define c89atomic_clear_explicit_8( dst, order) c89atomic_store_explicit_8 (dst, 0, order)
+ #endif
+ #if defined(C89ATOMIC_HAS_16)
+ #define c89atomic_clear_explicit_16(dst, order) c89atomic_store_explicit_16(dst, 0, order)
+ #endif
+ #if defined(C89ATOMIC_HAS_32)
+ #define c89atomic_clear_explicit_32(dst, order) c89atomic_store_explicit_32(dst, 0, order)
+ #endif
+ #if defined(C89ATOMIC_HAS_64)
+ #define c89atomic_clear_explicit_64(dst, order) c89atomic_store_explicit_64(dst, 0, order)
+ #endif
+ #if defined(C89ATOMIC_HAS_8)
+ typedef c89atomic_uint8 c89atomic_flag;
+ #define c89atomic_flag_test_and_set_explicit(ptr, order) (c89atomic_bool)c89atomic_test_and_set_explicit_8(ptr, order)
+ #define c89atomic_flag_clear_explicit(ptr, order) c89atomic_clear_explicit_8(ptr, order)
+ #define c89atoimc_flag_load_explicit(ptr, order) c89atomic_load_explicit_8(ptr, order)
+ #else
+ typedef c89atomic_uint32 c89atomic_flag;
+ #define c89atomic_flag_test_and_set_explicit(ptr, order) (c89atomic_bool)c89atomic_test_and_set_explicit_32(ptr, order)
+ #define c89atomic_flag_clear_explicit(ptr, order) c89atomic_clear_explicit_32(ptr, order)
+ #define c89atoimc_flag_load_explicit(ptr, order) c89atomic_load_explicit_32(ptr, order)
+ #endif
+#elif defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 7)))
+ #define C89ATOMIC_HAS_NATIVE_COMPARE_EXCHANGE
+ #define C89ATOMIC_HAS_NATIVE_IS_LOCK_FREE
+ #define c89atomic_memory_order_relaxed __ATOMIC_RELAXED
+ #define c89atomic_memory_order_consume __ATOMIC_CONSUME
+ #define c89atomic_memory_order_acquire __ATOMIC_ACQUIRE
+ #define c89atomic_memory_order_release __ATOMIC_RELEASE
+ #define c89atomic_memory_order_acq_rel __ATOMIC_ACQ_REL
+ #define c89atomic_memory_order_seq_cst __ATOMIC_SEQ_CST
+ #define c89atomic_compiler_fence() __asm__ __volatile__("":::"memory")
+ #define c89atomic_thread_fence(order) __atomic_thread_fence(order)
+ #define c89atomic_signal_fence(order) __atomic_signal_fence(order)
+ #define c89atomic_is_lock_free_8(ptr) __atomic_is_lock_free(1, ptr)
+ #define c89atomic_is_lock_free_16(ptr) __atomic_is_lock_free(2, ptr)
+ #define c89atomic_is_lock_free_32(ptr) __atomic_is_lock_free(4, ptr)
+ #define c89atomic_is_lock_free_64(ptr) __atomic_is_lock_free(8, ptr)
+ #define c89atomic_test_and_set_explicit_8( dst, order) __atomic_exchange_n(dst, 1, order)
+ #define c89atomic_test_and_set_explicit_16(dst, order) __atomic_exchange_n(dst, 1, order)
+ #define c89atomic_test_and_set_explicit_32(dst, order) __atomic_exchange_n(dst, 1, order)
+ #define c89atomic_test_and_set_explicit_64(dst, order) __atomic_exchange_n(dst, 1, order)
+ #define c89atomic_clear_explicit_8( dst, order) __atomic_store_n(dst, 0, order)
+ #define c89atomic_clear_explicit_16(dst, order) __atomic_store_n(dst, 0, order)
+ #define c89atomic_clear_explicit_32(dst, order) __atomic_store_n(dst, 0, order)
+ #define c89atomic_clear_explicit_64(dst, order) __atomic_store_n(dst, 0, order)
+ #define c89atomic_store_explicit_8( dst, src, order) __atomic_store_n(dst, src, order)
+ #define c89atomic_store_explicit_16(dst, src, order) __atomic_store_n(dst, src, order)
+ #define c89atomic_store_explicit_32(dst, src, order) __atomic_store_n(dst, src, order)
+ #define c89atomic_store_explicit_64(dst, src, order) __atomic_store_n(dst, src, order)
+ #define c89atomic_load_explicit_8( dst, order) __atomic_load_n(dst, order)
+ #define c89atomic_load_explicit_16(dst, order) __atomic_load_n(dst, order)
+ #define c89atomic_load_explicit_32(dst, order) __atomic_load_n(dst, order)
+ #define c89atomic_load_explicit_64(dst, order) __atomic_load_n(dst, order)
+ #define c89atomic_exchange_explicit_8( dst, src, order) __atomic_exchange_n(dst, src, order)
+ #define c89atomic_exchange_explicit_16(dst, src, order) __atomic_exchange_n(dst, src, order)
+ #define c89atomic_exchange_explicit_32(dst, src, order) __atomic_exchange_n(dst, src, order)
+ #define c89atomic_exchange_explicit_64(dst, src, order) __atomic_exchange_n(dst, src, order)
+ #define c89atomic_compare_exchange_strong_explicit_8( dst, expected, desired, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, desired, 0, successOrder, failureOrder)
+ #define c89atomic_compare_exchange_strong_explicit_16(dst, expected, desired, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, desired, 0, successOrder, failureOrder)
+ #define c89atomic_compare_exchange_strong_explicit_32(dst, expected, desired, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, desired, 0, successOrder, failureOrder)
+ #define c89atomic_compare_exchange_strong_explicit_64(dst, expected, desired, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, desired, 0, successOrder, failureOrder)
+ #define c89atomic_compare_exchange_weak_explicit_8( dst, expected, desired, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, desired, 1, successOrder, failureOrder)
+ #define c89atomic_compare_exchange_weak_explicit_16(dst, expected, desired, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, desired, 1, successOrder, failureOrder)
+ #define c89atomic_compare_exchange_weak_explicit_32(dst, expected, desired, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, desired, 1, successOrder, failureOrder)
+ #define c89atomic_compare_exchange_weak_explicit_64(dst, expected, desired, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, desired, 1, successOrder, failureOrder)
+ #define c89atomic_fetch_add_explicit_8( dst, src, order) __atomic_fetch_add(dst, src, order)
+ #define c89atomic_fetch_add_explicit_16(dst, src, order) __atomic_fetch_add(dst, src, order)
+ #define c89atomic_fetch_add_explicit_32(dst, src, order) __atomic_fetch_add(dst, src, order)
+ #define c89atomic_fetch_add_explicit_64(dst, src, order) __atomic_fetch_add(dst, src, order)
+ #define c89atomic_fetch_sub_explicit_8( dst, src, order) __atomic_fetch_sub(dst, src, order)
+ #define c89atomic_fetch_sub_explicit_16(dst, src, order) __atomic_fetch_sub(dst, src, order)
+ #define c89atomic_fetch_sub_explicit_32(dst, src, order) __atomic_fetch_sub(dst, src, order)
+ #define c89atomic_fetch_sub_explicit_64(dst, src, order) __atomic_fetch_sub(dst, src, order)
+ #define c89atomic_fetch_or_explicit_8( dst, src, order) __atomic_fetch_or(dst, src, order)
+ #define c89atomic_fetch_or_explicit_16(dst, src, order) __atomic_fetch_or(dst, src, order)
+ #define c89atomic_fetch_or_explicit_32(dst, src, order) __atomic_fetch_or(dst, src, order)
+ #define c89atomic_fetch_or_explicit_64(dst, src, order) __atomic_fetch_or(dst, src, order)
+ #define c89atomic_fetch_xor_explicit_8( dst, src, order) __atomic_fetch_xor(dst, src, order)
+ #define c89atomic_fetch_xor_explicit_16(dst, src, order) __atomic_fetch_xor(dst, src, order)
+ #define c89atomic_fetch_xor_explicit_32(dst, src, order) __atomic_fetch_xor(dst, src, order)
+ #define c89atomic_fetch_xor_explicit_64(dst, src, order) __atomic_fetch_xor(dst, src, order)
+ #define c89atomic_fetch_and_explicit_8( dst, src, order) __atomic_fetch_and(dst, src, order)
+ #define c89atomic_fetch_and_explicit_16(dst, src, order) __atomic_fetch_and(dst, src, order)
+ #define c89atomic_fetch_and_explicit_32(dst, src, order) __atomic_fetch_and(dst, src, order)
+ #define c89atomic_fetch_and_explicit_64(dst, src, order) __atomic_fetch_and(dst, src, order)
+ #define c89atomic_compare_and_swap_8 (dst, expected, desired) __sync_val_compare_and_swap(dst, expected, desired)
+ #define c89atomic_compare_and_swap_16(dst, expected, desired) __sync_val_compare_and_swap(dst, expected, desired)
+ #define c89atomic_compare_and_swap_32(dst, expected, desired) __sync_val_compare_and_swap(dst, expected, desired)
+ #define c89atomic_compare_and_swap_64(dst, expected, desired) __sync_val_compare_and_swap(dst, expected, desired)
+ typedef c89atomic_uint8 c89atomic_flag;
+ #define c89atomic_flag_test_and_set_explicit(dst, order) (c89atomic_bool)__atomic_test_and_set(dst, order)
+ #define c89atomic_flag_clear_explicit(dst, order) __atomic_clear(dst, order)
+ #define c89atoimc_flag_load_explicit(ptr, order) c89atomic_load_explicit_8(ptr, order)
+#else
+ #define c89atomic_memory_order_relaxed 1
+ #define c89atomic_memory_order_consume 2
+ #define c89atomic_memory_order_acquire 3
+ #define c89atomic_memory_order_release 4
+ #define c89atomic_memory_order_acq_rel 5
+ #define c89atomic_memory_order_seq_cst 6
+ #define c89atomic_compiler_fence() __asm__ __volatile__("":::"memory")
+ #if defined(__GNUC__)
+ #define c89atomic_thread_fence(order) __sync_synchronize(), (void)order
+ static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_exchange_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
+ {
+ if (order > c89atomic_memory_order_acquire) {
+ __sync_synchronize();
+ }
+ return __sync_lock_test_and_set(dst, src);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_exchange_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint16 oldValue;
+ do {
+ oldValue = *dst;
+ } while (__sync_val_compare_and_swap(dst, oldValue, src) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_exchange_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint32 oldValue;
+ do {
+ oldValue = *dst;
+ } while (__sync_val_compare_and_swap(dst, oldValue, src) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_exchange_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint64 oldValue;
+ do {
+ oldValue = *dst;
+ } while (__sync_val_compare_and_swap(dst, oldValue, src) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_fetch_add_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return __sync_fetch_and_add(dst, src);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_fetch_add_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return __sync_fetch_and_add(dst, src);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_fetch_add_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return __sync_fetch_and_add(dst, src);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_fetch_add_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return __sync_fetch_and_add(dst, src);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_fetch_sub_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return __sync_fetch_and_sub(dst, src);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_fetch_sub_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return __sync_fetch_and_sub(dst, src);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_fetch_sub_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return __sync_fetch_and_sub(dst, src);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_fetch_sub_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return __sync_fetch_and_sub(dst, src);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_fetch_or_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return __sync_fetch_and_or(dst, src);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_fetch_or_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return __sync_fetch_and_or(dst, src);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_fetch_or_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return __sync_fetch_and_or(dst, src);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_fetch_or_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return __sync_fetch_and_or(dst, src);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_fetch_xor_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return __sync_fetch_and_xor(dst, src);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_fetch_xor_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return __sync_fetch_and_xor(dst, src);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_fetch_xor_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return __sync_fetch_and_xor(dst, src);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_fetch_xor_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return __sync_fetch_and_xor(dst, src);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_fetch_and_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return __sync_fetch_and_and(dst, src);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_fetch_and_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return __sync_fetch_and_and(dst, src);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_fetch_and_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return __sync_fetch_and_and(dst, src);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_fetch_and_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
+ {
+ (void)order;
+ return __sync_fetch_and_and(dst, src);
+ }
+ #define c89atomic_compare_and_swap_8( dst, expected, desired) __sync_val_compare_and_swap(dst, expected, desired)
+ #define c89atomic_compare_and_swap_16(dst, expected, desired) __sync_val_compare_and_swap(dst, expected, desired)
+ #define c89atomic_compare_and_swap_32(dst, expected, desired) __sync_val_compare_and_swap(dst, expected, desired)
+ #define c89atomic_compare_and_swap_64(dst, expected, desired) __sync_val_compare_and_swap(dst, expected, desired)
+ #else
+ #if defined(C89ATOMIC_X86)
+ #define c89atomic_thread_fence(order) __asm__ __volatile__("lock; addl $0, (%%esp)" ::: "memory", "cc")
+ #elif defined(C89ATOMIC_X64)
+ #define c89atomic_thread_fence(order) __asm__ __volatile__("lock; addq $0, (%%rsp)" ::: "memory", "cc")
+ #else
+ #error Unsupported architecture. Please submit a feature request.
+ #endif
+ static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_compare_and_swap_8(volatile c89atomic_uint8* dst, c89atomic_uint8 expected, c89atomic_uint8 desired)
+ {
+ c89atomic_uint8 result;
+ #if defined(C89ATOMIC_X86) || defined(C89ATOMIC_X64)
+ __asm__ __volatile__("lock; cmpxchg %3, %0" : "+m"(*dst), "=a"(result) : "a"(expected), "d"(desired) : "cc");
+ #else
+ #error Unsupported architecture. Please submit a feature request.
+ #endif
+ return result;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_compare_and_swap_16(volatile c89atomic_uint16* dst, c89atomic_uint16 expected, c89atomic_uint16 desired)
+ {
+ c89atomic_uint16 result;
+ #if defined(C89ATOMIC_X86) || defined(C89ATOMIC_X64)
+ __asm__ __volatile__("lock; cmpxchg %3, %0" : "+m"(*dst), "=a"(result) : "a"(expected), "d"(desired) : "cc");
+ #else
+ #error Unsupported architecture. Please submit a feature request.
+ #endif
+ return result;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_compare_and_swap_32(volatile c89atomic_uint32* dst, c89atomic_uint32 expected, c89atomic_uint32 desired)
+ {
+ c89atomic_uint32 result;
+ #if defined(C89ATOMIC_X86) || defined(C89ATOMIC_X64)
+ __asm__ __volatile__("lock; cmpxchg %3, %0" : "+m"(*dst), "=a"(result) : "a"(expected), "d"(desired) : "cc");
+ #else
+ #error Unsupported architecture. Please submit a feature request.
+ #endif
+ return result;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_compare_and_swap_64(volatile c89atomic_uint64* dst, c89atomic_uint64 expected, c89atomic_uint64 desired)
+ {
+ volatile c89atomic_uint64 result;
+ #if defined(C89ATOMIC_X86)
+ c89atomic_uint32 resultEAX;
+ c89atomic_uint32 resultEDX;
+ __asm__ __volatile__("push %%ebx; xchg %5, %%ebx; lock; cmpxchg8b %0; pop %%ebx" : "+m"(*dst), "=a"(resultEAX), "=d"(resultEDX) : "a"(expected & 0xFFFFFFFF), "d"(expected >> 32), "r"(desired & 0xFFFFFFFF), "c"(desired >> 32) : "cc");
+ result = ((c89atomic_uint64)resultEDX << 32) | resultEAX;
+ #elif defined(C89ATOMIC_X64)
+ __asm__ __volatile__("lock; cmpxchg %3, %0" : "+m"(*dst), "=a"(result) : "a"(expected), "d"(desired) : "cc");
+ #else
+ #error Unsupported architecture. Please submit a feature request.
+ #endif
+ return result;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_exchange_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint8 result = 0;
+ (void)order;
+ #if defined(C89ATOMIC_X86) || defined(C89ATOMIC_X64)
+ __asm__ __volatile__("lock; xchg %1, %0" : "+m"(*dst), "=a"(result) : "a"(src));
+ #else
+ #error Unsupported architecture. Please submit a feature request.
+ #endif
+ return result;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_exchange_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint16 result = 0;
+ (void)order;
+ #if defined(C89ATOMIC_X86) || defined(C89ATOMIC_X64)
+ __asm__ __volatile__("lock; xchg %1, %0" : "+m"(*dst), "=a"(result) : "a"(src));
+ #else
+ #error Unsupported architecture. Please submit a feature request.
+ #endif
+ return result;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_exchange_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint32 result;
+ (void)order;
+ #if defined(C89ATOMIC_X86) || defined(C89ATOMIC_X64)
+ __asm__ __volatile__("lock; xchg %1, %0" : "+m"(*dst), "=a"(result) : "a"(src));
+ #else
+ #error Unsupported architecture. Please submit a feature request.
+ #endif
+ return result;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_exchange_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint64 result;
+ (void)order;
+ #if defined(C89ATOMIC_X86)
+ do {
+ result = *dst;
+ } while (c89atomic_compare_and_swap_64(dst, result, src) != result);
+ #elif defined(C89ATOMIC_X64)
+ __asm__ __volatile__("lock; xchg %1, %0" : "+m"(*dst), "=a"(result) : "a"(src));
+ #else
+ #error Unsupported architecture. Please submit a feature request.
+ #endif
+ return result;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_fetch_add_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint8 result;
+ (void)order;
+ #if defined(C89ATOMIC_X86) || defined(C89ATOMIC_X64)
+ __asm__ __volatile__("lock; xadd %1, %0" : "+m"(*dst), "=a"(result) : "a"(src) : "cc");
+ #else
+ #error Unsupported architecture. Please submit a feature request.
+ #endif
+ return result;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_fetch_add_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint16 result;
+ (void)order;
+ #if defined(C89ATOMIC_X86) || defined(C89ATOMIC_X64)
+ __asm__ __volatile__("lock; xadd %1, %0" : "+m"(*dst), "=a"(result) : "a"(src) : "cc");
+ #else
+ #error Unsupported architecture. Please submit a feature request.
+ #endif
+ return result;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_fetch_add_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint32 result;
+ (void)order;
+ #if defined(C89ATOMIC_X86) || defined(C89ATOMIC_X64)
+ __asm__ __volatile__("lock; xadd %1, %0" : "+m"(*dst), "=a"(result) : "a"(src) : "cc");
+ #else
+ #error Unsupported architecture. Please submit a feature request.
+ #endif
+ return result;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_fetch_add_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
+ {
+ #if defined(C89ATOMIC_X86)
+ c89atomic_uint64 oldValue;
+ c89atomic_uint64 newValue;
+ (void)order;
+ do {
+ oldValue = *dst;
+ newValue = oldValue + src;
+ } while (c89atomic_compare_and_swap_64(dst, oldValue, newValue) != oldValue);
+ return oldValue;
+ #elif defined(C89ATOMIC_X64)
+ c89atomic_uint64 result;
+ (void)order;
+ __asm__ __volatile__("lock; xadd %1, %0" : "+m"(*dst), "=a"(result) : "a"(src) : "cc");
+ return result;
+ #endif
+ }
+ static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_fetch_sub_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint8 oldValue;
+ c89atomic_uint8 newValue;
+ do {
+ oldValue = *dst;
+ newValue = (c89atomic_uint8)(oldValue - src);
+ } while (c89atomic_compare_and_swap_8(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_fetch_sub_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint16 oldValue;
+ c89atomic_uint16 newValue;
+ do {
+ oldValue = *dst;
+ newValue = (c89atomic_uint16)(oldValue - src);
+ } while (c89atomic_compare_and_swap_16(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_fetch_sub_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint32 oldValue;
+ c89atomic_uint32 newValue;
+ do {
+ oldValue = *dst;
+ newValue = oldValue - src;
+ } while (c89atomic_compare_and_swap_32(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_fetch_sub_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint64 oldValue;
+ c89atomic_uint64 newValue;
+ do {
+ oldValue = *dst;
+ newValue = oldValue - src;
+ } while (c89atomic_compare_and_swap_64(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_fetch_and_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint8 oldValue;
+ c89atomic_uint8 newValue;
+ do {
+ oldValue = *dst;
+ newValue = (c89atomic_uint8)(oldValue & src);
+ } while (c89atomic_compare_and_swap_8(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_fetch_and_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint16 oldValue;
+ c89atomic_uint16 newValue;
+ do {
+ oldValue = *dst;
+ newValue = (c89atomic_uint16)(oldValue & src);
+ } while (c89atomic_compare_and_swap_16(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_fetch_and_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint32 oldValue;
+ c89atomic_uint32 newValue;
+ do {
+ oldValue = *dst;
+ newValue = oldValue & src;
+ } while (c89atomic_compare_and_swap_32(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_fetch_and_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint64 oldValue;
+ c89atomic_uint64 newValue;
+ do {
+ oldValue = *dst;
+ newValue = oldValue & src;
+ } while (c89atomic_compare_and_swap_64(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_fetch_xor_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint8 oldValue;
+ c89atomic_uint8 newValue;
+ do {
+ oldValue = *dst;
+ newValue = (c89atomic_uint8)(oldValue ^ src);
+ } while (c89atomic_compare_and_swap_8(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_fetch_xor_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint16 oldValue;
+ c89atomic_uint16 newValue;
+ do {
+ oldValue = *dst;
+ newValue = (c89atomic_uint16)(oldValue ^ src);
+ } while (c89atomic_compare_and_swap_16(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_fetch_xor_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint32 oldValue;
+ c89atomic_uint32 newValue;
+ do {
+ oldValue = *dst;
+ newValue = oldValue ^ src;
+ } while (c89atomic_compare_and_swap_32(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_fetch_xor_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint64 oldValue;
+ c89atomic_uint64 newValue;
+ do {
+ oldValue = *dst;
+ newValue = oldValue ^ src;
+ } while (c89atomic_compare_and_swap_64(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_fetch_or_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint8 oldValue;
+ c89atomic_uint8 newValue;
+ do {
+ oldValue = *dst;
+ newValue = (c89atomic_uint8)(oldValue | src);
+ } while (c89atomic_compare_and_swap_8(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_fetch_or_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint16 oldValue;
+ c89atomic_uint16 newValue;
+ do {
+ oldValue = *dst;
+ newValue = (c89atomic_uint16)(oldValue | src);
+ } while (c89atomic_compare_and_swap_16(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_fetch_or_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint32 oldValue;
+ c89atomic_uint32 newValue;
+ do {
+ oldValue = *dst;
+ newValue = oldValue | src;
+ } while (c89atomic_compare_and_swap_32(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_fetch_or_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
+ {
+ c89atomic_uint64 oldValue;
+ c89atomic_uint64 newValue;
+ do {
+ oldValue = *dst;
+ newValue = oldValue | src;
+ } while (c89atomic_compare_and_swap_64(dst, oldValue, newValue) != oldValue);
+ (void)order;
+ return oldValue;
+ }
+ #endif
+ #define c89atomic_signal_fence(order) c89atomic_thread_fence(order)
+ static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_load_explicit_8(volatile const c89atomic_uint8* ptr, c89atomic_memory_order order)
+ {
+ (void)order;
+ return c89atomic_compare_and_swap_8((c89atomic_uint8*)ptr, 0, 0);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_load_explicit_16(volatile const c89atomic_uint16* ptr, c89atomic_memory_order order)
+ {
+ (void)order;
+ return c89atomic_compare_and_swap_16((c89atomic_uint16*)ptr, 0, 0);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_load_explicit_32(volatile const c89atomic_uint32* ptr, c89atomic_memory_order order)
+ {
+ (void)order;
+ return c89atomic_compare_and_swap_32((c89atomic_uint32*)ptr, 0, 0);
+ }
+ static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_load_explicit_64(volatile const c89atomic_uint64* ptr, c89atomic_memory_order order)
+ {
+ (void)order;
+ return c89atomic_compare_and_swap_64((c89atomic_uint64*)ptr, 0, 0);
+ }
+ #define c89atomic_store_explicit_8( dst, src, order) (void)c89atomic_exchange_explicit_8 (dst, src, order)
+ #define c89atomic_store_explicit_16(dst, src, order) (void)c89atomic_exchange_explicit_16(dst, src, order)
+ #define c89atomic_store_explicit_32(dst, src, order) (void)c89atomic_exchange_explicit_32(dst, src, order)
+ #define c89atomic_store_explicit_64(dst, src, order) (void)c89atomic_exchange_explicit_64(dst, src, order)
+ #define c89atomic_test_and_set_explicit_8( dst, order) c89atomic_exchange_explicit_8 (dst, 1, order)
+ #define c89atomic_test_and_set_explicit_16(dst, order) c89atomic_exchange_explicit_16(dst, 1, order)
+ #define c89atomic_test_and_set_explicit_32(dst, order) c89atomic_exchange_explicit_32(dst, 1, order)
+ #define c89atomic_test_and_set_explicit_64(dst, order) c89atomic_exchange_explicit_64(dst, 1, order)
+ #define c89atomic_clear_explicit_8( dst, order) c89atomic_store_explicit_8 (dst, 0, order)
+ #define c89atomic_clear_explicit_16(dst, order) c89atomic_store_explicit_16(dst, 0, order)
+ #define c89atomic_clear_explicit_32(dst, order) c89atomic_store_explicit_32(dst, 0, order)
+ #define c89atomic_clear_explicit_64(dst, order) c89atomic_store_explicit_64(dst, 0, order)
+ typedef c89atomic_uint8 c89atomic_flag;
+ #define c89atomic_flag_test_and_set_explicit(ptr, order) (c89atomic_bool)c89atomic_test_and_set_explicit_8(ptr, order)
+ #define c89atomic_flag_clear_explicit(ptr, order) c89atomic_clear_explicit_8(ptr, order)
+ #define c89atoimc_flag_load_explicit(ptr, order) c89atomic_load_explicit_8(ptr, order)
+#endif
+#if !defined(C89ATOMIC_HAS_NATIVE_COMPARE_EXCHANGE)
+ #if defined(C89ATOMIC_HAS_8)
+ c89atomic_bool c89atomic_compare_exchange_strong_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8* expected, c89atomic_uint8 desired, c89atomic_memory_order successOrder, c89atomic_memory_order failureOrder)
+ {
+ c89atomic_uint8 expectedValue;
+ c89atomic_uint8 result;
+ (void)successOrder;
+ (void)failureOrder;
+ expectedValue = c89atomic_load_explicit_8(expected, c89atomic_memory_order_seq_cst);
+ result = c89atomic_compare_and_swap_8(dst, expectedValue, desired);
+ if (result == expectedValue) {
+ return 1;
+ } else {
+ c89atomic_store_explicit_8(expected, result, failureOrder);
+ return 0;
+ }
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_16)
+ c89atomic_bool c89atomic_compare_exchange_strong_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16* expected, c89atomic_uint16 desired, c89atomic_memory_order successOrder, c89atomic_memory_order failureOrder)
+ {
+ c89atomic_uint16 expectedValue;
+ c89atomic_uint16 result;
+ (void)successOrder;
+ (void)failureOrder;
+ expectedValue = c89atomic_load_explicit_16(expected, c89atomic_memory_order_seq_cst);
+ result = c89atomic_compare_and_swap_16(dst, expectedValue, desired);
+ if (result == expectedValue) {
+ return 1;
+ } else {
+ c89atomic_store_explicit_16(expected, result, failureOrder);
+ return 0;
+ }
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_32)
+ c89atomic_bool c89atomic_compare_exchange_strong_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32* expected, c89atomic_uint32 desired, c89atomic_memory_order successOrder, c89atomic_memory_order failureOrder)
+ {
+ c89atomic_uint32 expectedValue;
+ c89atomic_uint32 result;
+ (void)successOrder;
+ (void)failureOrder;
+ expectedValue = c89atomic_load_explicit_32(expected, c89atomic_memory_order_seq_cst);
+ result = c89atomic_compare_and_swap_32(dst, expectedValue, desired);
+ if (result == expectedValue) {
+ return 1;
+ } else {
+ c89atomic_store_explicit_32(expected, result, failureOrder);
+ return 0;
+ }
+ }
+ #endif
+ #if defined(C89ATOMIC_HAS_64)
+ c89atomic_bool c89atomic_compare_exchange_strong_explicit_64(volatile c89atomic_uint64* dst, volatile c89atomic_uint64* expected, c89atomic_uint64 desired, c89atomic_memory_order successOrder, c89atomic_memory_order failureOrder)
+ {
+ c89atomic_uint64 expectedValue;
+ c89atomic_uint64 result;
+ (void)successOrder;
+ (void)failureOrder;
+ expectedValue = c89atomic_load_explicit_64(expected, c89atomic_memory_order_seq_cst);
+ result = c89atomic_compare_and_swap_64(dst, expectedValue, desired);
+ if (result == expectedValue) {
+ return 1;
+ } else {
+ c89atomic_store_explicit_64(expected, result, failureOrder);
+ return 0;
+ }
+ }
+ #endif
+ #define c89atomic_compare_exchange_weak_explicit_8( dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_strong_explicit_8 (dst, expected, desired, successOrder, failureOrder)
+ #define c89atomic_compare_exchange_weak_explicit_16(dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_strong_explicit_16(dst, expected, desired, successOrder, failureOrder)
+ #define c89atomic_compare_exchange_weak_explicit_32(dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_strong_explicit_32(dst, expected, desired, successOrder, failureOrder)
+ #define c89atomic_compare_exchange_weak_explicit_64(dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_strong_explicit_64(dst, expected, desired, successOrder, failureOrder)
+#endif
+#if !defined(C89ATOMIC_HAS_NATIVE_IS_LOCK_FREE)
+ static C89ATOMIC_INLINE c89atomic_bool c89atomic_is_lock_free_8(volatile void* ptr)
+ {
+ (void)ptr;
+ return 1;
+ }
+ static C89ATOMIC_INLINE c89atomic_bool c89atomic_is_lock_free_16(volatile void* ptr)
+ {
+ (void)ptr;
+ return 1;
+ }
+ static C89ATOMIC_INLINE c89atomic_bool c89atomic_is_lock_free_32(volatile void* ptr)
+ {
+ (void)ptr;
+ return 1;
+ }
+ static C89ATOMIC_INLINE c89atomic_bool c89atomic_is_lock_free_64(volatile void* ptr)
+ {
+ (void)ptr;
+ #if defined(C89ATOMIC_64BIT)
+ return 1;
+ #else
+ #if defined(C89ATOMIC_X86) || defined(C89ATOMIC_X64)
+ return 1;
+ #else
+ return 0;
+ #endif
+ #endif
+ }
+#endif
+#if defined(C89ATOMIC_64BIT)
+ static C89ATOMIC_INLINE c89atomic_bool c89atomic_is_lock_free_ptr(volatile void** ptr)
+ {
+ return c89atomic_is_lock_free_64((volatile c89atomic_uint64*)ptr);
+ }
+ static C89ATOMIC_INLINE void* c89atomic_load_explicit_ptr(volatile void** ptr, c89atomic_memory_order order)
+ {
+ return (void*)c89atomic_load_explicit_64((volatile c89atomic_uint64*)ptr, order);
+ }
+ static C89ATOMIC_INLINE void c89atomic_store_explicit_ptr(volatile void** dst, void* src, c89atomic_memory_order order)
+ {
+ c89atomic_store_explicit_64((volatile c89atomic_uint64*)dst, (c89atomic_uint64)src, order);
+ }
+ static C89ATOMIC_INLINE void* c89atomic_exchange_explicit_ptr(volatile void** dst, void* src, c89atomic_memory_order order)
+ {
+ return (void*)c89atomic_exchange_explicit_64((volatile c89atomic_uint64*)dst, (c89atomic_uint64)src, order);
+ }
+ static C89ATOMIC_INLINE c89atomic_bool c89atomic_compare_exchange_strong_explicit_ptr(volatile void** dst, void** expected, void* desired, c89atomic_memory_order successOrder, c89atomic_memory_order failureOrder)
+ {
+ return c89atomic_compare_exchange_strong_explicit_64((volatile c89atomic_uint64*)dst, (c89atomic_uint64*)expected, (c89atomic_uint64)desired, successOrder, failureOrder);
+ }
+ static C89ATOMIC_INLINE c89atomic_bool c89atomic_compare_exchange_weak_explicit_ptr(volatile void** dst, void** expected, void* desired, c89atomic_memory_order successOrder, c89atomic_memory_order failureOrder)
+ {
+ return c89atomic_compare_exchange_weak_explicit_64((volatile c89atomic_uint64*)dst, (c89atomic_uint64*)expected, (c89atomic_uint64)desired, successOrder, failureOrder);
+ }
+ static C89ATOMIC_INLINE void* c89atomic_compare_and_swap_ptr(volatile void** dst, void* expected, void* desired)
+ {
+ return (void*)c89atomic_compare_and_swap_64((volatile c89atomic_uint64*)dst, (c89atomic_uint64)expected, (c89atomic_uint64)desired);
+ }
+#elif defined(C89ATOMIC_32BIT)
+ static C89ATOMIC_INLINE c89atomic_bool c89atomic_is_lock_free_ptr(volatile void** ptr)
+ {
+ return c89atomic_is_lock_free_32((volatile c89atomic_uint32*)ptr);
+ }
+ static C89ATOMIC_INLINE void* c89atomic_load_explicit_ptr(volatile void** ptr, c89atomic_memory_order order)
+ {
+ return (void*)c89atomic_load_explicit_32((volatile c89atomic_uint32*)ptr, order);
+ }
+ static C89ATOMIC_INLINE void c89atomic_store_explicit_ptr(volatile void** dst, void* src, c89atomic_memory_order order)
+ {
+ c89atomic_store_explicit_32((volatile c89atomic_uint32*)dst, (c89atomic_uint32)src, order);
+ }
+ static C89ATOMIC_INLINE void* c89atomic_exchange_explicit_ptr(volatile void** dst, void* src, c89atomic_memory_order order)
+ {
+ return (void*)c89atomic_exchange_explicit_32((volatile c89atomic_uint32*)dst, (c89atomic_uint32)src, order);
+ }
+ static C89ATOMIC_INLINE c89atomic_bool c89atomic_compare_exchange_strong_explicit_ptr(volatile void** dst, void** expected, void* desired, c89atomic_memory_order successOrder, c89atomic_memory_order failureOrder)
+ {
+ return c89atomic_compare_exchange_strong_explicit_32((volatile c89atomic_uint32*)dst, (c89atomic_uint32*)expected, (c89atomic_uint32)desired, successOrder, failureOrder);
+ }
+ static C89ATOMIC_INLINE c89atomic_bool c89atomic_compare_exchange_weak_explicit_ptr(volatile void** dst, void** expected, void* desired, c89atomic_memory_order successOrder, c89atomic_memory_order failureOrder)
+ {
+ return c89atomic_compare_exchange_weak_explicit_32((volatile c89atomic_uint32*)dst, (c89atomic_uint32*)expected, (c89atomic_uint32)desired, successOrder, failureOrder);
+ }
+ static C89ATOMIC_INLINE void* c89atomic_compare_and_swap_ptr(volatile void** dst, void* expected, void* desired)
+ {
+ return (void*)c89atomic_compare_and_swap_32((volatile c89atomic_uint32*)dst, (c89atomic_uint32)expected, (c89atomic_uint32)desired);
+ }
+#else
+ #error Unsupported architecture.
+#endif
+#define c89atomic_flag_test_and_set(ptr) c89atomic_flag_test_and_set_explicit(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_flag_clear(ptr) c89atomic_flag_clear_explicit(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_store_ptr(dst, src) c89atomic_store_explicit_ptr((volatile void**)dst, (void*)src, c89atomic_memory_order_seq_cst)
+#define c89atomic_load_ptr(ptr) c89atomic_load_explicit_ptr((volatile void**)ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_exchange_ptr(dst, src) c89atomic_exchange_explicit_ptr((volatile void**)dst, (void*)src, c89atomic_memory_order_seq_cst)
+#define c89atomic_compare_exchange_strong_ptr(dst, expected, desired) c89atomic_compare_exchange_strong_explicit_ptr((volatile void**)dst, (void**)expected, (void*)desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
+#define c89atomic_compare_exchange_weak_ptr(dst, expected, desired) c89atomic_compare_exchange_weak_explicit_ptr((volatile void**)dst, (void**)expected, (void*)desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
+#define c89atomic_test_and_set_8( ptr) c89atomic_test_and_set_explicit_8( ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_test_and_set_16(ptr) c89atomic_test_and_set_explicit_16(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_test_and_set_32(ptr) c89atomic_test_and_set_explicit_32(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_test_and_set_64(ptr) c89atomic_test_and_set_explicit_64(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_clear_8( ptr) c89atomic_clear_explicit_8( ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_clear_16(ptr) c89atomic_clear_explicit_16(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_clear_32(ptr) c89atomic_clear_explicit_32(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_clear_64(ptr) c89atomic_clear_explicit_64(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_store_8( dst, src) c89atomic_store_explicit_8( dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_store_16(dst, src) c89atomic_store_explicit_16(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_store_32(dst, src) c89atomic_store_explicit_32(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_store_64(dst, src) c89atomic_store_explicit_64(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_load_8( ptr) c89atomic_load_explicit_8( ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_load_16(ptr) c89atomic_load_explicit_16(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_load_32(ptr) c89atomic_load_explicit_32(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_load_64(ptr) c89atomic_load_explicit_64(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_exchange_8( dst, src) c89atomic_exchange_explicit_8( dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_exchange_16(dst, src) c89atomic_exchange_explicit_16(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_exchange_32(dst, src) c89atomic_exchange_explicit_32(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_exchange_64(dst, src) c89atomic_exchange_explicit_64(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_compare_exchange_strong_8( dst, expected, desired) c89atomic_compare_exchange_strong_explicit_8( dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
+#define c89atomic_compare_exchange_strong_16(dst, expected, desired) c89atomic_compare_exchange_strong_explicit_16(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
+#define c89atomic_compare_exchange_strong_32(dst, expected, desired) c89atomic_compare_exchange_strong_explicit_32(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
+#define c89atomic_compare_exchange_strong_64(dst, expected, desired) c89atomic_compare_exchange_strong_explicit_64(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
+#define c89atomic_compare_exchange_weak_8( dst, expected, desired) c89atomic_compare_exchange_weak_explicit_8( dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
+#define c89atomic_compare_exchange_weak_16( dst, expected, desired) c89atomic_compare_exchange_weak_explicit_16(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
+#define c89atomic_compare_exchange_weak_32( dst, expected, desired) c89atomic_compare_exchange_weak_explicit_32(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
+#define c89atomic_compare_exchange_weak_64( dst, expected, desired) c89atomic_compare_exchange_weak_explicit_64(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_add_8( dst, src) c89atomic_fetch_add_explicit_8( dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_add_16(dst, src) c89atomic_fetch_add_explicit_16(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_add_32(dst, src) c89atomic_fetch_add_explicit_32(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_add_64(dst, src) c89atomic_fetch_add_explicit_64(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_sub_8( dst, src) c89atomic_fetch_sub_explicit_8( dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_sub_16(dst, src) c89atomic_fetch_sub_explicit_16(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_sub_32(dst, src) c89atomic_fetch_sub_explicit_32(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_sub_64(dst, src) c89atomic_fetch_sub_explicit_64(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_or_8( dst, src) c89atomic_fetch_or_explicit_8( dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_or_16(dst, src) c89atomic_fetch_or_explicit_16(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_or_32(dst, src) c89atomic_fetch_or_explicit_32(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_or_64(dst, src) c89atomic_fetch_or_explicit_64(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_xor_8( dst, src) c89atomic_fetch_xor_explicit_8( dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_xor_16(dst, src) c89atomic_fetch_xor_explicit_16(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_xor_32(dst, src) c89atomic_fetch_xor_explicit_32(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_xor_64(dst, src) c89atomic_fetch_xor_explicit_64(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_and_8( dst, src) c89atomic_fetch_and_explicit_8 (dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_and_16(dst, src) c89atomic_fetch_and_explicit_16(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_and_32(dst, src) c89atomic_fetch_and_explicit_32(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_and_64(dst, src) c89atomic_fetch_and_explicit_64(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_test_and_set_explicit_i8( ptr, order) (c89atomic_int8 )c89atomic_test_and_set_explicit_8( (c89atomic_uint8* )ptr, order)
+#define c89atomic_test_and_set_explicit_i16(ptr, order) (c89atomic_int16)c89atomic_test_and_set_explicit_16((c89atomic_uint16*)ptr, order)
+#define c89atomic_test_and_set_explicit_i32(ptr, order) (c89atomic_int32)c89atomic_test_and_set_explicit_32((c89atomic_uint32*)ptr, order)
+#define c89atomic_test_and_set_explicit_i64(ptr, order) (c89atomic_int64)c89atomic_test_and_set_explicit_64((c89atomic_uint64*)ptr, order)
+#define c89atomic_clear_explicit_i8( ptr, order) c89atomic_clear_explicit_8( (c89atomic_uint8* )ptr, order)
+#define c89atomic_clear_explicit_i16(ptr, order) c89atomic_clear_explicit_16((c89atomic_uint16*)ptr, order)
+#define c89atomic_clear_explicit_i32(ptr, order) c89atomic_clear_explicit_32((c89atomic_uint32*)ptr, order)
+#define c89atomic_clear_explicit_i64(ptr, order) c89atomic_clear_explicit_64((c89atomic_uint64*)ptr, order)
+#define c89atomic_store_explicit_i8( dst, src, order) (c89atomic_int8 )c89atomic_store_explicit_8( (c89atomic_uint8* )dst, (c89atomic_uint8 )src, order)
+#define c89atomic_store_explicit_i16(dst, src, order) (c89atomic_int16)c89atomic_store_explicit_16((c89atomic_uint16*)dst, (c89atomic_uint16)src, order)
+#define c89atomic_store_explicit_i32(dst, src, order) (c89atomic_int32)c89atomic_store_explicit_32((c89atomic_uint32*)dst, (c89atomic_uint32)src, order)
+#define c89atomic_store_explicit_i64(dst, src, order) (c89atomic_int64)c89atomic_store_explicit_64((c89atomic_uint64*)dst, (c89atomic_uint64)src, order)
+#define c89atomic_load_explicit_i8( ptr, order) (c89atomic_int8 )c89atomic_load_explicit_8( (c89atomic_uint8* )ptr, order)
+#define c89atomic_load_explicit_i16(ptr, order) (c89atomic_int16)c89atomic_load_explicit_16((c89atomic_uint16*)ptr, order)
+#define c89atomic_load_explicit_i32(ptr, order) (c89atomic_int32)c89atomic_load_explicit_32((c89atomic_uint32*)ptr, order)
+#define c89atomic_load_explicit_i64(ptr, order) (c89atomic_int64)c89atomic_load_explicit_64((c89atomic_uint64*)ptr, order)
+#define c89atomic_exchange_explicit_i8( dst, src, order) (c89atomic_int8 )c89atomic_exchange_explicit_8 ((c89atomic_uint8* )dst, (c89atomic_uint8 )src, order)
+#define c89atomic_exchange_explicit_i16(dst, src, order) (c89atomic_int16)c89atomic_exchange_explicit_16((c89atomic_uint16*)dst, (c89atomic_uint16)src, order)
+#define c89atomic_exchange_explicit_i32(dst, src, order) (c89atomic_int32)c89atomic_exchange_explicit_32((c89atomic_uint32*)dst, (c89atomic_uint32)src, order)
+#define c89atomic_exchange_explicit_i64(dst, src, order) (c89atomic_int64)c89atomic_exchange_explicit_64((c89atomic_uint64*)dst, (c89atomic_uint64)src, order)
+#define c89atomic_compare_exchange_strong_explicit_i8( dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_strong_explicit_8( (c89atomic_uint8* )dst, (c89atomic_uint8* )expected, (c89atomic_uint8 )desired, successOrder, failureOrder)
+#define c89atomic_compare_exchange_strong_explicit_i16(dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_strong_explicit_16((c89atomic_uint16*)dst, (c89atomic_uint16*)expected, (c89atomic_uint16)desired, successOrder, failureOrder)
+#define c89atomic_compare_exchange_strong_explicit_i32(dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_strong_explicit_32((c89atomic_uint32*)dst, (c89atomic_uint32*)expected, (c89atomic_uint32)desired, successOrder, failureOrder)
+#define c89atomic_compare_exchange_strong_explicit_i64(dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_strong_explicit_64((c89atomic_uint64*)dst, (c89atomic_uint64*)expected, (c89atomic_uint64)desired, successOrder, failureOrder)
+#define c89atomic_compare_exchange_weak_explicit_i8( dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_weak_explicit_8( (c89atomic_uint8* )dst, (c89atomic_uint8* )expected, (c89atomic_uint8 )desired, successOrder, failureOrder)
+#define c89atomic_compare_exchange_weak_explicit_i16(dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_weak_explicit_16((c89atomic_uint16*)dst, (c89atomic_uint16*)expected, (c89atomic_uint16)desired, successOrder, failureOrder)
+#define c89atomic_compare_exchange_weak_explicit_i32(dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_weak_explicit_32((c89atomic_uint32*)dst, (c89atomic_uint32*)expected, (c89atomic_uint32)desired, successOrder, failureOrder)
+#define c89atomic_compare_exchange_weak_explicit_i64(dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_weak_explicit_64((c89atomic_uint64*)dst, (c89atomic_uint64*)expected, (c89atomic_uint64)desired, successOrder, failureOrder)
+#define c89atomic_fetch_add_explicit_i8( dst, src, order) (c89atomic_int8 )c89atomic_fetch_add_explicit_8( (c89atomic_uint8* )dst, (c89atomic_uint8 )src, order)
+#define c89atomic_fetch_add_explicit_i16(dst, src, order) (c89atomic_int16)c89atomic_fetch_add_explicit_16((c89atomic_uint16*)dst, (c89atomic_uint16)src, order)
+#define c89atomic_fetch_add_explicit_i32(dst, src, order) (c89atomic_int32)c89atomic_fetch_add_explicit_32((c89atomic_uint32*)dst, (c89atomic_uint32)src, order)
+#define c89atomic_fetch_add_explicit_i64(dst, src, order) (c89atomic_int64)c89atomic_fetch_add_explicit_64((c89atomic_uint64*)dst, (c89atomic_uint64)src, order)
+#define c89atomic_fetch_sub_explicit_i8( dst, src, order) (c89atomic_int8 )c89atomic_fetch_sub_explicit_8( (c89atomic_uint8* )dst, (c89atomic_uint8 )src, order)
+#define c89atomic_fetch_sub_explicit_i16(dst, src, order) (c89atomic_int16)c89atomic_fetch_sub_explicit_16((c89atomic_uint16*)dst, (c89atomic_uint16)src, order)
+#define c89atomic_fetch_sub_explicit_i32(dst, src, order) (c89atomic_int32)c89atomic_fetch_sub_explicit_32((c89atomic_uint32*)dst, (c89atomic_uint32)src, order)
+#define c89atomic_fetch_sub_explicit_i64(dst, src, order) (c89atomic_int64)c89atomic_fetch_sub_explicit_64((c89atomic_uint64*)dst, (c89atomic_uint64)src, order)
+#define c89atomic_fetch_or_explicit_i8( dst, src, order) (c89atomic_int8 )c89atomic_fetch_or_explicit_8( (c89atomic_uint8* )dst, (c89atomic_uint8 )src, order)
+#define c89atomic_fetch_or_explicit_i16(dst, src, order) (c89atomic_int16)c89atomic_fetch_or_explicit_16((c89atomic_uint16*)dst, (c89atomic_uint16)src, order)
+#define c89atomic_fetch_or_explicit_i32(dst, src, order) (c89atomic_int32)c89atomic_fetch_or_explicit_32((c89atomic_uint32*)dst, (c89atomic_uint32)src, order)
+#define c89atomic_fetch_or_explicit_i64(dst, src, order) (c89atomic_int64)c89atomic_fetch_or_explicit_64((c89atomic_uint64*)dst, (c89atomic_uint64)src, order)
+#define c89atomic_fetch_xor_explicit_i8( dst, src, order) (c89atomic_int8 )c89atomic_fetch_xor_explicit_8( (c89atomic_uint8* )dst, (c89atomic_uint8 )src, order)
+#define c89atomic_fetch_xor_explicit_i16(dst, src, order) (c89atomic_int16)c89atomic_fetch_xor_explicit_16((c89atomic_uint16*)dst, (c89atomic_uint16)src, order)
+#define c89atomic_fetch_xor_explicit_i32(dst, src, order) (c89atomic_int32)c89atomic_fetch_xor_explicit_32((c89atomic_uint32*)dst, (c89atomic_uint32)src, order)
+#define c89atomic_fetch_xor_explicit_i64(dst, src, order) (c89atomic_int64)c89atomic_fetch_xor_explicit_64((c89atomic_uint64*)dst, (c89atomic_uint64)src, order)
+#define c89atomic_fetch_and_explicit_i8( dst, src, order) (c89atomic_int8 )c89atomic_fetch_and_explicit_8( (c89atomic_uint8* )dst, (c89atomic_uint8 )src, order)
+#define c89atomic_fetch_and_explicit_i16(dst, src, order) (c89atomic_int16)c89atomic_fetch_and_explicit_16((c89atomic_uint16*)dst, (c89atomic_uint16)src, order)
+#define c89atomic_fetch_and_explicit_i32(dst, src, order) (c89atomic_int32)c89atomic_fetch_and_explicit_32((c89atomic_uint32*)dst, (c89atomic_uint32)src, order)
+#define c89atomic_fetch_and_explicit_i64(dst, src, order) (c89atomic_int64)c89atomic_fetch_and_explicit_64((c89atomic_uint64*)dst, (c89atomic_uint64)src, order)
+#define c89atomic_test_and_set_i8( ptr) c89atomic_test_and_set_explicit_i8( ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_test_and_set_i16(ptr) c89atomic_test_and_set_explicit_i16(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_test_and_set_i32(ptr) c89atomic_test_and_set_explicit_i32(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_test_and_set_i64(ptr) c89atomic_test_and_set_explicit_i64(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_clear_i8( ptr) c89atomic_clear_explicit_i8( ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_clear_i16(ptr) c89atomic_clear_explicit_i16(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_clear_i32(ptr) c89atomic_clear_explicit_i32(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_clear_i64(ptr) c89atomic_clear_explicit_i64(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_store_i8( dst, src) c89atomic_store_explicit_i8( dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_store_i16(dst, src) c89atomic_store_explicit_i16(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_store_i32(dst, src) c89atomic_store_explicit_i32(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_store_i64(dst, src) c89atomic_store_explicit_i64(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_load_i8( ptr) c89atomic_load_explicit_i8( ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_load_i16(ptr) c89atomic_load_explicit_i16(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_load_i32(ptr) c89atomic_load_explicit_i32(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_load_i64(ptr) c89atomic_load_explicit_i64(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_exchange_i8( dst, src) c89atomic_exchange_explicit_i8( dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_exchange_i16(dst, src) c89atomic_exchange_explicit_i16(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_exchange_i32(dst, src) c89atomic_exchange_explicit_i32(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_exchange_i64(dst, src) c89atomic_exchange_explicit_i64(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_compare_exchange_strong_i8( dst, expected, desired) c89atomic_compare_exchange_strong_explicit_i8( dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
+#define c89atomic_compare_exchange_strong_i16(dst, expected, desired) c89atomic_compare_exchange_strong_explicit_i16(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
+#define c89atomic_compare_exchange_strong_i32(dst, expected, desired) c89atomic_compare_exchange_strong_explicit_i32(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
+#define c89atomic_compare_exchange_strong_i64(dst, expected, desired) c89atomic_compare_exchange_strong_explicit_i64(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
+#define c89atomic_compare_exchange_weak_i8( dst, expected, desired) c89atomic_compare_exchange_weak_explicit_i8( dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
+#define c89atomic_compare_exchange_weak_i16(dst, expected, desired) c89atomic_compare_exchange_weak_explicit_i16(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
+#define c89atomic_compare_exchange_weak_i32(dst, expected, desired) c89atomic_compare_exchange_weak_explicit_i32(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
+#define c89atomic_compare_exchange_weak_i64(dst, expected, desired) c89atomic_compare_exchange_weak_explicit_i64(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_add_i8( dst, src) c89atomic_fetch_add_explicit_i8( dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_add_i16(dst, src) c89atomic_fetch_add_explicit_i16(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_add_i32(dst, src) c89atomic_fetch_add_explicit_i32(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_add_i64(dst, src) c89atomic_fetch_add_explicit_i64(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_sub_i8( dst, src) c89atomic_fetch_sub_explicit_i8( dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_sub_i16(dst, src) c89atomic_fetch_sub_explicit_i16(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_sub_i32(dst, src) c89atomic_fetch_sub_explicit_i32(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_sub_i64(dst, src) c89atomic_fetch_sub_explicit_i64(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_or_i8( dst, src) c89atomic_fetch_or_explicit_i8( dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_or_i16(dst, src) c89atomic_fetch_or_explicit_i16(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_or_i32(dst, src) c89atomic_fetch_or_explicit_i32(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_or_i64(dst, src) c89atomic_fetch_or_explicit_i64(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_xor_i8( dst, src) c89atomic_fetch_xor_explicit_i8( dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_xor_i16(dst, src) c89atomic_fetch_xor_explicit_i16(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_xor_i32(dst, src) c89atomic_fetch_xor_explicit_i32(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_xor_i64(dst, src) c89atomic_fetch_xor_explicit_i64(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_and_i8( dst, src) c89atomic_fetch_and_explicit_i8( dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_and_i16(dst, src) c89atomic_fetch_and_explicit_i16(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_and_i32(dst, src) c89atomic_fetch_and_explicit_i32(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_fetch_and_i64(dst, src) c89atomic_fetch_and_explicit_i64(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_compare_and_swap_i8( dst, expected, dedsired) (c89atomic_int8 )c89atomic_compare_and_swap_8( (c89atomic_uint8* )dst, (c89atomic_uint8 )expected, (c89atomic_uint8 )dedsired)
+#define c89atomic_compare_and_swap_i16(dst, expected, dedsired) (c89atomic_int16)c89atomic_compare_and_swap_16((c89atomic_uint16*)dst, (c89atomic_uint16)expected, (c89atomic_uint16)dedsired)
+#define c89atomic_compare_and_swap_i32(dst, expected, dedsired) (c89atomic_int32)c89atomic_compare_and_swap_32((c89atomic_uint32*)dst, (c89atomic_uint32)expected, (c89atomic_uint32)dedsired)
+#define c89atomic_compare_and_swap_i64(dst, expected, dedsired) (c89atomic_int64)c89atomic_compare_and_swap_64((c89atomic_uint64*)dst, (c89atomic_uint64)expected, (c89atomic_uint64)dedsired)
+typedef union
+{
+ c89atomic_uint32 i;
+ float f;
+} c89atomic_if32;
+typedef union
+{
+ c89atomic_uint64 i;
+ double f;
+} c89atomic_if64;
+#define c89atomic_clear_explicit_f32(ptr, order) c89atomic_clear_explicit_32((c89atomic_uint32*)ptr, order)
+#define c89atomic_clear_explicit_f64(ptr, order) c89atomic_clear_explicit_64((c89atomic_uint64*)ptr, order)
+static C89ATOMIC_INLINE void c89atomic_store_explicit_f32(volatile float* dst, float src, c89atomic_memory_order order)
+{
+ c89atomic_if32 x;
+ x.f = src;
+ c89atomic_store_explicit_32((volatile c89atomic_uint32*)dst, x.i, order);
+}
+static C89ATOMIC_INLINE void c89atomic_store_explicit_f64(volatile double* dst, double src, c89atomic_memory_order order)
+{
+ c89atomic_if64 x;
+ x.f = src;
+ c89atomic_store_explicit_64((volatile c89atomic_uint64*)dst, x.i, order);
+}
+static C89ATOMIC_INLINE float c89atomic_load_explicit_f32(volatile const float* ptr, c89atomic_memory_order order)
+{
+ c89atomic_if32 r;
+ r.i = c89atomic_load_explicit_32((volatile const c89atomic_uint32*)ptr, order);
+ return r.f;
+}
+static C89ATOMIC_INLINE double c89atomic_load_explicit_f64(volatile const double* ptr, c89atomic_memory_order order)
+{
+ c89atomic_if64 r;
+ r.i = c89atomic_load_explicit_64((volatile const c89atomic_uint64*)ptr, order);
+ return r.f;
+}
+static C89ATOMIC_INLINE float c89atomic_exchange_explicit_f32(volatile float* dst, float src, c89atomic_memory_order order)
+{
+ c89atomic_if32 r;
+ c89atomic_if32 x;
+ x.f = src;
+ r.i = c89atomic_exchange_explicit_32((volatile c89atomic_uint32*)dst, x.i, order);
+ return r.f;
+}
+static C89ATOMIC_INLINE double c89atomic_exchange_explicit_f64(volatile double* dst, double src, c89atomic_memory_order order)
+{
+ c89atomic_if64 r;
+ c89atomic_if64 x;
+ x.f = src;
+ r.i = c89atomic_exchange_explicit_64((volatile c89atomic_uint64*)dst, x.i, order);
+ return r.f;
+}
+#define c89atomic_clear_f32(ptr) (float )c89atomic_clear_explicit_f32(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_clear_f64(ptr) (double)c89atomic_clear_explicit_f64(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_store_f32(dst, src) c89atomic_store_explicit_f32(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_store_f64(dst, src) c89atomic_store_explicit_f64(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_load_f32(ptr) (float )c89atomic_load_explicit_f32(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_load_f64(ptr) (double)c89atomic_load_explicit_f64(ptr, c89atomic_memory_order_seq_cst)
+#define c89atomic_exchange_f32(dst, src) (float )c89atomic_exchange_explicit_f32(dst, src, c89atomic_memory_order_seq_cst)
+#define c89atomic_exchange_f64(dst, src) (double)c89atomic_exchange_explicit_f64(dst, src, c89atomic_memory_order_seq_cst)
+typedef c89atomic_flag c89atomic_spinlock;
+static C89ATOMIC_INLINE void c89atomic_spinlock_lock(volatile c89atomic_spinlock* pSpinlock)
+{
+ for (;;) {
+ if (c89atomic_flag_test_and_set_explicit(pSpinlock, c89atomic_memory_order_acquire) == 0) {
+ break;
+ }
+ while (c89atoimc_flag_load_explicit(pSpinlock, c89atomic_memory_order_relaxed) == 1) {
+ }
+ }
+}
+static C89ATOMIC_INLINE void c89atomic_spinlock_unlock(volatile c89atomic_spinlock* pSpinlock)
+{
+ c89atomic_flag_clear_explicit(pSpinlock, c89atomic_memory_order_release);
+}
+#if defined(__cplusplus)
+}
+#endif
+#endif
+/* c89atomic.h end */
+
+
+
+MA_API ma_uint64 ma_calculate_frame_count_after_resampling(ma_uint32 sampleRateOut, ma_uint32 sampleRateIn, ma_uint64 frameCountIn)
+{
+ /* This is based on the calculation in ma_linear_resampler_get_expected_output_frame_count(). */
+ ma_uint64 outputFrameCount;
+ ma_uint64 preliminaryInputFrameCountFromFrac;
+ ma_uint64 preliminaryInputFrameCount;
+
+ if (sampleRateIn == 0 || sampleRateOut == 0 || frameCountIn == 0) {
+ return 0;
+ }
+
+ if (sampleRateOut == sampleRateIn) {
+ return frameCountIn;
+ }
+
+ outputFrameCount = (frameCountIn * sampleRateOut) / sampleRateIn;
+
+ preliminaryInputFrameCountFromFrac = (outputFrameCount * (sampleRateIn / sampleRateOut)) / sampleRateOut;
+ preliminaryInputFrameCount = (outputFrameCount * (sampleRateIn % sampleRateOut)) + preliminaryInputFrameCountFromFrac;
+
+ if (preliminaryInputFrameCount <= frameCountIn) {
+ outputFrameCount += 1;
+ }
+
+ return outputFrameCount;
+}
+
+#ifndef MA_DATA_CONVERTER_STACK_BUFFER_SIZE
+#define MA_DATA_CONVERTER_STACK_BUFFER_SIZE 4096
+#endif
+
+
+
+#if defined(MA_WIN32)
+static ma_result ma_result_from_GetLastError(DWORD error)
+{
+ switch (error)
+ {
+ case ERROR_SUCCESS: return MA_SUCCESS;
+ case ERROR_PATH_NOT_FOUND: return MA_DOES_NOT_EXIST;
+ case ERROR_TOO_MANY_OPEN_FILES: return MA_TOO_MANY_OPEN_FILES;
+ case ERROR_NOT_ENOUGH_MEMORY: return MA_OUT_OF_MEMORY;
+ case ERROR_DISK_FULL: return MA_NO_SPACE;
+ case ERROR_HANDLE_EOF: return MA_AT_END;
+ case ERROR_NEGATIVE_SEEK: return MA_BAD_SEEK;
+ case ERROR_INVALID_PARAMETER: return MA_INVALID_ARGS;
+ case ERROR_ACCESS_DENIED: return MA_ACCESS_DENIED;
+ case ERROR_SEM_TIMEOUT: return MA_TIMEOUT;
+ case ERROR_FILE_NOT_FOUND: return MA_DOES_NOT_EXIST;
+ default: break;
+ }
+
+ return MA_ERROR;
+}
+#endif /* MA_WIN32 */
+
+
+/*******************************************************************************
+
+Threading
+
+*******************************************************************************/
+static MA_INLINE ma_result ma_spinlock_lock_ex(volatile ma_spinlock* pSpinlock, ma_bool32 yield)
+{
+ if (pSpinlock == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ for (;;) {
+ if (c89atomic_exchange_explicit_32(pSpinlock, 1, c89atomic_memory_order_acquire) == 0) {
+ break;
+ }
+
+ while (c89atomic_load_explicit_32(pSpinlock, c89atomic_memory_order_relaxed) == 1) {
+ if (yield) {
+ ma_yield();
+ }
+ }
+ }
+
+ return MA_SUCCESS;
+}
+
+MA_API ma_result ma_spinlock_lock(volatile ma_spinlock* pSpinlock)
+{
+ return ma_spinlock_lock_ex(pSpinlock, MA_TRUE);
+}
+
+MA_API ma_result ma_spinlock_lock_noyield(volatile ma_spinlock* pSpinlock)
+{
+ return ma_spinlock_lock_ex(pSpinlock, MA_FALSE);
+}
+
+MA_API ma_result ma_spinlock_unlock(volatile ma_spinlock* pSpinlock)
+{
+ if (pSpinlock == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ c89atomic_store_explicit_32(pSpinlock, 0, c89atomic_memory_order_release);
+ return MA_SUCCESS;
+}
+
+
+#ifndef MA_NO_THREADING
+#ifdef MA_WIN32
+ #define MA_THREADCALL WINAPI
+ typedef unsigned long ma_thread_result;
+#else
+ #define MA_THREADCALL
+ typedef void* ma_thread_result;
+#endif
+typedef ma_thread_result (MA_THREADCALL * ma_thread_entry_proc)(void* pData);
+
+#ifdef MA_WIN32
+static int ma_thread_priority_to_win32(ma_thread_priority priority)
+{
+ switch (priority) {
+ case ma_thread_priority_idle: return THREAD_PRIORITY_IDLE;
+ case ma_thread_priority_lowest: return THREAD_PRIORITY_LOWEST;
+ case ma_thread_priority_low: return THREAD_PRIORITY_BELOW_NORMAL;
+ case ma_thread_priority_normal: return THREAD_PRIORITY_NORMAL;
+ case ma_thread_priority_high: return THREAD_PRIORITY_ABOVE_NORMAL;
+ case ma_thread_priority_highest: return THREAD_PRIORITY_HIGHEST;
+ case ma_thread_priority_realtime: return THREAD_PRIORITY_TIME_CRITICAL;
+ default: return THREAD_PRIORITY_NORMAL;
+ }
+}
+
+static ma_result ma_thread_create__win32(ma_thread* pThread, ma_thread_priority priority, size_t stackSize, ma_thread_entry_proc entryProc, void* pData)
+{
+ *pThread = CreateThread(NULL, stackSize, entryProc, pData, 0, NULL);
+ if (*pThread == NULL) {
+ return ma_result_from_GetLastError(GetLastError());
+ }
+
+ SetThreadPriority((HANDLE)*pThread, ma_thread_priority_to_win32(priority));
+
+ return MA_SUCCESS;
+}
+
+static void ma_thread_wait__win32(ma_thread* pThread)
+{
+ WaitForSingleObject((HANDLE)*pThread, INFINITE);
+ CloseHandle((HANDLE)*pThread);
+}
+
+
+static ma_result ma_mutex_init__win32(ma_mutex* pMutex)
+{
+ *pMutex = CreateEventW(NULL, FALSE, TRUE, NULL);
+ if (*pMutex == NULL) {
+ return ma_result_from_GetLastError(GetLastError());
+ }
+
+ return MA_SUCCESS;
+}
+
+static void ma_mutex_uninit__win32(ma_mutex* pMutex)
+{
+ CloseHandle((HANDLE)*pMutex);
+}
+
+static void ma_mutex_lock__win32(ma_mutex* pMutex)
+{
+ WaitForSingleObject((HANDLE)*pMutex, INFINITE);
+}
+
+static void ma_mutex_unlock__win32(ma_mutex* pMutex)
+{
+ SetEvent((HANDLE)*pMutex);
+}
+
+
+static ma_result ma_event_init__win32(ma_event* pEvent)
+{
+ *pEvent = CreateEventW(NULL, FALSE, FALSE, NULL);
+ if (*pEvent == NULL) {
+ return ma_result_from_GetLastError(GetLastError());
+ }
+
+ return MA_SUCCESS;
+}
+
+static void ma_event_uninit__win32(ma_event* pEvent)
+{
+ CloseHandle((HANDLE)*pEvent);
+}
+
+static ma_result ma_event_wait__win32(ma_event* pEvent)
+{
+ DWORD result = WaitForSingleObject((HANDLE)*pEvent, INFINITE);
+ if (result == WAIT_OBJECT_0) {
+ return MA_SUCCESS;
+ }
+
+ if (result == WAIT_TIMEOUT) {
+ return MA_TIMEOUT;
+ }
+
+ return ma_result_from_GetLastError(GetLastError());
+}
+
+static ma_result ma_event_signal__win32(ma_event* pEvent)
+{
+ BOOL result = SetEvent((HANDLE)*pEvent);
+ if (result == 0) {
+ return ma_result_from_GetLastError(GetLastError());
+ }
+
+ return MA_SUCCESS;
+}
+
+
+static ma_result ma_semaphore_init__win32(int initialValue, ma_semaphore* pSemaphore)
+{
+ *pSemaphore = CreateSemaphoreW(NULL, (LONG)initialValue, LONG_MAX, NULL);
+ if (*pSemaphore == NULL) {
+ return ma_result_from_GetLastError(GetLastError());
+ }
+
+ return MA_SUCCESS;
+}
+
+static void ma_semaphore_uninit__win32(ma_semaphore* pSemaphore)
+{
+ CloseHandle((HANDLE)*pSemaphore);
+}
+
+static ma_result ma_semaphore_wait__win32(ma_semaphore* pSemaphore)
+{
+ DWORD result = WaitForSingleObject((HANDLE)*pSemaphore, INFINITE);
+ if (result == WAIT_OBJECT_0) {
+ return MA_SUCCESS;
+ }
+
+ if (result == WAIT_TIMEOUT) {
+ return MA_TIMEOUT;
+ }
+
+ return ma_result_from_GetLastError(GetLastError());
+}
+
+static ma_result ma_semaphore_release__win32(ma_semaphore* pSemaphore)
+{
+ BOOL result = ReleaseSemaphore((HANDLE)*pSemaphore, 1, NULL);
+ if (result == 0) {
+ return ma_result_from_GetLastError(GetLastError());
+ }
+
+ return MA_SUCCESS;
+}
+#endif
+
+
+#ifdef MA_POSIX
+static ma_result ma_thread_create__posix(ma_thread* pThread, ma_thread_priority priority, size_t stackSize, ma_thread_entry_proc entryProc, void* pData)
+{
+ int result;
+ pthread_attr_t* pAttr = NULL;
+
+#if !defined(__EMSCRIPTEN__)
+ /* Try setting the thread priority. It's not critical if anything fails here. */
+ pthread_attr_t attr;
+ if (pthread_attr_init(&attr) == 0) {
+ int scheduler = -1;
+ if (priority == ma_thread_priority_idle) {
+#ifdef SCHED_IDLE
+ if (pthread_attr_setschedpolicy(&attr, SCHED_IDLE) == 0) {
+ scheduler = SCHED_IDLE;
+ }
+#endif
+ } else if (priority == ma_thread_priority_realtime) {
+#ifdef SCHED_FIFO
+ if (pthread_attr_setschedpolicy(&attr, SCHED_FIFO) == 0) {
+ scheduler = SCHED_FIFO;
+ }
+#endif
+#ifdef MA_LINUX
+ } else {
+ scheduler = sched_getscheduler(0);
+#endif
+ }
+
+ if (stackSize > 0) {
+ pthread_attr_setstacksize(&attr, stackSize);
+ }
+
+ if (scheduler != -1) {
+ int priorityMin = sched_get_priority_min(scheduler);
+ int priorityMax = sched_get_priority_max(scheduler);
+ int priorityStep = (priorityMax - priorityMin) / 7; /* 7 = number of priorities supported by miniaudio. */
+
+ struct sched_param sched;
+ if (pthread_attr_getschedparam(&attr, &sched) == 0) {
+ if (priority == ma_thread_priority_idle) {
+ sched.sched_priority = priorityMin;
+ } else if (priority == ma_thread_priority_realtime) {
+ sched.sched_priority = priorityMax;
+ } else {
+ sched.sched_priority += ((int)priority + 5) * priorityStep; /* +5 because the lowest priority is -5. */
+ if (sched.sched_priority < priorityMin) {
+ sched.sched_priority = priorityMin;
+ }
+ if (sched.sched_priority > priorityMax) {
+ sched.sched_priority = priorityMax;
+ }
+ }
+
+ if (pthread_attr_setschedparam(&attr, &sched) == 0) {
+ pAttr = &attr;
+ }
+ }
+ }
+ }
+#else
+ /* It's the emscripten build. We'll have a few unused parameters. */
+ (void)priority;
+ (void)stackSize;
+#endif
+
+ result = pthread_create((pthread_t*)pThread, pAttr, entryProc, pData);
+
+ /* The thread attributes object is no longer required. */
+ if (pAttr != NULL) {
+ pthread_attr_destroy(pAttr);
+ }
+
+ if (result != 0) {
+ return ma_result_from_errno(result);
+ }
+
+ return MA_SUCCESS;
+}
+
+static void ma_thread_wait__posix(ma_thread* pThread)
+{
+ pthread_join((pthread_t)*pThread, NULL);
+}
+
+
+static ma_result ma_mutex_init__posix(ma_mutex* pMutex)
+{
+ int result = pthread_mutex_init((pthread_mutex_t*)pMutex, NULL);
+ if (result != 0) {
+ return ma_result_from_errno(result);
+ }
+
+ return MA_SUCCESS;
+}
+
+static void ma_mutex_uninit__posix(ma_mutex* pMutex)
+{
+ pthread_mutex_destroy((pthread_mutex_t*)pMutex);
+}
+
+static void ma_mutex_lock__posix(ma_mutex* pMutex)
+{
+ pthread_mutex_lock((pthread_mutex_t*)pMutex);
+}
+
+static void ma_mutex_unlock__posix(ma_mutex* pMutex)
+{
+ pthread_mutex_unlock((pthread_mutex_t*)pMutex);
+}
+
+
+static ma_result ma_event_init__posix(ma_event* pEvent)
+{
+ int result;
+
+ result = pthread_mutex_init((pthread_mutex_t*)&pEvent->lock, NULL);
+ if (result != 0) {
+ return ma_result_from_errno(result);
+ }
+
+ result = pthread_cond_init((pthread_cond_t*)&pEvent->cond, NULL);
+ if (result != 0) {
+ pthread_mutex_destroy((pthread_mutex_t*)&pEvent->lock);
+ return ma_result_from_errno(result);
+ }
+
+ pEvent->value = 0;
+ return MA_SUCCESS;
+}
+
+static void ma_event_uninit__posix(ma_event* pEvent)
+{
+ pthread_cond_destroy((pthread_cond_t*)&pEvent->cond);
+ pthread_mutex_destroy((pthread_mutex_t*)&pEvent->lock);
+}
+
+static ma_result ma_event_wait__posix(ma_event* pEvent)
+{
+ pthread_mutex_lock((pthread_mutex_t*)&pEvent->lock);
+ {
+ while (pEvent->value == 0) {
+ pthread_cond_wait((pthread_cond_t*)&pEvent->cond, (pthread_mutex_t*)&pEvent->lock);
+ }
+ pEvent->value = 0; /* Auto-reset. */
+ }
+ pthread_mutex_unlock((pthread_mutex_t*)&pEvent->lock);
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_event_signal__posix(ma_event* pEvent)
+{
+ pthread_mutex_lock((pthread_mutex_t*)&pEvent->lock);
+ {
+ pEvent->value = 1;
+ pthread_cond_signal((pthread_cond_t*)&pEvent->cond);
+ }
+ pthread_mutex_unlock((pthread_mutex_t*)&pEvent->lock);
+
+ return MA_SUCCESS;
+}
+
+
+static ma_result ma_semaphore_init__posix(int initialValue, ma_semaphore* pSemaphore)
+{
+ int result;
+
+ if (pSemaphore == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ pSemaphore->value = initialValue;
+
+ result = pthread_mutex_init((pthread_mutex_t*)&pSemaphore->lock, NULL);
+ if (result != 0) {
+ return ma_result_from_errno(result); /* Failed to create mutex. */
+ }
+
+ result = pthread_cond_init((pthread_cond_t*)&pSemaphore->cond, NULL);
+ if (result != 0) {
+ pthread_mutex_destroy((pthread_mutex_t*)&pSemaphore->lock);
+ return ma_result_from_errno(result); /* Failed to create condition variable. */
+ }
+
+ return MA_SUCCESS;
+}
+
+static void ma_semaphore_uninit__posix(ma_semaphore* pSemaphore)
+{
+ if (pSemaphore == NULL) {
+ return;
+ }
+
+ pthread_cond_destroy((pthread_cond_t*)&pSemaphore->cond);
+ pthread_mutex_destroy((pthread_mutex_t*)&pSemaphore->lock);
+}
+
+static ma_result ma_semaphore_wait__posix(ma_semaphore* pSemaphore)
+{
+ if (pSemaphore == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ pthread_mutex_lock((pthread_mutex_t*)&pSemaphore->lock);
+ {
+ /* We need to wait on a condition variable before escaping. We can't return from this function until the semaphore has been signaled. */
+ while (pSemaphore->value == 0) {
+ pthread_cond_wait((pthread_cond_t*)&pSemaphore->cond, (pthread_mutex_t*)&pSemaphore->lock);
+ }
+
+ pSemaphore->value -= 1;
+ }
+ pthread_mutex_unlock((pthread_mutex_t*)&pSemaphore->lock);
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_semaphore_release__posix(ma_semaphore* pSemaphore)
+{
+ if (pSemaphore == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ pthread_mutex_lock((pthread_mutex_t*)&pSemaphore->lock);
+ {
+ pSemaphore->value += 1;
+ pthread_cond_signal((pthread_cond_t*)&pSemaphore->cond);
+ }
+ pthread_mutex_unlock((pthread_mutex_t*)&pSemaphore->lock);
+
+ return MA_SUCCESS;
+}
+#endif
+
+typedef struct
+{
+ ma_thread_entry_proc entryProc;
+ void* pData;
+ ma_allocation_callbacks allocationCallbacks;
+} ma_thread_proxy_data;
+
+static ma_thread_result MA_THREADCALL ma_thread_entry_proxy(void* pData)
+{
+ ma_thread_proxy_data* pProxyData = (ma_thread_proxy_data*)pData;
+ ma_thread_entry_proc entryProc;
+ void* pEntryProcData;
+ ma_thread_result result;
+
+ #if defined(MA_ON_THREAD_ENTRY)
+ MA_ON_THREAD_ENTRY
+ #endif
+
+ entryProc = pProxyData->entryProc;
+ pEntryProcData = pProxyData->pData;
+
+ /* Free the proxy data before getting into the real thread entry proc. */
+ ma_free(pProxyData, &pProxyData->allocationCallbacks);
+
+ result = entryProc(pEntryProcData);
+
+ #if defined(MA_ON_THREAD_EXIT)
+ MA_ON_THREAD_EXIT
+ #endif
+
+ return result;
+}
+
+static ma_result ma_thread_create(ma_thread* pThread, ma_thread_priority priority, size_t stackSize, ma_thread_entry_proc entryProc, void* pData, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ ma_result result;
+ ma_thread_proxy_data* pProxyData;
+
+ if (pThread == NULL || entryProc == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ pProxyData = (ma_thread_proxy_data*)ma_malloc(sizeof(*pProxyData), pAllocationCallbacks); /* Will be freed by the proxy entry proc. */
+ if (pProxyData == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+
+ pProxyData->entryProc = entryProc;
+ pProxyData->pData = pData;
+ ma_allocation_callbacks_init_copy(&pProxyData->allocationCallbacks, pAllocationCallbacks);
+
+#ifdef MA_WIN32
+ result = ma_thread_create__win32(pThread, priority, stackSize, ma_thread_entry_proxy, pProxyData);
+#endif
+#ifdef MA_POSIX
+ result = ma_thread_create__posix(pThread, priority, stackSize, ma_thread_entry_proxy, pProxyData);
+#endif
+
+ if (result != MA_SUCCESS) {
+ ma_free(pProxyData, pAllocationCallbacks);
+ return result;
+ }
+
+ return MA_SUCCESS;
+}
+
+static void ma_thread_wait(ma_thread* pThread)
+{
+ if (pThread == NULL) {
+ return;
+ }
+
+#ifdef MA_WIN32
+ ma_thread_wait__win32(pThread);
+#endif
+#ifdef MA_POSIX
+ ma_thread_wait__posix(pThread);
+#endif
+}
+
+
+MA_API ma_result ma_mutex_init(ma_mutex* pMutex)
+{
+ if (pMutex == NULL) {
+ MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */
+ return MA_INVALID_ARGS;
+ }
+
+#ifdef MA_WIN32
+ return ma_mutex_init__win32(pMutex);
+#endif
+#ifdef MA_POSIX
+ return ma_mutex_init__posix(pMutex);
+#endif
+}
+
+MA_API void ma_mutex_uninit(ma_mutex* pMutex)
+{
+ if (pMutex == NULL) {
+ return;
+ }
+
+#ifdef MA_WIN32
+ ma_mutex_uninit__win32(pMutex);
+#endif
+#ifdef MA_POSIX
+ ma_mutex_uninit__posix(pMutex);
+#endif
+}
+
+MA_API void ma_mutex_lock(ma_mutex* pMutex)
+{
+ if (pMutex == NULL) {
+ MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */
+ return;
+ }
+
+#ifdef MA_WIN32
+ ma_mutex_lock__win32(pMutex);
+#endif
+#ifdef MA_POSIX
+ ma_mutex_lock__posix(pMutex);
+#endif
+}
+
+MA_API void ma_mutex_unlock(ma_mutex* pMutex)
+{
+ if (pMutex == NULL) {
+ MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */
+ return;
+}
+
+#ifdef MA_WIN32
+ ma_mutex_unlock__win32(pMutex);
+#endif
+#ifdef MA_POSIX
+ ma_mutex_unlock__posix(pMutex);
+#endif
+}
+
+
+MA_API ma_result ma_event_init(ma_event* pEvent)
+{
+ if (pEvent == NULL) {
+ MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */
+ return MA_INVALID_ARGS;
+ }
+
+#ifdef MA_WIN32
+ return ma_event_init__win32(pEvent);
+#endif
+#ifdef MA_POSIX
+ return ma_event_init__posix(pEvent);
+#endif
+}
+
+#if 0
+static ma_result ma_event_alloc_and_init(ma_event** ppEvent, ma_allocation_callbacks* pAllocationCallbacks)
+{
+ ma_result result;
+ ma_event* pEvent;
+
+ if (ppEvent == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ *ppEvent = NULL;
+
+ pEvent = ma_malloc(sizeof(*pEvent), pAllocationCallbacks);
+ if (pEvent == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+
+ result = ma_event_init(pEvent);
+ if (result != MA_SUCCESS) {
+ ma_free(pEvent, pAllocationCallbacks);
+ return result;
+ }
+
+ *ppEvent = pEvent;
+ return result;
+}
+#endif
+
+MA_API void ma_event_uninit(ma_event* pEvent)
+{
+ if (pEvent == NULL) {
+ return;
+ }
+
+#ifdef MA_WIN32
+ ma_event_uninit__win32(pEvent);
+#endif
+#ifdef MA_POSIX
+ ma_event_uninit__posix(pEvent);
+#endif
+}
+
+#if 0
+static void ma_event_uninit_and_free(ma_event* pEvent, ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pEvent == NULL) {
+ return;
+ }
+
+ ma_event_uninit(pEvent);
+ ma_free(pEvent, pAllocationCallbacks);
+}
+#endif
+
+MA_API ma_result ma_event_wait(ma_event* pEvent)
+{
+ if (pEvent == NULL) {
+ MA_ASSERT(MA_FALSE); /* Fire an assert to the caller is aware of this bug. */
+ return MA_INVALID_ARGS;
+ }
+
+#ifdef MA_WIN32
+ return ma_event_wait__win32(pEvent);
+#endif
+#ifdef MA_POSIX
+ return ma_event_wait__posix(pEvent);
+#endif
+}
+
+MA_API ma_result ma_event_signal(ma_event* pEvent)
+{
+ if (pEvent == NULL) {
+ MA_ASSERT(MA_FALSE); /* Fire an assert to the caller is aware of this bug. */
+ return MA_INVALID_ARGS;
+ }
+
+#ifdef MA_WIN32
+ return ma_event_signal__win32(pEvent);
+#endif
+#ifdef MA_POSIX
+ return ma_event_signal__posix(pEvent);
+#endif
+}
+
+
+MA_API ma_result ma_semaphore_init(int initialValue, ma_semaphore* pSemaphore)
+{
+ if (pSemaphore == NULL) {
+ MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */
+ return MA_INVALID_ARGS;
+ }
+
+#ifdef MA_WIN32
+ return ma_semaphore_init__win32(initialValue, pSemaphore);
+#endif
+#ifdef MA_POSIX
+ return ma_semaphore_init__posix(initialValue, pSemaphore);
+#endif
+}
+
+MA_API void ma_semaphore_uninit(ma_semaphore* pSemaphore)
+{
+ if (pSemaphore == NULL) {
+ MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */
+ return;
+ }
+
+#ifdef MA_WIN32
+ ma_semaphore_uninit__win32(pSemaphore);
+#endif
+#ifdef MA_POSIX
+ ma_semaphore_uninit__posix(pSemaphore);
+#endif
+}
+
+MA_API ma_result ma_semaphore_wait(ma_semaphore* pSemaphore)
+{
+ if (pSemaphore == NULL) {
+ MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */
+ return MA_INVALID_ARGS;
+ }
+
+#ifdef MA_WIN32
+ return ma_semaphore_wait__win32(pSemaphore);
+#endif
+#ifdef MA_POSIX
+ return ma_semaphore_wait__posix(pSemaphore);
+#endif
+}
+
+MA_API ma_result ma_semaphore_release(ma_semaphore* pSemaphore)
+{
+ if (pSemaphore == NULL) {
+ MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */
+ return MA_INVALID_ARGS;
+ }
+
+#ifdef MA_WIN32
+ return ma_semaphore_release__win32(pSemaphore);
+#endif
+#ifdef MA_POSIX
+ return ma_semaphore_release__posix(pSemaphore);
+#endif
+}
+#else
+/* MA_NO_THREADING is set which means threading is disabled. Threading is required by some API families. If any of these are enabled we need to throw an error. */
+#ifndef MA_NO_DEVICE_IO
+#error "MA_NO_THREADING cannot be used without MA_NO_DEVICE_IO";
+#endif
+#endif /* MA_NO_THREADING */
+
+
+
+#define MA_FENCE_COUNTER_MAX 0x7FFFFFFF
+
+MA_API ma_result ma_fence_init(ma_fence* pFence)
+{
+ if (pFence == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ MA_ZERO_OBJECT(pFence);
+ pFence->counter = 0;
+
+ #ifndef MA_NO_THREADING
+ {
+ ma_result result;
+
+ result = ma_event_init(&pFence->e);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ }
+ #endif
+
+ return MA_SUCCESS;
+}
+
+MA_API void ma_fence_uninit(ma_fence* pFence)
+{
+ if (pFence == NULL) {
+ return;
+ }
+
+ #ifndef MA_NO_THREADING
+ {
+ ma_event_uninit(&pFence->e);
+ }
+ #endif
+
+ MA_ZERO_OBJECT(pFence);
+}
+
+MA_API ma_result ma_fence_acquire(ma_fence* pFence)
+{
+ if (pFence == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ for (;;) {
+ ma_uint32 oldCounter = c89atomic_load_32(&pFence->counter);
+ ma_uint32 newCounter = oldCounter + 1;
+
+ /* Make sure we're not about to exceed our maximum value. */
+ if (newCounter > MA_FENCE_COUNTER_MAX) {
+ MA_ASSERT(MA_FALSE);
+ return MA_OUT_OF_RANGE;
+ }
+
+ if (c89atomic_compare_exchange_weak_32(&pFence->counter, &oldCounter, newCounter)) {
+ return MA_SUCCESS;
+ } else {
+ if (oldCounter == MA_FENCE_COUNTER_MAX) {
+ MA_ASSERT(MA_FALSE);
+ return MA_OUT_OF_RANGE; /* The other thread took the last available slot. Abort. */
+ }
+ }
+ }
+
+ /* Should never get here. */
+ /*return MA_SUCCESS;*/
+}
+
+MA_API ma_result ma_fence_release(ma_fence* pFence)
+{
+ if (pFence == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ for (;;) {
+ ma_uint32 oldCounter = c89atomic_load_32(&pFence->counter);
+ ma_uint32 newCounter = oldCounter - 1;
+
+ if (oldCounter == 0) {
+ MA_ASSERT(MA_FALSE);
+ return MA_INVALID_OPERATION; /* Acquire/release mismatch. */
+ }
+
+ if (c89atomic_compare_exchange_weak_32(&pFence->counter, &oldCounter, newCounter)) {
+ #ifndef MA_NO_THREADING
+ {
+ if (newCounter == 0) {
+ ma_event_signal(&pFence->e); /* <-- ma_fence_wait() will be waiting on this. */
+ }
+ }
+ #endif
+
+ return MA_SUCCESS;
+ } else {
+ if (oldCounter == 0) {
+ MA_ASSERT(MA_FALSE);
+ return MA_INVALID_OPERATION; /* Another thread has taken the 0 slot. Acquire/release mismatch. */
+ }
+ }
+ }
+
+ /* Should never get here. */
+ /*return MA_SUCCESS;*/
+}
+
+MA_API ma_result ma_fence_wait(ma_fence* pFence)
+{
+ if (pFence == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ for (;;) {
+ ma_uint32 counter;
+
+ counter = c89atomic_load_32(&pFence->counter);
+ if (counter == 0) {
+ /*
+ Counter has hit zero. By the time we get here some other thread may have acquired the
+ fence again, but that is where the caller needs to take care with how they se the fence.
+ */
+ return MA_SUCCESS;
+ }
+
+ /* Getting here means the counter is > 0. We'll need to wait for something to happen. */
+ #ifndef MA_NO_THREADING
+ {
+ ma_result result;
+
+ result = ma_event_wait(&pFence->e);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ }
+ #endif
+ }
+
+ /* Should never get here. */
+ /*return MA_INVALID_OPERATION;*/
+}
+
+
+MA_API ma_result ma_async_notification_signal(ma_async_notification* pNotification)
+{
+ ma_async_notification_callbacks* pNotificationCallbacks = (ma_async_notification_callbacks*)pNotification;
+
+ if (pNotification == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (pNotificationCallbacks->onSignal == NULL) {
+ return MA_NOT_IMPLEMENTED;
+ }
+
+ pNotificationCallbacks->onSignal(pNotification);
+ return MA_INVALID_ARGS;
+}
+
+
+static void ma_async_notification_poll__on_signal(ma_async_notification* pNotification)
+{
+ ((ma_async_notification_poll*)pNotification)->signalled = MA_TRUE;
+}
+
+MA_API ma_result ma_async_notification_poll_init(ma_async_notification_poll* pNotificationPoll)
+{
+ if (pNotificationPoll == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ pNotificationPoll->cb.onSignal = ma_async_notification_poll__on_signal;
+ pNotificationPoll->signalled = MA_FALSE;
+
+ return MA_SUCCESS;
+}
+
+MA_API ma_bool32 ma_async_notification_poll_is_signalled(const ma_async_notification_poll* pNotificationPoll)
+{
+ if (pNotificationPoll == NULL) {
+ return MA_FALSE;
+ }
+
+ return pNotificationPoll->signalled;
+}
+
+
+static void ma_async_notification_event__on_signal(ma_async_notification* pNotification)
+{
+ ma_async_notification_event_signal((ma_async_notification_event*)pNotification);
+}
+
+MA_API ma_result ma_async_notification_event_init(ma_async_notification_event* pNotificationEvent)
+{
+ if (pNotificationEvent == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ pNotificationEvent->cb.onSignal = ma_async_notification_event__on_signal;
+
+ #ifndef MA_NO_THREADING
+ {
+ ma_result result;
+
+ result = ma_event_init(&pNotificationEvent->e);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ return MA_NOT_IMPLEMENTED; /* Threading is disabled. */
+ }
+ #endif
+}
+
+MA_API ma_result ma_async_notification_event_uninit(ma_async_notification_event* pNotificationEvent)
+{
+ if (pNotificationEvent == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ #ifndef MA_NO_THREADING
+ {
+ ma_event_uninit(&pNotificationEvent->e);
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ return MA_NOT_IMPLEMENTED; /* Threading is disabled. */
+ }
+ #endif
+}
+
+MA_API ma_result ma_async_notification_event_wait(ma_async_notification_event* pNotificationEvent)
+{
+ if (pNotificationEvent == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ #ifndef MA_NO_THREADING
+ {
+ return ma_event_wait(&pNotificationEvent->e);
+ }
+ #else
+ {
+ return MA_NOT_IMPLEMENTED; /* Threading is disabled. */
+ }
+ #endif
+}
+
+MA_API ma_result ma_async_notification_event_signal(ma_async_notification_event* pNotificationEvent)
+{
+ if (pNotificationEvent == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ #ifndef MA_NO_THREADING
+ {
+ return ma_event_signal(&pNotificationEvent->e);
+ }
+ #else
+ {
+ return MA_NOT_IMPLEMENTED; /* Threading is disabled. */
+ }
+ #endif
+}
+
+
+
+/************************************************************************************************************************************************************
+
+Job Queue
+
+************************************************************************************************************************************************************/
+MA_API ma_slot_allocator_config ma_slot_allocator_config_init(ma_uint32 capacity)
+{
+ ma_slot_allocator_config config;
+
+ MA_ZERO_OBJECT(&config);
+ config.capacity = capacity;
+
+ return config;
+}
+
+
+static MA_INLINE ma_uint32 ma_slot_allocator_calculate_group_capacity(ma_uint32 slotCapacity)
+{
+ ma_uint32 cap = slotCapacity / 32;
+ if ((slotCapacity % 32) != 0) {
+ cap += 1;
+ }
+
+ return cap;
+}
+
+static MA_INLINE ma_uint32 ma_slot_allocator_group_capacity(const ma_slot_allocator* pAllocator)
+{
+ return ma_slot_allocator_calculate_group_capacity(pAllocator->capacity);
+}
+
+
+typedef struct
+{
+ size_t sizeInBytes;
+ size_t groupsOffset;
+ size_t slotsOffset;
+} ma_slot_allocator_heap_layout;
+
+static ma_result ma_slot_allocator_get_heap_layout(const ma_slot_allocator_config* pConfig, ma_slot_allocator_heap_layout* pHeapLayout)
+{
+ MA_ASSERT(pHeapLayout != NULL);
+
+ MA_ZERO_OBJECT(pHeapLayout);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (pConfig->capacity == 0) {
+ return MA_INVALID_ARGS;
+ }
+
+ pHeapLayout->sizeInBytes = 0;
+
+ /* Groups. */
+ pHeapLayout->groupsOffset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += ma_align_64(ma_slot_allocator_calculate_group_capacity(pConfig->capacity) * sizeof(ma_slot_allocator_group));
+
+ /* Slots. */
+ pHeapLayout->slotsOffset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += ma_align_64(pConfig->capacity * sizeof(ma_uint32));
+
+ return MA_SUCCESS;
+}
+
+MA_API ma_result ma_slot_allocator_get_heap_size(const ma_slot_allocator_config* pConfig, size_t* pHeapSizeInBytes)
+{
+ ma_result result;
+ ma_slot_allocator_heap_layout layout;
+
+ if (pHeapSizeInBytes == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ *pHeapSizeInBytes = 0;
+
+ result = ma_slot_allocator_get_heap_layout(pConfig, &layout);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ *pHeapSizeInBytes = layout.sizeInBytes;
+
+ return result;
+}
+
+MA_API ma_result ma_slot_allocator_init_preallocated(const ma_slot_allocator_config* pConfig, void* pHeap, ma_slot_allocator* pAllocator)
+{
+ ma_result result;
+ ma_slot_allocator_heap_layout heapLayout;
+
+ if (pAllocator == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ MA_ZERO_OBJECT(pAllocator);
+
+ if (pHeap == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ result = ma_slot_allocator_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ pAllocator->_pHeap = pHeap;
+ MA_ZERO_MEMORY(pHeap, heapLayout.sizeInBytes);
+
+ pAllocator->pGroups = (ma_slot_allocator_group*)ma_offset_ptr(pHeap, heapLayout.groupsOffset);
+ pAllocator->pSlots = (ma_uint32*)ma_offset_ptr(pHeap, heapLayout.slotsOffset);
+ pAllocator->capacity = pConfig->capacity;
+
+ return MA_SUCCESS;
+}
+
+MA_API ma_result ma_slot_allocator_init(const ma_slot_allocator_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_slot_allocator* pAllocator)
+{
+ ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
+
+ result = ma_slot_allocator_get_heap_size(pConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to retrieve the size of the heap allocation. */
+ }
+
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+ } else {
+ pHeap = NULL;
+ }
+
+ result = ma_slot_allocator_init_preallocated(pConfig, pHeap, pAllocator);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
+ }
+
+ pAllocator->_ownsHeap = MA_TRUE;
+ return MA_SUCCESS;
+}
+
+MA_API void ma_slot_allocator_uninit(ma_slot_allocator* pAllocator, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pAllocator == NULL) {
+ return;
+ }
+
+ if (pAllocator->_ownsHeap) {
+ ma_free(pAllocator->_pHeap, pAllocationCallbacks);
+ }
+}
+
+MA_API ma_result ma_slot_allocator_alloc(ma_slot_allocator* pAllocator, ma_uint64* pSlot)
+{
+ ma_uint32 iAttempt;
+ const ma_uint32 maxAttempts = 2; /* The number of iterations to perform until returning MA_OUT_OF_MEMORY if no slots can be found. */
+
+ if (pAllocator == NULL || pSlot == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ for (iAttempt = 0; iAttempt < maxAttempts; iAttempt += 1) {
+ /* We need to acquire a suitable bitfield first. This is a bitfield that's got an available slot within it. */
+ ma_uint32 iGroup;
+ for (iGroup = 0; iGroup < ma_slot_allocator_group_capacity(pAllocator); iGroup += 1) {
+ /* CAS */
+ for (;;) {
+ ma_uint32 oldBitfield;
+ ma_uint32 newBitfield;
+ ma_uint32 bitOffset;
+
+ oldBitfield = c89atomic_load_32(&pAllocator->pGroups[iGroup].bitfield); /* <-- This copy must happen. The compiler must not optimize this away. */
+
+ /* Fast check to see if anything is available. */
+ if (oldBitfield == 0xFFFFFFFF) {
+ break; /* No available bits in this bitfield. */
+ }
+
+ bitOffset = ma_ffs_32(~oldBitfield);
+ MA_ASSERT(bitOffset < 32);
+
+ newBitfield = oldBitfield | (1 << bitOffset);
+
+ if (c89atomic_compare_and_swap_32(&pAllocator->pGroups[iGroup].bitfield, oldBitfield, newBitfield) == oldBitfield) {
+ ma_uint32 slotIndex;
+
+ /* Increment the counter as soon as possible to have other threads report out-of-memory sooner than later. */
+ c89atomic_fetch_add_32(&pAllocator->count, 1);
+
+ /* The slot index is required for constructing the output value. */
+ slotIndex = (iGroup << 5) + bitOffset; /* iGroup << 5 = iGroup * 32 */
+ if (slotIndex >= pAllocator->capacity) {
+ return MA_OUT_OF_MEMORY;
+ }
+
+ /* Increment the reference count before constructing the output value. */
+ pAllocator->pSlots[slotIndex] += 1;
+
+ /* Construct the output value. */
+ *pSlot = (((ma_uint64)pAllocator->pSlots[slotIndex] << 32) | slotIndex);
+
+ return MA_SUCCESS;
+ }
+ }
+ }
+
+ /* We weren't able to find a slot. If it's because we've reached our capacity we need to return MA_OUT_OF_MEMORY. Otherwise we need to do another iteration and try again. */
+ if (pAllocator->count < pAllocator->capacity) {
+ ma_yield();
+ } else {
+ return MA_OUT_OF_MEMORY;
+ }
+ }
+
+ /* We couldn't find a slot within the maximum number of attempts. */
+ return MA_OUT_OF_MEMORY;
+}
+
+MA_API ma_result ma_slot_allocator_free(ma_slot_allocator* pAllocator, ma_uint64 slot)
+{
+ ma_uint32 iGroup;
+ ma_uint32 iBit;
+
+ if (pAllocator == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ iGroup = (ma_uint32)((slot & 0xFFFFFFFF) >> 5); /* slot / 32 */
+ iBit = (ma_uint32)((slot & 0xFFFFFFFF) & 31); /* slot % 32 */
+
+ if (iGroup >= ma_slot_allocator_group_capacity(pAllocator)) {
+ return MA_INVALID_ARGS;
+ }
+
+ MA_ASSERT(iBit < 32); /* This must be true due to the logic we used to actually calculate it. */
+
+ while (c89atomic_load_32(&pAllocator->count) > 0) {
+ /* CAS */
+ ma_uint32 oldBitfield;
+ ma_uint32 newBitfield;
+
+ oldBitfield = c89atomic_load_32(&pAllocator->pGroups[iGroup].bitfield); /* <-- This copy must happen. The compiler must not optimize this away. */
+ newBitfield = oldBitfield & ~(1 << iBit);
+
+ /* Debugging for checking for double-frees. */
+ #if defined(MA_DEBUG_OUTPUT)
+ {
+ if ((oldBitfield & (1 << iBit)) == 0) {
+ MA_ASSERT(MA_FALSE); /* Double free detected.*/
+ }
+ }
+ #endif
+
+ if (c89atomic_compare_and_swap_32(&pAllocator->pGroups[iGroup].bitfield, oldBitfield, newBitfield) == oldBitfield) {
+ c89atomic_fetch_sub_32(&pAllocator->count, 1);
+ return MA_SUCCESS;
+ }
+ }
+
+ /* Getting here means there are no allocations available for freeing. */
+ return MA_INVALID_OPERATION;
+}
+
+
+#define MA_JOB_ID_NONE ~((ma_uint64)0)
+#define MA_JOB_SLOT_NONE (ma_uint16)(~0)
+
+static MA_INLINE ma_uint32 ma_job_extract_refcount(ma_uint64 toc)
+{
+ return (ma_uint32)(toc >> 32);
+}
+
+static MA_INLINE ma_uint16 ma_job_extract_slot(ma_uint64 toc)
+{
+ return (ma_uint16)(toc & 0x0000FFFF);
+}
+
+static MA_INLINE ma_uint16 ma_job_extract_code(ma_uint64 toc)
+{
+ return (ma_uint16)((toc & 0xFFFF0000) >> 16);
+}
+
+static MA_INLINE ma_uint64 ma_job_toc_to_allocation(ma_uint64 toc)
+{
+ return ((ma_uint64)ma_job_extract_refcount(toc) << 32) | (ma_uint64)ma_job_extract_slot(toc);
+}
+
+static MA_INLINE ma_uint64 ma_job_set_refcount(ma_uint64 toc, ma_uint32 refcount)
+{
+ /* Clear the reference count first. */
+ toc = toc & ~((ma_uint64)0xFFFFFFFF << 32);
+ toc = toc | ((ma_uint64)refcount << 32);
+
+ return toc;
+}
+
+
+MA_API ma_job ma_job_init(ma_uint16 code)
+{
+ ma_job job;
+
+ MA_ZERO_OBJECT(&job);
+ job.toc.breakup.code = code;
+ job.toc.breakup.slot = MA_JOB_SLOT_NONE; /* Temp value. Will be allocated when posted to a queue. */
+ job.next = MA_JOB_ID_NONE;
+
+ return job;
+}
+
+
+static ma_result ma_job_process__noop(ma_job* pJob);
+static ma_result ma_job_process__quit(ma_job* pJob);
+static ma_result ma_job_process__custom(ma_job* pJob);
+static ma_result ma_job_process__resource_manager__load_data_buffer_node(ma_job* pJob);
+static ma_result ma_job_process__resource_manager__free_data_buffer_node(ma_job* pJob);
+static ma_result ma_job_process__resource_manager__page_data_buffer_node(ma_job* pJob);
+static ma_result ma_job_process__resource_manager__load_data_buffer(ma_job* pJob);
+static ma_result ma_job_process__resource_manager__free_data_buffer(ma_job* pJob);
+static ma_result ma_job_process__resource_manager__load_data_stream(ma_job* pJob);
+static ma_result ma_job_process__resource_manager__free_data_stream(ma_job* pJob);
+static ma_result ma_job_process__resource_manager__page_data_stream(ma_job* pJob);
+static ma_result ma_job_process__resource_manager__seek_data_stream(ma_job* pJob);
+
+#if !defined(MA_NO_DEVICE_IO)
+static ma_result ma_job_process__device__aaudio_reroute(ma_job* pJob);
+#endif
+
+static ma_job_proc g_jobVTable[MA_JOB_TYPE_COUNT] =
+{
+ /* Miscellaneous. */
+ ma_job_process__quit, /* MA_JOB_TYPE_QUIT */
+ ma_job_process__custom, /* MA_JOB_TYPE_CUSTOM */
+
+ /* Resource Manager. */
+ ma_job_process__resource_manager__load_data_buffer_node, /* MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_BUFFER_NODE */
+ ma_job_process__resource_manager__free_data_buffer_node, /* MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_BUFFER_NODE */
+ ma_job_process__resource_manager__page_data_buffer_node, /* MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_BUFFER_NODE */
+ ma_job_process__resource_manager__load_data_buffer, /* MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_BUFFER */
+ ma_job_process__resource_manager__free_data_buffer, /* MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_BUFFER */
+ ma_job_process__resource_manager__load_data_stream, /* MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_STREAM */
+ ma_job_process__resource_manager__free_data_stream, /* MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_STREAM */
+ ma_job_process__resource_manager__page_data_stream, /* MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_STREAM */
+ ma_job_process__resource_manager__seek_data_stream, /* MA_JOB_TYPE_RESOURCE_MANAGER_SEEK_DATA_STREAM */
+
+ /* Device. */
+#if !defined(MA_NO_DEVICE_IO)
+ ma_job_process__device__aaudio_reroute /*MA_JOB_TYPE_DEVICE_AAUDIO_REROUTE*/
+#endif
+};
+
+MA_API ma_result ma_job_process(ma_job* pJob)
+{
+ if (pJob == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (pJob->toc.breakup.code > MA_JOB_TYPE_COUNT) {
+ return MA_INVALID_OPERATION;
+ }
+
+ return g_jobVTable[pJob->toc.breakup.code](pJob);
+}
+
+static ma_result ma_job_process__noop(ma_job* pJob)
+{
+ MA_ASSERT(pJob != NULL);
+
+ /* No-op. */
+ (void)pJob;
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_job_process__quit(ma_job* pJob)
+{
+ return ma_job_process__noop(pJob);
+}
+
+static ma_result ma_job_process__custom(ma_job* pJob)
+{
+ MA_ASSERT(pJob != NULL);
+
+ /* No-op if there's no callback. */
+ if (pJob->data.custom.proc == NULL) {
+ return MA_SUCCESS;
+ }
+
+ return pJob->data.custom.proc(pJob);
+}
+
+
+
+MA_API ma_job_queue_config ma_job_queue_config_init(ma_uint32 flags, ma_uint32 capacity)
+{
+ ma_job_queue_config config;
+
+ config.flags = flags;
+ config.capacity = capacity;
+
+ return config;
+}
+
+
+typedef struct
+{
+ size_t sizeInBytes;
+ size_t allocatorOffset;
+ size_t jobsOffset;
+} ma_job_queue_heap_layout;
+
+static ma_result ma_job_queue_get_heap_layout(const ma_job_queue_config* pConfig, ma_job_queue_heap_layout* pHeapLayout)
+{
+ ma_result result;
+
+ MA_ASSERT(pHeapLayout != NULL);
+
+ MA_ZERO_OBJECT(pHeapLayout);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (pConfig->capacity == 0) {
+ return MA_INVALID_ARGS;
+ }
+
+ pHeapLayout->sizeInBytes = 0;
+
+ /* Allocator. */
+ {
+ ma_slot_allocator_config allocatorConfig;
+ size_t allocatorHeapSizeInBytes;
+
+ allocatorConfig = ma_slot_allocator_config_init(pConfig->capacity);
+ result = ma_slot_allocator_get_heap_size(&allocatorConfig, &allocatorHeapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ pHeapLayout->allocatorOffset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += allocatorHeapSizeInBytes;
+ }
+
+ /* Jobs. */
+ pHeapLayout->jobsOffset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += ma_align_64(pConfig->capacity * sizeof(ma_job));
+
+ return MA_SUCCESS;
+}
+
+MA_API ma_result ma_job_queue_get_heap_size(const ma_job_queue_config* pConfig, size_t* pHeapSizeInBytes)
+{
+ ma_result result;
+ ma_job_queue_heap_layout layout;
+
+ if (pHeapSizeInBytes == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ *pHeapSizeInBytes = 0;
+
+ result = ma_job_queue_get_heap_layout(pConfig, &layout);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ *pHeapSizeInBytes = layout.sizeInBytes;
+
+ return MA_SUCCESS;
+}
+
+MA_API ma_result ma_job_queue_init_preallocated(const ma_job_queue_config* pConfig, void* pHeap, ma_job_queue* pQueue)
+{
+ ma_result result;
+ ma_job_queue_heap_layout heapLayout;
+ ma_slot_allocator_config allocatorConfig;
+
+ if (pQueue == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ MA_ZERO_OBJECT(pQueue);
+
+ result = ma_job_queue_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ pQueue->_pHeap = pHeap;
+ MA_ZERO_MEMORY(pHeap, heapLayout.sizeInBytes);
+
+ pQueue->flags = pConfig->flags;
+ pQueue->capacity = pConfig->capacity;
+ pQueue->pJobs = (ma_job*)ma_offset_ptr(pHeap, heapLayout.jobsOffset);
+
+ allocatorConfig = ma_slot_allocator_config_init(pConfig->capacity);
+ result = ma_slot_allocator_init_preallocated(&allocatorConfig, ma_offset_ptr(pHeap, heapLayout.allocatorOffset), &pQueue->allocator);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ /* We need a semaphore if we're running in non-blocking mode. If threading is disabled we need to return an error. */
+ if ((pQueue->flags & MA_JOB_QUEUE_FLAG_NON_BLOCKING) == 0) {
+ #ifndef MA_NO_THREADING
+ {
+ ma_semaphore_init(0, &pQueue->sem);
+ }
+ #else
+ {
+ /* Threading is disabled and we've requested non-blocking mode. */
+ return MA_INVALID_OPERATION;
+ }
+ #endif
+ }
+
+ /*
+ Our queue needs to be initialized with a free standing node. This should always be slot 0. Required for the lock free algorithm. The first job in the queue is
+ just a dummy item for giving us the first item in the list which is stored in the "next" member.
+ */
+ ma_slot_allocator_alloc(&pQueue->allocator, &pQueue->head); /* Will never fail. */
+ pQueue->pJobs[ma_job_extract_slot(pQueue->head)].next = MA_JOB_ID_NONE;
+ pQueue->tail = pQueue->head;
+
+ return MA_SUCCESS;
+}
+
+MA_API ma_result ma_job_queue_init(const ma_job_queue_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_job_queue* pQueue)
+{
+ ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
+
+ result = ma_job_queue_get_heap_size(pConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+ } else {
+ pHeap = NULL;
+ }
+
+ result = ma_job_queue_init_preallocated(pConfig, pHeap, pQueue);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
+ }
+
+ pQueue->_ownsHeap = MA_TRUE;
+ return MA_SUCCESS;
+}
+
+MA_API void ma_job_queue_uninit(ma_job_queue* pQueue, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pQueue == NULL) {
+ return;
+ }
+
+ /* All we need to do is uninitialize the semaphore. */
+ if ((pQueue->flags & MA_JOB_QUEUE_FLAG_NON_BLOCKING) == 0) {
+ #ifndef MA_NO_THREADING
+ {
+ ma_semaphore_uninit(&pQueue->sem);
+ }
+ #else
+ {
+ MA_ASSERT(MA_FALSE); /* Should never get here. Should have been checked at initialization time. */
+ }
+ #endif
+ }
+
+ ma_slot_allocator_uninit(&pQueue->allocator, pAllocationCallbacks);
+
+ if (pQueue->_ownsHeap) {
+ ma_free(pQueue->_pHeap, pAllocationCallbacks);
+ }
+}
+
+static ma_bool32 ma_job_queue_cas(volatile ma_uint64* dst, ma_uint64 expected, ma_uint64 desired)
+{
+ /* The new counter is taken from the expected value. */
+ return c89atomic_compare_and_swap_64(dst, expected, ma_job_set_refcount(desired, ma_job_extract_refcount(expected) + 1)) == expected;
+}
+
+MA_API ma_result ma_job_queue_post(ma_job_queue* pQueue, const ma_job* pJob)
+{
+ /*
+ Lock free queue implementation based on the paper by Michael and Scott: Nonblocking Algorithms and Preemption-Safe Locking on Multiprogrammed Shared Memory Multiprocessors
+ */
+ ma_result result;
+ ma_uint64 slot;
+ ma_uint64 tail;
+ ma_uint64 next;
+
+ if (pQueue == NULL || pJob == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ /* We need a new slot. */
+ result = ma_slot_allocator_alloc(&pQueue->allocator, &slot);
+ if (result != MA_SUCCESS) {
+ return result; /* Probably ran out of slots. If so, MA_OUT_OF_MEMORY will be returned. */
+ }
+
+ /* At this point we should have a slot to place the job. */
+ MA_ASSERT(ma_job_extract_slot(slot) < pQueue->capacity);
+
+ /* We need to put the job into memory before we do anything. */
+ pQueue->pJobs[ma_job_extract_slot(slot)] = *pJob;
+ pQueue->pJobs[ma_job_extract_slot(slot)].toc.allocation = slot; /* This will overwrite the job code. */
+ pQueue->pJobs[ma_job_extract_slot(slot)].toc.breakup.code = pJob->toc.breakup.code; /* The job code needs to be applied again because the line above overwrote it. */
+ pQueue->pJobs[ma_job_extract_slot(slot)].next = MA_JOB_ID_NONE; /* Reset for safety. */
+
+ #ifndef MA_USE_EXPERIMENTAL_LOCK_FREE_JOB_QUEUE
+ ma_spinlock_lock(&pQueue->lock);
+ #endif
+ {
+ /* The job is stored in memory so now we need to add it to our linked list. We only ever add items to the end of the list. */
+ for (;;) {
+ tail = c89atomic_load_64(&pQueue->tail);
+ next = c89atomic_load_64(&pQueue->pJobs[ma_job_extract_slot(tail)].next);
+
+ if (ma_job_toc_to_allocation(tail) == ma_job_toc_to_allocation(c89atomic_load_64(&pQueue->tail))) {
+ if (ma_job_extract_slot(next) == 0xFFFF) {
+ if (ma_job_queue_cas(&pQueue->pJobs[ma_job_extract_slot(tail)].next, next, slot)) {
+ break;
+ }
+ } else {
+ ma_job_queue_cas(&pQueue->tail, tail, ma_job_extract_slot(next));
+ }
+ }
+ }
+ ma_job_queue_cas(&pQueue->tail, tail, slot);
+ }
+ #ifndef MA_USE_EXPERIMENTAL_LOCK_FREE_JOB_QUEUE
+ ma_spinlock_unlock(&pQueue->lock);
+ #endif
+
+
+ /* Signal the semaphore as the last step if we're using synchronous mode. */
+ if ((pQueue->flags & MA_JOB_QUEUE_FLAG_NON_BLOCKING) == 0) {
+ #ifndef MA_NO_THREADING
+ {
+ ma_semaphore_release(&pQueue->sem);
+ }
+ #else
+ {
+ MA_ASSERT(MA_FALSE); /* Should never get here. Should have been checked at initialization time. */
+ }
+ #endif
+ }
+
+ return MA_SUCCESS;
+}
+
+MA_API ma_result ma_job_queue_next(ma_job_queue* pQueue, ma_job* pJob)
+{
+ ma_uint64 head;
+ ma_uint64 tail;
+ ma_uint64 next;
+
+ if (pQueue == NULL || pJob == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ /* If we're running in synchronous mode we'll need to wait on a semaphore. */
+ if ((pQueue->flags & MA_JOB_QUEUE_FLAG_NON_BLOCKING) == 0) {
+ #ifndef MA_NO_THREADING
+ {
+ ma_semaphore_wait(&pQueue->sem);
+ }
+ #else
+ {
+ MA_ASSERT(MA_FALSE); /* Should never get here. Should have been checked at initialization time. */
+ }
+ #endif
+ }
+
+ #ifndef MA_USE_EXPERIMENTAL_LOCK_FREE_JOB_QUEUE
+ ma_spinlock_lock(&pQueue->lock);
+ #endif
+ {
+ /*
+ BUG: In lock-free mode, multiple threads can be in this section of code. The "head" variable in the loop below
+ is stored. One thread can fall through to the freeing of this item while another is still using "head" for the
+ retrieval of the "next" variable.
+
+ The slot allocator might need to make use of some reference counting to ensure it's only truely freed when
+ there are no more references to the item. This must be fixed before removing these locks.
+ */
+
+ /* Now we need to remove the root item from the list. */
+ for (;;) {
+ head = c89atomic_load_64(&pQueue->head);
+ tail = c89atomic_load_64(&pQueue->tail);
+ next = c89atomic_load_64(&pQueue->pJobs[ma_job_extract_slot(head)].next);
+
+ if (ma_job_toc_to_allocation(head) == ma_job_toc_to_allocation(c89atomic_load_64(&pQueue->head))) {
+ if (ma_job_extract_slot(head) == ma_job_extract_slot(tail)) {
+ if (ma_job_extract_slot(next) == 0xFFFF) {
+ #ifndef MA_USE_EXPERIMENTAL_LOCK_FREE_JOB_QUEUE
+ ma_spinlock_unlock(&pQueue->lock);
+ #endif
+ return MA_NO_DATA_AVAILABLE;
+ }
+ ma_job_queue_cas(&pQueue->tail, tail, ma_job_extract_slot(next));
+ } else {
+ *pJob = pQueue->pJobs[ma_job_extract_slot(next)];
+ if (ma_job_queue_cas(&pQueue->head, head, ma_job_extract_slot(next))) {
+ break;
+ }
+ }
+ }
+ }
+ }
+ #ifndef MA_USE_EXPERIMENTAL_LOCK_FREE_JOB_QUEUE
+ ma_spinlock_unlock(&pQueue->lock);
+ #endif
+
+ ma_slot_allocator_free(&pQueue->allocator, head);
+
+ /*
+ If it's a quit job make sure it's put back on the queue to ensure other threads have an opportunity to detect it and terminate naturally. We
+ could instead just leave it on the queue, but that would involve fiddling with the lock-free code above and I want to keep that as simple as
+ possible.
+ */
+ if (pJob->toc.breakup.code == MA_JOB_TYPE_QUIT) {
+ ma_job_queue_post(pQueue, pJob);
+ return MA_CANCELLED; /* Return a cancelled status just in case the thread is checking return codes and not properly checking for a quit job. */
+ }
+
+ return MA_SUCCESS;
+}
+
+
+
+
+/************************************************************************************************************************************************************
+*************************************************************************************************************************************************************
+
+DEVICE I/O
+==========
+
+*************************************************************************************************************************************************************
+************************************************************************************************************************************************************/
+#ifndef MA_NO_DEVICE_IO
+#ifdef MA_WIN32
+ #include
+ #include
+ #include
+#endif
+
+#if defined(MA_APPLE) && (__MAC_OS_X_VERSION_MIN_REQUIRED < 101200)
+ #include /* For mach_absolute_time() */
+#endif
+
+#ifdef MA_POSIX
+ #include
+ #include
+ #include
+#endif
+
+/*
+Unfortunately using runtime linking for pthreads causes problems. This has occurred for me when testing on FreeBSD. When
+using runtime linking, deadlocks can occur (for me it happens when loading data from fread()). It turns out that doing
+compile-time linking fixes this. I'm not sure why this happens, but the safest way I can think of to fix this is to simply
+disable runtime linking by default. To enable runtime linking, #define this before the implementation of this file. I am
+not officially supporting this, but I'm leaving it here in case it's useful for somebody, somewhere.
+*/
+/*#define MA_USE_RUNTIME_LINKING_FOR_PTHREAD*/
+
+/* Disable run-time linking on certain backends. */
+#ifndef MA_NO_RUNTIME_LINKING
+ #if defined(MA_EMSCRIPTEN)
+ #define MA_NO_RUNTIME_LINKING
+ #endif
+#endif
+
+
+MA_API void ma_device_info_add_native_data_format(ma_device_info* pDeviceInfo, ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 flags)
+{
+ if (pDeviceInfo == NULL) {
+ return;
+ }
+
+ if (pDeviceInfo->nativeDataFormatCount < ma_countof(pDeviceInfo->nativeDataFormats)) {
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = format;
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = channels;
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = sampleRate;
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = flags;
+ pDeviceInfo->nativeDataFormatCount += 1;
+ }
+}
+
+
+MA_API const char* ma_get_backend_name(ma_backend backend)
+{
+ switch (backend)
+ {
+ case ma_backend_wasapi: return "WASAPI";
+ case ma_backend_dsound: return "DirectSound";
+ case ma_backend_winmm: return "WinMM";
+ case ma_backend_coreaudio: return "Core Audio";
+ case ma_backend_sndio: return "sndio";
+ case ma_backend_audio4: return "audio(4)";
+ case ma_backend_oss: return "OSS";
+ case ma_backend_pulseaudio: return "PulseAudio";
+ case ma_backend_alsa: return "ALSA";
+ case ma_backend_jack: return "JACK";
+ case ma_backend_aaudio: return "AAudio";
+ case ma_backend_opensl: return "OpenSL|ES";
+ case ma_backend_webaudio: return "Web Audio";
+ case ma_backend_custom: return "Custom";
+ case ma_backend_null: return "Null";
+ default: return "Unknown";
+ }
+}
+
+MA_API ma_bool32 ma_is_backend_enabled(ma_backend backend)
+{
+ /*
+ This looks a little bit gross, but we want all backends to be included in the switch to avoid warnings on some compilers
+ about some enums not being handled by the switch statement.
+ */
+ switch (backend)
+ {
+ case ma_backend_wasapi:
+ #if defined(MA_HAS_WASAPI)
+ return MA_TRUE;
+ #else
+ return MA_FALSE;
+ #endif
+ case ma_backend_dsound:
+ #if defined(MA_HAS_DSOUND)
+ return MA_TRUE;
+ #else
+ return MA_FALSE;
+ #endif
+ case ma_backend_winmm:
+ #if defined(MA_HAS_WINMM)
+ return MA_TRUE;
+ #else
+ return MA_FALSE;
+ #endif
+ case ma_backend_coreaudio:
+ #if defined(MA_HAS_COREAUDIO)
+ return MA_TRUE;
+ #else
+ return MA_FALSE;
+ #endif
+ case ma_backend_sndio:
+ #if defined(MA_HAS_SNDIO)
+ return MA_TRUE;
+ #else
+ return MA_FALSE;
+ #endif
+ case ma_backend_audio4:
+ #if defined(MA_HAS_AUDIO4)
+ return MA_TRUE;
+ #else
+ return MA_FALSE;
+ #endif
+ case ma_backend_oss:
+ #if defined(MA_HAS_OSS)
+ return MA_TRUE;
+ #else
+ return MA_FALSE;
+ #endif
+ case ma_backend_pulseaudio:
+ #if defined(MA_HAS_PULSEAUDIO)
+ return MA_TRUE;
+ #else
+ return MA_FALSE;
+ #endif
+ case ma_backend_alsa:
+ #if defined(MA_HAS_ALSA)
+ return MA_TRUE;
+ #else
+ return MA_FALSE;
+ #endif
+ case ma_backend_jack:
+ #if defined(MA_HAS_JACK)
+ return MA_TRUE;
+ #else
+ return MA_FALSE;
+ #endif
+ case ma_backend_aaudio:
+ #if defined(MA_HAS_AAUDIO)
+ return MA_TRUE;
+ #else
+ return MA_FALSE;
+ #endif
+ case ma_backend_opensl:
+ #if defined(MA_HAS_OPENSL)
+ return MA_TRUE;
+ #else
+ return MA_FALSE;
+ #endif
+ case ma_backend_webaudio:
+ #if defined(MA_HAS_WEBAUDIO)
+ return MA_TRUE;
+ #else
+ return MA_FALSE;
+ #endif
+ case ma_backend_custom:
+ #if defined(MA_HAS_CUSTOM)
+ return MA_TRUE;
+ #else
+ return MA_FALSE;
+ #endif
+ case ma_backend_null:
+ #if defined(MA_HAS_NULL)
+ return MA_TRUE;
+ #else
+ return MA_FALSE;
+ #endif
+
+ default: return MA_FALSE;
+ }
+}
+
+MA_API ma_result ma_get_enabled_backends(ma_backend* pBackends, size_t backendCap, size_t* pBackendCount)
+{
+ size_t backendCount;
+ size_t iBackend;
+ ma_result result = MA_SUCCESS;
+
+ if (pBackendCount == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ backendCount = 0;
+
+ for (iBackend = 0; iBackend <= ma_backend_null; iBackend += 1) {
+ ma_backend backend = (ma_backend)iBackend;
+
+ if (ma_is_backend_enabled(backend)) {
+ /* The backend is enabled. Try adding it to the list. If there's no room, MA_NO_SPACE needs to be returned. */
+ if (backendCount == backendCap) {
+ result = MA_NO_SPACE;
+ break;
+ } else {
+ pBackends[backendCount] = backend;
+ backendCount += 1;
+ }
+ }
+ }
+
+ if (pBackendCount != NULL) {
+ *pBackendCount = backendCount;
+ }
+
+ return result;
+}
+
+MA_API ma_bool32 ma_is_loopback_supported(ma_backend backend)
+{
+ switch (backend)
+ {
+ case ma_backend_wasapi: return MA_TRUE;
+ case ma_backend_dsound: return MA_FALSE;
+ case ma_backend_winmm: return MA_FALSE;
+ case ma_backend_coreaudio: return MA_FALSE;
+ case ma_backend_sndio: return MA_FALSE;
+ case ma_backend_audio4: return MA_FALSE;
+ case ma_backend_oss: return MA_FALSE;
+ case ma_backend_pulseaudio: return MA_FALSE;
+ case ma_backend_alsa: return MA_FALSE;
+ case ma_backend_jack: return MA_FALSE;
+ case ma_backend_aaudio: return MA_FALSE;
+ case ma_backend_opensl: return MA_FALSE;
+ case ma_backend_webaudio: return MA_FALSE;
+ case ma_backend_custom: return MA_FALSE; /* <-- Will depend on the implementation of the backend. */
+ case ma_backend_null: return MA_FALSE;
+ default: return MA_FALSE;
+ }
+}
+
+
+
+#ifdef MA_WIN32
+/* WASAPI error codes. */
+#define MA_AUDCLNT_E_NOT_INITIALIZED ((HRESULT)0x88890001)
+#define MA_AUDCLNT_E_ALREADY_INITIALIZED ((HRESULT)0x88890002)
+#define MA_AUDCLNT_E_WRONG_ENDPOINT_TYPE ((HRESULT)0x88890003)
+#define MA_AUDCLNT_E_DEVICE_INVALIDATED ((HRESULT)0x88890004)
+#define MA_AUDCLNT_E_NOT_STOPPED ((HRESULT)0x88890005)
+#define MA_AUDCLNT_E_BUFFER_TOO_LARGE ((HRESULT)0x88890006)
+#define MA_AUDCLNT_E_OUT_OF_ORDER ((HRESULT)0x88890007)
+#define MA_AUDCLNT_E_UNSUPPORTED_FORMAT ((HRESULT)0x88890008)
+#define MA_AUDCLNT_E_INVALID_SIZE ((HRESULT)0x88890009)
+#define MA_AUDCLNT_E_DEVICE_IN_USE ((HRESULT)0x8889000A)
+#define MA_AUDCLNT_E_BUFFER_OPERATION_PENDING ((HRESULT)0x8889000B)
+#define MA_AUDCLNT_E_THREAD_NOT_REGISTERED ((HRESULT)0x8889000C)
+#define MA_AUDCLNT_E_NO_SINGLE_PROCESS ((HRESULT)0x8889000D)
+#define MA_AUDCLNT_E_EXCLUSIVE_MODE_NOT_ALLOWED ((HRESULT)0x8889000E)
+#define MA_AUDCLNT_E_ENDPOINT_CREATE_FAILED ((HRESULT)0x8889000F)
+#define MA_AUDCLNT_E_SERVICE_NOT_RUNNING ((HRESULT)0x88890010)
+#define MA_AUDCLNT_E_EVENTHANDLE_NOT_EXPECTED ((HRESULT)0x88890011)
+#define MA_AUDCLNT_E_EXCLUSIVE_MODE_ONLY ((HRESULT)0x88890012)
+#define MA_AUDCLNT_E_BUFDURATION_PERIOD_NOT_EQUAL ((HRESULT)0x88890013)
+#define MA_AUDCLNT_E_EVENTHANDLE_NOT_SET ((HRESULT)0x88890014)
+#define MA_AUDCLNT_E_INCORRECT_BUFFER_SIZE ((HRESULT)0x88890015)
+#define MA_AUDCLNT_E_BUFFER_SIZE_ERROR ((HRESULT)0x88890016)
+#define MA_AUDCLNT_E_CPUUSAGE_EXCEEDED ((HRESULT)0x88890017)
+#define MA_AUDCLNT_E_BUFFER_ERROR ((HRESULT)0x88890018)
+#define MA_AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED ((HRESULT)0x88890019)
+#define MA_AUDCLNT_E_INVALID_DEVICE_PERIOD ((HRESULT)0x88890020)
+#define MA_AUDCLNT_E_INVALID_STREAM_FLAG ((HRESULT)0x88890021)
+#define MA_AUDCLNT_E_ENDPOINT_OFFLOAD_NOT_CAPABLE ((HRESULT)0x88890022)
+#define MA_AUDCLNT_E_OUT_OF_OFFLOAD_RESOURCES ((HRESULT)0x88890023)
+#define MA_AUDCLNT_E_OFFLOAD_MODE_ONLY ((HRESULT)0x88890024)
+#define MA_AUDCLNT_E_NONOFFLOAD_MODE_ONLY ((HRESULT)0x88890025)
+#define MA_AUDCLNT_E_RESOURCES_INVALIDATED ((HRESULT)0x88890026)
+#define MA_AUDCLNT_E_RAW_MODE_UNSUPPORTED ((HRESULT)0x88890027)
+#define MA_AUDCLNT_E_ENGINE_PERIODICITY_LOCKED ((HRESULT)0x88890028)
+#define MA_AUDCLNT_E_ENGINE_FORMAT_LOCKED ((HRESULT)0x88890029)
+#define MA_AUDCLNT_E_HEADTRACKING_ENABLED ((HRESULT)0x88890030)
+#define MA_AUDCLNT_E_HEADTRACKING_UNSUPPORTED ((HRESULT)0x88890040)
+#define MA_AUDCLNT_S_BUFFER_EMPTY ((HRESULT)0x08890001)
+#define MA_AUDCLNT_S_THREAD_ALREADY_REGISTERED ((HRESULT)0x08890002)
+#define MA_AUDCLNT_S_POSITION_STALLED ((HRESULT)0x08890003)
+
+#define MA_DS_OK ((HRESULT)0)
+#define MA_DS_NO_VIRTUALIZATION ((HRESULT)0x0878000A)
+#define MA_DSERR_ALLOCATED ((HRESULT)0x8878000A)
+#define MA_DSERR_CONTROLUNAVAIL ((HRESULT)0x8878001E)
+#define MA_DSERR_INVALIDPARAM ((HRESULT)0x80070057) /*E_INVALIDARG*/
+#define MA_DSERR_INVALIDCALL ((HRESULT)0x88780032)
+#define MA_DSERR_GENERIC ((HRESULT)0x80004005) /*E_FAIL*/
+#define MA_DSERR_PRIOLEVELNEEDED ((HRESULT)0x88780046)
+#define MA_DSERR_OUTOFMEMORY ((HRESULT)0x8007000E) /*E_OUTOFMEMORY*/
+#define MA_DSERR_BADFORMAT ((HRESULT)0x88780064)
+#define MA_DSERR_UNSUPPORTED ((HRESULT)0x80004001) /*E_NOTIMPL*/
+#define MA_DSERR_NODRIVER ((HRESULT)0x88780078)
+#define MA_DSERR_ALREADYINITIALIZED ((HRESULT)0x88780082)
+#define MA_DSERR_NOAGGREGATION ((HRESULT)0x80040110) /*CLASS_E_NOAGGREGATION*/
+#define MA_DSERR_BUFFERLOST ((HRESULT)0x88780096)
+#define MA_DSERR_OTHERAPPHASPRIO ((HRESULT)0x887800A0)
+#define MA_DSERR_UNINITIALIZED ((HRESULT)0x887800AA)
+#define MA_DSERR_NOINTERFACE ((HRESULT)0x80004002) /*E_NOINTERFACE*/
+#define MA_DSERR_ACCESSDENIED ((HRESULT)0x80070005) /*E_ACCESSDENIED*/
+#define MA_DSERR_BUFFERTOOSMALL ((HRESULT)0x887800B4)
+#define MA_DSERR_DS8_REQUIRED ((HRESULT)0x887800BE)
+#define MA_DSERR_SENDLOOP ((HRESULT)0x887800C8)
+#define MA_DSERR_BADSENDBUFFERGUID ((HRESULT)0x887800D2)
+#define MA_DSERR_OBJECTNOTFOUND ((HRESULT)0x88781161)
+#define MA_DSERR_FXUNAVAILABLE ((HRESULT)0x887800DC)
+
+static ma_result ma_result_from_HRESULT(HRESULT hr)
+{
+ switch (hr)
+ {
+ case NOERROR: return MA_SUCCESS;
+ /*case S_OK: return MA_SUCCESS;*/
+
+ case E_POINTER: return MA_INVALID_ARGS;
+ case E_UNEXPECTED: return MA_ERROR;
+ case E_NOTIMPL: return MA_NOT_IMPLEMENTED;
+ case E_OUTOFMEMORY: return MA_OUT_OF_MEMORY;
+ case E_INVALIDARG: return MA_INVALID_ARGS;
+ case E_NOINTERFACE: return MA_API_NOT_FOUND;
+ case E_HANDLE: return MA_INVALID_ARGS;
+ case E_ABORT: return MA_ERROR;
+ case E_FAIL: return MA_ERROR;
+ case E_ACCESSDENIED: return MA_ACCESS_DENIED;
+
+ /* WASAPI */
+ case MA_AUDCLNT_E_NOT_INITIALIZED: return MA_DEVICE_NOT_INITIALIZED;
+ case MA_AUDCLNT_E_ALREADY_INITIALIZED: return MA_DEVICE_ALREADY_INITIALIZED;
+ case MA_AUDCLNT_E_WRONG_ENDPOINT_TYPE: return MA_INVALID_ARGS;
+ case MA_AUDCLNT_E_DEVICE_INVALIDATED: return MA_UNAVAILABLE;
+ case MA_AUDCLNT_E_NOT_STOPPED: return MA_DEVICE_NOT_STOPPED;
+ case MA_AUDCLNT_E_BUFFER_TOO_LARGE: return MA_TOO_BIG;
+ case MA_AUDCLNT_E_OUT_OF_ORDER: return MA_INVALID_OPERATION;
+ case MA_AUDCLNT_E_UNSUPPORTED_FORMAT: return MA_FORMAT_NOT_SUPPORTED;
+ case MA_AUDCLNT_E_INVALID_SIZE: return MA_INVALID_ARGS;
+ case MA_AUDCLNT_E_DEVICE_IN_USE: return MA_BUSY;
+ case MA_AUDCLNT_E_BUFFER_OPERATION_PENDING: return MA_INVALID_OPERATION;
+ case MA_AUDCLNT_E_THREAD_NOT_REGISTERED: return MA_DOES_NOT_EXIST;
+ case MA_AUDCLNT_E_NO_SINGLE_PROCESS: return MA_INVALID_OPERATION;
+ case MA_AUDCLNT_E_EXCLUSIVE_MODE_NOT_ALLOWED: return MA_SHARE_MODE_NOT_SUPPORTED;
+ case MA_AUDCLNT_E_ENDPOINT_CREATE_FAILED: return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
+ case MA_AUDCLNT_E_SERVICE_NOT_RUNNING: return MA_NOT_CONNECTED;
+ case MA_AUDCLNT_E_EVENTHANDLE_NOT_EXPECTED: return MA_INVALID_ARGS;
+ case MA_AUDCLNT_E_EXCLUSIVE_MODE_ONLY: return MA_SHARE_MODE_NOT_SUPPORTED;
+ case MA_AUDCLNT_E_BUFDURATION_PERIOD_NOT_EQUAL: return MA_INVALID_ARGS;
+ case MA_AUDCLNT_E_EVENTHANDLE_NOT_SET: return MA_INVALID_ARGS;
+ case MA_AUDCLNT_E_INCORRECT_BUFFER_SIZE: return MA_INVALID_ARGS;
+ case MA_AUDCLNT_E_BUFFER_SIZE_ERROR: return MA_INVALID_ARGS;
+ case MA_AUDCLNT_E_CPUUSAGE_EXCEEDED: return MA_ERROR;
+ case MA_AUDCLNT_E_BUFFER_ERROR: return MA_ERROR;
+ case MA_AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED: return MA_INVALID_ARGS;
+ case MA_AUDCLNT_E_INVALID_DEVICE_PERIOD: return MA_INVALID_ARGS;
+ case MA_AUDCLNT_E_INVALID_STREAM_FLAG: return MA_INVALID_ARGS;
+ case MA_AUDCLNT_E_ENDPOINT_OFFLOAD_NOT_CAPABLE: return MA_INVALID_OPERATION;
+ case MA_AUDCLNT_E_OUT_OF_OFFLOAD_RESOURCES: return MA_OUT_OF_MEMORY;
+ case MA_AUDCLNT_E_OFFLOAD_MODE_ONLY: return MA_INVALID_OPERATION;
+ case MA_AUDCLNT_E_NONOFFLOAD_MODE_ONLY: return MA_INVALID_OPERATION;
+ case MA_AUDCLNT_E_RESOURCES_INVALIDATED: return MA_INVALID_DATA;
+ case MA_AUDCLNT_E_RAW_MODE_UNSUPPORTED: return MA_INVALID_OPERATION;
+ case MA_AUDCLNT_E_ENGINE_PERIODICITY_LOCKED: return MA_INVALID_OPERATION;
+ case MA_AUDCLNT_E_ENGINE_FORMAT_LOCKED: return MA_INVALID_OPERATION;
+ case MA_AUDCLNT_E_HEADTRACKING_ENABLED: return MA_INVALID_OPERATION;
+ case MA_AUDCLNT_E_HEADTRACKING_UNSUPPORTED: return MA_INVALID_OPERATION;
+ case MA_AUDCLNT_S_BUFFER_EMPTY: return MA_NO_SPACE;
+ case MA_AUDCLNT_S_THREAD_ALREADY_REGISTERED: return MA_ALREADY_EXISTS;
+ case MA_AUDCLNT_S_POSITION_STALLED: return MA_ERROR;
+
+ /* DirectSound */
+ /*case MA_DS_OK: return MA_SUCCESS;*/ /* S_OK */
+ case MA_DS_NO_VIRTUALIZATION: return MA_SUCCESS;
+ case MA_DSERR_ALLOCATED: return MA_ALREADY_IN_USE;
+ case MA_DSERR_CONTROLUNAVAIL: return MA_INVALID_OPERATION;
+ /*case MA_DSERR_INVALIDPARAM: return MA_INVALID_ARGS;*/ /* E_INVALIDARG */
+ case MA_DSERR_INVALIDCALL: return MA_INVALID_OPERATION;
+ /*case MA_DSERR_GENERIC: return MA_ERROR;*/ /* E_FAIL */
+ case MA_DSERR_PRIOLEVELNEEDED: return MA_INVALID_OPERATION;
+ /*case MA_DSERR_OUTOFMEMORY: return MA_OUT_OF_MEMORY;*/ /* E_OUTOFMEMORY */
+ case MA_DSERR_BADFORMAT: return MA_FORMAT_NOT_SUPPORTED;
+ /*case MA_DSERR_UNSUPPORTED: return MA_NOT_IMPLEMENTED;*/ /* E_NOTIMPL */
+ case MA_DSERR_NODRIVER: return MA_FAILED_TO_INIT_BACKEND;
+ case MA_DSERR_ALREADYINITIALIZED: return MA_DEVICE_ALREADY_INITIALIZED;
+ case MA_DSERR_NOAGGREGATION: return MA_ERROR;
+ case MA_DSERR_BUFFERLOST: return MA_UNAVAILABLE;
+ case MA_DSERR_OTHERAPPHASPRIO: return MA_ACCESS_DENIED;
+ case MA_DSERR_UNINITIALIZED: return MA_DEVICE_NOT_INITIALIZED;
+ /*case MA_DSERR_NOINTERFACE: return MA_API_NOT_FOUND;*/ /* E_NOINTERFACE */
+ /*case MA_DSERR_ACCESSDENIED: return MA_ACCESS_DENIED;*/ /* E_ACCESSDENIED */
+ case MA_DSERR_BUFFERTOOSMALL: return MA_NO_SPACE;
+ case MA_DSERR_DS8_REQUIRED: return MA_INVALID_OPERATION;
+ case MA_DSERR_SENDLOOP: return MA_DEADLOCK;
+ case MA_DSERR_BADSENDBUFFERGUID: return MA_INVALID_ARGS;
+ case MA_DSERR_OBJECTNOTFOUND: return MA_NO_DEVICE;
+ case MA_DSERR_FXUNAVAILABLE: return MA_UNAVAILABLE;
+
+ default: return MA_ERROR;
+ }
+}
+
+typedef HRESULT (WINAPI * MA_PFN_CoInitializeEx)(LPVOID pvReserved, DWORD dwCoInit);
+typedef void (WINAPI * MA_PFN_CoUninitialize)(void);
+typedef HRESULT (WINAPI * MA_PFN_CoCreateInstance)(REFCLSID rclsid, LPUNKNOWN pUnkOuter, DWORD dwClsContext, REFIID riid, LPVOID *ppv);
+typedef void (WINAPI * MA_PFN_CoTaskMemFree)(LPVOID pv);
+typedef HRESULT (WINAPI * MA_PFN_PropVariantClear)(PROPVARIANT *pvar);
+typedef int (WINAPI * MA_PFN_StringFromGUID2)(const GUID* const rguid, LPOLESTR lpsz, int cchMax);
+
+typedef HWND (WINAPI * MA_PFN_GetForegroundWindow)(void);
+typedef HWND (WINAPI * MA_PFN_GetDesktopWindow)(void);
+
+#if defined(MA_WIN32_DESKTOP)
+/* Microsoft documents these APIs as returning LSTATUS, but the Win32 API shipping with some compilers do not define it. It's just a LONG. */
+typedef LONG (WINAPI * MA_PFN_RegOpenKeyExA)(HKEY hKey, LPCSTR lpSubKey, DWORD ulOptions, REGSAM samDesired, PHKEY phkResult);
+typedef LONG (WINAPI * MA_PFN_RegCloseKey)(HKEY hKey);
+typedef LONG (WINAPI * MA_PFN_RegQueryValueExA)(HKEY hKey, LPCSTR lpValueName, LPDWORD lpReserved, LPDWORD lpType, LPBYTE lpData, LPDWORD lpcbData);
+#endif /* MA_WIN32_DESKTOP */
+#endif /* MA_WIN32 */
+
+
+#define MA_DEFAULT_PLAYBACK_DEVICE_NAME "Default Playback Device"
+#define MA_DEFAULT_CAPTURE_DEVICE_NAME "Default Capture Device"
+
+
+
+
+/*******************************************************************************
+
+Timing
+
+*******************************************************************************/
+#ifdef MA_WIN32
+ static LARGE_INTEGER g_ma_TimerFrequency; /* <-- Initialized to zero since it's static. */
+ void ma_timer_init(ma_timer* pTimer)
+ {
+ LARGE_INTEGER counter;
+
+ if (g_ma_TimerFrequency.QuadPart == 0) {
+ QueryPerformanceFrequency(&g_ma_TimerFrequency);
+ }
+
+ QueryPerformanceCounter(&counter);
+ pTimer->counter = counter.QuadPart;
+ }
+
+ double ma_timer_get_time_in_seconds(ma_timer* pTimer)
+ {
+ LARGE_INTEGER counter;
+ if (!QueryPerformanceCounter(&counter)) {
+ return 0;
+ }
+
+ return (double)(counter.QuadPart - pTimer->counter) / g_ma_TimerFrequency.QuadPart;
+ }
+#elif defined(MA_APPLE) && (__MAC_OS_X_VERSION_MIN_REQUIRED < 101200)
+ static ma_uint64 g_ma_TimerFrequency = 0;
+ static void ma_timer_init(ma_timer* pTimer)
+ {
+ mach_timebase_info_data_t baseTime;
+ mach_timebase_info(&baseTime);
+ g_ma_TimerFrequency = (baseTime.denom * 1e9) / baseTime.numer;
+
+ pTimer->counter = mach_absolute_time();
+ }
+
+ static double ma_timer_get_time_in_seconds(ma_timer* pTimer)
+ {
+ ma_uint64 newTimeCounter = mach_absolute_time();
+ ma_uint64 oldTimeCounter = pTimer->counter;
+
+ return (newTimeCounter - oldTimeCounter) / g_ma_TimerFrequency;
+ }
+#elif defined(MA_EMSCRIPTEN)
+ static MA_INLINE void ma_timer_init(ma_timer* pTimer)
+ {
+ pTimer->counterD = emscripten_get_now();
+ }
+
+ static MA_INLINE double ma_timer_get_time_in_seconds(ma_timer* pTimer)
+ {
+ return (emscripten_get_now() - pTimer->counterD) / 1000; /* Emscripten is in milliseconds. */
+ }
+#else
+ #if defined(_POSIX_C_SOURCE) && _POSIX_C_SOURCE >= 199309L
+ #if defined(CLOCK_MONOTONIC)
+ #define MA_CLOCK_ID CLOCK_MONOTONIC
+ #else
+ #define MA_CLOCK_ID CLOCK_REALTIME
+ #endif
+
+ static void ma_timer_init(ma_timer* pTimer)
+ {
+ struct timespec newTime;
+ clock_gettime(MA_CLOCK_ID, &newTime);
+
+ pTimer->counter = (newTime.tv_sec * 1000000000) + newTime.tv_nsec;
+ }
+
+ static double ma_timer_get_time_in_seconds(ma_timer* pTimer)
+ {
+ ma_uint64 newTimeCounter;
+ ma_uint64 oldTimeCounter;
+
+ struct timespec newTime;
+ clock_gettime(MA_CLOCK_ID, &newTime);
+
+ newTimeCounter = (newTime.tv_sec * 1000000000) + newTime.tv_nsec;
+ oldTimeCounter = pTimer->counter;
+
+ return (newTimeCounter - oldTimeCounter) / 1000000000.0;
+ }
+ #else
+ static void ma_timer_init(ma_timer* pTimer)
+ {
+ struct timeval newTime;
+ gettimeofday(&newTime, NULL);
+
+ pTimer->counter = (newTime.tv_sec * 1000000) + newTime.tv_usec;
+ }
+
+ static double ma_timer_get_time_in_seconds(ma_timer* pTimer)
+ {
+ ma_uint64 newTimeCounter;
+ ma_uint64 oldTimeCounter;
+
+ struct timeval newTime;
+ gettimeofday(&newTime, NULL);
+
+ newTimeCounter = (newTime.tv_sec * 1000000) + newTime.tv_usec;
+ oldTimeCounter = pTimer->counter;
+
+ return (newTimeCounter - oldTimeCounter) / 1000000.0;
+ }
+ #endif
+#endif
+
+
+/*******************************************************************************
+
+Dynamic Linking
+
+*******************************************************************************/
+MA_API ma_handle ma_dlopen(ma_context* pContext, const char* filename)
+{
+ ma_handle handle;
+
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "Loading library: %s\n", filename);
+
#ifdef _WIN32
-#ifdef _WIN64
-#define C89ATOMIC_64BIT
+#ifdef MA_WIN32_DESKTOP
+ handle = (ma_handle)LoadLibraryA(filename);
+#else
+ /* *sigh* It appears there is no ANSI version of LoadPackagedLibrary()... */
+ WCHAR filenameW[4096];
+ if (MultiByteToWideChar(CP_UTF8, 0, filename, -1, filenameW, sizeof(filenameW)) == 0) {
+ handle = NULL;
+ } else {
+ handle = (ma_handle)LoadPackagedLibrary(filenameW, 0);
+ }
+#endif
+#else
+ handle = (ma_handle)dlopen(filename, RTLD_NOW);
+#endif
+
+ /*
+ I'm not considering failure to load a library an error nor a warning because seamlessly falling through to a lower-priority
+ backend is a deliberate design choice. Instead I'm logging it as an informational message.
+ */
+ if (handle == NULL) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_INFO, "Failed to load library: %s\n", filename);
+ }
+
+ (void)pContext; /* It's possible for pContext to be unused. */
+ return handle;
+}
+
+MA_API void ma_dlclose(ma_context* pContext, ma_handle handle)
+{
+#ifdef _WIN32
+ FreeLibrary((HMODULE)handle);
+#else
+ dlclose((void*)handle);
+#endif
+
+ (void)pContext;
+}
+
+MA_API ma_proc ma_dlsym(ma_context* pContext, ma_handle handle, const char* symbol)
+{
+ ma_proc proc;
+
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "Loading symbol: %s\n", symbol);
+
+#ifdef _WIN32
+ proc = (ma_proc)GetProcAddress((HMODULE)handle, symbol);
+#else
+#if defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8))
+ #pragma GCC diagnostic push
+ #pragma GCC diagnostic ignored "-Wpedantic"
+#endif
+ proc = (ma_proc)dlsym((void*)handle, symbol);
+#if defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8))
+ #pragma GCC diagnostic pop
+#endif
+#endif
+
+ if (proc == NULL) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_WARNING, "Failed to load symbol: %s\n", symbol);
+ }
+
+ (void)pContext; /* It's possible for pContext to be unused. */
+ return proc;
+}
+
+
+#if 0
+static ma_uint32 ma_get_closest_standard_sample_rate(ma_uint32 sampleRateIn)
+{
+ ma_uint32 closestRate = 0;
+ ma_uint32 closestDiff = 0xFFFFFFFF;
+ size_t iStandardRate;
+
+ for (iStandardRate = 0; iStandardRate < ma_countof(g_maStandardSampleRatePriorities); ++iStandardRate) {
+ ma_uint32 standardRate = g_maStandardSampleRatePriorities[iStandardRate];
+ ma_uint32 diff;
+
+ if (sampleRateIn > standardRate) {
+ diff = sampleRateIn - standardRate;
+ } else {
+ diff = standardRate - sampleRateIn;
+ }
+
+ if (diff == 0) {
+ return standardRate; /* The input sample rate is a standard rate. */
+ }
+
+ if (closestDiff > diff) {
+ closestDiff = diff;
+ closestRate = standardRate;
+ }
+ }
+
+ return closestRate;
+}
+#endif
+
+
+static MA_INLINE unsigned int ma_device_disable_denormals(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
+
+ if (!pDevice->noDisableDenormals) {
+ return ma_disable_denormals();
+ } else {
+ return 0;
+ }
+}
+
+static MA_INLINE void ma_device_restore_denormals(ma_device* pDevice, unsigned int prevState)
+{
+ MA_ASSERT(pDevice != NULL);
+
+ if (!pDevice->noDisableDenormals) {
+ ma_restore_denormals(prevState);
+ } else {
+ /* Do nothing. */
+ (void)prevState;
+ }
+}
+
+static ma_device_notification ma_device_notification_init(ma_device* pDevice, ma_device_notification_type type)
+{
+ ma_device_notification notification;
+
+ MA_ZERO_OBJECT(¬ification);
+ notification.pDevice = pDevice;
+ notification.type = type;
+
+ return notification;
+}
+
+static void ma_device__on_notification(ma_device_notification notification)
+{
+ MA_ASSERT(notification.pDevice != NULL);
+
+ if (notification.pDevice->onNotification != NULL) {
+ notification.pDevice->onNotification(¬ification);
+ }
+
+ /* TEMP FOR COMPATIBILITY: If it's a stopped notification, fire the onStop callback as well. This is only for backwards compatibility and will be removed. */
+ if (notification.pDevice->onStop != NULL && notification.type == ma_device_notification_type_stopped) {
+ notification.pDevice->onStop(notification.pDevice);
+ }
+}
+
+void ma_device__on_notification_started(ma_device* pDevice)
+{
+ ma_device__on_notification(ma_device_notification_init(pDevice, ma_device_notification_type_started));
+}
+
+void ma_device__on_notification_stopped(ma_device* pDevice)
+{
+ ma_device__on_notification(ma_device_notification_init(pDevice, ma_device_notification_type_stopped));
+}
+
+void ma_device__on_notification_rerouted(ma_device* pDevice)
+{
+ ma_device__on_notification(ma_device_notification_init(pDevice, ma_device_notification_type_rerouted));
+}
+
+void ma_device__on_notification_interruption_began(ma_device* pDevice)
+{
+ ma_device__on_notification(ma_device_notification_init(pDevice, ma_device_notification_type_interruption_began));
+}
+
+void ma_device__on_notification_interruption_ended(ma_device* pDevice)
+{
+ ma_device__on_notification(ma_device_notification_init(pDevice, ma_device_notification_type_interruption_ended));
+}
+
+
+static void ma_device__on_data_inner(ma_device* pDevice, void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount)
+{
+ MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(pDevice->onData != NULL);
+
+ if (!pDevice->noPreSilencedOutputBuffer && pFramesOut != NULL) {
+ ma_silence_pcm_frames(pFramesOut, frameCount, pDevice->playback.format, pDevice->playback.channels);
+ }
+
+ pDevice->onData(pDevice, pFramesOut, pFramesIn, frameCount);
+}
+
+static void ma_device__on_data(ma_device* pDevice, void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount)
+{
+ MA_ASSERT(pDevice != NULL);
+
+ if (pDevice->noFixedSizedCallback) {
+ /* Fast path. Not using a fixed sized callback. Process directly from the specified buffers. */
+ ma_device__on_data_inner(pDevice, pFramesOut, pFramesIn, frameCount);
+ } else {
+ /* Slow path. Using a fixed sized callback. Need to use the intermediary buffer. */
+ ma_uint32 totalFramesProcessed = 0;
+
+ while (totalFramesProcessed < frameCount) {
+ ma_uint32 totalFramesRemaining = frameCount - totalFramesProcessed;
+ ma_uint32 framesToProcessThisIteration = 0;
+
+ if (pFramesIn != NULL) {
+ /* Capturing. Write to the intermediary buffer. If there's no room, fire the callback to empty it. */
+ if (pDevice->capture.intermediaryBufferLen < pDevice->capture.intermediaryBufferCap) {
+ /* There's some room left in the intermediary buffer. Write to it without firing the callback. */
+ framesToProcessThisIteration = totalFramesRemaining;
+ if (framesToProcessThisIteration > pDevice->capture.intermediaryBufferCap - pDevice->capture.intermediaryBufferLen) {
+ framesToProcessThisIteration = pDevice->capture.intermediaryBufferCap - pDevice->capture.intermediaryBufferLen;
+ }
+
+ ma_copy_pcm_frames(
+ ma_offset_pcm_frames_ptr(pDevice->capture.pIntermediaryBuffer, pDevice->capture.intermediaryBufferLen, pDevice->capture.format, pDevice->capture.channels),
+ ma_offset_pcm_frames_const_ptr(pFramesIn, totalFramesProcessed, pDevice->capture.format, pDevice->capture.channels),
+ framesToProcessThisIteration,
+ pDevice->capture.format, pDevice->capture.channels);
+
+ pDevice->capture.intermediaryBufferLen += framesToProcessThisIteration;
+ }
+
+ if (pDevice->capture.intermediaryBufferLen == pDevice->capture.intermediaryBufferCap) {
+ /* No room left in the intermediary buffer. Fire the data callback. */
+ if (pDevice->type == ma_device_type_duplex) {
+ /* We'll do the duplex data callback later after we've processed the playback data. */
+ } else {
+ ma_device__on_data_inner(pDevice, NULL, pDevice->capture.pIntermediaryBuffer, pDevice->capture.intermediaryBufferCap);
+
+ /* The intermediary buffer has just been drained. */
+ pDevice->capture.intermediaryBufferLen = 0;
+ }
+ }
+ }
+
+ if (pFramesOut != NULL) {
+ /* Playing back. Read from the intermediary buffer. If there's nothing in it, fire the callback to fill it. */
+ if (pDevice->playback.intermediaryBufferLen > 0) {
+ /* There's some content in the intermediary buffer. Read from that without firing the callback. */
+ if (pDevice->type == ma_device_type_duplex) {
+ /* The frames processed this iteration for a duplex device will always be based on the capture side. Leave it unmodified. */
+ } else {
+ framesToProcessThisIteration = totalFramesRemaining;
+ if (framesToProcessThisIteration > pDevice->playback.intermediaryBufferLen) {
+ framesToProcessThisIteration = pDevice->playback.intermediaryBufferLen;
+ }
+ }
+
+ ma_copy_pcm_frames(
+ ma_offset_pcm_frames_ptr(pFramesOut, totalFramesProcessed, pDevice->playback.format, pDevice->playback.channels),
+ ma_offset_pcm_frames_ptr(pDevice->playback.pIntermediaryBuffer, pDevice->playback.intermediaryBufferCap - pDevice->playback.intermediaryBufferLen, pDevice->playback.format, pDevice->playback.channels),
+ framesToProcessThisIteration,
+ pDevice->playback.format, pDevice->playback.channels);
+
+ pDevice->playback.intermediaryBufferLen -= framesToProcessThisIteration;
+ }
+
+ if (pDevice->playback.intermediaryBufferLen == 0) {
+ /* There's nothing in the intermediary buffer. Fire the data callback to fill it. */
+ if (pDevice->type == ma_device_type_duplex) {
+ /* In duplex mode, the data callback will be fired later. Nothing to do here. */
+ } else {
+ ma_device__on_data_inner(pDevice, pDevice->playback.pIntermediaryBuffer, NULL, pDevice->playback.intermediaryBufferCap);
+
+ /* The intermediary buffer has just been filled. */
+ pDevice->playback.intermediaryBufferLen = pDevice->playback.intermediaryBufferCap;
+ }
+ }
+ }
+
+ /* If we're in duplex mode we might need to do a refill of the data. */
+ if (pDevice->type == ma_device_type_duplex) {
+ if (pDevice->capture.intermediaryBufferLen == pDevice->capture.intermediaryBufferCap) {
+ ma_device__on_data_inner(pDevice, pDevice->playback.pIntermediaryBuffer, pDevice->capture.pIntermediaryBuffer, pDevice->capture.intermediaryBufferCap);
+
+ pDevice->playback.intermediaryBufferLen = pDevice->playback.intermediaryBufferCap; /* The playback buffer will have just been filled. */
+ pDevice->capture.intermediaryBufferLen = 0; /* The intermediary buffer has just been drained. */
+ }
+ }
+
+ /* Make sure this is only incremented once in the duplex case. */
+ totalFramesProcessed += framesToProcessThisIteration;
+ }
+ }
+}
+
+static void ma_device__handle_data_callback(ma_device* pDevice, void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount)
+{
+ float masterVolumeFactor;
+
+ ma_device_get_master_volume(pDevice, &masterVolumeFactor); /* Use ma_device_get_master_volume() to ensure the volume is loaded atomically. */
+
+ if (pDevice->onData) {
+ unsigned int prevDenormalState = ma_device_disable_denormals(pDevice);
+ {
+ /* Volume control of input makes things a bit awkward because the input buffer is read-only. We'll need to use a temp buffer and loop in this case. */
+ if (pFramesIn != NULL && masterVolumeFactor < 1) {
+ ma_uint8 tempFramesIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
+ ma_uint32 bpfCapture = ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels);
+ ma_uint32 bpfPlayback = ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
+ ma_uint32 totalFramesProcessed = 0;
+ while (totalFramesProcessed < frameCount) {
+ ma_uint32 framesToProcessThisIteration = frameCount - totalFramesProcessed;
+ if (framesToProcessThisIteration > sizeof(tempFramesIn)/bpfCapture) {
+ framesToProcessThisIteration = sizeof(tempFramesIn)/bpfCapture;
+ }
+
+ ma_copy_and_apply_volume_factor_pcm_frames(tempFramesIn, ma_offset_ptr(pFramesIn, totalFramesProcessed*bpfCapture), framesToProcessThisIteration, pDevice->capture.format, pDevice->capture.channels, masterVolumeFactor);
+
+ ma_device__on_data(pDevice, ma_offset_ptr(pFramesOut, totalFramesProcessed*bpfPlayback), tempFramesIn, framesToProcessThisIteration);
+
+ totalFramesProcessed += framesToProcessThisIteration;
+ }
+ } else {
+ ma_device__on_data(pDevice, pFramesOut, pFramesIn, frameCount);
+ }
+
+ /* Volume control and clipping for playback devices. */
+ if (pFramesOut != NULL) {
+ if (masterVolumeFactor < 1) {
+ if (pFramesIn == NULL) { /* <-- In full-duplex situations, the volume will have been applied to the input samples before the data callback. Applying it again post-callback will incorrectly compound it. */
+ ma_apply_volume_factor_pcm_frames(pFramesOut, frameCount, pDevice->playback.format, pDevice->playback.channels, masterVolumeFactor);
+ }
+ }
+
+ if (!pDevice->noClip && pDevice->playback.format == ma_format_f32) {
+ ma_clip_samples_f32((float*)pFramesOut, (const float*)pFramesOut, frameCount * pDevice->playback.channels); /* Intentionally specifying the same pointer for both input and output for in-place processing. */
+ }
+ }
+ }
+ ma_device_restore_denormals(pDevice, prevDenormalState);
+ }
+}
+
+
+
+/* A helper function for reading sample data from the client. */
+static void ma_device__read_frames_from_client(ma_device* pDevice, ma_uint32 frameCount, void* pFramesOut)
+{
+ MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(frameCount > 0);
+ MA_ASSERT(pFramesOut != NULL);
+
+ if (pDevice->playback.converter.isPassthrough) {
+ ma_device__handle_data_callback(pDevice, pFramesOut, NULL, frameCount);
+ } else {
+ ma_result result;
+ ma_uint64 totalFramesReadOut;
+ void* pRunningFramesOut;
+
+ totalFramesReadOut = 0;
+ pRunningFramesOut = pFramesOut;
+
+ /*
+ We run slightly different logic depending on whether or not we're using a heap-allocated
+ buffer for caching input data. This will be the case if the data converter does not have
+ the ability to retrieve the required input frame count for a given output frame count.
+ */
+ if (pDevice->playback.pInputCache != NULL) {
+ while (totalFramesReadOut < frameCount) {
+ ma_uint64 framesToReadThisIterationIn;
+ ma_uint64 framesToReadThisIterationOut;
+
+ /* If there's any data available in the cache, that needs to get processed first. */
+ if (pDevice->playback.inputCacheRemaining > 0) {
+ framesToReadThisIterationOut = (frameCount - totalFramesReadOut);
+ framesToReadThisIterationIn = framesToReadThisIterationOut;
+ if (framesToReadThisIterationIn > pDevice->playback.inputCacheRemaining) {
+ framesToReadThisIterationIn = pDevice->playback.inputCacheRemaining;
+ }
+
+ result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, ma_offset_pcm_frames_ptr(pDevice->playback.pInputCache, pDevice->playback.inputCacheConsumed, pDevice->playback.format, pDevice->playback.channels), &framesToReadThisIterationIn, pRunningFramesOut, &framesToReadThisIterationOut);
+ if (result != MA_SUCCESS) {
+ break;
+ }
+
+ pDevice->playback.inputCacheConsumed += framesToReadThisIterationIn;
+ pDevice->playback.inputCacheRemaining -= framesToReadThisIterationIn;
+
+ totalFramesReadOut += framesToReadThisIterationOut;
+ pRunningFramesOut = ma_offset_ptr(pRunningFramesOut, framesToReadThisIterationOut * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels));
+
+ if (framesToReadThisIterationIn == 0 && framesToReadThisIterationOut == 0) {
+ break; /* We're done. */
+ }
+ }
+
+ /* Getting here means there's no data in the cache and we need to fill it up with data from the client. */
+ if (pDevice->playback.inputCacheRemaining == 0) {
+ ma_device__handle_data_callback(pDevice, pDevice->playback.pInputCache, NULL, (ma_uint32)pDevice->playback.inputCacheCap);
+
+ pDevice->playback.inputCacheConsumed = 0;
+ pDevice->playback.inputCacheRemaining = pDevice->playback.inputCacheCap;
+ }
+ }
+ } else {
+ while (totalFramesReadOut < frameCount) {
+ ma_uint8 pIntermediaryBuffer[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In client format. */
+ ma_uint64 intermediaryBufferCap = sizeof(pIntermediaryBuffer) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
+ ma_uint64 framesToReadThisIterationIn;
+ ma_uint64 framesReadThisIterationIn;
+ ma_uint64 framesToReadThisIterationOut;
+ ma_uint64 framesReadThisIterationOut;
+ ma_uint64 requiredInputFrameCount;
+
+ framesToReadThisIterationOut = (frameCount - totalFramesReadOut);
+ framesToReadThisIterationIn = framesToReadThisIterationOut;
+ if (framesToReadThisIterationIn > intermediaryBufferCap) {
+ framesToReadThisIterationIn = intermediaryBufferCap;
+ }
+
+ ma_data_converter_get_required_input_frame_count(&pDevice->playback.converter, framesToReadThisIterationOut, &requiredInputFrameCount);
+ if (framesToReadThisIterationIn > requiredInputFrameCount) {
+ framesToReadThisIterationIn = requiredInputFrameCount;
+ }
+
+ if (framesToReadThisIterationIn > 0) {
+ ma_device__handle_data_callback(pDevice, pIntermediaryBuffer, NULL, (ma_uint32)framesToReadThisIterationIn);
+ }
+
+ /*
+ At this point we have our decoded data in input format and now we need to convert to output format. Note that even if we didn't read any
+ input frames, we still want to try processing frames because there may some output frames generated from cached input data.
+ */
+ framesReadThisIterationIn = framesToReadThisIterationIn;
+ framesReadThisIterationOut = framesToReadThisIterationOut;
+ result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, pIntermediaryBuffer, &framesReadThisIterationIn, pRunningFramesOut, &framesReadThisIterationOut);
+ if (result != MA_SUCCESS) {
+ break;
+ }
+
+ totalFramesReadOut += framesReadThisIterationOut;
+ pRunningFramesOut = ma_offset_ptr(pRunningFramesOut, framesReadThisIterationOut * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels));
+
+ if (framesReadThisIterationIn == 0 && framesReadThisIterationOut == 0) {
+ break; /* We're done. */
+ }
+ }
+ }
+ }
+}
+
+/* A helper for sending sample data to the client. */
+static void ma_device__send_frames_to_client(ma_device* pDevice, ma_uint32 frameCountInDeviceFormat, const void* pFramesInDeviceFormat)
+{
+ MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(frameCountInDeviceFormat > 0);
+ MA_ASSERT(pFramesInDeviceFormat != NULL);
+
+ if (pDevice->capture.converter.isPassthrough) {
+ ma_device__handle_data_callback(pDevice, NULL, pFramesInDeviceFormat, frameCountInDeviceFormat);
+ } else {
+ ma_result result;
+ ma_uint8 pFramesInClientFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
+ ma_uint64 framesInClientFormatCap = sizeof(pFramesInClientFormat) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels);
+ ma_uint64 totalDeviceFramesProcessed = 0;
+ ma_uint64 totalClientFramesProcessed = 0;
+ const void* pRunningFramesInDeviceFormat = pFramesInDeviceFormat;
+
+ /* We just keep going until we've exhaused all of our input frames and cannot generate any more output frames. */
+ for (;;) {
+ ma_uint64 deviceFramesProcessedThisIteration;
+ ma_uint64 clientFramesProcessedThisIteration;
+
+ deviceFramesProcessedThisIteration = (frameCountInDeviceFormat - totalDeviceFramesProcessed);
+ clientFramesProcessedThisIteration = framesInClientFormatCap;
+
+ result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningFramesInDeviceFormat, &deviceFramesProcessedThisIteration, pFramesInClientFormat, &clientFramesProcessedThisIteration);
+ if (result != MA_SUCCESS) {
+ break;
+ }
+
+ if (clientFramesProcessedThisIteration > 0) {
+ ma_device__handle_data_callback(pDevice, NULL, pFramesInClientFormat, (ma_uint32)clientFramesProcessedThisIteration); /* Safe cast. */
+ }
+
+ pRunningFramesInDeviceFormat = ma_offset_ptr(pRunningFramesInDeviceFormat, deviceFramesProcessedThisIteration * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels));
+ totalDeviceFramesProcessed += deviceFramesProcessedThisIteration;
+ totalClientFramesProcessed += clientFramesProcessedThisIteration;
+
+ if (deviceFramesProcessedThisIteration == 0 && clientFramesProcessedThisIteration == 0) {
+ break; /* We're done. */
+ }
+ }
+ }
+}
+
+static ma_result ma_device__handle_duplex_callback_capture(ma_device* pDevice, ma_uint32 frameCountInDeviceFormat, const void* pFramesInDeviceFormat, ma_pcm_rb* pRB)
+{
+ ma_result result;
+ ma_uint32 totalDeviceFramesProcessed = 0;
+ const void* pRunningFramesInDeviceFormat = pFramesInDeviceFormat;
+
+ MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(frameCountInDeviceFormat > 0);
+ MA_ASSERT(pFramesInDeviceFormat != NULL);
+ MA_ASSERT(pRB != NULL);
+
+ /* Write to the ring buffer. The ring buffer is in the client format which means we need to convert. */
+ for (;;) {
+ ma_uint32 framesToProcessInDeviceFormat = (frameCountInDeviceFormat - totalDeviceFramesProcessed);
+ ma_uint32 framesToProcessInClientFormat = MA_DATA_CONVERTER_STACK_BUFFER_SIZE / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels);
+ ma_uint64 framesProcessedInDeviceFormat;
+ ma_uint64 framesProcessedInClientFormat;
+ void* pFramesInClientFormat;
+
+ result = ma_pcm_rb_acquire_write(pRB, &framesToProcessInClientFormat, &pFramesInClientFormat);
+ if (result != MA_SUCCESS) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "Failed to acquire capture PCM frames from ring buffer.");
+ break;
+ }
+
+ if (framesToProcessInClientFormat == 0) {
+ if (ma_pcm_rb_pointer_distance(pRB) == (ma_int32)ma_pcm_rb_get_subbuffer_size(pRB)) {
+ break; /* Overrun. Not enough room in the ring buffer for input frame. Excess frames are dropped. */
+ }
+ }
+
+ /* Convert. */
+ framesProcessedInDeviceFormat = framesToProcessInDeviceFormat;
+ framesProcessedInClientFormat = framesToProcessInClientFormat;
+ result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningFramesInDeviceFormat, &framesProcessedInDeviceFormat, pFramesInClientFormat, &framesProcessedInClientFormat);
+ if (result != MA_SUCCESS) {
+ break;
+ }
+
+ result = ma_pcm_rb_commit_write(pRB, (ma_uint32)framesProcessedInClientFormat); /* Safe cast. */
+ if (result != MA_SUCCESS) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "Failed to commit capture PCM frames to ring buffer.");
+ break;
+ }
+
+ pRunningFramesInDeviceFormat = ma_offset_ptr(pRunningFramesInDeviceFormat, framesProcessedInDeviceFormat * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels));
+ totalDeviceFramesProcessed += (ma_uint32)framesProcessedInDeviceFormat; /* Safe cast. */
+
+ /* We're done when we're unable to process any client nor device frames. */
+ if (framesProcessedInClientFormat == 0 && framesProcessedInDeviceFormat == 0) {
+ break; /* Done. */
+ }
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device__handle_duplex_callback_playback(ma_device* pDevice, ma_uint32 frameCount, void* pFramesInInternalFormat, ma_pcm_rb* pRB)
+{
+ ma_result result;
+ ma_uint8 silentInputFrames[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
+ ma_uint32 totalFramesReadOut = 0;
+
+ MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(frameCount > 0);
+ MA_ASSERT(pFramesInInternalFormat != NULL);
+ MA_ASSERT(pRB != NULL);
+ MA_ASSERT(pDevice->playback.pInputCache != NULL);
+
+ /*
+ Sitting in the ring buffer should be captured data from the capture callback in external format. If there's not enough data in there for
+ the whole frameCount frames we just use silence instead for the input data.
+ */
+ MA_ZERO_MEMORY(silentInputFrames, sizeof(silentInputFrames));
+
+ while (totalFramesReadOut < frameCount && ma_device_is_started(pDevice)) {
+ /*
+ We should have a buffer allocated on the heap. Any playback frames still sitting in there
+ need to be sent to the internal device before we process any more data from the client.
+ */
+ if (pDevice->playback.inputCacheRemaining > 0) {
+ ma_uint64 framesConvertedIn = pDevice->playback.inputCacheRemaining;
+ ma_uint64 framesConvertedOut = (frameCount - totalFramesReadOut);
+ ma_data_converter_process_pcm_frames(&pDevice->playback.converter, ma_offset_pcm_frames_ptr(pDevice->playback.pInputCache, pDevice->playback.inputCacheConsumed, pDevice->playback.format, pDevice->playback.channels), &framesConvertedIn, pFramesInInternalFormat, &framesConvertedOut);
+
+ pDevice->playback.inputCacheConsumed += framesConvertedIn;
+ pDevice->playback.inputCacheRemaining -= framesConvertedIn;
+
+ totalFramesReadOut += (ma_uint32)framesConvertedOut; /* Safe cast. */
+ pFramesInInternalFormat = ma_offset_ptr(pFramesInInternalFormat, framesConvertedOut * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels));
+ }
+
+ /* If there's no more data in the cache we'll need to fill it with some. */
+ if (totalFramesReadOut < frameCount && pDevice->playback.inputCacheRemaining == 0) {
+ ma_uint32 inputFrameCount;
+ void* pInputFrames;
+
+ inputFrameCount = (ma_uint32)pDevice->playback.inputCacheCap;
+ result = ma_pcm_rb_acquire_read(pRB, &inputFrameCount, &pInputFrames);
+ if (result == MA_SUCCESS) {
+ if (inputFrameCount > 0) {
+ ma_device__handle_data_callback(pDevice, pDevice->playback.pInputCache, pInputFrames, inputFrameCount);
+ } else {
+ if (ma_pcm_rb_pointer_distance(pRB) == 0) {
+ break; /* Underrun. */
+ }
+ }
+ } else {
+ /* No capture data available. Feed in silence. */
+ inputFrameCount = (ma_uint32)ma_min(pDevice->playback.inputCacheCap, sizeof(silentInputFrames) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels));
+ ma_device__handle_data_callback(pDevice, pDevice->playback.pInputCache, silentInputFrames, inputFrameCount);
+ }
+
+ pDevice->playback.inputCacheConsumed = 0;
+ pDevice->playback.inputCacheRemaining = inputFrameCount;
+
+ result = ma_pcm_rb_commit_read(pRB, inputFrameCount);
+ if (result != MA_SUCCESS) {
+ return result; /* Should never happen. */
+ }
+ }
+ }
+
+ return MA_SUCCESS;
+}
+
+/* A helper for changing the state of the device. */
+static MA_INLINE void ma_device__set_state(ma_device* pDevice, ma_device_state newState)
+{
+ c89atomic_exchange_i32((ma_int32*)&pDevice->state, (ma_int32)newState);
+}
+
+
+#ifdef MA_WIN32
+ GUID MA_GUID_KSDATAFORMAT_SUBTYPE_PCM = {0x00000001, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}};
+ GUID MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT = {0x00000003, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}};
+ /*GUID MA_GUID_KSDATAFORMAT_SUBTYPE_ALAW = {0x00000006, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}};*/
+ /*GUID MA_GUID_KSDATAFORMAT_SUBTYPE_MULAW = {0x00000007, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}};*/
+#endif
+
+
+
+MA_API ma_uint32 ma_get_format_priority_index(ma_format format) /* Lower = better. */
+{
+ ma_uint32 i;
+ for (i = 0; i < ma_countof(g_maFormatPriorities); ++i) {
+ if (g_maFormatPriorities[i] == format) {
+ return i;
+ }
+ }
+
+ /* Getting here means the format could not be found or is equal to ma_format_unknown. */
+ return (ma_uint32)-1;
+}
+
+static ma_result ma_device__post_init_setup(ma_device* pDevice, ma_device_type deviceType);
+
+static ma_bool32 ma_device_descriptor_is_valid(const ma_device_descriptor* pDeviceDescriptor)
+{
+ if (pDeviceDescriptor == NULL) {
+ return MA_FALSE;
+ }
+
+ if (pDeviceDescriptor->format == ma_format_unknown) {
+ return MA_FALSE;
+ }
+
+ if (pDeviceDescriptor->channels == 0 || pDeviceDescriptor->channels > MA_MAX_CHANNELS) {
+ return MA_FALSE;
+ }
+
+ if (pDeviceDescriptor->sampleRate == 0) {
+ return MA_FALSE;
+ }
+
+ return MA_TRUE;
+}
+
+
+static ma_result ma_device_audio_thread__default_read_write(ma_device* pDevice)
+{
+ ma_result result = MA_SUCCESS;
+ ma_bool32 exitLoop = MA_FALSE;
+ ma_uint8 capturedDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
+ ma_uint8 playbackDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
+ ma_uint32 capturedDeviceDataCapInFrames = 0;
+ ma_uint32 playbackDeviceDataCapInFrames = 0;
+
+ MA_ASSERT(pDevice != NULL);
+
+ /* Just some quick validation on the device type and the available callbacks. */
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) {
+ if (pDevice->pContext->callbacks.onDeviceRead == NULL) {
+ return MA_NOT_IMPLEMENTED;
+ }
+
+ capturedDeviceDataCapInFrames = sizeof(capturedDeviceData) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
+ }
+
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ if (pDevice->pContext->callbacks.onDeviceWrite == NULL) {
+ return MA_NOT_IMPLEMENTED;
+ }
+
+ playbackDeviceDataCapInFrames = sizeof(playbackDeviceData) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
+ }
+
+ /* NOTE: The device was started outside of this function, in the worker thread. */
+
+ while (ma_device_get_state(pDevice) == ma_device_state_started && !exitLoop) {
+ switch (pDevice->type) {
+ case ma_device_type_duplex:
+ {
+ /* The process is: onDeviceRead() -> convert -> callback -> convert -> onDeviceWrite() */
+ ma_uint32 totalCapturedDeviceFramesProcessed = 0;
+ ma_uint32 capturedDevicePeriodSizeInFrames = ma_min(pDevice->capture.internalPeriodSizeInFrames, pDevice->playback.internalPeriodSizeInFrames);
+
+ while (totalCapturedDeviceFramesProcessed < capturedDevicePeriodSizeInFrames) {
+ ma_uint32 capturedDeviceFramesRemaining;
+ ma_uint32 capturedDeviceFramesProcessed;
+ ma_uint32 capturedDeviceFramesToProcess;
+ ma_uint32 capturedDeviceFramesToTryProcessing = capturedDevicePeriodSizeInFrames - totalCapturedDeviceFramesProcessed;
+ if (capturedDeviceFramesToTryProcessing > capturedDeviceDataCapInFrames) {
+ capturedDeviceFramesToTryProcessing = capturedDeviceDataCapInFrames;
+ }
+
+ result = pDevice->pContext->callbacks.onDeviceRead(pDevice, capturedDeviceData, capturedDeviceFramesToTryProcessing, &capturedDeviceFramesToProcess);
+ if (result != MA_SUCCESS) {
+ exitLoop = MA_TRUE;
+ break;
+ }
+
+ capturedDeviceFramesRemaining = capturedDeviceFramesToProcess;
+ capturedDeviceFramesProcessed = 0;
+
+ /* At this point we have our captured data in device format and we now need to convert it to client format. */
+ for (;;) {
+ ma_uint8 capturedClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
+ ma_uint8 playbackClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
+ ma_uint32 capturedClientDataCapInFrames = sizeof(capturedClientData) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels);
+ ma_uint32 playbackClientDataCapInFrames = sizeof(playbackClientData) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
+ ma_uint64 capturedClientFramesToProcessThisIteration = ma_min(capturedClientDataCapInFrames, playbackClientDataCapInFrames);
+ ma_uint64 capturedDeviceFramesToProcessThisIteration = capturedDeviceFramesRemaining;
+ ma_uint8* pRunningCapturedDeviceFrames = ma_offset_ptr(capturedDeviceData, capturedDeviceFramesProcessed * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels));
+
+ /* Convert capture data from device format to client format. */
+ result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningCapturedDeviceFrames, &capturedDeviceFramesToProcessThisIteration, capturedClientData, &capturedClientFramesToProcessThisIteration);
+ if (result != MA_SUCCESS) {
+ break;
+ }
+
+ /*
+ If we weren't able to generate any output frames it must mean we've exhaused all of our input. The only time this would not be the case is if capturedClientData was too small
+ which should never be the case when it's of the size MA_DATA_CONVERTER_STACK_BUFFER_SIZE.
+ */
+ if (capturedClientFramesToProcessThisIteration == 0) {
+ break;
+ }
+
+ ma_device__handle_data_callback(pDevice, playbackClientData, capturedClientData, (ma_uint32)capturedClientFramesToProcessThisIteration); /* Safe cast .*/
+
+ capturedDeviceFramesProcessed += (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */
+ capturedDeviceFramesRemaining -= (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */
+
+ /* At this point the playbackClientData buffer should be holding data that needs to be written to the device. */
+ for (;;) {
+ ma_uint64 convertedClientFrameCount = capturedClientFramesToProcessThisIteration;
+ ma_uint64 convertedDeviceFrameCount = playbackDeviceDataCapInFrames;
+ result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, playbackClientData, &convertedClientFrameCount, playbackDeviceData, &convertedDeviceFrameCount);
+ if (result != MA_SUCCESS) {
+ break;
+ }
+
+ result = pDevice->pContext->callbacks.onDeviceWrite(pDevice, playbackDeviceData, (ma_uint32)convertedDeviceFrameCount, NULL); /* Safe cast. */
+ if (result != MA_SUCCESS) {
+ exitLoop = MA_TRUE;
+ break;
+ }
+
+ capturedClientFramesToProcessThisIteration -= (ma_uint32)convertedClientFrameCount; /* Safe cast. */
+ if (capturedClientFramesToProcessThisIteration == 0) {
+ break;
+ }
+ }
+
+ /* In case an error happened from ma_device_write__null()... */
+ if (result != MA_SUCCESS) {
+ exitLoop = MA_TRUE;
+ break;
+ }
+ }
+
+ /* Make sure we don't get stuck in the inner loop. */
+ if (capturedDeviceFramesProcessed == 0) {
+ break;
+ }
+
+ totalCapturedDeviceFramesProcessed += capturedDeviceFramesProcessed;
+ }
+ } break;
+
+ case ma_device_type_capture:
+ case ma_device_type_loopback:
+ {
+ ma_uint32 periodSizeInFrames = pDevice->capture.internalPeriodSizeInFrames;
+ ma_uint32 framesReadThisPeriod = 0;
+ while (framesReadThisPeriod < periodSizeInFrames) {
+ ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesReadThisPeriod;
+ ma_uint32 framesProcessed;
+ ma_uint32 framesToReadThisIteration = framesRemainingInPeriod;
+ if (framesToReadThisIteration > capturedDeviceDataCapInFrames) {
+ framesToReadThisIteration = capturedDeviceDataCapInFrames;
+ }
+
+ result = pDevice->pContext->callbacks.onDeviceRead(pDevice, capturedDeviceData, framesToReadThisIteration, &framesProcessed);
+ if (result != MA_SUCCESS) {
+ exitLoop = MA_TRUE;
+ break;
+ }
+
+ /* Make sure we don't get stuck in the inner loop. */
+ if (framesProcessed == 0) {
+ break;
+ }
+
+ ma_device__send_frames_to_client(pDevice, framesProcessed, capturedDeviceData);
+
+ framesReadThisPeriod += framesProcessed;
+ }
+ } break;
+
+ case ma_device_type_playback:
+ {
+ /* We write in chunks of the period size, but use a stack allocated buffer for the intermediary. */
+ ma_uint32 periodSizeInFrames = pDevice->playback.internalPeriodSizeInFrames;
+ ma_uint32 framesWrittenThisPeriod = 0;
+ while (framesWrittenThisPeriod < periodSizeInFrames) {
+ ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesWrittenThisPeriod;
+ ma_uint32 framesProcessed;
+ ma_uint32 framesToWriteThisIteration = framesRemainingInPeriod;
+ if (framesToWriteThisIteration > playbackDeviceDataCapInFrames) {
+ framesToWriteThisIteration = playbackDeviceDataCapInFrames;
+ }
+
+ ma_device__read_frames_from_client(pDevice, framesToWriteThisIteration, playbackDeviceData);
+
+ result = pDevice->pContext->callbacks.onDeviceWrite(pDevice, playbackDeviceData, framesToWriteThisIteration, &framesProcessed);
+ if (result != MA_SUCCESS) {
+ exitLoop = MA_TRUE;
+ break;
+ }
+
+ /* Make sure we don't get stuck in the inner loop. */
+ if (framesProcessed == 0) {
+ break;
+ }
+
+ framesWrittenThisPeriod += framesProcessed;
+ }
+ } break;
+
+ /* Should never get here. */
+ default: break;
+ }
+ }
+
+ return result;
+}
+
+
+
+/*******************************************************************************
+
+Null Backend
+
+*******************************************************************************/
+#ifdef MA_HAS_NULL
+
+#define MA_DEVICE_OP_NONE__NULL 0
+#define MA_DEVICE_OP_START__NULL 1
+#define MA_DEVICE_OP_SUSPEND__NULL 2
+#define MA_DEVICE_OP_KILL__NULL 3
+
+static ma_thread_result MA_THREADCALL ma_device_thread__null(void* pData)
+{
+ ma_device* pDevice = (ma_device*)pData;
+ MA_ASSERT(pDevice != NULL);
+
+ for (;;) { /* Keep the thread alive until the device is uninitialized. */
+ ma_uint32 operation;
+
+ /* Wait for an operation to be requested. */
+ ma_event_wait(&pDevice->null_device.operationEvent);
+
+ /* At this point an event should have been triggered. */
+ operation = pDevice->null_device.operation;
+
+ /* Starting the device needs to put the thread into a loop. */
+ if (operation == MA_DEVICE_OP_START__NULL) {
+ /* Reset the timer just in case. */
+ ma_timer_init(&pDevice->null_device.timer);
+
+ /* Getting here means a suspend or kill operation has been requested. */
+ pDevice->null_device.operationResult = MA_SUCCESS;
+ ma_event_signal(&pDevice->null_device.operationCompletionEvent);
+ ma_semaphore_release(&pDevice->null_device.operationSemaphore);
+ continue;
+ }
+
+ /* Suspending the device means we need to stop the timer and just continue the loop. */
+ if (operation == MA_DEVICE_OP_SUSPEND__NULL) {
+ /* We need to add the current run time to the prior run time, then reset the timer. */
+ pDevice->null_device.priorRunTime += ma_timer_get_time_in_seconds(&pDevice->null_device.timer);
+ ma_timer_init(&pDevice->null_device.timer);
+
+ /* We're done. */
+ pDevice->null_device.operationResult = MA_SUCCESS;
+ ma_event_signal(&pDevice->null_device.operationCompletionEvent);
+ ma_semaphore_release(&pDevice->null_device.operationSemaphore);
+ continue;
+ }
+
+ /* Killing the device means we need to get out of this loop so that this thread can terminate. */
+ if (operation == MA_DEVICE_OP_KILL__NULL) {
+ pDevice->null_device.operationResult = MA_SUCCESS;
+ ma_event_signal(&pDevice->null_device.operationCompletionEvent);
+ ma_semaphore_release(&pDevice->null_device.operationSemaphore);
+ break;
+ }
+
+ /* Getting a signal on a "none" operation probably means an error. Return invalid operation. */
+ if (operation == MA_DEVICE_OP_NONE__NULL) {
+ MA_ASSERT(MA_FALSE); /* <-- Trigger this in debug mode to ensure developers are aware they're doing something wrong (or there's a bug in a miniaudio). */
+ pDevice->null_device.operationResult = MA_INVALID_OPERATION;
+ ma_event_signal(&pDevice->null_device.operationCompletionEvent);
+ ma_semaphore_release(&pDevice->null_device.operationSemaphore);
+ continue; /* Continue the loop. Don't terminate. */
+ }
+ }
+
+ return (ma_thread_result)0;
+}
+
+static ma_result ma_device_do_operation__null(ma_device* pDevice, ma_uint32 operation)
+{
+ ma_result result;
+
+ /*
+ TODO: Need to review this and consider just using mutual exclusion. I think the original motivation
+ for this was to just post the event to a queue and return immediately, but that has since changed
+ and now this function is synchronous. I think this can be simplified to just use a mutex.
+ */
+
+ /*
+ The first thing to do is wait for an operation slot to become available. We only have a single slot for this, but we could extend this later
+ to support queing of operations.
+ */
+ result = ma_semaphore_wait(&pDevice->null_device.operationSemaphore);
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to wait for the event. */
+ }
+
+ /*
+ When we get here it means the background thread is not referencing the operation code and it can be changed. After changing this we need to
+ signal an event to the worker thread to let it know that it can start work.
+ */
+ pDevice->null_device.operation = operation;
+
+ /* Once the operation code has been set, the worker thread can start work. */
+ if (ma_event_signal(&pDevice->null_device.operationEvent) != MA_SUCCESS) {
+ return MA_ERROR;
+ }
+
+ /* We want everything to be synchronous so we're going to wait for the worker thread to complete it's operation. */
+ if (ma_event_wait(&pDevice->null_device.operationCompletionEvent) != MA_SUCCESS) {
+ return MA_ERROR;
+ }
+
+ return pDevice->null_device.operationResult;
+}
+
+static ma_uint64 ma_device_get_total_run_time_in_frames__null(ma_device* pDevice)
+{
+ ma_uint32 internalSampleRate;
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ internalSampleRate = pDevice->capture.internalSampleRate;
+ } else {
+ internalSampleRate = pDevice->playback.internalSampleRate;
+ }
+
+ return (ma_uint64)((pDevice->null_device.priorRunTime + ma_timer_get_time_in_seconds(&pDevice->null_device.timer)) * internalSampleRate);
+}
+
+static ma_result ma_context_enumerate_devices__null(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+{
+ ma_bool32 cbResult = MA_TRUE;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(callback != NULL);
+
+ /* Playback. */
+ if (cbResult) {
+ ma_device_info deviceInfo;
+ MA_ZERO_OBJECT(&deviceInfo);
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), "NULL Playback Device", (size_t)-1);
+ deviceInfo.isDefault = MA_TRUE; /* Only one playback and capture device for the null backend, so might as well mark as default. */
+ cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
+ }
+
+ /* Capture. */
+ if (cbResult) {
+ ma_device_info deviceInfo;
+ MA_ZERO_OBJECT(&deviceInfo);
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), "NULL Capture Device", (size_t)-1);
+ deviceInfo.isDefault = MA_TRUE; /* Only one playback and capture device for the null backend, so might as well mark as default. */
+ cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
+ }
+
+ (void)cbResult; /* Silence a static analysis warning. */
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_get_device_info__null(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+{
+ MA_ASSERT(pContext != NULL);
+
+ if (pDeviceID != NULL && pDeviceID->nullbackend != 0) {
+ return MA_NO_DEVICE; /* Don't know the device. */
+ }
+
+ /* Name / Description */
+ if (deviceType == ma_device_type_playback) {
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), "NULL Playback Device", (size_t)-1);
+ } else {
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), "NULL Capture Device", (size_t)-1);
+ }
+
+ pDeviceInfo->isDefault = MA_TRUE; /* Only one playback and capture device for the null backend, so might as well mark as default. */
+
+ /* Support everything on the null backend. */
+ pDeviceInfo->nativeDataFormats[0].format = ma_format_unknown;
+ pDeviceInfo->nativeDataFormats[0].channels = 0;
+ pDeviceInfo->nativeDataFormats[0].sampleRate = 0;
+ pDeviceInfo->nativeDataFormats[0].flags = 0;
+ pDeviceInfo->nativeDataFormatCount = 1;
+
+ (void)pContext;
+ return MA_SUCCESS;
+}
+
+
+static ma_result ma_device_uninit__null(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
+
+ /* Keep it clean and wait for the device thread to finish before returning. */
+ ma_device_do_operation__null(pDevice, MA_DEVICE_OP_KILL__NULL);
+
+ /* Wait for the thread to finish before continuing. */
+ ma_thread_wait(&pDevice->null_device.deviceThread);
+
+ /* At this point the loop in the device thread is as good as terminated so we can uninitialize our events. */
+ ma_semaphore_uninit(&pDevice->null_device.operationSemaphore);
+ ma_event_uninit(&pDevice->null_device.operationCompletionEvent);
+ ma_event_uninit(&pDevice->null_device.operationEvent);
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_init__null(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+{
+ ma_result result;
+
+ MA_ASSERT(pDevice != NULL);
+
+ MA_ZERO_OBJECT(&pDevice->null_device);
+
+ if (pConfig->deviceType == ma_device_type_loopback) {
+ return MA_DEVICE_TYPE_NOT_SUPPORTED;
+ }
+
+ /* The null backend supports everything exactly as we specify it. */
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
+ pDescriptorCapture->format = (pDescriptorCapture->format != ma_format_unknown) ? pDescriptorCapture->format : MA_DEFAULT_FORMAT;
+ pDescriptorCapture->channels = (pDescriptorCapture->channels != 0) ? pDescriptorCapture->channels : MA_DEFAULT_CHANNELS;
+ pDescriptorCapture->sampleRate = (pDescriptorCapture->sampleRate != 0) ? pDescriptorCapture->sampleRate : MA_DEFAULT_SAMPLE_RATE;
+
+ if (pDescriptorCapture->channelMap[0] == MA_CHANNEL_NONE) {
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pDescriptorCapture->channelMap, ma_countof(pDescriptorCapture->channelMap), pDescriptorCapture->channels);
+ }
+
+ pDescriptorCapture->periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptorCapture, pDescriptorCapture->sampleRate, pConfig->performanceProfile);
+ }
+
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ pDescriptorPlayback->format = (pDescriptorPlayback->format != ma_format_unknown) ? pDescriptorPlayback->format : MA_DEFAULT_FORMAT;
+ pDescriptorPlayback->channels = (pDescriptorPlayback->channels != 0) ? pDescriptorPlayback->channels : MA_DEFAULT_CHANNELS;
+ pDescriptorPlayback->sampleRate = (pDescriptorPlayback->sampleRate != 0) ? pDescriptorPlayback->sampleRate : MA_DEFAULT_SAMPLE_RATE;
+
+ if (pDescriptorPlayback->channelMap[0] == MA_CHANNEL_NONE) {
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pDescriptorPlayback->channelMap, ma_countof(pDescriptorCapture->channelMap), pDescriptorPlayback->channels);
+ }
+
+ pDescriptorPlayback->periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptorPlayback, pDescriptorPlayback->sampleRate, pConfig->performanceProfile);
+ }
+
+ /*
+ In order to get timing right, we need to create a thread that does nothing but keeps track of the timer. This timer is started when the
+ first period is "written" to it, and then stopped in ma_device_stop__null().
+ */
+ result = ma_event_init(&pDevice->null_device.operationEvent);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ result = ma_event_init(&pDevice->null_device.operationCompletionEvent);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ result = ma_semaphore_init(1, &pDevice->null_device.operationSemaphore); /* <-- It's important that the initial value is set to 1. */
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ result = ma_thread_create(&pDevice->null_device.deviceThread, pDevice->pContext->threadPriority, 0, ma_device_thread__null, pDevice, &pDevice->pContext->allocationCallbacks);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_start__null(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
+
+ ma_device_do_operation__null(pDevice, MA_DEVICE_OP_START__NULL);
+
+ c89atomic_exchange_32(&pDevice->null_device.isStarted, MA_TRUE);
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_stop__null(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
+
+ ma_device_do_operation__null(pDevice, MA_DEVICE_OP_SUSPEND__NULL);
+
+ c89atomic_exchange_32(&pDevice->null_device.isStarted, MA_FALSE);
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_write__null(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten)
+{
+ ma_result result = MA_SUCCESS;
+ ma_uint32 totalPCMFramesProcessed;
+ ma_bool32 wasStartedOnEntry;
+
+ if (pFramesWritten != NULL) {
+ *pFramesWritten = 0;
+ }
+
+ wasStartedOnEntry = c89atomic_load_32(&pDevice->null_device.isStarted);
+
+ /* Keep going until everything has been read. */
+ totalPCMFramesProcessed = 0;
+ while (totalPCMFramesProcessed < frameCount) {
+ ma_uint64 targetFrame;
+
+ /* If there are any frames remaining in the current period, consume those first. */
+ if (pDevice->null_device.currentPeriodFramesRemainingPlayback > 0) {
+ ma_uint32 framesRemaining = (frameCount - totalPCMFramesProcessed);
+ ma_uint32 framesToProcess = pDevice->null_device.currentPeriodFramesRemainingPlayback;
+ if (framesToProcess > framesRemaining) {
+ framesToProcess = framesRemaining;
+ }
+
+ /* We don't actually do anything with pPCMFrames, so just mark it as unused to prevent a warning. */
+ (void)pPCMFrames;
+
+ pDevice->null_device.currentPeriodFramesRemainingPlayback -= framesToProcess;
+ totalPCMFramesProcessed += framesToProcess;
+ }
+
+ /* If we've consumed the current period we'll need to mark it as such an ensure the device is started if it's not already. */
+ if (pDevice->null_device.currentPeriodFramesRemainingPlayback == 0) {
+ pDevice->null_device.currentPeriodFramesRemainingPlayback = 0;
+
+ if (!c89atomic_load_32(&pDevice->null_device.isStarted) && !wasStartedOnEntry) {
+ result = ma_device_start__null(pDevice);
+ if (result != MA_SUCCESS) {
+ break;
+ }
+ }
+ }
+
+ /* If we've consumed the whole buffer we can return now. */
+ MA_ASSERT(totalPCMFramesProcessed <= frameCount);
+ if (totalPCMFramesProcessed == frameCount) {
+ break;
+ }
+
+ /* Getting here means we've still got more frames to consume, we but need to wait for it to become available. */
+ targetFrame = pDevice->null_device.lastProcessedFramePlayback;
+ for (;;) {
+ ma_uint64 currentFrame;
+
+ /* Stop waiting if the device has been stopped. */
+ if (!c89atomic_load_32(&pDevice->null_device.isStarted)) {
+ break;
+ }
+
+ currentFrame = ma_device_get_total_run_time_in_frames__null(pDevice);
+ if (currentFrame >= targetFrame) {
+ break;
+ }
+
+ /* Getting here means we haven't yet reached the target sample, so continue waiting. */
+ ma_sleep(10);
+ }
+
+ pDevice->null_device.lastProcessedFramePlayback += pDevice->playback.internalPeriodSizeInFrames;
+ pDevice->null_device.currentPeriodFramesRemainingPlayback = pDevice->playback.internalPeriodSizeInFrames;
+ }
+
+ if (pFramesWritten != NULL) {
+ *pFramesWritten = totalPCMFramesProcessed;
+ }
+
+ return result;
+}
+
+static ma_result ma_device_read__null(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead)
+{
+ ma_result result = MA_SUCCESS;
+ ma_uint32 totalPCMFramesProcessed;
+
+ if (pFramesRead != NULL) {
+ *pFramesRead = 0;
+ }
+
+ /* Keep going until everything has been read. */
+ totalPCMFramesProcessed = 0;
+ while (totalPCMFramesProcessed < frameCount) {
+ ma_uint64 targetFrame;
+
+ /* If there are any frames remaining in the current period, consume those first. */
+ if (pDevice->null_device.currentPeriodFramesRemainingCapture > 0) {
+ ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
+ ma_uint32 framesRemaining = (frameCount - totalPCMFramesProcessed);
+ ma_uint32 framesToProcess = pDevice->null_device.currentPeriodFramesRemainingCapture;
+ if (framesToProcess > framesRemaining) {
+ framesToProcess = framesRemaining;
+ }
+
+ /* We need to ensure the output buffer is zeroed. */
+ MA_ZERO_MEMORY(ma_offset_ptr(pPCMFrames, totalPCMFramesProcessed*bpf), framesToProcess*bpf);
+
+ pDevice->null_device.currentPeriodFramesRemainingCapture -= framesToProcess;
+ totalPCMFramesProcessed += framesToProcess;
+ }
+
+ /* If we've consumed the current period we'll need to mark it as such an ensure the device is started if it's not already. */
+ if (pDevice->null_device.currentPeriodFramesRemainingCapture == 0) {
+ pDevice->null_device.currentPeriodFramesRemainingCapture = 0;
+ }
+
+ /* If we've consumed the whole buffer we can return now. */
+ MA_ASSERT(totalPCMFramesProcessed <= frameCount);
+ if (totalPCMFramesProcessed == frameCount) {
+ break;
+ }
+
+ /* Getting here means we've still got more frames to consume, we but need to wait for it to become available. */
+ targetFrame = pDevice->null_device.lastProcessedFrameCapture + pDevice->capture.internalPeriodSizeInFrames;
+ for (;;) {
+ ma_uint64 currentFrame;
+
+ /* Stop waiting if the device has been stopped. */
+ if (!c89atomic_load_32(&pDevice->null_device.isStarted)) {
+ break;
+ }
+
+ currentFrame = ma_device_get_total_run_time_in_frames__null(pDevice);
+ if (currentFrame >= targetFrame) {
+ break;
+ }
+
+ /* Getting here means we haven't yet reached the target sample, so continue waiting. */
+ ma_sleep(10);
+ }
+
+ pDevice->null_device.lastProcessedFrameCapture += pDevice->capture.internalPeriodSizeInFrames;
+ pDevice->null_device.currentPeriodFramesRemainingCapture = pDevice->capture.internalPeriodSizeInFrames;
+ }
+
+ if (pFramesRead != NULL) {
+ *pFramesRead = totalPCMFramesProcessed;
+ }
+
+ return result;
+}
+
+static ma_result ma_context_uninit__null(ma_context* pContext)
+{
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pContext->backend == ma_backend_null);
+
+ (void)pContext;
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_init__null(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+{
+ MA_ASSERT(pContext != NULL);
+
+ (void)pConfig;
+ (void)pContext;
+
+ pCallbacks->onContextInit = ma_context_init__null;
+ pCallbacks->onContextUninit = ma_context_uninit__null;
+ pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__null;
+ pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__null;
+ pCallbacks->onDeviceInit = ma_device_init__null;
+ pCallbacks->onDeviceUninit = ma_device_uninit__null;
+ pCallbacks->onDeviceStart = ma_device_start__null;
+ pCallbacks->onDeviceStop = ma_device_stop__null;
+ pCallbacks->onDeviceRead = ma_device_read__null;
+ pCallbacks->onDeviceWrite = ma_device_write__null;
+ pCallbacks->onDeviceDataLoop = NULL; /* Our backend is asynchronous with a blocking read-write API which means we can get miniaudio to deal with the audio thread. */
+
+ /* The null backend always works. */
+ return MA_SUCCESS;
+}
+#endif
+
+
+
+/*******************************************************************************
+
+WIN32 COMMON
+
+*******************************************************************************/
+#if defined(MA_WIN32)
+#if defined(MA_WIN32_DESKTOP)
+ #define ma_CoInitializeEx(pContext, pvReserved, dwCoInit) ((MA_PFN_CoInitializeEx)pContext->win32.CoInitializeEx)(pvReserved, dwCoInit)
+ #define ma_CoUninitialize(pContext) ((MA_PFN_CoUninitialize)pContext->win32.CoUninitialize)()
+ #define ma_CoCreateInstance(pContext, rclsid, pUnkOuter, dwClsContext, riid, ppv) ((MA_PFN_CoCreateInstance)pContext->win32.CoCreateInstance)(rclsid, pUnkOuter, dwClsContext, riid, ppv)
+ #define ma_CoTaskMemFree(pContext, pv) ((MA_PFN_CoTaskMemFree)pContext->win32.CoTaskMemFree)(pv)
+ #define ma_PropVariantClear(pContext, pvar) ((MA_PFN_PropVariantClear)pContext->win32.PropVariantClear)(pvar)
+#else
+ #define ma_CoInitializeEx(pContext, pvReserved, dwCoInit) CoInitializeEx(pvReserved, dwCoInit)
+ #define ma_CoUninitialize(pContext) CoUninitialize()
+ #define ma_CoCreateInstance(pContext, rclsid, pUnkOuter, dwClsContext, riid, ppv) CoCreateInstance(rclsid, pUnkOuter, dwClsContext, riid, ppv)
+ #define ma_CoTaskMemFree(pContext, pv) CoTaskMemFree(pv)
+ #define ma_PropVariantClear(pContext, pvar) PropVariantClear(pvar)
+#endif
+
+#if !defined(MAXULONG_PTR) && !defined(__WATCOMC__)
+typedef size_t DWORD_PTR;
+#endif
+
+#if !defined(WAVE_FORMAT_44M08)
+#define WAVE_FORMAT_44M08 0x00000100
+#define WAVE_FORMAT_44S08 0x00000200
+#define WAVE_FORMAT_44M16 0x00000400
+#define WAVE_FORMAT_44S16 0x00000800
+#define WAVE_FORMAT_48M08 0x00001000
+#define WAVE_FORMAT_48S08 0x00002000
+#define WAVE_FORMAT_48M16 0x00004000
+#define WAVE_FORMAT_48S16 0x00008000
+#define WAVE_FORMAT_96M08 0x00010000
+#define WAVE_FORMAT_96S08 0x00020000
+#define WAVE_FORMAT_96M16 0x00040000
+#define WAVE_FORMAT_96S16 0x00080000
+#endif
+
+#ifndef SPEAKER_FRONT_LEFT
+#define SPEAKER_FRONT_LEFT 0x1
+#define SPEAKER_FRONT_RIGHT 0x2
+#define SPEAKER_FRONT_CENTER 0x4
+#define SPEAKER_LOW_FREQUENCY 0x8
+#define SPEAKER_BACK_LEFT 0x10
+#define SPEAKER_BACK_RIGHT 0x20
+#define SPEAKER_FRONT_LEFT_OF_CENTER 0x40
+#define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80
+#define SPEAKER_BACK_CENTER 0x100
+#define SPEAKER_SIDE_LEFT 0x200
+#define SPEAKER_SIDE_RIGHT 0x400
+#define SPEAKER_TOP_CENTER 0x800
+#define SPEAKER_TOP_FRONT_LEFT 0x1000
+#define SPEAKER_TOP_FRONT_CENTER 0x2000
+#define SPEAKER_TOP_FRONT_RIGHT 0x4000
+#define SPEAKER_TOP_BACK_LEFT 0x8000
+#define SPEAKER_TOP_BACK_CENTER 0x10000
+#define SPEAKER_TOP_BACK_RIGHT 0x20000
+#endif
+
+/*
+The SDK that comes with old versions of MSVC (VC6, for example) does not appear to define WAVEFORMATEXTENSIBLE. We
+define our own implementation in this case.
+*/
+#if (defined(_MSC_VER) && !defined(_WAVEFORMATEXTENSIBLE_)) || defined(__DMC__)
+typedef struct
+{
+ WAVEFORMATEX Format;
+ union
+ {
+ WORD wValidBitsPerSample;
+ WORD wSamplesPerBlock;
+ WORD wReserved;
+ } Samples;
+ DWORD dwChannelMask;
+ GUID SubFormat;
+} WAVEFORMATEXTENSIBLE;
+#endif
+
+#ifndef WAVE_FORMAT_EXTENSIBLE
+#define WAVE_FORMAT_EXTENSIBLE 0xFFFE
+#endif
+
+#ifndef WAVE_FORMAT_IEEE_FLOAT
+#define WAVE_FORMAT_IEEE_FLOAT 0x0003
+#endif
+
+/* Converts an individual Win32-style channel identifier (SPEAKER_FRONT_LEFT, etc.) to miniaudio. */
+static ma_uint8 ma_channel_id_to_ma__win32(DWORD id)
+{
+ switch (id)
+ {
+ case SPEAKER_FRONT_LEFT: return MA_CHANNEL_FRONT_LEFT;
+ case SPEAKER_FRONT_RIGHT: return MA_CHANNEL_FRONT_RIGHT;
+ case SPEAKER_FRONT_CENTER: return MA_CHANNEL_FRONT_CENTER;
+ case SPEAKER_LOW_FREQUENCY: return MA_CHANNEL_LFE;
+ case SPEAKER_BACK_LEFT: return MA_CHANNEL_BACK_LEFT;
+ case SPEAKER_BACK_RIGHT: return MA_CHANNEL_BACK_RIGHT;
+ case SPEAKER_FRONT_LEFT_OF_CENTER: return MA_CHANNEL_FRONT_LEFT_CENTER;
+ case SPEAKER_FRONT_RIGHT_OF_CENTER: return MA_CHANNEL_FRONT_RIGHT_CENTER;
+ case SPEAKER_BACK_CENTER: return MA_CHANNEL_BACK_CENTER;
+ case SPEAKER_SIDE_LEFT: return MA_CHANNEL_SIDE_LEFT;
+ case SPEAKER_SIDE_RIGHT: return MA_CHANNEL_SIDE_RIGHT;
+ case SPEAKER_TOP_CENTER: return MA_CHANNEL_TOP_CENTER;
+ case SPEAKER_TOP_FRONT_LEFT: return MA_CHANNEL_TOP_FRONT_LEFT;
+ case SPEAKER_TOP_FRONT_CENTER: return MA_CHANNEL_TOP_FRONT_CENTER;
+ case SPEAKER_TOP_FRONT_RIGHT: return MA_CHANNEL_TOP_FRONT_RIGHT;
+ case SPEAKER_TOP_BACK_LEFT: return MA_CHANNEL_TOP_BACK_LEFT;
+ case SPEAKER_TOP_BACK_CENTER: return MA_CHANNEL_TOP_BACK_CENTER;
+ case SPEAKER_TOP_BACK_RIGHT: return MA_CHANNEL_TOP_BACK_RIGHT;
+ default: return 0;
+ }
+}
+
+/* Converts an individual miniaudio channel identifier (MA_CHANNEL_FRONT_LEFT, etc.) to Win32-style. */
+static DWORD ma_channel_id_to_win32(DWORD id)
+{
+ switch (id)
+ {
+ case MA_CHANNEL_MONO: return SPEAKER_FRONT_CENTER;
+ case MA_CHANNEL_FRONT_LEFT: return SPEAKER_FRONT_LEFT;
+ case MA_CHANNEL_FRONT_RIGHT: return SPEAKER_FRONT_RIGHT;
+ case MA_CHANNEL_FRONT_CENTER: return SPEAKER_FRONT_CENTER;
+ case MA_CHANNEL_LFE: return SPEAKER_LOW_FREQUENCY;
+ case MA_CHANNEL_BACK_LEFT: return SPEAKER_BACK_LEFT;
+ case MA_CHANNEL_BACK_RIGHT: return SPEAKER_BACK_RIGHT;
+ case MA_CHANNEL_FRONT_LEFT_CENTER: return SPEAKER_FRONT_LEFT_OF_CENTER;
+ case MA_CHANNEL_FRONT_RIGHT_CENTER: return SPEAKER_FRONT_RIGHT_OF_CENTER;
+ case MA_CHANNEL_BACK_CENTER: return SPEAKER_BACK_CENTER;
+ case MA_CHANNEL_SIDE_LEFT: return SPEAKER_SIDE_LEFT;
+ case MA_CHANNEL_SIDE_RIGHT: return SPEAKER_SIDE_RIGHT;
+ case MA_CHANNEL_TOP_CENTER: return SPEAKER_TOP_CENTER;
+ case MA_CHANNEL_TOP_FRONT_LEFT: return SPEAKER_TOP_FRONT_LEFT;
+ case MA_CHANNEL_TOP_FRONT_CENTER: return SPEAKER_TOP_FRONT_CENTER;
+ case MA_CHANNEL_TOP_FRONT_RIGHT: return SPEAKER_TOP_FRONT_RIGHT;
+ case MA_CHANNEL_TOP_BACK_LEFT: return SPEAKER_TOP_BACK_LEFT;
+ case MA_CHANNEL_TOP_BACK_CENTER: return SPEAKER_TOP_BACK_CENTER;
+ case MA_CHANNEL_TOP_BACK_RIGHT: return SPEAKER_TOP_BACK_RIGHT;
+ default: return 0;
+ }
+}
+
+/* Converts a channel mapping to a Win32-style channel mask. */
+static DWORD ma_channel_map_to_channel_mask__win32(const ma_channel* pChannelMap, ma_uint32 channels)
+{
+ DWORD dwChannelMask = 0;
+ ma_uint32 iChannel;
+
+ for (iChannel = 0; iChannel < channels; ++iChannel) {
+ dwChannelMask |= ma_channel_id_to_win32(pChannelMap[iChannel]);
+ }
+
+ return dwChannelMask;
+}
+
+/* Converts a Win32-style channel mask to a miniaudio channel map. */
+static void ma_channel_mask_to_channel_map__win32(DWORD dwChannelMask, ma_uint32 channels, ma_channel* pChannelMap)
+{
+ if (channels == 1 && dwChannelMask == 0) {
+ pChannelMap[0] = MA_CHANNEL_MONO;
+ } else if (channels == 2 && dwChannelMask == 0) {
+ pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
+ pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
+ } else {
+ if (channels == 1 && (dwChannelMask & SPEAKER_FRONT_CENTER) != 0) {
+ pChannelMap[0] = MA_CHANNEL_MONO;
+ } else {
+ /* Just iterate over each bit. */
+ ma_uint32 iChannel = 0;
+ ma_uint32 iBit;
+
+ for (iBit = 0; iBit < 32 && iChannel < channels; ++iBit) {
+ DWORD bitValue = (dwChannelMask & (1UL << iBit));
+ if (bitValue != 0) {
+ /* The bit is set. */
+ pChannelMap[iChannel] = ma_channel_id_to_ma__win32(bitValue);
+ iChannel += 1;
+ }
+ }
+ }
+ }
+}
+
+#ifdef __cplusplus
+static ma_bool32 ma_is_guid_equal(const void* a, const void* b)
+{
+ return IsEqualGUID(*(const GUID*)a, *(const GUID*)b);
+}
+#else
+#define ma_is_guid_equal(a, b) IsEqualGUID((const GUID*)a, (const GUID*)b)
+#endif
+
+static MA_INLINE ma_bool32 ma_is_guid_null(const void* guid)
+{
+ static GUID nullguid = {0x00000000, 0x0000, 0x0000, {0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00}};
+ return ma_is_guid_equal(guid, &nullguid);
+}
+
+static ma_format ma_format_from_WAVEFORMATEX(const WAVEFORMATEX* pWF)
+{
+ MA_ASSERT(pWF != NULL);
+
+ if (pWF->wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
+ const WAVEFORMATEXTENSIBLE* pWFEX = (const WAVEFORMATEXTENSIBLE*)pWF;
+ if (ma_is_guid_equal(&pWFEX->SubFormat, &MA_GUID_KSDATAFORMAT_SUBTYPE_PCM)) {
+ if (pWFEX->Samples.wValidBitsPerSample == 32) {
+ return ma_format_s32;
+ }
+ if (pWFEX->Samples.wValidBitsPerSample == 24) {
+ if (pWFEX->Format.wBitsPerSample == 32) {
+ /*return ma_format_s24_32;*/
+ }
+ if (pWFEX->Format.wBitsPerSample == 24) {
+ return ma_format_s24;
+ }
+ }
+ if (pWFEX->Samples.wValidBitsPerSample == 16) {
+ return ma_format_s16;
+ }
+ if (pWFEX->Samples.wValidBitsPerSample == 8) {
+ return ma_format_u8;
+ }
+ }
+ if (ma_is_guid_equal(&pWFEX->SubFormat, &MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)) {
+ if (pWFEX->Samples.wValidBitsPerSample == 32) {
+ return ma_format_f32;
+ }
+ /*
+ if (pWFEX->Samples.wValidBitsPerSample == 64) {
+ return ma_format_f64;
+ }
+ */
+ }
+ } else {
+ if (pWF->wFormatTag == WAVE_FORMAT_PCM) {
+ if (pWF->wBitsPerSample == 32) {
+ return ma_format_s32;
+ }
+ if (pWF->wBitsPerSample == 24) {
+ return ma_format_s24;
+ }
+ if (pWF->wBitsPerSample == 16) {
+ return ma_format_s16;
+ }
+ if (pWF->wBitsPerSample == 8) {
+ return ma_format_u8;
+ }
+ }
+ if (pWF->wFormatTag == WAVE_FORMAT_IEEE_FLOAT) {
+ if (pWF->wBitsPerSample == 32) {
+ return ma_format_f32;
+ }
+ if (pWF->wBitsPerSample == 64) {
+ /*return ma_format_f64;*/
+ }
+ }
+ }
+
+ return ma_format_unknown;
+}
+#endif
+
+
+/*******************************************************************************
+
+WASAPI Backend
+
+*******************************************************************************/
+#ifdef MA_HAS_WASAPI
+#if 0
+#if defined(_MSC_VER)
+ #pragma warning(push)
+ #pragma warning(disable:4091) /* 'typedef ': ignored on left of '' when no variable is declared */
+#endif
+#include
+#include
+#if defined(_MSC_VER)
+ #pragma warning(pop)
+#endif
+#endif /* 0 */
+
+static ma_result ma_device_reroute__wasapi(ma_device* pDevice, ma_device_type deviceType);
+
+/* Some compilers don't define VerifyVersionInfoW. Need to write this ourselves. */
+#define MA_WIN32_WINNT_VISTA 0x0600
+#define MA_VER_MINORVERSION 0x01
+#define MA_VER_MAJORVERSION 0x02
+#define MA_VER_SERVICEPACKMAJOR 0x20
+#define MA_VER_GREATER_EQUAL 0x03
+
+typedef struct {
+ DWORD dwOSVersionInfoSize;
+ DWORD dwMajorVersion;
+ DWORD dwMinorVersion;
+ DWORD dwBuildNumber;
+ DWORD dwPlatformId;
+ WCHAR szCSDVersion[128];
+ WORD wServicePackMajor;
+ WORD wServicePackMinor;
+ WORD wSuiteMask;
+ BYTE wProductType;
+ BYTE wReserved;
+} ma_OSVERSIONINFOEXW;
+
+typedef BOOL (WINAPI * ma_PFNVerifyVersionInfoW) (ma_OSVERSIONINFOEXW* lpVersionInfo, DWORD dwTypeMask, DWORDLONG dwlConditionMask);
+typedef ULONGLONG (WINAPI * ma_PFNVerSetConditionMask)(ULONGLONG dwlConditionMask, DWORD dwTypeBitMask, BYTE dwConditionMask);
+
+
+#ifndef PROPERTYKEY_DEFINED
+#define PROPERTYKEY_DEFINED
+#ifndef __WATCOMC__
+typedef struct
+{
+ GUID fmtid;
+ DWORD pid;
+} PROPERTYKEY;
+#endif
+#endif
+
+/* Some compilers don't define PropVariantInit(). We just do this ourselves since it's just a memset(). */
+static MA_INLINE void ma_PropVariantInit(PROPVARIANT* pProp)
+{
+ MA_ZERO_OBJECT(pProp);
+}
+
+
+static const PROPERTYKEY MA_PKEY_Device_FriendlyName = {{0xA45C254E, 0xDF1C, 0x4EFD, {0x80, 0x20, 0x67, 0xD1, 0x46, 0xA8, 0x50, 0xE0}}, 14};
+static const PROPERTYKEY MA_PKEY_AudioEngine_DeviceFormat = {{0xF19F064D, 0x82C, 0x4E27, {0xBC, 0x73, 0x68, 0x82, 0xA1, 0xBB, 0x8E, 0x4C}}, 0};
+
+static const IID MA_IID_IUnknown = {0x00000000, 0x0000, 0x0000, {0xC0, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x46}}; /* 00000000-0000-0000-C000-000000000046 */
+#if !defined(MA_WIN32_DESKTOP) && !defined(MA_WIN32_GDK)
+static const IID MA_IID_IAgileObject = {0x94EA2B94, 0xE9CC, 0x49E0, {0xC0, 0xFF, 0xEE, 0x64, 0xCA, 0x8F, 0x5B, 0x90}}; /* 94EA2B94-E9CC-49E0-C0FF-EE64CA8F5B90 */
+#endif
+
+static const IID MA_IID_IAudioClient = {0x1CB9AD4C, 0xDBFA, 0x4C32, {0xB1, 0x78, 0xC2, 0xF5, 0x68, 0xA7, 0x03, 0xB2}}; /* 1CB9AD4C-DBFA-4C32-B178-C2F568A703B2 = __uuidof(IAudioClient) */
+static const IID MA_IID_IAudioClient2 = {0x726778CD, 0xF60A, 0x4EDA, {0x82, 0xDE, 0xE4, 0x76, 0x10, 0xCD, 0x78, 0xAA}}; /* 726778CD-F60A-4EDA-82DE-E47610CD78AA = __uuidof(IAudioClient2) */
+static const IID MA_IID_IAudioClient3 = {0x7ED4EE07, 0x8E67, 0x4CD4, {0x8C, 0x1A, 0x2B, 0x7A, 0x59, 0x87, 0xAD, 0x42}}; /* 7ED4EE07-8E67-4CD4-8C1A-2B7A5987AD42 = __uuidof(IAudioClient3) */
+static const IID MA_IID_IAudioRenderClient = {0xF294ACFC, 0x3146, 0x4483, {0xA7, 0xBF, 0xAD, 0xDC, 0xA7, 0xC2, 0x60, 0xE2}}; /* F294ACFC-3146-4483-A7BF-ADDCA7C260E2 = __uuidof(IAudioRenderClient) */
+static const IID MA_IID_IAudioCaptureClient = {0xC8ADBD64, 0xE71E, 0x48A0, {0xA4, 0xDE, 0x18, 0x5C, 0x39, 0x5C, 0xD3, 0x17}}; /* C8ADBD64-E71E-48A0-A4DE-185C395CD317 = __uuidof(IAudioCaptureClient) */
+static const IID MA_IID_IMMNotificationClient = {0x7991EEC9, 0x7E89, 0x4D85, {0x83, 0x90, 0x6C, 0x70, 0x3C, 0xEC, 0x60, 0xC0}}; /* 7991EEC9-7E89-4D85-8390-6C703CEC60C0 = __uuidof(IMMNotificationClient) */
+#if !defined(MA_WIN32_DESKTOP) && !defined(MA_WIN32_GDK)
+static const IID MA_IID_DEVINTERFACE_AUDIO_RENDER = {0xE6327CAD, 0xDCEC, 0x4949, {0xAE, 0x8A, 0x99, 0x1E, 0x97, 0x6A, 0x79, 0xD2}}; /* E6327CAD-DCEC-4949-AE8A-991E976A79D2 */
+static const IID MA_IID_DEVINTERFACE_AUDIO_CAPTURE = {0x2EEF81BE, 0x33FA, 0x4800, {0x96, 0x70, 0x1C, 0xD4, 0x74, 0x97, 0x2C, 0x3F}}; /* 2EEF81BE-33FA-4800-9670-1CD474972C3F */
+static const IID MA_IID_IActivateAudioInterfaceCompletionHandler = {0x41D949AB, 0x9862, 0x444A, {0x80, 0xF6, 0xC2, 0x61, 0x33, 0x4D, 0xA5, 0xEB}}; /* 41D949AB-9862-444A-80F6-C261334DA5EB */
+#endif
+
+static const IID MA_CLSID_MMDeviceEnumerator_Instance = {0xBCDE0395, 0xE52F, 0x467C, {0x8E, 0x3D, 0xC4, 0x57, 0x92, 0x91, 0x69, 0x2E}}; /* BCDE0395-E52F-467C-8E3D-C4579291692E = __uuidof(MMDeviceEnumerator) */
+static const IID MA_IID_IMMDeviceEnumerator_Instance = {0xA95664D2, 0x9614, 0x4F35, {0xA7, 0x46, 0xDE, 0x8D, 0xB6, 0x36, 0x17, 0xE6}}; /* A95664D2-9614-4F35-A746-DE8DB63617E6 = __uuidof(IMMDeviceEnumerator) */
+#ifdef __cplusplus
+#define MA_CLSID_MMDeviceEnumerator MA_CLSID_MMDeviceEnumerator_Instance
+#define MA_IID_IMMDeviceEnumerator MA_IID_IMMDeviceEnumerator_Instance
+#else
+#define MA_CLSID_MMDeviceEnumerator &MA_CLSID_MMDeviceEnumerator_Instance
+#define MA_IID_IMMDeviceEnumerator &MA_IID_IMMDeviceEnumerator_Instance
+#endif
+
+typedef struct ma_IUnknown ma_IUnknown;
+#if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK)
+#define MA_MM_DEVICE_STATE_ACTIVE 1
+#define MA_MM_DEVICE_STATE_DISABLED 2
+#define MA_MM_DEVICE_STATE_NOTPRESENT 4
+#define MA_MM_DEVICE_STATE_UNPLUGGED 8
+
+typedef struct ma_IMMDeviceEnumerator ma_IMMDeviceEnumerator;
+typedef struct ma_IMMDeviceCollection ma_IMMDeviceCollection;
+typedef struct ma_IMMDevice ma_IMMDevice;
+#else
+typedef struct ma_IActivateAudioInterfaceCompletionHandler ma_IActivateAudioInterfaceCompletionHandler;
+typedef struct ma_IActivateAudioInterfaceAsyncOperation ma_IActivateAudioInterfaceAsyncOperation;
+#endif
+typedef struct ma_IPropertyStore ma_IPropertyStore;
+typedef struct ma_IAudioClient ma_IAudioClient;
+typedef struct ma_IAudioClient2 ma_IAudioClient2;
+typedef struct ma_IAudioClient3 ma_IAudioClient3;
+typedef struct ma_IAudioRenderClient ma_IAudioRenderClient;
+typedef struct ma_IAudioCaptureClient ma_IAudioCaptureClient;
+
+typedef ma_int64 MA_REFERENCE_TIME;
+
+#define MA_AUDCLNT_STREAMFLAGS_CROSSPROCESS 0x00010000
+#define MA_AUDCLNT_STREAMFLAGS_LOOPBACK 0x00020000
+#define MA_AUDCLNT_STREAMFLAGS_EVENTCALLBACK 0x00040000
+#define MA_AUDCLNT_STREAMFLAGS_NOPERSIST 0x00080000
+#define MA_AUDCLNT_STREAMFLAGS_RATEADJUST 0x00100000
+#define MA_AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY 0x08000000
+#define MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM 0x80000000
+#define MA_AUDCLNT_SESSIONFLAGS_EXPIREWHENUNOWNED 0x10000000
+#define MA_AUDCLNT_SESSIONFLAGS_DISPLAY_HIDE 0x20000000
+#define MA_AUDCLNT_SESSIONFLAGS_DISPLAY_HIDEWHENEXPIRED 0x40000000
+
+/* Buffer flags. */
+#define MA_AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY 1
+#define MA_AUDCLNT_BUFFERFLAGS_SILENT 2
+#define MA_AUDCLNT_BUFFERFLAGS_TIMESTAMP_ERROR 4
+
+typedef enum
+{
+ ma_eRender = 0,
+ ma_eCapture = 1,
+ ma_eAll = 2
+} ma_EDataFlow;
+
+typedef enum
+{
+ ma_eConsole = 0,
+ ma_eMultimedia = 1,
+ ma_eCommunications = 2
+} ma_ERole;
+
+typedef enum
+{
+ MA_AUDCLNT_SHAREMODE_SHARED,
+ MA_AUDCLNT_SHAREMODE_EXCLUSIVE
+} MA_AUDCLNT_SHAREMODE;
+
+typedef enum
+{
+ MA_AudioCategory_Other = 0 /* <-- miniaudio is only caring about Other. */
+} MA_AUDIO_STREAM_CATEGORY;
+
+typedef struct
+{
+ ma_uint32 cbSize;
+ BOOL bIsOffload;
+ MA_AUDIO_STREAM_CATEGORY eCategory;
+} ma_AudioClientProperties;
+
+/* IUnknown */
+typedef struct
+{
+ /* IUnknown */
+ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IUnknown* pThis, const IID* const riid, void** ppObject);
+ ULONG (STDMETHODCALLTYPE * AddRef) (ma_IUnknown* pThis);
+ ULONG (STDMETHODCALLTYPE * Release) (ma_IUnknown* pThis);
+} ma_IUnknownVtbl;
+struct ma_IUnknown
+{
+ ma_IUnknownVtbl* lpVtbl;
+};
+static MA_INLINE HRESULT ma_IUnknown_QueryInterface(ma_IUnknown* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
+static MA_INLINE ULONG ma_IUnknown_AddRef(ma_IUnknown* pThis) { return pThis->lpVtbl->AddRef(pThis); }
+static MA_INLINE ULONG ma_IUnknown_Release(ma_IUnknown* pThis) { return pThis->lpVtbl->Release(pThis); }
+
+#if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK)
+ /* IMMNotificationClient */
+ typedef struct
+ {
+ /* IUnknown */
+ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IMMNotificationClient* pThis, const IID* const riid, void** ppObject);
+ ULONG (STDMETHODCALLTYPE * AddRef) (ma_IMMNotificationClient* pThis);
+ ULONG (STDMETHODCALLTYPE * Release) (ma_IMMNotificationClient* pThis);
+
+ /* IMMNotificationClient */
+ HRESULT (STDMETHODCALLTYPE * OnDeviceStateChanged) (ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID, DWORD dwNewState);
+ HRESULT (STDMETHODCALLTYPE * OnDeviceAdded) (ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID);
+ HRESULT (STDMETHODCALLTYPE * OnDeviceRemoved) (ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID);
+ HRESULT (STDMETHODCALLTYPE * OnDefaultDeviceChanged)(ma_IMMNotificationClient* pThis, ma_EDataFlow dataFlow, ma_ERole role, LPCWSTR pDefaultDeviceID);
+ HRESULT (STDMETHODCALLTYPE * OnPropertyValueChanged)(ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID, const PROPERTYKEY key);
+ } ma_IMMNotificationClientVtbl;
+
+ /* IMMDeviceEnumerator */
+ typedef struct
+ {
+ /* IUnknown */
+ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IMMDeviceEnumerator* pThis, const IID* const riid, void** ppObject);
+ ULONG (STDMETHODCALLTYPE * AddRef) (ma_IMMDeviceEnumerator* pThis);
+ ULONG (STDMETHODCALLTYPE * Release) (ma_IMMDeviceEnumerator* pThis);
+
+ /* IMMDeviceEnumerator */
+ HRESULT (STDMETHODCALLTYPE * EnumAudioEndpoints) (ma_IMMDeviceEnumerator* pThis, ma_EDataFlow dataFlow, DWORD dwStateMask, ma_IMMDeviceCollection** ppDevices);
+ HRESULT (STDMETHODCALLTYPE * GetDefaultAudioEndpoint) (ma_IMMDeviceEnumerator* pThis, ma_EDataFlow dataFlow, ma_ERole role, ma_IMMDevice** ppEndpoint);
+ HRESULT (STDMETHODCALLTYPE * GetDevice) (ma_IMMDeviceEnumerator* pThis, LPCWSTR pID, ma_IMMDevice** ppDevice);
+ HRESULT (STDMETHODCALLTYPE * RegisterEndpointNotificationCallback) (ma_IMMDeviceEnumerator* pThis, ma_IMMNotificationClient* pClient);
+ HRESULT (STDMETHODCALLTYPE * UnregisterEndpointNotificationCallback)(ma_IMMDeviceEnumerator* pThis, ma_IMMNotificationClient* pClient);
+ } ma_IMMDeviceEnumeratorVtbl;
+ struct ma_IMMDeviceEnumerator
+ {
+ ma_IMMDeviceEnumeratorVtbl* lpVtbl;
+ };
+ static MA_INLINE HRESULT ma_IMMDeviceEnumerator_QueryInterface(ma_IMMDeviceEnumerator* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
+ static MA_INLINE ULONG ma_IMMDeviceEnumerator_AddRef(ma_IMMDeviceEnumerator* pThis) { return pThis->lpVtbl->AddRef(pThis); }
+ static MA_INLINE ULONG ma_IMMDeviceEnumerator_Release(ma_IMMDeviceEnumerator* pThis) { return pThis->lpVtbl->Release(pThis); }
+ static MA_INLINE HRESULT ma_IMMDeviceEnumerator_EnumAudioEndpoints(ma_IMMDeviceEnumerator* pThis, ma_EDataFlow dataFlow, DWORD dwStateMask, ma_IMMDeviceCollection** ppDevices) { return pThis->lpVtbl->EnumAudioEndpoints(pThis, dataFlow, dwStateMask, ppDevices); }
+ static MA_INLINE HRESULT ma_IMMDeviceEnumerator_GetDefaultAudioEndpoint(ma_IMMDeviceEnumerator* pThis, ma_EDataFlow dataFlow, ma_ERole role, ma_IMMDevice** ppEndpoint) { return pThis->lpVtbl->GetDefaultAudioEndpoint(pThis, dataFlow, role, ppEndpoint); }
+ static MA_INLINE HRESULT ma_IMMDeviceEnumerator_GetDevice(ma_IMMDeviceEnumerator* pThis, LPCWSTR pID, ma_IMMDevice** ppDevice) { return pThis->lpVtbl->GetDevice(pThis, pID, ppDevice); }
+ static MA_INLINE HRESULT ma_IMMDeviceEnumerator_RegisterEndpointNotificationCallback(ma_IMMDeviceEnumerator* pThis, ma_IMMNotificationClient* pClient) { return pThis->lpVtbl->RegisterEndpointNotificationCallback(pThis, pClient); }
+ static MA_INLINE HRESULT ma_IMMDeviceEnumerator_UnregisterEndpointNotificationCallback(ma_IMMDeviceEnumerator* pThis, ma_IMMNotificationClient* pClient) { return pThis->lpVtbl->UnregisterEndpointNotificationCallback(pThis, pClient); }
+
+
+ /* IMMDeviceCollection */
+ typedef struct
+ {
+ /* IUnknown */
+ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IMMDeviceCollection* pThis, const IID* const riid, void** ppObject);
+ ULONG (STDMETHODCALLTYPE * AddRef) (ma_IMMDeviceCollection* pThis);
+ ULONG (STDMETHODCALLTYPE * Release) (ma_IMMDeviceCollection* pThis);
+
+ /* IMMDeviceCollection */
+ HRESULT (STDMETHODCALLTYPE * GetCount)(ma_IMMDeviceCollection* pThis, UINT* pDevices);
+ HRESULT (STDMETHODCALLTYPE * Item) (ma_IMMDeviceCollection* pThis, UINT nDevice, ma_IMMDevice** ppDevice);
+ } ma_IMMDeviceCollectionVtbl;
+ struct ma_IMMDeviceCollection
+ {
+ ma_IMMDeviceCollectionVtbl* lpVtbl;
+ };
+ static MA_INLINE HRESULT ma_IMMDeviceCollection_QueryInterface(ma_IMMDeviceCollection* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
+ static MA_INLINE ULONG ma_IMMDeviceCollection_AddRef(ma_IMMDeviceCollection* pThis) { return pThis->lpVtbl->AddRef(pThis); }
+ static MA_INLINE ULONG ma_IMMDeviceCollection_Release(ma_IMMDeviceCollection* pThis) { return pThis->lpVtbl->Release(pThis); }
+ static MA_INLINE HRESULT ma_IMMDeviceCollection_GetCount(ma_IMMDeviceCollection* pThis, UINT* pDevices) { return pThis->lpVtbl->GetCount(pThis, pDevices); }
+ static MA_INLINE HRESULT ma_IMMDeviceCollection_Item(ma_IMMDeviceCollection* pThis, UINT nDevice, ma_IMMDevice** ppDevice) { return pThis->lpVtbl->Item(pThis, nDevice, ppDevice); }
+
+
+ /* IMMDevice */
+ typedef struct
+ {
+ /* IUnknown */
+ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IMMDevice* pThis, const IID* const riid, void** ppObject);
+ ULONG (STDMETHODCALLTYPE * AddRef) (ma_IMMDevice* pThis);
+ ULONG (STDMETHODCALLTYPE * Release) (ma_IMMDevice* pThis);
+
+ /* IMMDevice */
+ HRESULT (STDMETHODCALLTYPE * Activate) (ma_IMMDevice* pThis, const IID* const iid, DWORD dwClsCtx, PROPVARIANT* pActivationParams, void** ppInterface);
+ HRESULT (STDMETHODCALLTYPE * OpenPropertyStore)(ma_IMMDevice* pThis, DWORD stgmAccess, ma_IPropertyStore** ppProperties);
+ HRESULT (STDMETHODCALLTYPE * GetId) (ma_IMMDevice* pThis, LPWSTR *pID);
+ HRESULT (STDMETHODCALLTYPE * GetState) (ma_IMMDevice* pThis, DWORD *pState);
+ } ma_IMMDeviceVtbl;
+ struct ma_IMMDevice
+ {
+ ma_IMMDeviceVtbl* lpVtbl;
+ };
+ static MA_INLINE HRESULT ma_IMMDevice_QueryInterface(ma_IMMDevice* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
+ static MA_INLINE ULONG ma_IMMDevice_AddRef(ma_IMMDevice* pThis) { return pThis->lpVtbl->AddRef(pThis); }
+ static MA_INLINE ULONG ma_IMMDevice_Release(ma_IMMDevice* pThis) { return pThis->lpVtbl->Release(pThis); }
+ static MA_INLINE HRESULT ma_IMMDevice_Activate(ma_IMMDevice* pThis, const IID* const iid, DWORD dwClsCtx, PROPVARIANT* pActivationParams, void** ppInterface) { return pThis->lpVtbl->Activate(pThis, iid, dwClsCtx, pActivationParams, ppInterface); }
+ static MA_INLINE HRESULT ma_IMMDevice_OpenPropertyStore(ma_IMMDevice* pThis, DWORD stgmAccess, ma_IPropertyStore** ppProperties) { return pThis->lpVtbl->OpenPropertyStore(pThis, stgmAccess, ppProperties); }
+ static MA_INLINE HRESULT ma_IMMDevice_GetId(ma_IMMDevice* pThis, LPWSTR *pID) { return pThis->lpVtbl->GetId(pThis, pID); }
+ static MA_INLINE HRESULT ma_IMMDevice_GetState(ma_IMMDevice* pThis, DWORD *pState) { return pThis->lpVtbl->GetState(pThis, pState); }
+#else
+ /* IActivateAudioInterfaceAsyncOperation */
+ typedef struct
+ {
+ /* IUnknown */
+ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IActivateAudioInterfaceAsyncOperation* pThis, const IID* const riid, void** ppObject);
+ ULONG (STDMETHODCALLTYPE * AddRef) (ma_IActivateAudioInterfaceAsyncOperation* pThis);
+ ULONG (STDMETHODCALLTYPE * Release) (ma_IActivateAudioInterfaceAsyncOperation* pThis);
+
+ /* IActivateAudioInterfaceAsyncOperation */
+ HRESULT (STDMETHODCALLTYPE * GetActivateResult)(ma_IActivateAudioInterfaceAsyncOperation* pThis, HRESULT *pActivateResult, ma_IUnknown** ppActivatedInterface);
+ } ma_IActivateAudioInterfaceAsyncOperationVtbl;
+ struct ma_IActivateAudioInterfaceAsyncOperation
+ {
+ ma_IActivateAudioInterfaceAsyncOperationVtbl* lpVtbl;
+ };
+ static MA_INLINE HRESULT ma_IActivateAudioInterfaceAsyncOperation_QueryInterface(ma_IActivateAudioInterfaceAsyncOperation* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
+ static MA_INLINE ULONG ma_IActivateAudioInterfaceAsyncOperation_AddRef(ma_IActivateAudioInterfaceAsyncOperation* pThis) { return pThis->lpVtbl->AddRef(pThis); }
+ static MA_INLINE ULONG ma_IActivateAudioInterfaceAsyncOperation_Release(ma_IActivateAudioInterfaceAsyncOperation* pThis) { return pThis->lpVtbl->Release(pThis); }
+ static MA_INLINE HRESULT ma_IActivateAudioInterfaceAsyncOperation_GetActivateResult(ma_IActivateAudioInterfaceAsyncOperation* pThis, HRESULT *pActivateResult, ma_IUnknown** ppActivatedInterface) { return pThis->lpVtbl->GetActivateResult(pThis, pActivateResult, ppActivatedInterface); }
+#endif
+
+/* IPropertyStore */
+typedef struct
+{
+ /* IUnknown */
+ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IPropertyStore* pThis, const IID* const riid, void** ppObject);
+ ULONG (STDMETHODCALLTYPE * AddRef) (ma_IPropertyStore* pThis);
+ ULONG (STDMETHODCALLTYPE * Release) (ma_IPropertyStore* pThis);
+
+ /* IPropertyStore */
+ HRESULT (STDMETHODCALLTYPE * GetCount)(ma_IPropertyStore* pThis, DWORD* pPropCount);
+ HRESULT (STDMETHODCALLTYPE * GetAt) (ma_IPropertyStore* pThis, DWORD propIndex, PROPERTYKEY* pPropKey);
+ HRESULT (STDMETHODCALLTYPE * GetValue)(ma_IPropertyStore* pThis, const PROPERTYKEY* const pKey, PROPVARIANT* pPropVar);
+ HRESULT (STDMETHODCALLTYPE * SetValue)(ma_IPropertyStore* pThis, const PROPERTYKEY* const pKey, const PROPVARIANT* const pPropVar);
+ HRESULT (STDMETHODCALLTYPE * Commit) (ma_IPropertyStore* pThis);
+} ma_IPropertyStoreVtbl;
+struct ma_IPropertyStore
+{
+ ma_IPropertyStoreVtbl* lpVtbl;
+};
+static MA_INLINE HRESULT ma_IPropertyStore_QueryInterface(ma_IPropertyStore* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
+static MA_INLINE ULONG ma_IPropertyStore_AddRef(ma_IPropertyStore* pThis) { return pThis->lpVtbl->AddRef(pThis); }
+static MA_INLINE ULONG ma_IPropertyStore_Release(ma_IPropertyStore* pThis) { return pThis->lpVtbl->Release(pThis); }
+static MA_INLINE HRESULT ma_IPropertyStore_GetCount(ma_IPropertyStore* pThis, DWORD* pPropCount) { return pThis->lpVtbl->GetCount(pThis, pPropCount); }
+static MA_INLINE HRESULT ma_IPropertyStore_GetAt(ma_IPropertyStore* pThis, DWORD propIndex, PROPERTYKEY* pPropKey) { return pThis->lpVtbl->GetAt(pThis, propIndex, pPropKey); }
+static MA_INLINE HRESULT ma_IPropertyStore_GetValue(ma_IPropertyStore* pThis, const PROPERTYKEY* const pKey, PROPVARIANT* pPropVar) { return pThis->lpVtbl->GetValue(pThis, pKey, pPropVar); }
+static MA_INLINE HRESULT ma_IPropertyStore_SetValue(ma_IPropertyStore* pThis, const PROPERTYKEY* const pKey, const PROPVARIANT* const pPropVar) { return pThis->lpVtbl->SetValue(pThis, pKey, pPropVar); }
+static MA_INLINE HRESULT ma_IPropertyStore_Commit(ma_IPropertyStore* pThis) { return pThis->lpVtbl->Commit(pThis); }
+
+
+/* IAudioClient */
+typedef struct
+{
+ /* IUnknown */
+ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioClient* pThis, const IID* const riid, void** ppObject);
+ ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioClient* pThis);
+ ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioClient* pThis);
+
+ /* IAudioClient */
+ HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IAudioClient* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid);
+ HRESULT (STDMETHODCALLTYPE * GetBufferSize) (ma_IAudioClient* pThis, ma_uint32* pNumBufferFrames);
+ HRESULT (STDMETHODCALLTYPE * GetStreamLatency) (ma_IAudioClient* pThis, MA_REFERENCE_TIME* pLatency);
+ HRESULT (STDMETHODCALLTYPE * GetCurrentPadding)(ma_IAudioClient* pThis, ma_uint32* pNumPaddingFrames);
+ HRESULT (STDMETHODCALLTYPE * IsFormatSupported)(ma_IAudioClient* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch);
+ HRESULT (STDMETHODCALLTYPE * GetMixFormat) (ma_IAudioClient* pThis, WAVEFORMATEX** ppDeviceFormat);
+ HRESULT (STDMETHODCALLTYPE * GetDevicePeriod) (ma_IAudioClient* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod);
+ HRESULT (STDMETHODCALLTYPE * Start) (ma_IAudioClient* pThis);
+ HRESULT (STDMETHODCALLTYPE * Stop) (ma_IAudioClient* pThis);
+ HRESULT (STDMETHODCALLTYPE * Reset) (ma_IAudioClient* pThis);
+ HRESULT (STDMETHODCALLTYPE * SetEventHandle) (ma_IAudioClient* pThis, HANDLE eventHandle);
+ HRESULT (STDMETHODCALLTYPE * GetService) (ma_IAudioClient* pThis, const IID* const riid, void** pp);
+} ma_IAudioClientVtbl;
+struct ma_IAudioClient
+{
+ ma_IAudioClientVtbl* lpVtbl;
+};
+static MA_INLINE HRESULT ma_IAudioClient_QueryInterface(ma_IAudioClient* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
+static MA_INLINE ULONG ma_IAudioClient_AddRef(ma_IAudioClient* pThis) { return pThis->lpVtbl->AddRef(pThis); }
+static MA_INLINE ULONG ma_IAudioClient_Release(ma_IAudioClient* pThis) { return pThis->lpVtbl->Release(pThis); }
+static MA_INLINE HRESULT ma_IAudioClient_Initialize(ma_IAudioClient* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid) { return pThis->lpVtbl->Initialize(pThis, shareMode, streamFlags, bufferDuration, periodicity, pFormat, pAudioSessionGuid); }
+static MA_INLINE HRESULT ma_IAudioClient_GetBufferSize(ma_IAudioClient* pThis, ma_uint32* pNumBufferFrames) { return pThis->lpVtbl->GetBufferSize(pThis, pNumBufferFrames); }
+static MA_INLINE HRESULT ma_IAudioClient_GetStreamLatency(ma_IAudioClient* pThis, MA_REFERENCE_TIME* pLatency) { return pThis->lpVtbl->GetStreamLatency(pThis, pLatency); }
+static MA_INLINE HRESULT ma_IAudioClient_GetCurrentPadding(ma_IAudioClient* pThis, ma_uint32* pNumPaddingFrames) { return pThis->lpVtbl->GetCurrentPadding(pThis, pNumPaddingFrames); }
+static MA_INLINE HRESULT ma_IAudioClient_IsFormatSupported(ma_IAudioClient* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch) { return pThis->lpVtbl->IsFormatSupported(pThis, shareMode, pFormat, ppClosestMatch); }
+static MA_INLINE HRESULT ma_IAudioClient_GetMixFormat(ma_IAudioClient* pThis, WAVEFORMATEX** ppDeviceFormat) { return pThis->lpVtbl->GetMixFormat(pThis, ppDeviceFormat); }
+static MA_INLINE HRESULT ma_IAudioClient_GetDevicePeriod(ma_IAudioClient* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod) { return pThis->lpVtbl->GetDevicePeriod(pThis, pDefaultDevicePeriod, pMinimumDevicePeriod); }
+static MA_INLINE HRESULT ma_IAudioClient_Start(ma_IAudioClient* pThis) { return pThis->lpVtbl->Start(pThis); }
+static MA_INLINE HRESULT ma_IAudioClient_Stop(ma_IAudioClient* pThis) { return pThis->lpVtbl->Stop(pThis); }
+static MA_INLINE HRESULT ma_IAudioClient_Reset(ma_IAudioClient* pThis) { return pThis->lpVtbl->Reset(pThis); }
+static MA_INLINE HRESULT ma_IAudioClient_SetEventHandle(ma_IAudioClient* pThis, HANDLE eventHandle) { return pThis->lpVtbl->SetEventHandle(pThis, eventHandle); }
+static MA_INLINE HRESULT ma_IAudioClient_GetService(ma_IAudioClient* pThis, const IID* const riid, void** pp) { return pThis->lpVtbl->GetService(pThis, riid, pp); }
+
+/* IAudioClient2 */
+typedef struct
+{
+ /* IUnknown */
+ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioClient2* pThis, const IID* const riid, void** ppObject);
+ ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioClient2* pThis);
+ ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioClient2* pThis);
+
+ /* IAudioClient */
+ HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IAudioClient2* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid);
+ HRESULT (STDMETHODCALLTYPE * GetBufferSize) (ma_IAudioClient2* pThis, ma_uint32* pNumBufferFrames);
+ HRESULT (STDMETHODCALLTYPE * GetStreamLatency) (ma_IAudioClient2* pThis, MA_REFERENCE_TIME* pLatency);
+ HRESULT (STDMETHODCALLTYPE * GetCurrentPadding)(ma_IAudioClient2* pThis, ma_uint32* pNumPaddingFrames);
+ HRESULT (STDMETHODCALLTYPE * IsFormatSupported)(ma_IAudioClient2* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch);
+ HRESULT (STDMETHODCALLTYPE * GetMixFormat) (ma_IAudioClient2* pThis, WAVEFORMATEX** ppDeviceFormat);
+ HRESULT (STDMETHODCALLTYPE * GetDevicePeriod) (ma_IAudioClient2* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod);
+ HRESULT (STDMETHODCALLTYPE * Start) (ma_IAudioClient2* pThis);
+ HRESULT (STDMETHODCALLTYPE * Stop) (ma_IAudioClient2* pThis);
+ HRESULT (STDMETHODCALLTYPE * Reset) (ma_IAudioClient2* pThis);
+ HRESULT (STDMETHODCALLTYPE * SetEventHandle) (ma_IAudioClient2* pThis, HANDLE eventHandle);
+ HRESULT (STDMETHODCALLTYPE * GetService) (ma_IAudioClient2* pThis, const IID* const riid, void** pp);
+
+ /* IAudioClient2 */
+ HRESULT (STDMETHODCALLTYPE * IsOffloadCapable) (ma_IAudioClient2* pThis, MA_AUDIO_STREAM_CATEGORY category, BOOL* pOffloadCapable);
+ HRESULT (STDMETHODCALLTYPE * SetClientProperties)(ma_IAudioClient2* pThis, const ma_AudioClientProperties* pProperties);
+ HRESULT (STDMETHODCALLTYPE * GetBufferSizeLimits)(ma_IAudioClient2* pThis, const WAVEFORMATEX* pFormat, BOOL eventDriven, MA_REFERENCE_TIME* pMinBufferDuration, MA_REFERENCE_TIME* pMaxBufferDuration);
+} ma_IAudioClient2Vtbl;
+struct ma_IAudioClient2
+{
+ ma_IAudioClient2Vtbl* lpVtbl;
+};
+static MA_INLINE HRESULT ma_IAudioClient2_QueryInterface(ma_IAudioClient2* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
+static MA_INLINE ULONG ma_IAudioClient2_AddRef(ma_IAudioClient2* pThis) { return pThis->lpVtbl->AddRef(pThis); }
+static MA_INLINE ULONG ma_IAudioClient2_Release(ma_IAudioClient2* pThis) { return pThis->lpVtbl->Release(pThis); }
+static MA_INLINE HRESULT ma_IAudioClient2_Initialize(ma_IAudioClient2* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid) { return pThis->lpVtbl->Initialize(pThis, shareMode, streamFlags, bufferDuration, periodicity, pFormat, pAudioSessionGuid); }
+static MA_INLINE HRESULT ma_IAudioClient2_GetBufferSize(ma_IAudioClient2* pThis, ma_uint32* pNumBufferFrames) { return pThis->lpVtbl->GetBufferSize(pThis, pNumBufferFrames); }
+static MA_INLINE HRESULT ma_IAudioClient2_GetStreamLatency(ma_IAudioClient2* pThis, MA_REFERENCE_TIME* pLatency) { return pThis->lpVtbl->GetStreamLatency(pThis, pLatency); }
+static MA_INLINE HRESULT ma_IAudioClient2_GetCurrentPadding(ma_IAudioClient2* pThis, ma_uint32* pNumPaddingFrames) { return pThis->lpVtbl->GetCurrentPadding(pThis, pNumPaddingFrames); }
+static MA_INLINE HRESULT ma_IAudioClient2_IsFormatSupported(ma_IAudioClient2* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch) { return pThis->lpVtbl->IsFormatSupported(pThis, shareMode, pFormat, ppClosestMatch); }
+static MA_INLINE HRESULT ma_IAudioClient2_GetMixFormat(ma_IAudioClient2* pThis, WAVEFORMATEX** ppDeviceFormat) { return pThis->lpVtbl->GetMixFormat(pThis, ppDeviceFormat); }
+static MA_INLINE HRESULT ma_IAudioClient2_GetDevicePeriod(ma_IAudioClient2* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod) { return pThis->lpVtbl->GetDevicePeriod(pThis, pDefaultDevicePeriod, pMinimumDevicePeriod); }
+static MA_INLINE HRESULT ma_IAudioClient2_Start(ma_IAudioClient2* pThis) { return pThis->lpVtbl->Start(pThis); }
+static MA_INLINE HRESULT ma_IAudioClient2_Stop(ma_IAudioClient2* pThis) { return pThis->lpVtbl->Stop(pThis); }
+static MA_INLINE HRESULT ma_IAudioClient2_Reset(ma_IAudioClient2* pThis) { return pThis->lpVtbl->Reset(pThis); }
+static MA_INLINE HRESULT ma_IAudioClient2_SetEventHandle(ma_IAudioClient2* pThis, HANDLE eventHandle) { return pThis->lpVtbl->SetEventHandle(pThis, eventHandle); }
+static MA_INLINE HRESULT ma_IAudioClient2_GetService(ma_IAudioClient2* pThis, const IID* const riid, void** pp) { return pThis->lpVtbl->GetService(pThis, riid, pp); }
+static MA_INLINE HRESULT ma_IAudioClient2_IsOffloadCapable(ma_IAudioClient2* pThis, MA_AUDIO_STREAM_CATEGORY category, BOOL* pOffloadCapable) { return pThis->lpVtbl->IsOffloadCapable(pThis, category, pOffloadCapable); }
+static MA_INLINE HRESULT ma_IAudioClient2_SetClientProperties(ma_IAudioClient2* pThis, const ma_AudioClientProperties* pProperties) { return pThis->lpVtbl->SetClientProperties(pThis, pProperties); }
+static MA_INLINE HRESULT ma_IAudioClient2_GetBufferSizeLimits(ma_IAudioClient2* pThis, const WAVEFORMATEX* pFormat, BOOL eventDriven, MA_REFERENCE_TIME* pMinBufferDuration, MA_REFERENCE_TIME* pMaxBufferDuration) { return pThis->lpVtbl->GetBufferSizeLimits(pThis, pFormat, eventDriven, pMinBufferDuration, pMaxBufferDuration); }
+
+
+/* IAudioClient3 */
+typedef struct
+{
+ /* IUnknown */
+ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioClient3* pThis, const IID* const riid, void** ppObject);
+ ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioClient3* pThis);
+ ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioClient3* pThis);
+
+ /* IAudioClient */
+ HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IAudioClient3* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid);
+ HRESULT (STDMETHODCALLTYPE * GetBufferSize) (ma_IAudioClient3* pThis, ma_uint32* pNumBufferFrames);
+ HRESULT (STDMETHODCALLTYPE * GetStreamLatency) (ma_IAudioClient3* pThis, MA_REFERENCE_TIME* pLatency);
+ HRESULT (STDMETHODCALLTYPE * GetCurrentPadding)(ma_IAudioClient3* pThis, ma_uint32* pNumPaddingFrames);
+ HRESULT (STDMETHODCALLTYPE * IsFormatSupported)(ma_IAudioClient3* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch);
+ HRESULT (STDMETHODCALLTYPE * GetMixFormat) (ma_IAudioClient3* pThis, WAVEFORMATEX** ppDeviceFormat);
+ HRESULT (STDMETHODCALLTYPE * GetDevicePeriod) (ma_IAudioClient3* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod);
+ HRESULT (STDMETHODCALLTYPE * Start) (ma_IAudioClient3* pThis);
+ HRESULT (STDMETHODCALLTYPE * Stop) (ma_IAudioClient3* pThis);
+ HRESULT (STDMETHODCALLTYPE * Reset) (ma_IAudioClient3* pThis);
+ HRESULT (STDMETHODCALLTYPE * SetEventHandle) (ma_IAudioClient3* pThis, HANDLE eventHandle);
+ HRESULT (STDMETHODCALLTYPE * GetService) (ma_IAudioClient3* pThis, const IID* const riid, void** pp);
+
+ /* IAudioClient2 */
+ HRESULT (STDMETHODCALLTYPE * IsOffloadCapable) (ma_IAudioClient3* pThis, MA_AUDIO_STREAM_CATEGORY category, BOOL* pOffloadCapable);
+ HRESULT (STDMETHODCALLTYPE * SetClientProperties)(ma_IAudioClient3* pThis, const ma_AudioClientProperties* pProperties);
+ HRESULT (STDMETHODCALLTYPE * GetBufferSizeLimits)(ma_IAudioClient3* pThis, const WAVEFORMATEX* pFormat, BOOL eventDriven, MA_REFERENCE_TIME* pMinBufferDuration, MA_REFERENCE_TIME* pMaxBufferDuration);
+
+ /* IAudioClient3 */
+ HRESULT (STDMETHODCALLTYPE * GetSharedModeEnginePeriod) (ma_IAudioClient3* pThis, const WAVEFORMATEX* pFormat, ma_uint32* pDefaultPeriodInFrames, ma_uint32* pFundamentalPeriodInFrames, ma_uint32* pMinPeriodInFrames, ma_uint32* pMaxPeriodInFrames);
+ HRESULT (STDMETHODCALLTYPE * GetCurrentSharedModeEnginePeriod)(ma_IAudioClient3* pThis, WAVEFORMATEX** ppFormat, ma_uint32* pCurrentPeriodInFrames);
+ HRESULT (STDMETHODCALLTYPE * InitializeSharedAudioStream) (ma_IAudioClient3* pThis, DWORD streamFlags, ma_uint32 periodInFrames, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid);
+} ma_IAudioClient3Vtbl;
+struct ma_IAudioClient3
+{
+ ma_IAudioClient3Vtbl* lpVtbl;
+};
+static MA_INLINE HRESULT ma_IAudioClient3_QueryInterface(ma_IAudioClient3* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
+static MA_INLINE ULONG ma_IAudioClient3_AddRef(ma_IAudioClient3* pThis) { return pThis->lpVtbl->AddRef(pThis); }
+static MA_INLINE ULONG ma_IAudioClient3_Release(ma_IAudioClient3* pThis) { return pThis->lpVtbl->Release(pThis); }
+static MA_INLINE HRESULT ma_IAudioClient3_Initialize(ma_IAudioClient3* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid) { return pThis->lpVtbl->Initialize(pThis, shareMode, streamFlags, bufferDuration, periodicity, pFormat, pAudioSessionGuid); }
+static MA_INLINE HRESULT ma_IAudioClient3_GetBufferSize(ma_IAudioClient3* pThis, ma_uint32* pNumBufferFrames) { return pThis->lpVtbl->GetBufferSize(pThis, pNumBufferFrames); }
+static MA_INLINE HRESULT ma_IAudioClient3_GetStreamLatency(ma_IAudioClient3* pThis, MA_REFERENCE_TIME* pLatency) { return pThis->lpVtbl->GetStreamLatency(pThis, pLatency); }
+static MA_INLINE HRESULT ma_IAudioClient3_GetCurrentPadding(ma_IAudioClient3* pThis, ma_uint32* pNumPaddingFrames) { return pThis->lpVtbl->GetCurrentPadding(pThis, pNumPaddingFrames); }
+static MA_INLINE HRESULT ma_IAudioClient3_IsFormatSupported(ma_IAudioClient3* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch) { return pThis->lpVtbl->IsFormatSupported(pThis, shareMode, pFormat, ppClosestMatch); }
+static MA_INLINE HRESULT ma_IAudioClient3_GetMixFormat(ma_IAudioClient3* pThis, WAVEFORMATEX** ppDeviceFormat) { return pThis->lpVtbl->GetMixFormat(pThis, ppDeviceFormat); }
+static MA_INLINE HRESULT ma_IAudioClient3_GetDevicePeriod(ma_IAudioClient3* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod) { return pThis->lpVtbl->GetDevicePeriod(pThis, pDefaultDevicePeriod, pMinimumDevicePeriod); }
+static MA_INLINE HRESULT ma_IAudioClient3_Start(ma_IAudioClient3* pThis) { return pThis->lpVtbl->Start(pThis); }
+static MA_INLINE HRESULT ma_IAudioClient3_Stop(ma_IAudioClient3* pThis) { return pThis->lpVtbl->Stop(pThis); }
+static MA_INLINE HRESULT ma_IAudioClient3_Reset(ma_IAudioClient3* pThis) { return pThis->lpVtbl->Reset(pThis); }
+static MA_INLINE HRESULT ma_IAudioClient3_SetEventHandle(ma_IAudioClient3* pThis, HANDLE eventHandle) { return pThis->lpVtbl->SetEventHandle(pThis, eventHandle); }
+static MA_INLINE HRESULT ma_IAudioClient3_GetService(ma_IAudioClient3* pThis, const IID* const riid, void** pp) { return pThis->lpVtbl->GetService(pThis, riid, pp); }
+static MA_INLINE HRESULT ma_IAudioClient3_IsOffloadCapable(ma_IAudioClient3* pThis, MA_AUDIO_STREAM_CATEGORY category, BOOL* pOffloadCapable) { return pThis->lpVtbl->IsOffloadCapable(pThis, category, pOffloadCapable); }
+static MA_INLINE HRESULT ma_IAudioClient3_SetClientProperties(ma_IAudioClient3* pThis, const ma_AudioClientProperties* pProperties) { return pThis->lpVtbl->SetClientProperties(pThis, pProperties); }
+static MA_INLINE HRESULT ma_IAudioClient3_GetBufferSizeLimits(ma_IAudioClient3* pThis, const WAVEFORMATEX* pFormat, BOOL eventDriven, MA_REFERENCE_TIME* pMinBufferDuration, MA_REFERENCE_TIME* pMaxBufferDuration) { return pThis->lpVtbl->GetBufferSizeLimits(pThis, pFormat, eventDriven, pMinBufferDuration, pMaxBufferDuration); }
+static MA_INLINE HRESULT ma_IAudioClient3_GetSharedModeEnginePeriod(ma_IAudioClient3* pThis, const WAVEFORMATEX* pFormat, ma_uint32* pDefaultPeriodInFrames, ma_uint32* pFundamentalPeriodInFrames, ma_uint32* pMinPeriodInFrames, ma_uint32* pMaxPeriodInFrames) { return pThis->lpVtbl->GetSharedModeEnginePeriod(pThis, pFormat, pDefaultPeriodInFrames, pFundamentalPeriodInFrames, pMinPeriodInFrames, pMaxPeriodInFrames); }
+static MA_INLINE HRESULT ma_IAudioClient3_GetCurrentSharedModeEnginePeriod(ma_IAudioClient3* pThis, WAVEFORMATEX** ppFormat, ma_uint32* pCurrentPeriodInFrames) { return pThis->lpVtbl->GetCurrentSharedModeEnginePeriod(pThis, ppFormat, pCurrentPeriodInFrames); }
+static MA_INLINE HRESULT ma_IAudioClient3_InitializeSharedAudioStream(ma_IAudioClient3* pThis, DWORD streamFlags, ma_uint32 periodInFrames, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGUID) { return pThis->lpVtbl->InitializeSharedAudioStream(pThis, streamFlags, periodInFrames, pFormat, pAudioSessionGUID); }
+
+
+/* IAudioRenderClient */
+typedef struct
+{
+ /* IUnknown */
+ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioRenderClient* pThis, const IID* const riid, void** ppObject);
+ ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioRenderClient* pThis);
+ ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioRenderClient* pThis);
+
+ /* IAudioRenderClient */
+ HRESULT (STDMETHODCALLTYPE * GetBuffer) (ma_IAudioRenderClient* pThis, ma_uint32 numFramesRequested, BYTE** ppData);
+ HRESULT (STDMETHODCALLTYPE * ReleaseBuffer)(ma_IAudioRenderClient* pThis, ma_uint32 numFramesWritten, DWORD dwFlags);
+} ma_IAudioRenderClientVtbl;
+struct ma_IAudioRenderClient
+{
+ ma_IAudioRenderClientVtbl* lpVtbl;
+};
+static MA_INLINE HRESULT ma_IAudioRenderClient_QueryInterface(ma_IAudioRenderClient* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
+static MA_INLINE ULONG ma_IAudioRenderClient_AddRef(ma_IAudioRenderClient* pThis) { return pThis->lpVtbl->AddRef(pThis); }
+static MA_INLINE ULONG ma_IAudioRenderClient_Release(ma_IAudioRenderClient* pThis) { return pThis->lpVtbl->Release(pThis); }
+static MA_INLINE HRESULT ma_IAudioRenderClient_GetBuffer(ma_IAudioRenderClient* pThis, ma_uint32 numFramesRequested, BYTE** ppData) { return pThis->lpVtbl->GetBuffer(pThis, numFramesRequested, ppData); }
+static MA_INLINE HRESULT ma_IAudioRenderClient_ReleaseBuffer(ma_IAudioRenderClient* pThis, ma_uint32 numFramesWritten, DWORD dwFlags) { return pThis->lpVtbl->ReleaseBuffer(pThis, numFramesWritten, dwFlags); }
+
+
+/* IAudioCaptureClient */
+typedef struct
+{
+ /* IUnknown */
+ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioCaptureClient* pThis, const IID* const riid, void** ppObject);
+ ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioCaptureClient* pThis);
+ ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioCaptureClient* pThis);
+
+ /* IAudioRenderClient */
+ HRESULT (STDMETHODCALLTYPE * GetBuffer) (ma_IAudioCaptureClient* pThis, BYTE** ppData, ma_uint32* pNumFramesToRead, DWORD* pFlags, ma_uint64* pDevicePosition, ma_uint64* pQPCPosition);
+ HRESULT (STDMETHODCALLTYPE * ReleaseBuffer) (ma_IAudioCaptureClient* pThis, ma_uint32 numFramesRead);
+ HRESULT (STDMETHODCALLTYPE * GetNextPacketSize)(ma_IAudioCaptureClient* pThis, ma_uint32* pNumFramesInNextPacket);
+} ma_IAudioCaptureClientVtbl;
+struct ma_IAudioCaptureClient
+{
+ ma_IAudioCaptureClientVtbl* lpVtbl;
+};
+static MA_INLINE HRESULT ma_IAudioCaptureClient_QueryInterface(ma_IAudioCaptureClient* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
+static MA_INLINE ULONG ma_IAudioCaptureClient_AddRef(ma_IAudioCaptureClient* pThis) { return pThis->lpVtbl->AddRef(pThis); }
+static MA_INLINE ULONG ma_IAudioCaptureClient_Release(ma_IAudioCaptureClient* pThis) { return pThis->lpVtbl->Release(pThis); }
+static MA_INLINE HRESULT ma_IAudioCaptureClient_GetBuffer(ma_IAudioCaptureClient* pThis, BYTE** ppData, ma_uint32* pNumFramesToRead, DWORD* pFlags, ma_uint64* pDevicePosition, ma_uint64* pQPCPosition) { return pThis->lpVtbl->GetBuffer(pThis, ppData, pNumFramesToRead, pFlags, pDevicePosition, pQPCPosition); }
+static MA_INLINE HRESULT ma_IAudioCaptureClient_ReleaseBuffer(ma_IAudioCaptureClient* pThis, ma_uint32 numFramesRead) { return pThis->lpVtbl->ReleaseBuffer(pThis, numFramesRead); }
+static MA_INLINE HRESULT ma_IAudioCaptureClient_GetNextPacketSize(ma_IAudioCaptureClient* pThis, ma_uint32* pNumFramesInNextPacket) { return pThis->lpVtbl->GetNextPacketSize(pThis, pNumFramesInNextPacket); }
+
+#if !defined(MA_WIN32_DESKTOP) && !defined(MA_WIN32_GDK)
+#include
+typedef struct ma_completion_handler_uwp ma_completion_handler_uwp;
+
+typedef struct
+{
+ /* IUnknown */
+ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_completion_handler_uwp* pThis, const IID* const riid, void** ppObject);
+ ULONG (STDMETHODCALLTYPE * AddRef) (ma_completion_handler_uwp* pThis);
+ ULONG (STDMETHODCALLTYPE * Release) (ma_completion_handler_uwp* pThis);
+
+ /* IActivateAudioInterfaceCompletionHandler */
+ HRESULT (STDMETHODCALLTYPE * ActivateCompleted)(ma_completion_handler_uwp* pThis, ma_IActivateAudioInterfaceAsyncOperation* pActivateOperation);
+} ma_completion_handler_uwp_vtbl;
+struct ma_completion_handler_uwp
+{
+ ma_completion_handler_uwp_vtbl* lpVtbl;
+ MA_ATOMIC(4, ma_uint32) counter;
+ HANDLE hEvent;
+};
+
+static HRESULT STDMETHODCALLTYPE ma_completion_handler_uwp_QueryInterface(ma_completion_handler_uwp* pThis, const IID* const riid, void** ppObject)
+{
+ /*
+ We need to "implement" IAgileObject which is just an indicator that's used internally by WASAPI for some multithreading management. To
+ "implement" this, we just make sure we return pThis when the IAgileObject is requested.
+ */
+ if (!ma_is_guid_equal(riid, &MA_IID_IUnknown) && !ma_is_guid_equal(riid, &MA_IID_IActivateAudioInterfaceCompletionHandler) && !ma_is_guid_equal(riid, &MA_IID_IAgileObject)) {
+ *ppObject = NULL;
+ return E_NOINTERFACE;
+ }
+
+ /* Getting here means the IID is IUnknown or IMMNotificationClient. */
+ *ppObject = (void*)pThis;
+ ((ma_completion_handler_uwp_vtbl*)pThis->lpVtbl)->AddRef(pThis);
+ return S_OK;
+}
+
+static ULONG STDMETHODCALLTYPE ma_completion_handler_uwp_AddRef(ma_completion_handler_uwp* pThis)
+{
+ return (ULONG)c89atomic_fetch_add_32(&pThis->counter, 1) + 1;
+}
+
+static ULONG STDMETHODCALLTYPE ma_completion_handler_uwp_Release(ma_completion_handler_uwp* pThis)
+{
+ ma_uint32 newRefCount = c89atomic_fetch_sub_32(&pThis->counter, 1) - 1;
+ if (newRefCount == 0) {
+ return 0; /* We don't free anything here because we never allocate the object on the heap. */
+ }
+
+ return (ULONG)newRefCount;
+}
+
+static HRESULT STDMETHODCALLTYPE ma_completion_handler_uwp_ActivateCompleted(ma_completion_handler_uwp* pThis, ma_IActivateAudioInterfaceAsyncOperation* pActivateOperation)
+{
+ (void)pActivateOperation;
+ SetEvent(pThis->hEvent);
+ return S_OK;
+}
+
+
+static ma_completion_handler_uwp_vtbl g_maCompletionHandlerVtblInstance = {
+ ma_completion_handler_uwp_QueryInterface,
+ ma_completion_handler_uwp_AddRef,
+ ma_completion_handler_uwp_Release,
+ ma_completion_handler_uwp_ActivateCompleted
+};
+
+static ma_result ma_completion_handler_uwp_init(ma_completion_handler_uwp* pHandler)
+{
+ MA_ASSERT(pHandler != NULL);
+ MA_ZERO_OBJECT(pHandler);
+
+ pHandler->lpVtbl = &g_maCompletionHandlerVtblInstance;
+ pHandler->counter = 1;
+ pHandler->hEvent = CreateEventW(NULL, FALSE, FALSE, NULL);
+ if (pHandler->hEvent == NULL) {
+ return ma_result_from_GetLastError(GetLastError());
+ }
+
+ return MA_SUCCESS;
+}
+
+static void ma_completion_handler_uwp_uninit(ma_completion_handler_uwp* pHandler)
+{
+ if (pHandler->hEvent != NULL) {
+ CloseHandle(pHandler->hEvent);
+ }
+}
+
+static void ma_completion_handler_uwp_wait(ma_completion_handler_uwp* pHandler)
+{
+ WaitForSingleObject(pHandler->hEvent, INFINITE);
+}
+#endif /* !MA_WIN32_DESKTOP */
+
+/* We need a virtual table for our notification client object that's used for detecting changes to the default device. */
+#if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK)
+static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_QueryInterface(ma_IMMNotificationClient* pThis, const IID* const riid, void** ppObject)
+{
+ /*
+ We care about two interfaces - IUnknown and IMMNotificationClient. If the requested IID is something else
+ we just return E_NOINTERFACE. Otherwise we need to increment the reference counter and return S_OK.
+ */
+ if (!ma_is_guid_equal(riid, &MA_IID_IUnknown) && !ma_is_guid_equal(riid, &MA_IID_IMMNotificationClient)) {
+ *ppObject = NULL;
+ return E_NOINTERFACE;
+ }
+
+ /* Getting here means the IID is IUnknown or IMMNotificationClient. */
+ *ppObject = (void*)pThis;
+ ((ma_IMMNotificationClientVtbl*)pThis->lpVtbl)->AddRef(pThis);
+ return S_OK;
+}
+
+static ULONG STDMETHODCALLTYPE ma_IMMNotificationClient_AddRef(ma_IMMNotificationClient* pThis)
+{
+ return (ULONG)c89atomic_fetch_add_32(&pThis->counter, 1) + 1;
+}
+
+static ULONG STDMETHODCALLTYPE ma_IMMNotificationClient_Release(ma_IMMNotificationClient* pThis)
+{
+ ma_uint32 newRefCount = c89atomic_fetch_sub_32(&pThis->counter, 1) - 1;
+ if (newRefCount == 0) {
+ return 0; /* We don't free anything here because we never allocate the object on the heap. */
+ }
+
+ return (ULONG)newRefCount;
+}
+
+static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDeviceStateChanged(ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID, DWORD dwNewState)
+{
+ ma_bool32 isThisDevice = MA_FALSE;
+ ma_bool32 isCapture = MA_FALSE;
+ ma_bool32 isPlayback = MA_FALSE;
+
+#ifdef MA_DEBUG_OUTPUT
+ /*ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "IMMNotificationClient_OnDeviceStateChanged(pDeviceID=%S, dwNewState=%u)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)", (unsigned int)dwNewState);*/
+#endif
+
+ /*
+ There have been reports of a hang when a playback device is disconnected. The idea with this code is to explicitly stop the device if we detect
+ that the device is disabled or has been unplugged.
+ */
+ if (pThis->pDevice->wasapi.allowCaptureAutoStreamRouting && (pThis->pDevice->type == ma_device_type_capture || pThis->pDevice->type == ma_device_type_duplex || pThis->pDevice->type == ma_device_type_loopback)) {
+ isCapture = MA_TRUE;
+ if (wcscmp(pThis->pDevice->capture.id.wasapi, pDeviceID) == 0) {
+ isThisDevice = MA_TRUE;
+ }
+ }
+
+ if (pThis->pDevice->wasapi.allowPlaybackAutoStreamRouting && (pThis->pDevice->type == ma_device_type_playback || pThis->pDevice->type == ma_device_type_duplex)) {
+ isPlayback = MA_TRUE;
+ if (wcscmp(pThis->pDevice->playback.id.wasapi, pDeviceID) == 0) {
+ isThisDevice = MA_TRUE;
+ }
+ }
+
+
+ /*
+ If the device ID matches our device we need to mark our device as detached and stop it. When a
+ device is added in OnDeviceAdded(), we'll restart it. We only mark it as detached if the device
+ was started at the time of being removed.
+ */
+ if (isThisDevice) {
+ if ((dwNewState & MA_MM_DEVICE_STATE_ACTIVE) == 0) {
+ /*
+ Unplugged or otherwise unavailable. Mark as detached if we were in a playing state. We'll
+ use this to determine whether or not we need to automatically start the device when it's
+ plugged back in again.
+ */
+ if (ma_device_get_state(pThis->pDevice) == ma_device_state_started) {
+ if (isPlayback) {
+ pThis->pDevice->wasapi.isDetachedPlayback = MA_TRUE;
+ }
+ if (isCapture) {
+ pThis->pDevice->wasapi.isDetachedCapture = MA_TRUE;
+ }
+
+ ma_device_stop(pThis->pDevice);
+ }
+ }
+
+ if ((dwNewState & MA_MM_DEVICE_STATE_ACTIVE) != 0) {
+ /* The device was activated. If we were detached, we need to start it again. */
+ ma_bool8 tryRestartingDevice = MA_FALSE;
+
+ if (isPlayback) {
+ if (pThis->pDevice->wasapi.isDetachedPlayback) {
+ pThis->pDevice->wasapi.isDetachedPlayback = MA_FALSE;
+ ma_device_reroute__wasapi(pThis->pDevice, ma_device_type_playback);
+ tryRestartingDevice = MA_TRUE;
+ }
+ }
+
+ if (isCapture) {
+ if (pThis->pDevice->wasapi.isDetachedCapture) {
+ pThis->pDevice->wasapi.isDetachedCapture = MA_FALSE;
+ ma_device_reroute__wasapi(pThis->pDevice, (pThis->pDevice->type == ma_device_type_loopback) ? ma_device_type_loopback : ma_device_type_capture);
+ tryRestartingDevice = MA_TRUE;
+ }
+ }
+
+ if (tryRestartingDevice) {
+ if (pThis->pDevice->wasapi.isDetachedPlayback == MA_FALSE && pThis->pDevice->wasapi.isDetachedCapture == MA_FALSE) {
+ ma_device_start(pThis->pDevice);
+ }
+ }
+ }
+ }
+
+ return S_OK;
+}
+
+static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDeviceAdded(ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID)
+{
+#ifdef MA_DEBUG_OUTPUT
+ /*ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "IMMNotificationClient_OnDeviceAdded(pDeviceID=%S)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)");*/
+#endif
+
+ /* We don't need to worry about this event for our purposes. */
+ (void)pThis;
+ (void)pDeviceID;
+ return S_OK;
+}
+
+static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDeviceRemoved(ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID)
+{
+#ifdef MA_DEBUG_OUTPUT
+ /*ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "IMMNotificationClient_OnDeviceRemoved(pDeviceID=%S)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)");*/
+#endif
+
+ /* We don't need to worry about this event for our purposes. */
+ (void)pThis;
+ (void)pDeviceID;
+ return S_OK;
+}
+
+static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDefaultDeviceChanged(ma_IMMNotificationClient* pThis, ma_EDataFlow dataFlow, ma_ERole role, LPCWSTR pDefaultDeviceID)
+{
+#ifdef MA_DEBUG_OUTPUT
+ /*ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "IMMNotificationClient_OnDefaultDeviceChanged(dataFlow=%d, role=%d, pDefaultDeviceID=%S)\n", dataFlow, role, (pDefaultDeviceID != NULL) ? pDefaultDeviceID : L"(NULL)");*/
+#endif
+
+ /* We only ever use the eConsole role in miniaudio. */
+ if (role != ma_eConsole) {
+ ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Stream rerouting: role != eConsole\n");
+ return S_OK;
+ }
+
+ /* We only care about devices with the same data flow and role as the current device. */
+ if ((pThis->pDevice->type == ma_device_type_playback && dataFlow != ma_eRender) ||
+ (pThis->pDevice->type == ma_device_type_capture && dataFlow != ma_eCapture)) {
+ ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Stream rerouting abandoned because dataFlow does match device type.\n");
+ return S_OK;
+ }
+
+ /* Don't do automatic stream routing if we're not allowed. */
+ if ((dataFlow == ma_eRender && pThis->pDevice->wasapi.allowPlaybackAutoStreamRouting == MA_FALSE) ||
+ (dataFlow == ma_eCapture && pThis->pDevice->wasapi.allowCaptureAutoStreamRouting == MA_FALSE)) {
+ ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Stream rerouting abandoned because automatic stream routing has been disabled by the device config.\n");
+ return S_OK;
+ }
+
+ /*
+ Not currently supporting automatic stream routing in exclusive mode. This is not working correctly on my machine due to
+ AUDCLNT_E_DEVICE_IN_USE errors when reinitializing the device. If this is a bug in miniaudio, we can try re-enabling this once
+ it's fixed.
+ */
+ if ((dataFlow == ma_eRender && pThis->pDevice->playback.shareMode == ma_share_mode_exclusive) ||
+ (dataFlow == ma_eCapture && pThis->pDevice->capture.shareMode == ma_share_mode_exclusive)) {
+ ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Stream rerouting abandoned because the device shared mode is exclusive.\n");
+ return S_OK;
+ }
+
+
+
+
+ /*
+ Second attempt at device rerouting. We're going to retrieve the device's state at the time of
+ the route change. We're then going to stop the device, reinitialize the device, and then start
+ it again if the state before stopping was ma_device_state_started.
+ */
+ {
+ ma_uint32 previousState = ma_device_get_state(pThis->pDevice);
+ ma_bool8 restartDevice = MA_FALSE;
+
+ if (previousState == ma_device_state_started) {
+ ma_device_stop(pThis->pDevice);
+ restartDevice = MA_TRUE;
+ }
+
+ if (pDefaultDeviceID != NULL) { /* <-- The input device ID will be null if there's no other device available. */
+ if (dataFlow == ma_eRender) {
+ ma_device_reroute__wasapi(pThis->pDevice, ma_device_type_playback);
+
+ if (pThis->pDevice->wasapi.isDetachedPlayback) {
+ pThis->pDevice->wasapi.isDetachedPlayback = MA_FALSE;
+
+ if (pThis->pDevice->type == ma_device_type_duplex && pThis->pDevice->wasapi.isDetachedCapture) {
+ restartDevice = MA_FALSE; /* It's a duplex device and the capture side is detached. We cannot be restarting the device just yet. */
+ } else {
+ restartDevice = MA_TRUE; /* It's not a duplex device, or the capture side is also attached so we can go ahead and restart the device. */
+ }
+ }
+ } else {
+ ma_device_reroute__wasapi(pThis->pDevice, (pThis->pDevice->type == ma_device_type_loopback) ? ma_device_type_loopback : ma_device_type_capture);
+
+ if (pThis->pDevice->wasapi.isDetachedCapture) {
+ pThis->pDevice->wasapi.isDetachedCapture = MA_FALSE;
+
+ if (pThis->pDevice->type == ma_device_type_duplex && pThis->pDevice->wasapi.isDetachedPlayback) {
+ restartDevice = MA_FALSE; /* It's a duplex device and the playback side is detached. We cannot be restarting the device just yet. */
+ } else {
+ restartDevice = MA_TRUE; /* It's not a duplex device, or the playback side is also attached so we can go ahead and restart the device. */
+ }
+ }
+ }
+
+ if (restartDevice) {
+ ma_device_start(pThis->pDevice);
+ }
+ }
+ }
+
+ return S_OK;
+}
+
+static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnPropertyValueChanged(ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID, const PROPERTYKEY key)
+{
+#ifdef MA_DEBUG_OUTPUT
+ /*ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "IMMNotificationClient_OnPropertyValueChanged(pDeviceID=%S)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)");*/
+#endif
+
+ (void)pThis;
+ (void)pDeviceID;
+ (void)key;
+ return S_OK;
+}
+
+static ma_IMMNotificationClientVtbl g_maNotificationCientVtbl = {
+ ma_IMMNotificationClient_QueryInterface,
+ ma_IMMNotificationClient_AddRef,
+ ma_IMMNotificationClient_Release,
+ ma_IMMNotificationClient_OnDeviceStateChanged,
+ ma_IMMNotificationClient_OnDeviceAdded,
+ ma_IMMNotificationClient_OnDeviceRemoved,
+ ma_IMMNotificationClient_OnDefaultDeviceChanged,
+ ma_IMMNotificationClient_OnPropertyValueChanged
+};
+#endif /* MA_WIN32_DESKTOP */
+
+#if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK)
+typedef ma_IMMDevice ma_WASAPIDeviceInterface;
+#else
+typedef ma_IUnknown ma_WASAPIDeviceInterface;
+#endif
+
+
+#define MA_CONTEXT_COMMAND_QUIT__WASAPI 1
+#define MA_CONTEXT_COMMAND_CREATE_IAUDIOCLIENT__WASAPI 2
+#define MA_CONTEXT_COMMAND_RELEASE_IAUDIOCLIENT__WASAPI 3
+
+static ma_context_command__wasapi ma_context_init_command__wasapi(int code)
+{
+ ma_context_command__wasapi cmd;
+
+ MA_ZERO_OBJECT(&cmd);
+ cmd.code = code;
+
+ return cmd;
+}
+
+static ma_result ma_context_post_command__wasapi(ma_context* pContext, const ma_context_command__wasapi* pCmd)
+{
+ /* For now we are doing everything synchronously, but I might relax this later if the need arises. */
+ ma_result result;
+ ma_bool32 isUsingLocalEvent = MA_FALSE;
+ ma_event localEvent;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pCmd != NULL);
+
+ if (pCmd->pEvent == NULL) {
+ isUsingLocalEvent = MA_TRUE;
+
+ result = ma_event_init(&localEvent);
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to create the event for this command. */
+ }
+ }
+
+ /* Here is where we add the command to the list. If there's not enough room we'll spin until there is. */
+ ma_mutex_lock(&pContext->wasapi.commandLock);
+ {
+ ma_uint32 index;
+
+ /* Spin until we've got some space available. */
+ while (pContext->wasapi.commandCount == ma_countof(pContext->wasapi.commands)) {
+ ma_yield();
+ }
+
+ /* Space is now available. Can safely add to the list. */
+ index = (pContext->wasapi.commandIndex + pContext->wasapi.commandCount) % ma_countof(pContext->wasapi.commands);
+ pContext->wasapi.commands[index] = *pCmd;
+ pContext->wasapi.commands[index].pEvent = &localEvent;
+ pContext->wasapi.commandCount += 1;
+
+ /* Now that the command has been added, release the semaphore so ma_context_next_command__wasapi() can return. */
+ ma_semaphore_release(&pContext->wasapi.commandSem);
+ }
+ ma_mutex_unlock(&pContext->wasapi.commandLock);
+
+ if (isUsingLocalEvent) {
+ ma_event_wait(&localEvent);
+ ma_event_uninit(&localEvent);
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_next_command__wasapi(ma_context* pContext, ma_context_command__wasapi* pCmd)
+{
+ ma_result result = MA_SUCCESS;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pCmd != NULL);
+
+ result = ma_semaphore_wait(&pContext->wasapi.commandSem);
+ if (result == MA_SUCCESS) {
+ ma_mutex_lock(&pContext->wasapi.commandLock);
+ {
+ *pCmd = pContext->wasapi.commands[pContext->wasapi.commandIndex];
+ pContext->wasapi.commandIndex = (pContext->wasapi.commandIndex + 1) % ma_countof(pContext->wasapi.commands);
+ pContext->wasapi.commandCount -= 1;
+ }
+ ma_mutex_unlock(&pContext->wasapi.commandLock);
+ }
+
+ return result;
+}
+
+static ma_thread_result MA_THREADCALL ma_context_command_thread__wasapi(void* pUserData)
+{
+ ma_result result;
+ ma_context* pContext = (ma_context*)pUserData;
+ MA_ASSERT(pContext != NULL);
+
+ for (;;) {
+ ma_context_command__wasapi cmd;
+ result = ma_context_next_command__wasapi(pContext, &cmd);
+ if (result != MA_SUCCESS) {
+ break;
+ }
+
+ switch (cmd.code)
+ {
+ case MA_CONTEXT_COMMAND_QUIT__WASAPI:
+ {
+ /* Do nothing. Handled after the switch. */
+ } break;
+
+ case MA_CONTEXT_COMMAND_CREATE_IAUDIOCLIENT__WASAPI:
+ {
+ if (cmd.data.createAudioClient.deviceType == ma_device_type_playback) {
+ *cmd.data.createAudioClient.pResult = ma_result_from_HRESULT(ma_IAudioClient_GetService((ma_IAudioClient*)cmd.data.createAudioClient.pAudioClient, &MA_IID_IAudioRenderClient, cmd.data.createAudioClient.ppAudioClientService));
+ } else {
+ *cmd.data.createAudioClient.pResult = ma_result_from_HRESULT(ma_IAudioClient_GetService((ma_IAudioClient*)cmd.data.createAudioClient.pAudioClient, &MA_IID_IAudioCaptureClient, cmd.data.createAudioClient.ppAudioClientService));
+ }
+ } break;
+
+ case MA_CONTEXT_COMMAND_RELEASE_IAUDIOCLIENT__WASAPI:
+ {
+ if (cmd.data.releaseAudioClient.deviceType == ma_device_type_playback) {
+ if (cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientPlayback != NULL) {
+ ma_IAudioClient_Release((ma_IAudioClient*)cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientPlayback);
+ cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientPlayback = NULL;
+ }
+ }
+
+ if (cmd.data.releaseAudioClient.deviceType == ma_device_type_capture) {
+ if (cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientCapture != NULL) {
+ ma_IAudioClient_Release((ma_IAudioClient*)cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientCapture);
+ cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientCapture = NULL;
+ }
+ }
+ } break;
+
+ default:
+ {
+ /* Unknown command. Ignore it, but trigger an assert in debug mode so we're aware of it. */
+ MA_ASSERT(MA_FALSE);
+ } break;
+ }
+
+ if (cmd.pEvent != NULL) {
+ ma_event_signal(cmd.pEvent);
+ }
+
+ if (cmd.code == MA_CONTEXT_COMMAND_QUIT__WASAPI) {
+ break; /* Received a quit message. Get out of here. */
+ }
+ }
+
+ return (ma_thread_result)0;
+}
+
+static ma_result ma_device_create_IAudioClient_service__wasapi(ma_context* pContext, ma_device_type deviceType, ma_IAudioClient* pAudioClient, void** ppAudioClientService)
+{
+ ma_result result;
+ ma_result cmdResult;
+ ma_context_command__wasapi cmd = ma_context_init_command__wasapi(MA_CONTEXT_COMMAND_CREATE_IAUDIOCLIENT__WASAPI);
+ cmd.data.createAudioClient.deviceType = deviceType;
+ cmd.data.createAudioClient.pAudioClient = (void*)pAudioClient;
+ cmd.data.createAudioClient.ppAudioClientService = ppAudioClientService;
+ cmd.data.createAudioClient.pResult = &cmdResult; /* Declared locally, but won't be dereferenced after this function returns since execution of the command will wait here. */
+
+ result = ma_context_post_command__wasapi(pContext, &cmd); /* This will not return until the command has actually been run. */
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ return *cmd.data.createAudioClient.pResult;
+}
+
+#if 0 /* Not used at the moment, but leaving here for future use. */
+static ma_result ma_device_release_IAudioClient_service__wasapi(ma_device* pDevice, ma_device_type deviceType)
+{
+ ma_result result;
+ ma_context_command__wasapi cmd = ma_context_init_command__wasapi(MA_CONTEXT_COMMAND_RELEASE_IAUDIOCLIENT__WASAPI);
+ cmd.data.releaseAudioClient.pDevice = pDevice;
+ cmd.data.releaseAudioClient.deviceType = deviceType;
+
+ result = ma_context_post_command__wasapi(pDevice->pContext, &cmd); /* This will not return until the command has actually been run. */
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ return MA_SUCCESS;
+}
+#endif
+
+
+static void ma_add_native_data_format_to_device_info_from_WAVEFORMATEX(const WAVEFORMATEX* pWF, ma_share_mode shareMode, ma_device_info* pInfo)
+{
+ MA_ASSERT(pWF != NULL);
+ MA_ASSERT(pInfo != NULL);
+
+ if (pInfo->nativeDataFormatCount >= ma_countof(pInfo->nativeDataFormats)) {
+ return; /* Too many data formats. Need to ignore this one. Don't think this should ever happen with WASAPI. */
+ }
+
+ pInfo->nativeDataFormats[pInfo->nativeDataFormatCount].format = ma_format_from_WAVEFORMATEX(pWF);
+ pInfo->nativeDataFormats[pInfo->nativeDataFormatCount].channels = pWF->nChannels;
+ pInfo->nativeDataFormats[pInfo->nativeDataFormatCount].sampleRate = pWF->nSamplesPerSec;
+ pInfo->nativeDataFormats[pInfo->nativeDataFormatCount].flags = (shareMode == ma_share_mode_exclusive) ? MA_DATA_FORMAT_FLAG_EXCLUSIVE_MODE : 0;
+ pInfo->nativeDataFormatCount += 1;
+}
+
+static ma_result ma_context_get_device_info_from_IAudioClient__wasapi(ma_context* pContext, /*ma_IMMDevice**/void* pMMDevice, ma_IAudioClient* pAudioClient, ma_device_info* pInfo)
+{
+ HRESULT hr;
+ WAVEFORMATEX* pWF = NULL;
+
+ MA_ASSERT(pAudioClient != NULL);
+ MA_ASSERT(pInfo != NULL);
+
+ /* Shared Mode. We use GetMixFormat() here. */
+ hr = ma_IAudioClient_GetMixFormat((ma_IAudioClient*)pAudioClient, (WAVEFORMATEX**)&pWF);
+ if (SUCCEEDED(hr)) {
+ ma_add_native_data_format_to_device_info_from_WAVEFORMATEX(pWF, ma_share_mode_shared, pInfo);
+ } else {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve mix format for device info retrieval.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ /*
+ Exlcusive Mode. We repeatedly call IsFormatSupported() here. This is not currently supported on
+ UWP. Failure to retrieve the exclusive mode format is not considered an error, so from here on
+ out, MA_SUCCESS is guaranteed to be returned.
+ */
+ #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK)
+ {
+ ma_IPropertyStore *pProperties;
+
+ /*
+ The first thing to do is get the format from PKEY_AudioEngine_DeviceFormat. This should give us a channel count we assume is
+ correct which will simplify our searching.
+ */
+ hr = ma_IMMDevice_OpenPropertyStore((ma_IMMDevice*)pMMDevice, STGM_READ, &pProperties);
+ if (SUCCEEDED(hr)) {
+ PROPVARIANT var;
+ ma_PropVariantInit(&var);
+
+ hr = ma_IPropertyStore_GetValue(pProperties, &MA_PKEY_AudioEngine_DeviceFormat, &var);
+ if (SUCCEEDED(hr)) {
+ pWF = (WAVEFORMATEX*)var.blob.pBlobData;
+
+ /*
+ In my testing, the format returned by PKEY_AudioEngine_DeviceFormat is suitable for exclusive mode so we check this format
+ first. If this fails, fall back to a search.
+ */
+ hr = ma_IAudioClient_IsFormatSupported((ma_IAudioClient*)pAudioClient, MA_AUDCLNT_SHAREMODE_EXCLUSIVE, pWF, NULL);
+ if (SUCCEEDED(hr)) {
+ /* The format returned by PKEY_AudioEngine_DeviceFormat is supported. */
+ ma_add_native_data_format_to_device_info_from_WAVEFORMATEX(pWF, ma_share_mode_exclusive, pInfo);
+ } else {
+ /*
+ The format returned by PKEY_AudioEngine_DeviceFormat is not supported, so fall back to a search. We assume the channel
+ count returned by MA_PKEY_AudioEngine_DeviceFormat is valid and correct. For simplicity we're only returning one format.
+ */
+ ma_uint32 channels = pWF->nChannels;
+ ma_channel defaultChannelMap[MA_MAX_CHANNELS];
+ WAVEFORMATEXTENSIBLE wf;
+ ma_bool32 found;
+ ma_uint32 iFormat;
+
+ /* Make sure we don't overflow the channel map. */
+ if (channels > MA_MAX_CHANNELS) {
+ channels = MA_MAX_CHANNELS;
+ }
+
+ ma_channel_map_init_standard(ma_standard_channel_map_microsoft, defaultChannelMap, ma_countof(defaultChannelMap), channels);
+
+ MA_ZERO_OBJECT(&wf);
+ wf.Format.cbSize = sizeof(wf);
+ wf.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
+ wf.Format.nChannels = (WORD)channels;
+ wf.dwChannelMask = ma_channel_map_to_channel_mask__win32(defaultChannelMap, channels);
+
+ found = MA_FALSE;
+ for (iFormat = 0; iFormat < ma_countof(g_maFormatPriorities); ++iFormat) {
+ ma_format format = g_maFormatPriorities[iFormat];
+ ma_uint32 iSampleRate;
+
+ wf.Format.wBitsPerSample = (WORD)(ma_get_bytes_per_sample(format)*8);
+ wf.Format.nBlockAlign = (WORD)(wf.Format.nChannels * wf.Format.wBitsPerSample / 8);
+ wf.Format.nAvgBytesPerSec = wf.Format.nBlockAlign * wf.Format.nSamplesPerSec;
+ wf.Samples.wValidBitsPerSample = /*(format == ma_format_s24_32) ? 24 :*/ wf.Format.wBitsPerSample;
+ if (format == ma_format_f32) {
+ wf.SubFormat = MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
+ } else {
+ wf.SubFormat = MA_GUID_KSDATAFORMAT_SUBTYPE_PCM;
+ }
+
+ for (iSampleRate = 0; iSampleRate < ma_countof(g_maStandardSampleRatePriorities); ++iSampleRate) {
+ wf.Format.nSamplesPerSec = g_maStandardSampleRatePriorities[iSampleRate];
+
+ hr = ma_IAudioClient_IsFormatSupported((ma_IAudioClient*)pAudioClient, MA_AUDCLNT_SHAREMODE_EXCLUSIVE, (WAVEFORMATEX*)&wf, NULL);
+ if (SUCCEEDED(hr)) {
+ ma_add_native_data_format_to_device_info_from_WAVEFORMATEX((WAVEFORMATEX*)&wf, ma_share_mode_exclusive, pInfo);
+ found = MA_TRUE;
+ break;
+ }
+ }
+
+ if (found) {
+ break;
+ }
+ }
+
+ ma_PropVariantClear(pContext, &var);
+
+ if (!found) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_WARNING, "[WASAPI] Failed to find suitable device format for device info retrieval.");
+ }
+ }
+ } else {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_WARNING, "[WASAPI] Failed to retrieve device format for device info retrieval.");
+ }
+
+ ma_IPropertyStore_Release(pProperties);
+ } else {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_WARNING, "[WASAPI] Failed to open property store for device info retrieval.");
+ }
+ }
+ #endif
+
+ return MA_SUCCESS;
+}
+
+#if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK)
+static ma_EDataFlow ma_device_type_to_EDataFlow(ma_device_type deviceType)
+{
+ if (deviceType == ma_device_type_playback) {
+ return ma_eRender;
+ } else if (deviceType == ma_device_type_capture) {
+ return ma_eCapture;
+ } else {
+ MA_ASSERT(MA_FALSE);
+ return ma_eRender; /* Should never hit this. */
+ }
+}
+
+static ma_result ma_context_create_IMMDeviceEnumerator__wasapi(ma_context* pContext, ma_IMMDeviceEnumerator** ppDeviceEnumerator)
+{
+ HRESULT hr;
+ ma_IMMDeviceEnumerator* pDeviceEnumerator;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(ppDeviceEnumerator != NULL);
+
+ *ppDeviceEnumerator = NULL; /* Safety. */
+
+ hr = ma_CoCreateInstance(pContext, MA_CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, MA_IID_IMMDeviceEnumerator, (void**)&pDeviceEnumerator);
+ if (FAILED(hr)) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create device enumerator.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ *ppDeviceEnumerator = pDeviceEnumerator;
+
+ return MA_SUCCESS;
+}
+
+static LPWSTR ma_context_get_default_device_id_from_IMMDeviceEnumerator__wasapi(ma_context* pContext, ma_IMMDeviceEnumerator* pDeviceEnumerator, ma_device_type deviceType)
+{
+ HRESULT hr;
+ ma_IMMDevice* pMMDefaultDevice = NULL;
+ LPWSTR pDefaultDeviceID = NULL;
+ ma_EDataFlow dataFlow;
+ ma_ERole role;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pDeviceEnumerator != NULL);
+
+ (void)pContext;
+
+ /* Grab the EDataFlow type from the device type. */
+ dataFlow = ma_device_type_to_EDataFlow(deviceType);
+
+ /* The role is always eConsole, but we may make this configurable later. */
+ role = ma_eConsole;
+
+ hr = ma_IMMDeviceEnumerator_GetDefaultAudioEndpoint(pDeviceEnumerator, dataFlow, role, &pMMDefaultDevice);
+ if (FAILED(hr)) {
+ return NULL;
+ }
+
+ hr = ma_IMMDevice_GetId(pMMDefaultDevice, &pDefaultDeviceID);
+
+ ma_IMMDevice_Release(pMMDefaultDevice);
+ pMMDefaultDevice = NULL;
+
+ if (FAILED(hr)) {
+ return NULL;
+ }
+
+ return pDefaultDeviceID;
+}
+
+static LPWSTR ma_context_get_default_device_id__wasapi(ma_context* pContext, ma_device_type deviceType) /* Free the returned pointer with ma_CoTaskMemFree() */
+{
+ ma_result result;
+ ma_IMMDeviceEnumerator* pDeviceEnumerator;
+ LPWSTR pDefaultDeviceID = NULL;
+
+ MA_ASSERT(pContext != NULL);
+
+ result = ma_context_create_IMMDeviceEnumerator__wasapi(pContext, &pDeviceEnumerator);
+ if (result != MA_SUCCESS) {
+ return NULL;
+ }
+
+ pDefaultDeviceID = ma_context_get_default_device_id_from_IMMDeviceEnumerator__wasapi(pContext, pDeviceEnumerator, deviceType);
+
+ ma_IMMDeviceEnumerator_Release(pDeviceEnumerator);
+ return pDefaultDeviceID;
+}
+
+static ma_result ma_context_get_MMDevice__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_IMMDevice** ppMMDevice)
+{
+ ma_IMMDeviceEnumerator* pDeviceEnumerator;
+ HRESULT hr;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(ppMMDevice != NULL);
+
+ hr = ma_CoCreateInstance(pContext, MA_CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, MA_IID_IMMDeviceEnumerator, (void**)&pDeviceEnumerator);
+ if (FAILED(hr)) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create IMMDeviceEnumerator.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ if (pDeviceID == NULL) {
+ hr = ma_IMMDeviceEnumerator_GetDefaultAudioEndpoint(pDeviceEnumerator, (deviceType == ma_device_type_capture) ? ma_eCapture : ma_eRender, ma_eConsole, ppMMDevice);
+ } else {
+ hr = ma_IMMDeviceEnumerator_GetDevice(pDeviceEnumerator, pDeviceID->wasapi, ppMMDevice);
+ }
+
+ ma_IMMDeviceEnumerator_Release(pDeviceEnumerator);
+ if (FAILED(hr)) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve IMMDevice.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_get_device_id_from_MMDevice__wasapi(ma_context* pContext, ma_IMMDevice* pMMDevice, ma_device_id* pDeviceID)
+{
+ LPWSTR pDeviceIDString;
+ HRESULT hr;
+
+ MA_ASSERT(pDeviceID != NULL);
+
+ hr = ma_IMMDevice_GetId(pMMDevice, &pDeviceIDString);
+ if (SUCCEEDED(hr)) {
+ size_t idlen = wcslen(pDeviceIDString);
+ if (idlen+1 > ma_countof(pDeviceID->wasapi)) {
+ ma_CoTaskMemFree(pContext, pDeviceIDString);
+ MA_ASSERT(MA_FALSE); /* NOTE: If this is triggered, please report it. It means the format of the ID must haved change and is too long to fit in our fixed sized buffer. */
+ return MA_ERROR;
+ }
+
+ MA_COPY_MEMORY(pDeviceID->wasapi, pDeviceIDString, idlen * sizeof(wchar_t));
+ pDeviceID->wasapi[idlen] = '\0';
+
+ ma_CoTaskMemFree(pContext, pDeviceIDString);
+
+ return MA_SUCCESS;
+ }
+
+ return MA_ERROR;
+}
+
+static ma_result ma_context_get_device_info_from_MMDevice__wasapi(ma_context* pContext, ma_IMMDevice* pMMDevice, LPWSTR pDefaultDeviceID, ma_bool32 onlySimpleInfo, ma_device_info* pInfo)
+{
+ ma_result result;
+ HRESULT hr;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pMMDevice != NULL);
+ MA_ASSERT(pInfo != NULL);
+
+ /* ID. */
+ result = ma_context_get_device_id_from_MMDevice__wasapi(pContext, pMMDevice, &pInfo->id);
+ if (result == MA_SUCCESS) {
+ if (pDefaultDeviceID != NULL) {
+ if (wcscmp(pInfo->id.wasapi, pDefaultDeviceID) == 0) {
+ pInfo->isDefault = MA_TRUE;
+ }
+ }
+ }
+
+ /* Description / Friendly Name */
+ {
+ ma_IPropertyStore *pProperties;
+ hr = ma_IMMDevice_OpenPropertyStore(pMMDevice, STGM_READ, &pProperties);
+ if (SUCCEEDED(hr)) {
+ PROPVARIANT var;
+
+ ma_PropVariantInit(&var);
+ hr = ma_IPropertyStore_GetValue(pProperties, &MA_PKEY_Device_FriendlyName, &var);
+ if (SUCCEEDED(hr)) {
+ WideCharToMultiByte(CP_UTF8, 0, var.pwszVal, -1, pInfo->name, sizeof(pInfo->name), 0, FALSE);
+ ma_PropVariantClear(pContext, &var);
+ }
+
+ ma_IPropertyStore_Release(pProperties);
+ }
+ }
+
+ /* Format */
+ if (!onlySimpleInfo) {
+ ma_IAudioClient* pAudioClient;
+ hr = ma_IMMDevice_Activate(pMMDevice, &MA_IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&pAudioClient);
+ if (SUCCEEDED(hr)) {
+ result = ma_context_get_device_info_from_IAudioClient__wasapi(pContext, pMMDevice, pAudioClient, pInfo);
+
+ ma_IAudioClient_Release(pAudioClient);
+ return result;
+ } else {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to activate audio client for device info retrieval.");
+ return ma_result_from_HRESULT(hr);
+ }
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_enumerate_devices_by_type__wasapi(ma_context* pContext, ma_IMMDeviceEnumerator* pDeviceEnumerator, ma_device_type deviceType, ma_enum_devices_callback_proc callback, void* pUserData)
+{
+ ma_result result = MA_SUCCESS;
+ UINT deviceCount;
+ HRESULT hr;
+ ma_uint32 iDevice;
+ LPWSTR pDefaultDeviceID = NULL;
+ ma_IMMDeviceCollection* pDeviceCollection = NULL;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(callback != NULL);
+
+ /* Grab the default device. We use this to know whether or not flag the returned device info as being the default. */
+ pDefaultDeviceID = ma_context_get_default_device_id_from_IMMDeviceEnumerator__wasapi(pContext, pDeviceEnumerator, deviceType);
+
+ /* We need to enumerate the devices which returns a device collection. */
+ hr = ma_IMMDeviceEnumerator_EnumAudioEndpoints(pDeviceEnumerator, ma_device_type_to_EDataFlow(deviceType), MA_MM_DEVICE_STATE_ACTIVE, &pDeviceCollection);
+ if (SUCCEEDED(hr)) {
+ hr = ma_IMMDeviceCollection_GetCount(pDeviceCollection, &deviceCount);
+ if (FAILED(hr)) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to get device count.");
+ result = ma_result_from_HRESULT(hr);
+ goto done;
+ }
+
+ for (iDevice = 0; iDevice < deviceCount; ++iDevice) {
+ ma_device_info deviceInfo;
+ ma_IMMDevice* pMMDevice;
+
+ MA_ZERO_OBJECT(&deviceInfo);
+
+ hr = ma_IMMDeviceCollection_Item(pDeviceCollection, iDevice, &pMMDevice);
+ if (SUCCEEDED(hr)) {
+ result = ma_context_get_device_info_from_MMDevice__wasapi(pContext, pMMDevice, pDefaultDeviceID, MA_TRUE, &deviceInfo); /* MA_TRUE = onlySimpleInfo. */
+
+ ma_IMMDevice_Release(pMMDevice);
+ if (result == MA_SUCCESS) {
+ ma_bool32 cbResult = callback(pContext, deviceType, &deviceInfo, pUserData);
+ if (cbResult == MA_FALSE) {
+ break;
+ }
+ }
+ }
+ }
+ }
+
+done:
+ if (pDefaultDeviceID != NULL) {
+ ma_CoTaskMemFree(pContext, pDefaultDeviceID);
+ pDefaultDeviceID = NULL;
+ }
+
+ if (pDeviceCollection != NULL) {
+ ma_IMMDeviceCollection_Release(pDeviceCollection);
+ pDeviceCollection = NULL;
+ }
+
+ return result;
+}
+
+static ma_result ma_context_get_IAudioClient_Desktop__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_IAudioClient** ppAudioClient, ma_IMMDevice** ppMMDevice)
+{
+ ma_result result;
+ HRESULT hr;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(ppAudioClient != NULL);
+ MA_ASSERT(ppMMDevice != NULL);
+
+ result = ma_context_get_MMDevice__wasapi(pContext, deviceType, pDeviceID, ppMMDevice);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ hr = ma_IMMDevice_Activate(*ppMMDevice, &MA_IID_IAudioClient, CLSCTX_ALL, NULL, (void**)ppAudioClient);
+ if (FAILED(hr)) {
+ return ma_result_from_HRESULT(hr);
+ }
+
+ return MA_SUCCESS;
+}
+#else
+static ma_result ma_context_get_IAudioClient_UWP__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_IAudioClient** ppAudioClient, ma_IUnknown** ppActivatedInterface)
+{
+ ma_IActivateAudioInterfaceAsyncOperation *pAsyncOp = NULL;
+ ma_completion_handler_uwp completionHandler;
+ IID iid;
+ LPOLESTR iidStr;
+ HRESULT hr;
+ ma_result result;
+ HRESULT activateResult;
+ ma_IUnknown* pActivatedInterface;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(ppAudioClient != NULL);
+
+ if (pDeviceID != NULL) {
+ MA_COPY_MEMORY(&iid, pDeviceID->wasapi, sizeof(iid));
+ } else {
+ if (deviceType == ma_device_type_playback) {
+ iid = MA_IID_DEVINTERFACE_AUDIO_RENDER;
+ } else {
+ iid = MA_IID_DEVINTERFACE_AUDIO_CAPTURE;
+ }
+ }
+
+#if defined(__cplusplus)
+ hr = StringFromIID(iid, &iidStr);
+#else
+ hr = StringFromIID(&iid, &iidStr);
+#endif
+ if (FAILED(hr)) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to convert device IID to string for ActivateAudioInterfaceAsync(). Out of memory.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ result = ma_completion_handler_uwp_init(&completionHandler);
+ if (result != MA_SUCCESS) {
+ ma_CoTaskMemFree(pContext, iidStr);
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create event for waiting for ActivateAudioInterfaceAsync().");
+ return result;
+ }
+
+#if defined(__cplusplus)
+ hr = ActivateAudioInterfaceAsync(iidStr, MA_IID_IAudioClient, NULL, (IActivateAudioInterfaceCompletionHandler*)&completionHandler, (IActivateAudioInterfaceAsyncOperation**)&pAsyncOp);
+#else
+ hr = ActivateAudioInterfaceAsync(iidStr, &MA_IID_IAudioClient, NULL, (IActivateAudioInterfaceCompletionHandler*)&completionHandler, (IActivateAudioInterfaceAsyncOperation**)&pAsyncOp);
+#endif
+ if (FAILED(hr)) {
+ ma_completion_handler_uwp_uninit(&completionHandler);
+ ma_CoTaskMemFree(pContext, iidStr);
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] ActivateAudioInterfaceAsync() failed.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ ma_CoTaskMemFree(pContext, iidStr);
+
+ /* Wait for the async operation for finish. */
+ ma_completion_handler_uwp_wait(&completionHandler);
+ ma_completion_handler_uwp_uninit(&completionHandler);
+
+ hr = ma_IActivateAudioInterfaceAsyncOperation_GetActivateResult(pAsyncOp, &activateResult, &pActivatedInterface);
+ ma_IActivateAudioInterfaceAsyncOperation_Release(pAsyncOp);
+
+ if (FAILED(hr) || FAILED(activateResult)) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to activate device.");
+ return FAILED(hr) ? ma_result_from_HRESULT(hr) : ma_result_from_HRESULT(activateResult);
+ }
+
+ /* Here is where we grab the IAudioClient interface. */
+ hr = ma_IUnknown_QueryInterface(pActivatedInterface, &MA_IID_IAudioClient, (void**)ppAudioClient);
+ if (FAILED(hr)) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to query IAudioClient interface.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ if (ppActivatedInterface) {
+ *ppActivatedInterface = pActivatedInterface;
+ } else {
+ ma_IUnknown_Release(pActivatedInterface);
+ }
+
+ return MA_SUCCESS;
+}
+#endif
+
+static ma_result ma_context_get_IAudioClient__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_IAudioClient** ppAudioClient, ma_WASAPIDeviceInterface** ppDeviceInterface)
+{
+#if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK)
+ return ma_context_get_IAudioClient_Desktop__wasapi(pContext, deviceType, pDeviceID, ppAudioClient, ppDeviceInterface);
+#else
+ return ma_context_get_IAudioClient_UWP__wasapi(pContext, deviceType, pDeviceID, ppAudioClient, ppDeviceInterface);
+#endif
+}
+
+
+static ma_result ma_context_enumerate_devices__wasapi(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+{
+ /* Different enumeration for desktop and UWP. */
+#if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK)
+ /* Desktop */
+ HRESULT hr;
+ ma_IMMDeviceEnumerator* pDeviceEnumerator;
+
+ hr = ma_CoCreateInstance(pContext, MA_CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, MA_IID_IMMDeviceEnumerator, (void**)&pDeviceEnumerator);
+ if (FAILED(hr)) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create device enumerator.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ ma_context_enumerate_devices_by_type__wasapi(pContext, pDeviceEnumerator, ma_device_type_playback, callback, pUserData);
+ ma_context_enumerate_devices_by_type__wasapi(pContext, pDeviceEnumerator, ma_device_type_capture, callback, pUserData);
+
+ ma_IMMDeviceEnumerator_Release(pDeviceEnumerator);
+#else
+ /*
+ UWP
+
+ The MMDevice API is only supported on desktop applications. For now, while I'm still figuring out how to properly enumerate
+ over devices without using MMDevice, I'm restricting devices to defaults.
+
+ Hint: DeviceInformation::FindAllAsync() with DeviceClass.AudioCapture/AudioRender. https://blogs.windows.com/buildingapps/2014/05/15/real-time-audio-in-windows-store-and-windows-phone-apps/
+ */
+ if (callback) {
+ ma_bool32 cbResult = MA_TRUE;
+
+ /* Playback. */
+ if (cbResult) {
+ ma_device_info deviceInfo;
+ MA_ZERO_OBJECT(&deviceInfo);
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
+ deviceInfo.isDefault = MA_TRUE;
+ cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
+ }
+
+ /* Capture. */
+ if (cbResult) {
+ ma_device_info deviceInfo;
+ MA_ZERO_OBJECT(&deviceInfo);
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
+ deviceInfo.isDefault = MA_TRUE;
+ cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
+ }
+ }
+#endif
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_get_device_info__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+{
+#if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK)
+ ma_result result;
+ ma_IMMDevice* pMMDevice = NULL;
+ LPWSTR pDefaultDeviceID = NULL;
+
+ result = ma_context_get_MMDevice__wasapi(pContext, deviceType, pDeviceID, &pMMDevice);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ /* We need the default device ID so we can set the isDefault flag in the device info. */
+ pDefaultDeviceID = ma_context_get_default_device_id__wasapi(pContext, deviceType);
+
+ result = ma_context_get_device_info_from_MMDevice__wasapi(pContext, pMMDevice, pDefaultDeviceID, MA_FALSE, pDeviceInfo); /* MA_FALSE = !onlySimpleInfo. */
+
+ if (pDefaultDeviceID != NULL) {
+ ma_CoTaskMemFree(pContext, pDefaultDeviceID);
+ pDefaultDeviceID = NULL;
+ }
+
+ ma_IMMDevice_Release(pMMDevice);
+
+ return result;
+#else
+ ma_IAudioClient* pAudioClient;
+ ma_result result;
+
+ /* UWP currently only uses default devices. */
+ if (deviceType == ma_device_type_playback) {
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
+ } else {
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
+ }
+
+ result = ma_context_get_IAudioClient_UWP__wasapi(pContext, deviceType, pDeviceID, &pAudioClient, NULL);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ result = ma_context_get_device_info_from_IAudioClient__wasapi(pContext, NULL, pAudioClient, pDeviceInfo);
+
+ pDeviceInfo->isDefault = MA_TRUE; /* UWP only supports default devices. */
+
+ ma_IAudioClient_Release(pAudioClient);
+ return result;
+#endif
+}
+
+static ma_result ma_device_uninit__wasapi(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
+
+#if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK)
+ if (pDevice->wasapi.pDeviceEnumerator) {
+ ((ma_IMMDeviceEnumerator*)pDevice->wasapi.pDeviceEnumerator)->lpVtbl->UnregisterEndpointNotificationCallback((ma_IMMDeviceEnumerator*)pDevice->wasapi.pDeviceEnumerator, &pDevice->wasapi.notificationClient);
+ ma_IMMDeviceEnumerator_Release((ma_IMMDeviceEnumerator*)pDevice->wasapi.pDeviceEnumerator);
+ }
+#endif
+
+ if (pDevice->wasapi.pRenderClient) {
+ if (pDevice->wasapi.pMappedBufferPlayback != NULL) {
+ ma_IAudioRenderClient_ReleaseBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, pDevice->wasapi.mappedBufferPlaybackCap, 0);
+ pDevice->wasapi.pMappedBufferPlayback = NULL;
+ pDevice->wasapi.mappedBufferPlaybackCap = 0;
+ pDevice->wasapi.mappedBufferPlaybackLen = 0;
+ }
+
+ ma_IAudioRenderClient_Release((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient);
+ }
+ if (pDevice->wasapi.pCaptureClient) {
+ if (pDevice->wasapi.pMappedBufferCapture != NULL) {
+ ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, pDevice->wasapi.mappedBufferCaptureCap);
+ pDevice->wasapi.pMappedBufferCapture = NULL;
+ pDevice->wasapi.mappedBufferCaptureCap = 0;
+ pDevice->wasapi.mappedBufferCaptureLen = 0;
+ }
+
+ ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient);
+ }
+
+ if (pDevice->wasapi.pAudioClientPlayback) {
+ ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback);
+ }
+ if (pDevice->wasapi.pAudioClientCapture) {
+ ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture);
+ }
+
+ if (pDevice->wasapi.hEventPlayback) {
+ CloseHandle(pDevice->wasapi.hEventPlayback);
+ }
+ if (pDevice->wasapi.hEventCapture) {
+ CloseHandle(pDevice->wasapi.hEventCapture);
+ }
+
+ return MA_SUCCESS;
+}
+
+
+typedef struct
+{
+ /* Input. */
+ ma_format formatIn;
+ ma_uint32 channelsIn;
+ ma_uint32 sampleRateIn;
+ ma_channel channelMapIn[MA_MAX_CHANNELS];
+ ma_uint32 periodSizeInFramesIn;
+ ma_uint32 periodSizeInMillisecondsIn;
+ ma_uint32 periodsIn;
+ ma_share_mode shareMode;
+ ma_performance_profile performanceProfile;
+ ma_bool32 noAutoConvertSRC;
+ ma_bool32 noDefaultQualitySRC;
+ ma_bool32 noHardwareOffloading;
+
+ /* Output. */
+ ma_IAudioClient* pAudioClient;
+ ma_IAudioRenderClient* pRenderClient;
+ ma_IAudioCaptureClient* pCaptureClient;
+ ma_format formatOut;
+ ma_uint32 channelsOut;
+ ma_uint32 sampleRateOut;
+ ma_channel channelMapOut[MA_MAX_CHANNELS];
+ ma_uint32 periodSizeInFramesOut;
+ ma_uint32 periodsOut;
+ ma_bool32 usingAudioClient3;
+ char deviceName[256];
+ ma_device_id id;
+} ma_device_init_internal_data__wasapi;
+
+static ma_result ma_device_init_internal__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_init_internal_data__wasapi* pData)
+{
+ HRESULT hr;
+ ma_result result = MA_SUCCESS;
+ const char* errorMsg = "";
+ MA_AUDCLNT_SHAREMODE shareMode = MA_AUDCLNT_SHAREMODE_SHARED;
+ DWORD streamFlags = 0;
+ MA_REFERENCE_TIME periodDurationInMicroseconds;
+ ma_bool32 wasInitializedUsingIAudioClient3 = MA_FALSE;
+ WAVEFORMATEXTENSIBLE wf;
+ ma_WASAPIDeviceInterface* pDeviceInterface = NULL;
+ ma_IAudioClient2* pAudioClient2;
+ ma_uint32 nativeSampleRate;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pData != NULL);
+
+ /* This function is only used to initialize one device type: either playback, capture or loopback. Never full-duplex. */
+ if (deviceType == ma_device_type_duplex) {
+ return MA_INVALID_ARGS;
+ }
+
+ pData->pAudioClient = NULL;
+ pData->pRenderClient = NULL;
+ pData->pCaptureClient = NULL;
+
+ streamFlags = MA_AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
+ if (!pData->noAutoConvertSRC && pData->sampleRateIn != 0 && pData->shareMode != ma_share_mode_exclusive) { /* <-- Exclusive streams must use the native sample rate. */
+ streamFlags |= MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM;
+ }
+ if (!pData->noDefaultQualitySRC && pData->sampleRateIn != 0 && (streamFlags & MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM) != 0) {
+ streamFlags |= MA_AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY;
+ }
+ if (deviceType == ma_device_type_loopback) {
+ streamFlags |= MA_AUDCLNT_STREAMFLAGS_LOOPBACK;
+ }
+
+ result = ma_context_get_IAudioClient__wasapi(pContext, deviceType, pDeviceID, &pData->pAudioClient, &pDeviceInterface);
+ if (result != MA_SUCCESS) {
+ goto done;
+ }
+
+ MA_ZERO_OBJECT(&wf);
+
+ /* Try enabling hardware offloading. */
+ if (!pData->noHardwareOffloading) {
+ hr = ma_IAudioClient_QueryInterface(pData->pAudioClient, &MA_IID_IAudioClient2, (void**)&pAudioClient2);
+ if (SUCCEEDED(hr)) {
+ BOOL isHardwareOffloadingSupported = 0;
+ hr = ma_IAudioClient2_IsOffloadCapable(pAudioClient2, MA_AudioCategory_Other, &isHardwareOffloadingSupported);
+ if (SUCCEEDED(hr) && isHardwareOffloadingSupported) {
+ ma_AudioClientProperties clientProperties;
+ MA_ZERO_OBJECT(&clientProperties);
+ clientProperties.cbSize = sizeof(clientProperties);
+ clientProperties.bIsOffload = 1;
+ clientProperties.eCategory = MA_AudioCategory_Other;
+ ma_IAudioClient2_SetClientProperties(pAudioClient2, &clientProperties);
+ }
+
+ pAudioClient2->lpVtbl->Release(pAudioClient2);
+ }
+ }
+
+ /* Here is where we try to determine the best format to use with the device. If the client if wanting exclusive mode, first try finding the best format for that. If this fails, fall back to shared mode. */
+ result = MA_FORMAT_NOT_SUPPORTED;
+ if (pData->shareMode == ma_share_mode_exclusive) {
+ #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK)
+ /* In exclusive mode on desktop we always use the backend's native format. */
+ ma_IPropertyStore* pStore = NULL;
+ hr = ma_IMMDevice_OpenPropertyStore(pDeviceInterface, STGM_READ, &pStore);
+ if (SUCCEEDED(hr)) {
+ PROPVARIANT prop;
+ ma_PropVariantInit(&prop);
+ hr = ma_IPropertyStore_GetValue(pStore, &MA_PKEY_AudioEngine_DeviceFormat, &prop);
+ if (SUCCEEDED(hr)) {
+ WAVEFORMATEX* pActualFormat = (WAVEFORMATEX*)prop.blob.pBlobData;
+ hr = ma_IAudioClient_IsFormatSupported((ma_IAudioClient*)pData->pAudioClient, MA_AUDCLNT_SHAREMODE_EXCLUSIVE, pActualFormat, NULL);
+ if (SUCCEEDED(hr)) {
+ MA_COPY_MEMORY(&wf, pActualFormat, sizeof(WAVEFORMATEXTENSIBLE));
+ }
+
+ ma_PropVariantClear(pContext, &prop);
+ }
+
+ ma_IPropertyStore_Release(pStore);
+ }
+ #else
+ /*
+ I do not know how to query the device's native format on UWP so for now I'm just disabling support for
+ exclusive mode. The alternative is to enumerate over different formats and check IsFormatSupported()
+ until you find one that works.
+
+ TODO: Add support for exclusive mode to UWP.
+ */
+ hr = S_FALSE;
+ #endif
+
+ if (hr == S_OK) {
+ shareMode = MA_AUDCLNT_SHAREMODE_EXCLUSIVE;
+ result = MA_SUCCESS;
+ } else {
+ result = MA_SHARE_MODE_NOT_SUPPORTED;
+ }
+ } else {
+ /* In shared mode we are always using the format reported by the operating system. */
+ WAVEFORMATEXTENSIBLE* pNativeFormat = NULL;
+ hr = ma_IAudioClient_GetMixFormat((ma_IAudioClient*)pData->pAudioClient, (WAVEFORMATEX**)&pNativeFormat);
+ if (hr != S_OK) {
+ result = MA_FORMAT_NOT_SUPPORTED;
+ } else {
+ MA_COPY_MEMORY(&wf, pNativeFormat, sizeof(wf));
+ result = MA_SUCCESS;
+ }
+
+ ma_CoTaskMemFree(pContext, pNativeFormat);
+
+ shareMode = MA_AUDCLNT_SHAREMODE_SHARED;
+ }
+
+ /* Return an error if we still haven't found a format. */
+ if (result != MA_SUCCESS) {
+ errorMsg = "[WASAPI] Failed to find best device mix format.";
+ goto done;
+ }
+
+ /*
+ Override the native sample rate with the one requested by the caller, but only if we're not using the default sample rate. We'll use
+ WASAPI to perform the sample rate conversion.
+ */
+ nativeSampleRate = wf.Format.nSamplesPerSec;
+ if (streamFlags & MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM) {
+ wf.Format.nSamplesPerSec = (pData->sampleRateIn != 0) ? pData->sampleRateIn : MA_DEFAULT_SAMPLE_RATE;
+ wf.Format.nAvgBytesPerSec = wf.Format.nSamplesPerSec * wf.Format.nBlockAlign;
+ }
+
+ pData->formatOut = ma_format_from_WAVEFORMATEX((WAVEFORMATEX*)&wf);
+ if (pData->formatOut == ma_format_unknown) {
+ /*
+ The format isn't supported. This is almost certainly because the exclusive mode format isn't supported by miniaudio. We need to return MA_SHARE_MODE_NOT_SUPPORTED
+ in this case so that the caller can detect it and fall back to shared mode if desired. We should never get here if shared mode was requested, but just for
+ completeness we'll check for it and return MA_FORMAT_NOT_SUPPORTED.
+ */
+ if (shareMode == MA_AUDCLNT_SHAREMODE_EXCLUSIVE) {
+ result = MA_SHARE_MODE_NOT_SUPPORTED;
+ } else {
+ result = MA_FORMAT_NOT_SUPPORTED;
+ }
+
+ errorMsg = "[WASAPI] Native format not supported.";
+ goto done;
+ }
+
+ pData->channelsOut = wf.Format.nChannels;
+ pData->sampleRateOut = wf.Format.nSamplesPerSec;
+
+ /* Get the internal channel map based on the channel mask. */
+ ma_channel_mask_to_channel_map__win32(wf.dwChannelMask, pData->channelsOut, pData->channelMapOut);
+
+ /* Period size. */
+ pData->periodsOut = (pData->periodsIn != 0) ? pData->periodsIn : MA_DEFAULT_PERIODS;
+ pData->periodSizeInFramesOut = pData->periodSizeInFramesIn;
+ if (pData->periodSizeInFramesOut == 0) {
+ if (pData->periodSizeInMillisecondsIn == 0) {
+ if (pData->performanceProfile == ma_performance_profile_low_latency) {
+ pData->periodSizeInFramesOut = ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY, wf.Format.nSamplesPerSec);
+ } else {
+ pData->periodSizeInFramesOut = ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE, wf.Format.nSamplesPerSec);
+ }
+ } else {
+ pData->periodSizeInFramesOut = ma_calculate_buffer_size_in_frames_from_milliseconds(pData->periodSizeInMillisecondsIn, wf.Format.nSamplesPerSec);
+ }
+ }
+
+ periodDurationInMicroseconds = ((ma_uint64)pData->periodSizeInFramesOut * 1000 * 1000) / wf.Format.nSamplesPerSec;
+
+
+ /* Slightly different initialization for shared and exclusive modes. We try exclusive mode first, and if it fails, fall back to shared mode. */
+ if (shareMode == MA_AUDCLNT_SHAREMODE_EXCLUSIVE) {
+ MA_REFERENCE_TIME bufferDuration = periodDurationInMicroseconds * pData->periodsOut * 10;
+
+ /*
+ If the periodicy is too small, Initialize() will fail with AUDCLNT_E_INVALID_DEVICE_PERIOD. In this case we should just keep increasing
+ it and trying it again.
+ */
+ hr = E_FAIL;
+ for (;;) {
+ hr = ma_IAudioClient_Initialize((ma_IAudioClient*)pData->pAudioClient, shareMode, streamFlags, bufferDuration, bufferDuration, (WAVEFORMATEX*)&wf, NULL);
+ if (hr == MA_AUDCLNT_E_INVALID_DEVICE_PERIOD) {
+ if (bufferDuration > 500*10000) {
+ break;
+ } else {
+ if (bufferDuration == 0) { /* <-- Just a sanity check to prevent an infinit loop. Should never happen, but it makes me feel better. */
+ break;
+ }
+
+ bufferDuration = bufferDuration * 2;
+ continue;
+ }
+ } else {
+ break;
+ }
+ }
+
+ if (hr == MA_AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED) {
+ ma_uint32 bufferSizeInFrames;
+ hr = ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pData->pAudioClient, &bufferSizeInFrames);
+ if (SUCCEEDED(hr)) {
+ bufferDuration = (MA_REFERENCE_TIME)((10000.0 * 1000 / wf.Format.nSamplesPerSec * bufferSizeInFrames) + 0.5);
+
+ /* Unfortunately we need to release and re-acquire the audio client according to MSDN. Seems silly - why not just call IAudioClient_Initialize() again?! */
+ ma_IAudioClient_Release((ma_IAudioClient*)pData->pAudioClient);
+
+ #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK)
+ hr = ma_IMMDevice_Activate(pDeviceInterface, &MA_IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&pData->pAudioClient);
+ #else
+ hr = ma_IUnknown_QueryInterface(pDeviceInterface, &MA_IID_IAudioClient, (void**)&pData->pAudioClient);
+ #endif
+
+ if (SUCCEEDED(hr)) {
+ hr = ma_IAudioClient_Initialize((ma_IAudioClient*)pData->pAudioClient, shareMode, streamFlags, bufferDuration, bufferDuration, (WAVEFORMATEX*)&wf, NULL);
+ }
+ }
+ }
+
+ if (FAILED(hr)) {
+ /* Failed to initialize in exclusive mode. Don't fall back to shared mode - instead tell the client about it. They can reinitialize in shared mode if they want. */
+ if (hr == E_ACCESSDENIED) {
+ errorMsg = "[WASAPI] Failed to initialize device in exclusive mode. Access denied.", result = MA_ACCESS_DENIED;
+ } else if (hr == MA_AUDCLNT_E_DEVICE_IN_USE) {
+ errorMsg = "[WASAPI] Failed to initialize device in exclusive mode. Device in use.", result = MA_BUSY;
+ } else {
+ errorMsg = "[WASAPI] Failed to initialize device in exclusive mode."; result = ma_result_from_HRESULT(hr);
+ }
+ goto done;
+ }
+ }
+
+ if (shareMode == MA_AUDCLNT_SHAREMODE_SHARED) {
+ /*
+ Low latency shared mode via IAudioClient3.
+
+ NOTE
+ ====
+ Contrary to the documentation on MSDN (https://docs.microsoft.com/en-us/windows/win32/api/audioclient/nf-audioclient-iaudioclient3-initializesharedaudiostream), the
+ use of AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM and AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY with IAudioClient3_InitializeSharedAudioStream() absolutely does not work. Using
+ any of these flags will result in HRESULT code 0x88890021. The other problem is that calling IAudioClient3_GetSharedModeEnginePeriod() with a sample rate different to
+ that returned by IAudioClient_GetMixFormat() also results in an error. I'm therefore disabling low-latency shared mode with AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM.
+ */
+ #ifndef MA_WASAPI_NO_LOW_LATENCY_SHARED_MODE
+ {
+ if ((streamFlags & MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM) == 0 || nativeSampleRate == wf.Format.nSamplesPerSec) {
+ ma_IAudioClient3* pAudioClient3 = NULL;
+ hr = ma_IAudioClient_QueryInterface(pData->pAudioClient, &MA_IID_IAudioClient3, (void**)&pAudioClient3);
+ if (SUCCEEDED(hr)) {
+ ma_uint32 defaultPeriodInFrames;
+ ma_uint32 fundamentalPeriodInFrames;
+ ma_uint32 minPeriodInFrames;
+ ma_uint32 maxPeriodInFrames;
+ hr = ma_IAudioClient3_GetSharedModeEnginePeriod(pAudioClient3, (WAVEFORMATEX*)&wf, &defaultPeriodInFrames, &fundamentalPeriodInFrames, &minPeriodInFrames, &maxPeriodInFrames);
+ if (SUCCEEDED(hr)) {
+ ma_uint32 desiredPeriodInFrames = pData->periodSizeInFramesOut;
+ ma_uint32 actualPeriodInFrames = desiredPeriodInFrames;
+
+ /* Make sure the period size is a multiple of fundamentalPeriodInFrames. */
+ actualPeriodInFrames = actualPeriodInFrames / fundamentalPeriodInFrames;
+ actualPeriodInFrames = actualPeriodInFrames * fundamentalPeriodInFrames;
+
+ /* The period needs to be clamped between minPeriodInFrames and maxPeriodInFrames. */
+ actualPeriodInFrames = ma_clamp(actualPeriodInFrames, minPeriodInFrames, maxPeriodInFrames);
+
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[WASAPI] Trying IAudioClient3_InitializeSharedAudioStream(actualPeriodInFrames=%d)\n", actualPeriodInFrames);
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, " defaultPeriodInFrames=%d\n", defaultPeriodInFrames);
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, " fundamentalPeriodInFrames=%d\n", fundamentalPeriodInFrames);
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, " minPeriodInFrames=%d\n", minPeriodInFrames);
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, " maxPeriodInFrames=%d\n", maxPeriodInFrames);
+
+ /* If the client requested a largish buffer than we don't actually want to use low latency shared mode because it forces small buffers. */
+ if (actualPeriodInFrames >= desiredPeriodInFrames) {
+ /*
+ MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM | MA_AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY must not be in the stream flags. If either of these are specified,
+ IAudioClient3_InitializeSharedAudioStream() will fail.
+ */
+ hr = ma_IAudioClient3_InitializeSharedAudioStream(pAudioClient3, streamFlags & ~(MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM | MA_AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY), actualPeriodInFrames, (WAVEFORMATEX*)&wf, NULL);
+ if (SUCCEEDED(hr)) {
+ wasInitializedUsingIAudioClient3 = MA_TRUE;
+ pData->periodSizeInFramesOut = actualPeriodInFrames;
+
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[WASAPI] Using IAudioClient3\n");
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, " periodSizeInFramesOut=%d\n", pData->periodSizeInFramesOut);
+ } else {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[WASAPI] IAudioClient3_InitializeSharedAudioStream failed. Falling back to IAudioClient.\n");
+ }
+ } else {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[WASAPI] Not using IAudioClient3 because the desired period size is larger than the maximum supported by IAudioClient3.\n");
+ }
+ } else {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[WASAPI] IAudioClient3_GetSharedModeEnginePeriod failed. Falling back to IAudioClient.\n");
+ }
+
+ ma_IAudioClient3_Release(pAudioClient3);
+ pAudioClient3 = NULL;
+ }
+ }
+ }
+ #else
+ {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[WASAPI] Not using IAudioClient3 because MA_WASAPI_NO_LOW_LATENCY_SHARED_MODE is enabled.\n");
+ }
+ #endif
+
+ /* If we don't have an IAudioClient3 then we need to use the normal initialization routine. */
+ if (!wasInitializedUsingIAudioClient3) {
+ MA_REFERENCE_TIME bufferDuration = periodDurationInMicroseconds * pData->periodsOut * 10; /* <-- Multiply by 10 for microseconds to 100-nanoseconds. */
+ hr = ma_IAudioClient_Initialize((ma_IAudioClient*)pData->pAudioClient, shareMode, streamFlags, bufferDuration, 0, (WAVEFORMATEX*)&wf, NULL);
+ if (FAILED(hr)) {
+ if (hr == E_ACCESSDENIED) {
+ errorMsg = "[WASAPI] Failed to initialize device. Access denied.", result = MA_ACCESS_DENIED;
+ } else if (hr == MA_AUDCLNT_E_DEVICE_IN_USE) {
+ errorMsg = "[WASAPI] Failed to initialize device. Device in use.", result = MA_BUSY;
+ } else {
+ errorMsg = "[WASAPI] Failed to initialize device.", result = ma_result_from_HRESULT(hr);
+ }
+
+ goto done;
+ }
+ }
+ }
+
+ if (!wasInitializedUsingIAudioClient3) {
+ ma_uint32 bufferSizeInFrames;
+ hr = ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pData->pAudioClient, &bufferSizeInFrames);
+ if (FAILED(hr)) {
+ errorMsg = "[WASAPI] Failed to get audio client's actual buffer size.", result = ma_result_from_HRESULT(hr);
+ goto done;
+ }
+
+ pData->periodSizeInFramesOut = bufferSizeInFrames / pData->periodsOut;
+ }
+
+ pData->usingAudioClient3 = wasInitializedUsingIAudioClient3;
+
+
+ if (deviceType == ma_device_type_playback) {
+ result = ma_device_create_IAudioClient_service__wasapi(pContext, deviceType, (ma_IAudioClient*)pData->pAudioClient, (void**)&pData->pRenderClient);
+ } else {
+ result = ma_device_create_IAudioClient_service__wasapi(pContext, deviceType, (ma_IAudioClient*)pData->pAudioClient, (void**)&pData->pCaptureClient);
+ }
+
+ /*if (FAILED(hr)) {*/
+ if (result != MA_SUCCESS) {
+ errorMsg = "[WASAPI] Failed to get audio client service.";
+ goto done;
+ }
+
+
+ /* Grab the name of the device. */
+ #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK)
+ {
+ ma_IPropertyStore *pProperties;
+ hr = ma_IMMDevice_OpenPropertyStore(pDeviceInterface, STGM_READ, &pProperties);
+ if (SUCCEEDED(hr)) {
+ PROPVARIANT varName;
+ ma_PropVariantInit(&varName);
+ hr = ma_IPropertyStore_GetValue(pProperties, &MA_PKEY_Device_FriendlyName, &varName);
+ if (SUCCEEDED(hr)) {
+ WideCharToMultiByte(CP_UTF8, 0, varName.pwszVal, -1, pData->deviceName, sizeof(pData->deviceName), 0, FALSE);
+ ma_PropVariantClear(pContext, &varName);
+ }
+
+ ma_IPropertyStore_Release(pProperties);
+ }
+ }
+ #endif
+
+ /*
+ For the WASAPI backend we need to know the actual IDs of the device in order to do automatic
+ stream routing so that IDs can be compared and we can determine which device has been detached
+ and whether or not it matches with our ma_device.
+ */
+ #if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK)
+ {
+ /* Desktop */
+ ma_context_get_device_id_from_MMDevice__wasapi(pContext, pDeviceInterface, &pData->id);
+ }
+ #else
+ {
+ /* UWP */
+ /* TODO: Implement me. Need to figure out how to get the ID of the default device. */
+ }
+ #endif
+
+done:
+ /* Clean up. */
+#if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK)
+ if (pDeviceInterface != NULL) {
+ ma_IMMDevice_Release(pDeviceInterface);
+ }
+#else
+ if (pDeviceInterface != NULL) {
+ ma_IUnknown_Release(pDeviceInterface);
+ }
+#endif
+
+ if (result != MA_SUCCESS) {
+ if (pData->pRenderClient) {
+ ma_IAudioRenderClient_Release((ma_IAudioRenderClient*)pData->pRenderClient);
+ pData->pRenderClient = NULL;
+ }
+ if (pData->pCaptureClient) {
+ ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pData->pCaptureClient);
+ pData->pCaptureClient = NULL;
+ }
+ if (pData->pAudioClient) {
+ ma_IAudioClient_Release((ma_IAudioClient*)pData->pAudioClient);
+ pData->pAudioClient = NULL;
+ }
+
+ if (errorMsg != NULL && errorMsg[0] != '\0') {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "%s", errorMsg);
+ }
+
+ return result;
+ } else {
+ return MA_SUCCESS;
+ }
+}
+
+static ma_result ma_device_reinit__wasapi(ma_device* pDevice, ma_device_type deviceType)
+{
+ ma_device_init_internal_data__wasapi data;
+ ma_result result;
+
+ MA_ASSERT(pDevice != NULL);
+
+ /* We only re-initialize the playback or capture device. Never a full-duplex device. */
+ if (deviceType == ma_device_type_duplex) {
+ return MA_INVALID_ARGS;
+ }
+
+
+ /*
+ Before reinitializing the device we need to free the previous audio clients.
+
+ There's a known memory leak here. We will be calling this from the routing change callback that
+ is fired by WASAPI. If we attempt to release the IAudioClient we will deadlock. In my opinion
+ this is a bug. I'm not sure what I need to do to handle this cleanly, but I think we'll probably
+ need some system where we post an event, but delay the execution of it until the callback has
+ returned. I'm not sure how to do this reliably, however. I have set up some infrastructure for
+ a command thread which might be useful for this.
+ */
+ if (deviceType == ma_device_type_capture || deviceType == ma_device_type_loopback) {
+ if (pDevice->wasapi.pCaptureClient) {
+ ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient);
+ pDevice->wasapi.pCaptureClient = NULL;
+ }
+
+ if (pDevice->wasapi.pAudioClientCapture) {
+ /*ma_device_release_IAudioClient_service__wasapi(pDevice, ma_device_type_capture);*/
+ pDevice->wasapi.pAudioClientCapture = NULL;
+ }
+ }
+
+ if (deviceType == ma_device_type_playback) {
+ if (pDevice->wasapi.pRenderClient) {
+ ma_IAudioRenderClient_Release((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient);
+ pDevice->wasapi.pRenderClient = NULL;
+ }
+
+ if (pDevice->wasapi.pAudioClientPlayback) {
+ /*ma_device_release_IAudioClient_service__wasapi(pDevice, ma_device_type_playback);*/
+ pDevice->wasapi.pAudioClientPlayback = NULL;
+ }
+ }
+
+
+ if (deviceType == ma_device_type_playback) {
+ data.formatIn = pDevice->playback.format;
+ data.channelsIn = pDevice->playback.channels;
+ MA_COPY_MEMORY(data.channelMapIn, pDevice->playback.channelMap, sizeof(pDevice->playback.channelMap));
+ data.shareMode = pDevice->playback.shareMode;
+ } else {
+ data.formatIn = pDevice->capture.format;
+ data.channelsIn = pDevice->capture.channels;
+ MA_COPY_MEMORY(data.channelMapIn, pDevice->capture.channelMap, sizeof(pDevice->capture.channelMap));
+ data.shareMode = pDevice->capture.shareMode;
+ }
+
+ data.sampleRateIn = pDevice->sampleRate;
+ data.periodSizeInFramesIn = pDevice->wasapi.originalPeriodSizeInFrames;
+ data.periodSizeInMillisecondsIn = pDevice->wasapi.originalPeriodSizeInMilliseconds;
+ data.periodsIn = pDevice->wasapi.originalPeriods;
+ data.performanceProfile = pDevice->wasapi.originalPerformanceProfile;
+ data.noAutoConvertSRC = pDevice->wasapi.noAutoConvertSRC;
+ data.noDefaultQualitySRC = pDevice->wasapi.noDefaultQualitySRC;
+ data.noHardwareOffloading = pDevice->wasapi.noHardwareOffloading;
+ result = ma_device_init_internal__wasapi(pDevice->pContext, deviceType, NULL, &data);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ /* At this point we have some new objects ready to go. We need to uninitialize the previous ones and then set the new ones. */
+ if (deviceType == ma_device_type_capture || deviceType == ma_device_type_loopback) {
+ pDevice->wasapi.pAudioClientCapture = data.pAudioClient;
+ pDevice->wasapi.pCaptureClient = data.pCaptureClient;
+
+ pDevice->capture.internalFormat = data.formatOut;
+ pDevice->capture.internalChannels = data.channelsOut;
+ pDevice->capture.internalSampleRate = data.sampleRateOut;
+ MA_COPY_MEMORY(pDevice->capture.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut));
+ pDevice->capture.internalPeriodSizeInFrames = data.periodSizeInFramesOut;
+ pDevice->capture.internalPeriods = data.periodsOut;
+ ma_strcpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), data.deviceName);
+
+ ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, pDevice->wasapi.hEventCapture);
+
+ pDevice->wasapi.periodSizeInFramesCapture = data.periodSizeInFramesOut;
+ ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, &pDevice->wasapi.actualBufferSizeInFramesCapture);
+
+ /* We must always have a valid ID. */
+ ma_wcscpy_s(pDevice->capture.id.wasapi, sizeof(pDevice->capture.id.wasapi), data.id.wasapi);
+ }
+
+ if (deviceType == ma_device_type_playback) {
+ pDevice->wasapi.pAudioClientPlayback = data.pAudioClient;
+ pDevice->wasapi.pRenderClient = data.pRenderClient;
+
+ pDevice->playback.internalFormat = data.formatOut;
+ pDevice->playback.internalChannels = data.channelsOut;
+ pDevice->playback.internalSampleRate = data.sampleRateOut;
+ MA_COPY_MEMORY(pDevice->playback.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut));
+ pDevice->playback.internalPeriodSizeInFrames = data.periodSizeInFramesOut;
+ pDevice->playback.internalPeriods = data.periodsOut;
+ ma_strcpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), data.deviceName);
+
+ ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, pDevice->wasapi.hEventPlayback);
+
+ pDevice->wasapi.periodSizeInFramesPlayback = data.periodSizeInFramesOut;
+ ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &pDevice->wasapi.actualBufferSizeInFramesPlayback);
+
+ /* We must always have a valid ID because rerouting will look at it. */
+ ma_wcscpy_s(pDevice->playback.id.wasapi, sizeof(pDevice->playback.id.wasapi), data.id.wasapi);
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_init__wasapi(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+{
+ ma_result result = MA_SUCCESS;
+
+#if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK)
+ HRESULT hr;
+ ma_IMMDeviceEnumerator* pDeviceEnumerator;
+#endif
+
+ MA_ASSERT(pDevice != NULL);
+
+ MA_ZERO_OBJECT(&pDevice->wasapi);
+ pDevice->wasapi.noAutoConvertSRC = pConfig->wasapi.noAutoConvertSRC;
+ pDevice->wasapi.noDefaultQualitySRC = pConfig->wasapi.noDefaultQualitySRC;
+ pDevice->wasapi.noHardwareOffloading = pConfig->wasapi.noHardwareOffloading;
+
+ /* Exclusive mode is not allowed with loopback. */
+ if (pConfig->deviceType == ma_device_type_loopback && pConfig->playback.shareMode == ma_share_mode_exclusive) {
+ return MA_INVALID_DEVICE_CONFIG;
+ }
+
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex || pConfig->deviceType == ma_device_type_loopback) {
+ ma_device_init_internal_data__wasapi data;
+ data.formatIn = pDescriptorCapture->format;
+ data.channelsIn = pDescriptorCapture->channels;
+ data.sampleRateIn = pDescriptorCapture->sampleRate;
+ MA_COPY_MEMORY(data.channelMapIn, pDescriptorCapture->channelMap, sizeof(pDescriptorCapture->channelMap));
+ data.periodSizeInFramesIn = pDescriptorCapture->periodSizeInFrames;
+ data.periodSizeInMillisecondsIn = pDescriptorCapture->periodSizeInMilliseconds;
+ data.periodsIn = pDescriptorCapture->periodCount;
+ data.shareMode = pDescriptorCapture->shareMode;
+ data.performanceProfile = pConfig->performanceProfile;
+ data.noAutoConvertSRC = pConfig->wasapi.noAutoConvertSRC;
+ data.noDefaultQualitySRC = pConfig->wasapi.noDefaultQualitySRC;
+ data.noHardwareOffloading = pConfig->wasapi.noHardwareOffloading;
+
+ result = ma_device_init_internal__wasapi(pDevice->pContext, (pConfig->deviceType == ma_device_type_loopback) ? ma_device_type_loopback : ma_device_type_capture, pDescriptorCapture->pDeviceID, &data);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ pDevice->wasapi.pAudioClientCapture = data.pAudioClient;
+ pDevice->wasapi.pCaptureClient = data.pCaptureClient;
+ pDevice->wasapi.originalPeriodSizeInMilliseconds = pDescriptorCapture->periodSizeInMilliseconds;
+ pDevice->wasapi.originalPeriodSizeInFrames = pDescriptorCapture->periodSizeInFrames;
+ pDevice->wasapi.originalPeriods = pDescriptorCapture->periodCount;
+ pDevice->wasapi.originalPerformanceProfile = pConfig->performanceProfile;
+
+ /*
+ The event for capture needs to be manual reset for the same reason as playback. We keep the initial state set to unsignaled,
+ however, because we want to block until we actually have something for the first call to ma_device_read().
+ */
+ pDevice->wasapi.hEventCapture = CreateEventW(NULL, FALSE, FALSE, NULL); /* Auto reset, unsignaled by default. */
+ if (pDevice->wasapi.hEventCapture == NULL) {
+ result = ma_result_from_GetLastError(GetLastError());
+
+ if (pDevice->wasapi.pCaptureClient != NULL) {
+ ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient);
+ pDevice->wasapi.pCaptureClient = NULL;
+ }
+ if (pDevice->wasapi.pAudioClientCapture != NULL) {
+ ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture);
+ pDevice->wasapi.pAudioClientCapture = NULL;
+ }
+
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create event for capture.");
+ return result;
+ }
+ ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, pDevice->wasapi.hEventCapture);
+
+ pDevice->wasapi.periodSizeInFramesCapture = data.periodSizeInFramesOut;
+ ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, &pDevice->wasapi.actualBufferSizeInFramesCapture);
+
+ /* We must always have a valid ID. */
+ ma_wcscpy_s(pDevice->capture.id.wasapi, sizeof(pDevice->capture.id.wasapi), data.id.wasapi);
+
+ /* The descriptor needs to be updated with actual values. */
+ pDescriptorCapture->format = data.formatOut;
+ pDescriptorCapture->channels = data.channelsOut;
+ pDescriptorCapture->sampleRate = data.sampleRateOut;
+ MA_COPY_MEMORY(pDescriptorCapture->channelMap, data.channelMapOut, sizeof(data.channelMapOut));
+ pDescriptorCapture->periodSizeInFrames = data.periodSizeInFramesOut;
+ pDescriptorCapture->periodCount = data.periodsOut;
+ }
+
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ ma_device_init_internal_data__wasapi data;
+ data.formatIn = pDescriptorPlayback->format;
+ data.channelsIn = pDescriptorPlayback->channels;
+ data.sampleRateIn = pDescriptorPlayback->sampleRate;
+ MA_COPY_MEMORY(data.channelMapIn, pDescriptorPlayback->channelMap, sizeof(pDescriptorPlayback->channelMap));
+ data.periodSizeInFramesIn = pDescriptorPlayback->periodSizeInFrames;
+ data.periodSizeInMillisecondsIn = pDescriptorPlayback->periodSizeInMilliseconds;
+ data.periodsIn = pDescriptorPlayback->periodCount;
+ data.shareMode = pDescriptorPlayback->shareMode;
+ data.performanceProfile = pConfig->performanceProfile;
+ data.noAutoConvertSRC = pConfig->wasapi.noAutoConvertSRC;
+ data.noDefaultQualitySRC = pConfig->wasapi.noDefaultQualitySRC;
+ data.noHardwareOffloading = pConfig->wasapi.noHardwareOffloading;
+
+ result = ma_device_init_internal__wasapi(pDevice->pContext, ma_device_type_playback, pDescriptorPlayback->pDeviceID, &data);
+ if (result != MA_SUCCESS) {
+ if (pConfig->deviceType == ma_device_type_duplex) {
+ if (pDevice->wasapi.pCaptureClient != NULL) {
+ ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient);
+ pDevice->wasapi.pCaptureClient = NULL;
+ }
+ if (pDevice->wasapi.pAudioClientCapture != NULL) {
+ ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture);
+ pDevice->wasapi.pAudioClientCapture = NULL;
+ }
+
+ CloseHandle(pDevice->wasapi.hEventCapture);
+ pDevice->wasapi.hEventCapture = NULL;
+ }
+ return result;
+ }
+
+ pDevice->wasapi.pAudioClientPlayback = data.pAudioClient;
+ pDevice->wasapi.pRenderClient = data.pRenderClient;
+ pDevice->wasapi.originalPeriodSizeInMilliseconds = pDescriptorPlayback->periodSizeInMilliseconds;
+ pDevice->wasapi.originalPeriodSizeInFrames = pDescriptorPlayback->periodSizeInFrames;
+ pDevice->wasapi.originalPeriods = pDescriptorPlayback->periodCount;
+ pDevice->wasapi.originalPerformanceProfile = pConfig->performanceProfile;
+
+ /*
+ The event for playback is needs to be manual reset because we want to explicitly control the fact that it becomes signalled
+ only after the whole available space has been filled, never before.
+
+ The playback event also needs to be initially set to a signaled state so that the first call to ma_device_write() is able
+ to get passed WaitForMultipleObjects().
+ */
+ pDevice->wasapi.hEventPlayback = CreateEventW(NULL, FALSE, TRUE, NULL); /* Auto reset, signaled by default. */
+ if (pDevice->wasapi.hEventPlayback == NULL) {
+ result = ma_result_from_GetLastError(GetLastError());
+
+ if (pConfig->deviceType == ma_device_type_duplex) {
+ if (pDevice->wasapi.pCaptureClient != NULL) {
+ ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient);
+ pDevice->wasapi.pCaptureClient = NULL;
+ }
+ if (pDevice->wasapi.pAudioClientCapture != NULL) {
+ ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture);
+ pDevice->wasapi.pAudioClientCapture = NULL;
+ }
+
+ CloseHandle(pDevice->wasapi.hEventCapture);
+ pDevice->wasapi.hEventCapture = NULL;
+ }
+
+ if (pDevice->wasapi.pRenderClient != NULL) {
+ ma_IAudioRenderClient_Release((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient);
+ pDevice->wasapi.pRenderClient = NULL;
+ }
+ if (pDevice->wasapi.pAudioClientPlayback != NULL) {
+ ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback);
+ pDevice->wasapi.pAudioClientPlayback = NULL;
+ }
+
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create event for playback.");
+ return result;
+ }
+ ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, pDevice->wasapi.hEventPlayback);
+
+ pDevice->wasapi.periodSizeInFramesPlayback = data.periodSizeInFramesOut;
+ ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &pDevice->wasapi.actualBufferSizeInFramesPlayback);
+
+ /* We must always have a valid ID because rerouting will look at it. */
+ ma_wcscpy_s(pDevice->playback.id.wasapi, sizeof(pDevice->playback.id.wasapi), data.id.wasapi);
+
+ /* The descriptor needs to be updated with actual values. */
+ pDescriptorPlayback->format = data.formatOut;
+ pDescriptorPlayback->channels = data.channelsOut;
+ pDescriptorPlayback->sampleRate = data.sampleRateOut;
+ MA_COPY_MEMORY(pDescriptorPlayback->channelMap, data.channelMapOut, sizeof(data.channelMapOut));
+ pDescriptorPlayback->periodSizeInFrames = data.periodSizeInFramesOut;
+ pDescriptorPlayback->periodCount = data.periodsOut;
+ }
+
+ /*
+ We need to register a notification client to detect when the device has been disabled, unplugged or re-routed (when the default device changes). When
+ we are connecting to the default device we want to do automatic stream routing when the device is disabled or unplugged. Otherwise we want to just
+ stop the device outright and let the application handle it.
+ */
+#if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK)
+ if (pConfig->wasapi.noAutoStreamRouting == MA_FALSE) {
+ if ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.pDeviceID == NULL) {
+ pDevice->wasapi.allowCaptureAutoStreamRouting = MA_TRUE;
+ }
+ if ((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.pDeviceID == NULL) {
+ pDevice->wasapi.allowPlaybackAutoStreamRouting = MA_TRUE;
+ }
+ }
+
+ hr = ma_CoCreateInstance(pDevice->pContext, MA_CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, MA_IID_IMMDeviceEnumerator, (void**)&pDeviceEnumerator);
+ if (FAILED(hr)) {
+ ma_device_uninit__wasapi(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create device enumerator.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ pDevice->wasapi.notificationClient.lpVtbl = (void*)&g_maNotificationCientVtbl;
+ pDevice->wasapi.notificationClient.counter = 1;
+ pDevice->wasapi.notificationClient.pDevice = pDevice;
+
+ hr = pDeviceEnumerator->lpVtbl->RegisterEndpointNotificationCallback(pDeviceEnumerator, &pDevice->wasapi.notificationClient);
+ if (SUCCEEDED(hr)) {
+ pDevice->wasapi.pDeviceEnumerator = (ma_ptr)pDeviceEnumerator;
+ } else {
+ /* Not the end of the world if we fail to register the notification callback. We just won't support automatic stream routing. */
+ ma_IMMDeviceEnumerator_Release(pDeviceEnumerator);
+ }
+#endif
+
+ c89atomic_exchange_32(&pDevice->wasapi.isStartedCapture, MA_FALSE);
+ c89atomic_exchange_32(&pDevice->wasapi.isStartedPlayback, MA_FALSE);
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device__get_available_frames__wasapi(ma_device* pDevice, ma_IAudioClient* pAudioClient, ma_uint32* pFrameCount)
+{
+ ma_uint32 paddingFramesCount;
+ HRESULT hr;
+ ma_share_mode shareMode;
+
+ MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(pFrameCount != NULL);
+
+ *pFrameCount = 0;
+
+ if ((ma_ptr)pAudioClient != pDevice->wasapi.pAudioClientPlayback && (ma_ptr)pAudioClient != pDevice->wasapi.pAudioClientCapture) {
+ return MA_INVALID_OPERATION;
+ }
+
+ /*
+ I've had a report that GetCurrentPadding() is returning a frame count of 0 which is preventing
+ higher level function calls from doing anything because it thinks nothing is available. I have
+ taken a look at the documentation and it looks like this is unnecessary in exclusive mode.
+
+ From Microsoft's documentation:
+
+ For an exclusive-mode rendering or capture stream that was initialized with the
+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK flag, the client typically has no use for the padding
+ value reported by GetCurrentPadding. Instead, the client accesses an entire buffer during
+ each processing pass.
+
+ Considering this, I'm going to skip GetCurrentPadding() for exclusive mode and just report the
+ entire buffer. This depends on the caller making sure they wait on the event handler.
+ */
+ shareMode = ((ma_ptr)pAudioClient == pDevice->wasapi.pAudioClientPlayback) ? pDevice->playback.shareMode : pDevice->capture.shareMode;
+ if (shareMode == ma_share_mode_shared) {
+ /* Shared mode. */
+ hr = ma_IAudioClient_GetCurrentPadding(pAudioClient, &paddingFramesCount);
+ if (FAILED(hr)) {
+ return ma_result_from_HRESULT(hr);
+ }
+
+ if ((ma_ptr)pAudioClient == pDevice->wasapi.pAudioClientPlayback) {
+ *pFrameCount = pDevice->wasapi.actualBufferSizeInFramesPlayback - paddingFramesCount;
+ } else {
+ *pFrameCount = paddingFramesCount;
+ }
+ } else {
+ /* Exclusive mode. */
+ if ((ma_ptr)pAudioClient == pDevice->wasapi.pAudioClientPlayback) {
+ *pFrameCount = pDevice->wasapi.actualBufferSizeInFramesPlayback;
+ } else {
+ *pFrameCount = pDevice->wasapi.actualBufferSizeInFramesCapture;
+ }
+ }
+
+ return MA_SUCCESS;
+}
+
+
+static ma_result ma_device_reroute__wasapi(ma_device* pDevice, ma_device_type deviceType)
+{
+ ma_result result;
+
+ if (deviceType == ma_device_type_duplex) {
+ return MA_INVALID_ARGS;
+ }
+
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "=== CHANGING DEVICE ===\n");
+
+ result = ma_device_reinit__wasapi(pDevice, deviceType);
+ if (result != MA_SUCCESS) {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[WASAPI] Reinitializing device after route change failed.\n");
+ return result;
+ }
+
+ ma_device__post_init_setup(pDevice, deviceType);
+
+ ma_device__on_notification_rerouted(pDevice);
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_start__wasapi(ma_device* pDevice)
+{
+ HRESULT hr;
+
+ MA_ASSERT(pDevice != NULL);
+
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) {
+ hr = ma_IAudioClient_Start((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture);
+ if (FAILED(hr)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to start internal capture device.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ c89atomic_exchange_32(&pDevice->wasapi.isStartedCapture, MA_TRUE);
+ }
+
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ hr = ma_IAudioClient_Start((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback);
+ if (FAILED(hr)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to start internal playback device.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ c89atomic_exchange_32(&pDevice->wasapi.isStartedPlayback, MA_TRUE);
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_stop__wasapi(ma_device* pDevice)
+{
+ ma_result result;
+ HRESULT hr;
+
+ MA_ASSERT(pDevice != NULL);
+
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) {
+ hr = ma_IAudioClient_Stop((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture);
+ if (FAILED(hr)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to stop internal capture device.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ /* The audio client needs to be reset otherwise restarting will fail. */
+ hr = ma_IAudioClient_Reset((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture);
+ if (FAILED(hr)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to reset internal capture device.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ /* If we have a mapped buffer we need to release it. */
+ if (pDevice->wasapi.pMappedBufferCapture != NULL) {
+ ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, pDevice->wasapi.mappedBufferCaptureCap);
+ pDevice->wasapi.pMappedBufferCapture = NULL;
+ pDevice->wasapi.mappedBufferCaptureCap = 0;
+ pDevice->wasapi.mappedBufferCaptureLen = 0;
+ }
+
+ c89atomic_exchange_32(&pDevice->wasapi.isStartedCapture, MA_FALSE);
+ }
+
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ /*
+ The buffer needs to be drained before stopping the device. Not doing this will result in the last few frames not getting output to
+ the speakers. This is a problem for very short sounds because it'll result in a significant portion of it not getting played.
+ */
+ if (c89atomic_load_32(&pDevice->wasapi.isStartedPlayback)) {
+ /* We need to make sure we put a timeout here or else we'll risk getting stuck in a deadlock in some cases. */
+ DWORD waitTime = pDevice->wasapi.actualBufferSizeInFramesPlayback / pDevice->playback.internalSampleRate;
+
+ if (pDevice->playback.shareMode == ma_share_mode_exclusive) {
+ WaitForSingleObject(pDevice->wasapi.hEventPlayback, waitTime);
+ } else {
+ ma_uint32 prevFramesAvaialablePlayback = (ma_uint32)-1;
+ ma_uint32 framesAvailablePlayback;
+ for (;;) {
+ result = ma_device__get_available_frames__wasapi(pDevice, (ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &framesAvailablePlayback);
+ if (result != MA_SUCCESS) {
+ break;
+ }
+
+ if (framesAvailablePlayback >= pDevice->wasapi.actualBufferSizeInFramesPlayback) {
+ break;
+ }
+
+ /*
+ Just a safety check to avoid an infinite loop. If this iteration results in a situation where the number of available frames
+ has not changed, get out of the loop. I don't think this should ever happen, but I think it's nice to have just in case.
+ */
+ if (framesAvailablePlayback == prevFramesAvaialablePlayback) {
+ break;
+ }
+ prevFramesAvaialablePlayback = framesAvailablePlayback;
+
+ WaitForSingleObject(pDevice->wasapi.hEventPlayback, waitTime);
+ ResetEvent(pDevice->wasapi.hEventPlayback); /* Manual reset. */
+ }
+ }
+ }
+
+ hr = ma_IAudioClient_Stop((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback);
+ if (FAILED(hr)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to stop internal playback device.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ /* The audio client needs to be reset otherwise restarting will fail. */
+ hr = ma_IAudioClient_Reset((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback);
+ if (FAILED(hr)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to reset internal playback device.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ if (pDevice->wasapi.pMappedBufferPlayback != NULL) {
+ ma_IAudioRenderClient_ReleaseBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, pDevice->wasapi.mappedBufferPlaybackCap, 0);
+ pDevice->wasapi.pMappedBufferPlayback = NULL;
+ pDevice->wasapi.mappedBufferPlaybackCap = 0;
+ pDevice->wasapi.mappedBufferPlaybackLen = 0;
+ }
+
+ c89atomic_exchange_32(&pDevice->wasapi.isStartedPlayback, MA_FALSE);
+ }
+
+ return MA_SUCCESS;
+}
+
+
+#ifndef MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS
+#define MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS 5000
+#endif
+
+static ma_result ma_device_read__wasapi(ma_device* pDevice, void* pFrames, ma_uint32 frameCount, ma_uint32* pFramesRead)
+{
+ ma_result result = MA_SUCCESS;
+ ma_uint32 totalFramesProcessed = 0;
+
+ /*
+ When reading, we need to get a buffer and process all of it before releasing it. Because the
+ frame count (frameCount) can be different to the size of the buffer, we'll need to cache the
+ pointer to the buffer.
+ */
+
+ /* Keep running until we've processed the requested number of frames. */
+ while (ma_device_get_state(pDevice) == ma_device_state_started && totalFramesProcessed < frameCount) {
+ ma_uint32 framesRemaining = frameCount - totalFramesProcessed;
+
+ /* If we have a mapped data buffer, consume that first. */
+ if (pDevice->wasapi.pMappedBufferCapture != NULL) {
+ /* We have a cached data pointer so consume that before grabbing another one from WASAPI. */
+ ma_uint32 framesToProcessNow = framesRemaining;
+ if (framesToProcessNow > pDevice->wasapi.mappedBufferCaptureLen) {
+ framesToProcessNow = pDevice->wasapi.mappedBufferCaptureLen;
+ }
+
+ /* Now just copy the data over to the output buffer. */
+ ma_copy_pcm_frames(
+ ma_offset_pcm_frames_ptr(pFrames, totalFramesProcessed, pDevice->capture.internalFormat, pDevice->capture.internalChannels),
+ ma_offset_pcm_frames_const_ptr(pDevice->wasapi.pMappedBufferCapture, pDevice->wasapi.mappedBufferCaptureCap - pDevice->wasapi.mappedBufferCaptureLen, pDevice->capture.internalFormat, pDevice->capture.internalChannels),
+ framesToProcessNow,
+ pDevice->capture.internalFormat, pDevice->capture.internalChannels
+ );
+
+ totalFramesProcessed += framesToProcessNow;
+ pDevice->wasapi.mappedBufferCaptureLen -= framesToProcessNow;
+
+ /* If the data buffer has been fully consumed we need to release it. */
+ if (pDevice->wasapi.mappedBufferCaptureLen == 0) {
+ ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, pDevice->wasapi.mappedBufferCaptureCap);
+ pDevice->wasapi.pMappedBufferCapture = NULL;
+ pDevice->wasapi.mappedBufferCaptureCap = 0;
+ }
+ } else {
+ /* We don't have any cached data pointer, so grab another one. */
+ HRESULT hr;
+ DWORD flags;
+
+ /* First just ask WASAPI for a data buffer. If it's not available, we'll wait for more. */
+ hr = ma_IAudioCaptureClient_GetBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, (BYTE**)&pDevice->wasapi.pMappedBufferCapture, &pDevice->wasapi.mappedBufferCaptureCap, &flags, NULL, NULL);
+ if (hr == S_OK) {
+ /* We got a data buffer. Continue to the next loop iteration which will then read from the mapped pointer. */
+
+ /* Overrun detection. */
+ if ((flags & MA_AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY) != 0) {
+ /* Glitched. Probably due to an overrun. */
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Data discontinuity (possible overrun). Attempting recovery. mappedBufferCaptureCap=%d\n", pDevice->wasapi.mappedBufferCaptureCap);
+
+ /*
+ If we got an overrun it probably means we're straddling the end of the buffer. In order to prevent
+ a never-ending sequence of glitches we're going to recover by completely clearing out the capture
+ buffer.
+ */
+ {
+ ma_uint32 iterationCount = 4; /* Safety to prevent an infinite loop. */
+ ma_uint32 i;
+
+ for (i = 0; i < iterationCount; i += 1) {
+ hr = ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, pDevice->wasapi.mappedBufferCaptureCap);
+ if (FAILED(hr)) {
+ break;
+ }
+
+ hr = ma_IAudioCaptureClient_GetBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, (BYTE**)&pDevice->wasapi.pMappedBufferCapture, &pDevice->wasapi.mappedBufferCaptureCap, &flags, NULL, NULL);
+ if (hr == MA_AUDCLNT_S_BUFFER_EMPTY || FAILED(hr)) {
+ break;
+ }
+ }
+ }
+
+ /* We should not have a valid buffer at this point so make sure everything is empty. */
+ pDevice->wasapi.pMappedBufferCapture = NULL;
+ pDevice->wasapi.mappedBufferCaptureCap = 0;
+ pDevice->wasapi.mappedBufferCaptureLen = 0;
+ } else {
+ /* The data is clean. */
+ pDevice->wasapi.mappedBufferCaptureLen = pDevice->wasapi.mappedBufferCaptureCap;
+
+ if (flags != 0) {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Capture Flags: %ld\n", flags);
+ }
+ }
+
+ continue;
+ } else {
+ if (hr == MA_AUDCLNT_S_BUFFER_EMPTY || hr == MA_AUDCLNT_E_BUFFER_ERROR) {
+ /*
+ No data is available. We need to wait for more. There's two situations to consider
+ here. The first is normal capture mode. If this times out it probably means the
+ microphone isn't delivering data for whatever reason. In this case we'll just
+ abort the read and return whatever we were able to get. The other situations is
+ loopback mode, in which case a timeout probably just means the nothing is playing
+ through the speakers.
+ */
+ if (WaitForSingleObject(pDevice->wasapi.hEventCapture, MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS) != WAIT_OBJECT_0) {
+ if (pDevice->type == ma_device_type_loopback) {
+ continue; /* Keep waiting in loopback mode. */
+ } else {
+ result = MA_ERROR;
+ break; /* Wait failed. */
+ }
+ }
+
+ /* At this point we should be able to loop back to the start of the loop and try retrieving a data buffer again. */
+ } else {
+ /* An error occured and we need to abort. */
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from capture device in preparation for reading from the device. HRESULT = %d. Stopping device.\n", (int)hr);
+ result = ma_result_from_HRESULT(hr);
+ break;
+ }
+ }
+ }
+ }
+
+ /*
+ If we were unable to process the entire requested frame count, but we still have a mapped buffer,
+ there's a good chance either an error occurred or the device was stopped mid-read. In this case
+ we'll need to make sure the buffer is released.
+ */
+ if (totalFramesProcessed < frameCount && pDevice->wasapi.pMappedBufferCapture != NULL) {
+ ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, pDevice->wasapi.mappedBufferCaptureCap);
+ pDevice->wasapi.pMappedBufferCapture = NULL;
+ pDevice->wasapi.mappedBufferCaptureCap = 0;
+ pDevice->wasapi.mappedBufferCaptureLen = 0;
+ }
+
+ if (pFramesRead != NULL) {
+ *pFramesRead = totalFramesProcessed;
+ }
+
+ return result;
+}
+
+static ma_result ma_device_write__wasapi(ma_device* pDevice, const void* pFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten)
+{
+ ma_result result = MA_SUCCESS;
+ ma_uint32 totalFramesProcessed = 0;
+
+ /* Keep writing to the device until it's stopped or we've consumed all of our input. */
+ while (ma_device_get_state(pDevice) == ma_device_state_started && totalFramesProcessed < frameCount) {
+ ma_uint32 framesRemaining = frameCount - totalFramesProcessed;
+
+ /*
+ We're going to do this in a similar way to capture. We'll first check if the cached data pointer
+ is valid, and if so, read from that. Otherwise We will call IAudioRenderClient_GetBuffer() with
+ a requested buffer size equal to our actual period size. If it returns AUDCLNT_E_BUFFER_TOO_LARGE
+ it means we need to wait for some data to become available.
+ */
+ if (pDevice->wasapi.pMappedBufferPlayback != NULL) {
+ /* We still have some space available in the mapped data buffer. Write to it. */
+ ma_uint32 framesToProcessNow = framesRemaining;
+ if (framesToProcessNow > (pDevice->wasapi.mappedBufferPlaybackCap - pDevice->wasapi.mappedBufferPlaybackLen)) {
+ framesToProcessNow = (pDevice->wasapi.mappedBufferPlaybackCap - pDevice->wasapi.mappedBufferPlaybackLen);
+ }
+
+ /* Now just copy the data over to the output buffer. */
+ ma_copy_pcm_frames(
+ ma_offset_pcm_frames_ptr(pDevice->wasapi.pMappedBufferPlayback, pDevice->wasapi.mappedBufferPlaybackLen, pDevice->playback.internalFormat, pDevice->playback.internalChannels),
+ ma_offset_pcm_frames_const_ptr(pFrames, totalFramesProcessed, pDevice->playback.internalFormat, pDevice->playback.internalChannels),
+ framesToProcessNow,
+ pDevice->playback.internalFormat, pDevice->playback.internalChannels
+ );
+
+ totalFramesProcessed += framesToProcessNow;
+ pDevice->wasapi.mappedBufferPlaybackLen += framesToProcessNow;
+
+ /* If the data buffer has been fully consumed we need to release it. */
+ if (pDevice->wasapi.mappedBufferPlaybackLen == pDevice->wasapi.mappedBufferPlaybackCap) {
+ ma_IAudioRenderClient_ReleaseBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, pDevice->wasapi.mappedBufferPlaybackCap, 0);
+ pDevice->wasapi.pMappedBufferPlayback = NULL;
+ pDevice->wasapi.mappedBufferPlaybackCap = 0;
+ pDevice->wasapi.mappedBufferPlaybackLen = 0;
+
+ /*
+ In exclusive mode we need to wait here. Exclusive mode is weird because GetBuffer() never
+ seems to return AUDCLNT_E_BUFFER_TOO_LARGE, which is what we normally use to determine
+ whether or not we need to wait for more data.
+ */
+ if (pDevice->playback.shareMode == ma_share_mode_exclusive) {
+ if (WaitForSingleObject(pDevice->wasapi.hEventPlayback, MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS) != WAIT_OBJECT_0) {
+ result = MA_ERROR;
+ break; /* Wait failed. Probably timed out. */
+ }
+ }
+ }
+ } else {
+ /* We don't have a mapped data buffer so we'll need to get one. */
+ HRESULT hr;
+ ma_uint32 bufferSizeInFrames;
+
+ /* Special rules for exclusive mode. */
+ if (pDevice->playback.shareMode == ma_share_mode_exclusive) {
+ bufferSizeInFrames = pDevice->wasapi.actualBufferSizeInFramesPlayback;
+ } else {
+ bufferSizeInFrames = pDevice->wasapi.periodSizeInFramesPlayback;
+ }
+
+ hr = ma_IAudioRenderClient_GetBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, bufferSizeInFrames, (BYTE**)&pDevice->wasapi.pMappedBufferPlayback);
+ if (hr == S_OK) {
+ /* We have data available. */
+ pDevice->wasapi.mappedBufferPlaybackCap = bufferSizeInFrames;
+ pDevice->wasapi.mappedBufferPlaybackLen = 0;
+ } else {
+ if (hr == MA_AUDCLNT_E_BUFFER_TOO_LARGE || hr == MA_AUDCLNT_E_BUFFER_ERROR) {
+ /* Not enough data available. We need to wait for more. */
+ if (WaitForSingleObject(pDevice->wasapi.hEventPlayback, MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS) != WAIT_OBJECT_0) {
+ result = MA_ERROR;
+ break; /* Wait failed. Probably timed out. */
+ }
+ } else {
+ /* Some error occurred. We'll need to abort. */
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from playback device in preparation for writing to the device. HRESULT = %d. Stopping device.\n", (int)hr);
+ result = ma_result_from_HRESULT(hr);
+ break;
+ }
+ }
+ }
+ }
+
+ if (pFramesWritten != NULL) {
+ *pFramesWritten = totalFramesProcessed;
+ }
+
+ return result;
+}
+
+static ma_result ma_device_data_loop_wakeup__wasapi(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
+
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) {
+ SetEvent((HANDLE)pDevice->wasapi.hEventCapture);
+ }
+
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ SetEvent((HANDLE)pDevice->wasapi.hEventPlayback);
+ }
+
+ return MA_SUCCESS;
+}
+
+
+static ma_result ma_context_uninit__wasapi(ma_context* pContext)
+{
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pContext->backend == ma_backend_wasapi);
+
+ if (pContext->wasapi.commandThread != NULL) {
+ ma_context_command__wasapi cmd = ma_context_init_command__wasapi(MA_CONTEXT_COMMAND_QUIT__WASAPI);
+ ma_context_post_command__wasapi(pContext, &cmd);
+ ma_thread_wait(&pContext->wasapi.commandThread);
+
+ /* Only after the thread has been terminated can we uninitialize the sync objects for the command thread. */
+ ma_semaphore_uninit(&pContext->wasapi.commandSem);
+ ma_mutex_uninit(&pContext->wasapi.commandLock);
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_init__wasapi(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+{
+ ma_result result = MA_SUCCESS;
+
+ MA_ASSERT(pContext != NULL);
+
+ (void)pConfig;
+
+#ifdef MA_WIN32_DESKTOP
+ /*
+ WASAPI is only supported in Vista SP1 and newer. The reason for SP1 and not the base version of Vista is that event-driven
+ exclusive mode does not work until SP1.
+
+ Unfortunately older compilers don't define these functions so we need to dynamically load them in order to avoid a link error.
+ */
+ {
+ ma_OSVERSIONINFOEXW osvi;
+ ma_handle kernel32DLL;
+ ma_PFNVerifyVersionInfoW _VerifyVersionInfoW;
+ ma_PFNVerSetConditionMask _VerSetConditionMask;
+
+ kernel32DLL = ma_dlopen(pContext, "kernel32.dll");
+ if (kernel32DLL == NULL) {
+ return MA_NO_BACKEND;
+ }
+
+ _VerifyVersionInfoW = (ma_PFNVerifyVersionInfoW )ma_dlsym(pContext, kernel32DLL, "VerifyVersionInfoW");
+ _VerSetConditionMask = (ma_PFNVerSetConditionMask)ma_dlsym(pContext, kernel32DLL, "VerSetConditionMask");
+ if (_VerifyVersionInfoW == NULL || _VerSetConditionMask == NULL) {
+ ma_dlclose(pContext, kernel32DLL);
+ return MA_NO_BACKEND;
+ }
+
+ MA_ZERO_OBJECT(&osvi);
+ osvi.dwOSVersionInfoSize = sizeof(osvi);
+ osvi.dwMajorVersion = ((MA_WIN32_WINNT_VISTA >> 8) & 0xFF);
+ osvi.dwMinorVersion = ((MA_WIN32_WINNT_VISTA >> 0) & 0xFF);
+ osvi.wServicePackMajor = 1;
+ if (_VerifyVersionInfoW(&osvi, MA_VER_MAJORVERSION | MA_VER_MINORVERSION | MA_VER_SERVICEPACKMAJOR, _VerSetConditionMask(_VerSetConditionMask(_VerSetConditionMask(0, MA_VER_MAJORVERSION, MA_VER_GREATER_EQUAL), MA_VER_MINORVERSION, MA_VER_GREATER_EQUAL), MA_VER_SERVICEPACKMAJOR, MA_VER_GREATER_EQUAL))) {
+ result = MA_SUCCESS;
+ } else {
+ result = MA_NO_BACKEND;
+ }
+
+ ma_dlclose(pContext, kernel32DLL);
+ }
+#endif
+
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ MA_ZERO_OBJECT(&pContext->wasapi);
+
+ /*
+ Annoyingly, WASAPI does not allow you to release an IAudioClient object from a different thread
+ than the one that retrieved it with GetService(). This can result in a deadlock in two
+ situations:
+
+ 1) When calling ma_device_uninit() from a different thread to ma_device_init(); and
+ 2) When uninitializing and reinitializing the internal IAudioClient object in response to
+ automatic stream routing.
+
+ We could define ma_device_uninit() such that it must be called on the same thread as
+ ma_device_init(). We could also just not release the IAudioClient when performing automatic
+ stream routing to avoid the deadlock. Neither of these are acceptable solutions in my view so
+ we're going to have to work around this with a worker thread. This is not ideal, but I can't
+ think of a better way to do this.
+
+ More information about this can be found here:
+
+ https://docs.microsoft.com/en-us/windows/win32/api/audioclient/nn-audioclient-iaudiorenderclient
+
+ Note this section:
+
+ When releasing an IAudioRenderClient interface instance, the client must call the interface's
+ Release method from the same thread as the call to IAudioClient::GetService that created the
+ object.
+ */
+ {
+ result = ma_mutex_init(&pContext->wasapi.commandLock);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ result = ma_semaphore_init(0, &pContext->wasapi.commandSem);
+ if (result != MA_SUCCESS) {
+ ma_mutex_uninit(&pContext->wasapi.commandLock);
+ return result;
+ }
+
+ result = ma_thread_create(&pContext->wasapi.commandThread, ma_thread_priority_normal, 0, ma_context_command_thread__wasapi, pContext, &pContext->allocationCallbacks);
+ if (result != MA_SUCCESS) {
+ ma_semaphore_uninit(&pContext->wasapi.commandSem);
+ ma_mutex_uninit(&pContext->wasapi.commandLock);
+ return result;
+ }
+ }
+
+
+ pCallbacks->onContextInit = ma_context_init__wasapi;
+ pCallbacks->onContextUninit = ma_context_uninit__wasapi;
+ pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__wasapi;
+ pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__wasapi;
+ pCallbacks->onDeviceInit = ma_device_init__wasapi;
+ pCallbacks->onDeviceUninit = ma_device_uninit__wasapi;
+ pCallbacks->onDeviceStart = ma_device_start__wasapi;
+ pCallbacks->onDeviceStop = ma_device_stop__wasapi;
+ pCallbacks->onDeviceRead = ma_device_read__wasapi;
+ pCallbacks->onDeviceWrite = ma_device_write__wasapi;
+ pCallbacks->onDeviceDataLoop = NULL;
+ pCallbacks->onDeviceDataLoopWakeup = ma_device_data_loop_wakeup__wasapi;
+
+ return MA_SUCCESS;
+}
+#endif
+
+/******************************************************************************
+
+DirectSound Backend
+
+******************************************************************************/
+#ifdef MA_HAS_DSOUND
+/*#include */
+
+/*static const GUID MA_GUID_IID_DirectSoundNotify = {0xb0210783, 0x89cd, 0x11d0, {0xaf, 0x08, 0x00, 0xa0, 0xc9, 0x25, 0xcd, 0x16}};*/
+
+/* miniaudio only uses priority or exclusive modes. */
+#define MA_DSSCL_NORMAL 1
+#define MA_DSSCL_PRIORITY 2
+#define MA_DSSCL_EXCLUSIVE 3
+#define MA_DSSCL_WRITEPRIMARY 4
+
+#define MA_DSCAPS_PRIMARYMONO 0x00000001
+#define MA_DSCAPS_PRIMARYSTEREO 0x00000002
+#define MA_DSCAPS_PRIMARY8BIT 0x00000004
+#define MA_DSCAPS_PRIMARY16BIT 0x00000008
+#define MA_DSCAPS_CONTINUOUSRATE 0x00000010
+#define MA_DSCAPS_EMULDRIVER 0x00000020
+#define MA_DSCAPS_CERTIFIED 0x00000040
+#define MA_DSCAPS_SECONDARYMONO 0x00000100
+#define MA_DSCAPS_SECONDARYSTEREO 0x00000200
+#define MA_DSCAPS_SECONDARY8BIT 0x00000400
+#define MA_DSCAPS_SECONDARY16BIT 0x00000800
+
+#define MA_DSBCAPS_PRIMARYBUFFER 0x00000001
+#define MA_DSBCAPS_STATIC 0x00000002
+#define MA_DSBCAPS_LOCHARDWARE 0x00000004
+#define MA_DSBCAPS_LOCSOFTWARE 0x00000008
+#define MA_DSBCAPS_CTRL3D 0x00000010
+#define MA_DSBCAPS_CTRLFREQUENCY 0x00000020
+#define MA_DSBCAPS_CTRLPAN 0x00000040
+#define MA_DSBCAPS_CTRLVOLUME 0x00000080
+#define MA_DSBCAPS_CTRLPOSITIONNOTIFY 0x00000100
+#define MA_DSBCAPS_CTRLFX 0x00000200
+#define MA_DSBCAPS_STICKYFOCUS 0x00004000
+#define MA_DSBCAPS_GLOBALFOCUS 0x00008000
+#define MA_DSBCAPS_GETCURRENTPOSITION2 0x00010000
+#define MA_DSBCAPS_MUTE3DATMAXDISTANCE 0x00020000
+#define MA_DSBCAPS_LOCDEFER 0x00040000
+#define MA_DSBCAPS_TRUEPLAYPOSITION 0x00080000
+
+#define MA_DSBPLAY_LOOPING 0x00000001
+#define MA_DSBPLAY_LOCHARDWARE 0x00000002
+#define MA_DSBPLAY_LOCSOFTWARE 0x00000004
+#define MA_DSBPLAY_TERMINATEBY_TIME 0x00000008
+#define MA_DSBPLAY_TERMINATEBY_DISTANCE 0x00000010
+#define MA_DSBPLAY_TERMINATEBY_PRIORITY 0x00000020
+
+#define MA_DSCBSTART_LOOPING 0x00000001
+
+typedef struct
+{
+ DWORD dwSize;
+ DWORD dwFlags;
+ DWORD dwBufferBytes;
+ DWORD dwReserved;
+ WAVEFORMATEX* lpwfxFormat;
+ GUID guid3DAlgorithm;
+} MA_DSBUFFERDESC;
+
+typedef struct
+{
+ DWORD dwSize;
+ DWORD dwFlags;
+ DWORD dwBufferBytes;
+ DWORD dwReserved;
+ WAVEFORMATEX* lpwfxFormat;
+ DWORD dwFXCount;
+ void* lpDSCFXDesc; /* <-- miniaudio doesn't use this, so set to void*. */
+} MA_DSCBUFFERDESC;
+
+typedef struct
+{
+ DWORD dwSize;
+ DWORD dwFlags;
+ DWORD dwMinSecondarySampleRate;
+ DWORD dwMaxSecondarySampleRate;
+ DWORD dwPrimaryBuffers;
+ DWORD dwMaxHwMixingAllBuffers;
+ DWORD dwMaxHwMixingStaticBuffers;
+ DWORD dwMaxHwMixingStreamingBuffers;
+ DWORD dwFreeHwMixingAllBuffers;
+ DWORD dwFreeHwMixingStaticBuffers;
+ DWORD dwFreeHwMixingStreamingBuffers;
+ DWORD dwMaxHw3DAllBuffers;
+ DWORD dwMaxHw3DStaticBuffers;
+ DWORD dwMaxHw3DStreamingBuffers;
+ DWORD dwFreeHw3DAllBuffers;
+ DWORD dwFreeHw3DStaticBuffers;
+ DWORD dwFreeHw3DStreamingBuffers;
+ DWORD dwTotalHwMemBytes;
+ DWORD dwFreeHwMemBytes;
+ DWORD dwMaxContigFreeHwMemBytes;
+ DWORD dwUnlockTransferRateHwBuffers;
+ DWORD dwPlayCpuOverheadSwBuffers;
+ DWORD dwReserved1;
+ DWORD dwReserved2;
+} MA_DSCAPS;
+
+typedef struct
+{
+ DWORD dwSize;
+ DWORD dwFlags;
+ DWORD dwBufferBytes;
+ DWORD dwUnlockTransferRate;
+ DWORD dwPlayCpuOverhead;
+} MA_DSBCAPS;
+
+typedef struct
+{
+ DWORD dwSize;
+ DWORD dwFlags;
+ DWORD dwFormats;
+ DWORD dwChannels;
+} MA_DSCCAPS;
+
+typedef struct
+{
+ DWORD dwSize;
+ DWORD dwFlags;
+ DWORD dwBufferBytes;
+ DWORD dwReserved;
+} MA_DSCBCAPS;
+
+typedef struct
+{
+ DWORD dwOffset;
+ HANDLE hEventNotify;
+} MA_DSBPOSITIONNOTIFY;
+
+typedef struct ma_IDirectSound ma_IDirectSound;
+typedef struct ma_IDirectSoundBuffer ma_IDirectSoundBuffer;
+typedef struct ma_IDirectSoundCapture ma_IDirectSoundCapture;
+typedef struct ma_IDirectSoundCaptureBuffer ma_IDirectSoundCaptureBuffer;
+typedef struct ma_IDirectSoundNotify ma_IDirectSoundNotify;
+
+
+/*
+COM objects. The way these work is that you have a vtable (a list of function pointers, kind of
+like how C++ works internally), and then you have a structure with a single member, which is a
+pointer to the vtable. The vtable is where the methods of the object are defined. Methods need
+to be in a specific order, and parent classes need to have their methods declared first.
+*/
+
+/* IDirectSound */
+typedef struct
+{
+ /* IUnknown */
+ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSound* pThis, const IID* const riid, void** ppObject);
+ ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSound* pThis);
+ ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSound* pThis);
+
+ /* IDirectSound */
+ HRESULT (STDMETHODCALLTYPE * CreateSoundBuffer) (ma_IDirectSound* pThis, const MA_DSBUFFERDESC* pDSBufferDesc, ma_IDirectSoundBuffer** ppDSBuffer, void* pUnkOuter);
+ HRESULT (STDMETHODCALLTYPE * GetCaps) (ma_IDirectSound* pThis, MA_DSCAPS* pDSCaps);
+ HRESULT (STDMETHODCALLTYPE * DuplicateSoundBuffer)(ma_IDirectSound* pThis, ma_IDirectSoundBuffer* pDSBufferOriginal, ma_IDirectSoundBuffer** ppDSBufferDuplicate);
+ HRESULT (STDMETHODCALLTYPE * SetCooperativeLevel) (ma_IDirectSound* pThis, HWND hwnd, DWORD dwLevel);
+ HRESULT (STDMETHODCALLTYPE * Compact) (ma_IDirectSound* pThis);
+ HRESULT (STDMETHODCALLTYPE * GetSpeakerConfig) (ma_IDirectSound* pThis, DWORD* pSpeakerConfig);
+ HRESULT (STDMETHODCALLTYPE * SetSpeakerConfig) (ma_IDirectSound* pThis, DWORD dwSpeakerConfig);
+ HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IDirectSound* pThis, const GUID* pGuidDevice);
+} ma_IDirectSoundVtbl;
+struct ma_IDirectSound
+{
+ ma_IDirectSoundVtbl* lpVtbl;
+};
+static MA_INLINE HRESULT ma_IDirectSound_QueryInterface(ma_IDirectSound* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
+static MA_INLINE ULONG ma_IDirectSound_AddRef(ma_IDirectSound* pThis) { return pThis->lpVtbl->AddRef(pThis); }
+static MA_INLINE ULONG ma_IDirectSound_Release(ma_IDirectSound* pThis) { return pThis->lpVtbl->Release(pThis); }
+static MA_INLINE HRESULT ma_IDirectSound_CreateSoundBuffer(ma_IDirectSound* pThis, const MA_DSBUFFERDESC* pDSBufferDesc, ma_IDirectSoundBuffer** ppDSBuffer, void* pUnkOuter) { return pThis->lpVtbl->CreateSoundBuffer(pThis, pDSBufferDesc, ppDSBuffer, pUnkOuter); }
+static MA_INLINE HRESULT ma_IDirectSound_GetCaps(ma_IDirectSound* pThis, MA_DSCAPS* pDSCaps) { return pThis->lpVtbl->GetCaps(pThis, pDSCaps); }
+static MA_INLINE HRESULT ma_IDirectSound_DuplicateSoundBuffer(ma_IDirectSound* pThis, ma_IDirectSoundBuffer* pDSBufferOriginal, ma_IDirectSoundBuffer** ppDSBufferDuplicate) { return pThis->lpVtbl->DuplicateSoundBuffer(pThis, pDSBufferOriginal, ppDSBufferDuplicate); }
+static MA_INLINE HRESULT ma_IDirectSound_SetCooperativeLevel(ma_IDirectSound* pThis, HWND hwnd, DWORD dwLevel) { return pThis->lpVtbl->SetCooperativeLevel(pThis, hwnd, dwLevel); }
+static MA_INLINE HRESULT ma_IDirectSound_Compact(ma_IDirectSound* pThis) { return pThis->lpVtbl->Compact(pThis); }
+static MA_INLINE HRESULT ma_IDirectSound_GetSpeakerConfig(ma_IDirectSound* pThis, DWORD* pSpeakerConfig) { return pThis->lpVtbl->GetSpeakerConfig(pThis, pSpeakerConfig); }
+static MA_INLINE HRESULT ma_IDirectSound_SetSpeakerConfig(ma_IDirectSound* pThis, DWORD dwSpeakerConfig) { return pThis->lpVtbl->SetSpeakerConfig(pThis, dwSpeakerConfig); }
+static MA_INLINE HRESULT ma_IDirectSound_Initialize(ma_IDirectSound* pThis, const GUID* pGuidDevice) { return pThis->lpVtbl->Initialize(pThis, pGuidDevice); }
+
+
+/* IDirectSoundBuffer */
+typedef struct
+{
+ /* IUnknown */
+ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSoundBuffer* pThis, const IID* const riid, void** ppObject);
+ ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSoundBuffer* pThis);
+ ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSoundBuffer* pThis);
+
+ /* IDirectSoundBuffer */
+ HRESULT (STDMETHODCALLTYPE * GetCaps) (ma_IDirectSoundBuffer* pThis, MA_DSBCAPS* pDSBufferCaps);
+ HRESULT (STDMETHODCALLTYPE * GetCurrentPosition)(ma_IDirectSoundBuffer* pThis, DWORD* pCurrentPlayCursor, DWORD* pCurrentWriteCursor);
+ HRESULT (STDMETHODCALLTYPE * GetFormat) (ma_IDirectSoundBuffer* pThis, WAVEFORMATEX* pFormat, DWORD dwSizeAllocated, DWORD* pSizeWritten);
+ HRESULT (STDMETHODCALLTYPE * GetVolume) (ma_IDirectSoundBuffer* pThis, LONG* pVolume);
+ HRESULT (STDMETHODCALLTYPE * GetPan) (ma_IDirectSoundBuffer* pThis, LONG* pPan);
+ HRESULT (STDMETHODCALLTYPE * GetFrequency) (ma_IDirectSoundBuffer* pThis, DWORD* pFrequency);
+ HRESULT (STDMETHODCALLTYPE * GetStatus) (ma_IDirectSoundBuffer* pThis, DWORD* pStatus);
+ HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IDirectSoundBuffer* pThis, ma_IDirectSound* pDirectSound, const MA_DSBUFFERDESC* pDSBufferDesc);
+ HRESULT (STDMETHODCALLTYPE * Lock) (ma_IDirectSoundBuffer* pThis, DWORD dwOffset, DWORD dwBytes, void** ppAudioPtr1, DWORD* pAudioBytes1, void** ppAudioPtr2, DWORD* pAudioBytes2, DWORD dwFlags);
+ HRESULT (STDMETHODCALLTYPE * Play) (ma_IDirectSoundBuffer* pThis, DWORD dwReserved1, DWORD dwPriority, DWORD dwFlags);
+ HRESULT (STDMETHODCALLTYPE * SetCurrentPosition)(ma_IDirectSoundBuffer* pThis, DWORD dwNewPosition);
+ HRESULT (STDMETHODCALLTYPE * SetFormat) (ma_IDirectSoundBuffer* pThis, const WAVEFORMATEX* pFormat);
+ HRESULT (STDMETHODCALLTYPE * SetVolume) (ma_IDirectSoundBuffer* pThis, LONG volume);
+ HRESULT (STDMETHODCALLTYPE * SetPan) (ma_IDirectSoundBuffer* pThis, LONG pan);
+ HRESULT (STDMETHODCALLTYPE * SetFrequency) (ma_IDirectSoundBuffer* pThis, DWORD dwFrequency);
+ HRESULT (STDMETHODCALLTYPE * Stop) (ma_IDirectSoundBuffer* pThis);
+ HRESULT (STDMETHODCALLTYPE * Unlock) (ma_IDirectSoundBuffer* pThis, void* pAudioPtr1, DWORD dwAudioBytes1, void* pAudioPtr2, DWORD dwAudioBytes2);
+ HRESULT (STDMETHODCALLTYPE * Restore) (ma_IDirectSoundBuffer* pThis);
+} ma_IDirectSoundBufferVtbl;
+struct ma_IDirectSoundBuffer
+{
+ ma_IDirectSoundBufferVtbl* lpVtbl;
+};
+static MA_INLINE HRESULT ma_IDirectSoundBuffer_QueryInterface(ma_IDirectSoundBuffer* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
+static MA_INLINE ULONG ma_IDirectSoundBuffer_AddRef(ma_IDirectSoundBuffer* pThis) { return pThis->lpVtbl->AddRef(pThis); }
+static MA_INLINE ULONG ma_IDirectSoundBuffer_Release(ma_IDirectSoundBuffer* pThis) { return pThis->lpVtbl->Release(pThis); }
+static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetCaps(ma_IDirectSoundBuffer* pThis, MA_DSBCAPS* pDSBufferCaps) { return pThis->lpVtbl->GetCaps(pThis, pDSBufferCaps); }
+static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetCurrentPosition(ma_IDirectSoundBuffer* pThis, DWORD* pCurrentPlayCursor, DWORD* pCurrentWriteCursor) { return pThis->lpVtbl->GetCurrentPosition(pThis, pCurrentPlayCursor, pCurrentWriteCursor); }
+static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetFormat(ma_IDirectSoundBuffer* pThis, WAVEFORMATEX* pFormat, DWORD dwSizeAllocated, DWORD* pSizeWritten) { return pThis->lpVtbl->GetFormat(pThis, pFormat, dwSizeAllocated, pSizeWritten); }
+static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetVolume(ma_IDirectSoundBuffer* pThis, LONG* pVolume) { return pThis->lpVtbl->GetVolume(pThis, pVolume); }
+static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetPan(ma_IDirectSoundBuffer* pThis, LONG* pPan) { return pThis->lpVtbl->GetPan(pThis, pPan); }
+static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetFrequency(ma_IDirectSoundBuffer* pThis, DWORD* pFrequency) { return pThis->lpVtbl->GetFrequency(pThis, pFrequency); }
+static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetStatus(ma_IDirectSoundBuffer* pThis, DWORD* pStatus) { return pThis->lpVtbl->GetStatus(pThis, pStatus); }
+static MA_INLINE HRESULT ma_IDirectSoundBuffer_Initialize(ma_IDirectSoundBuffer* pThis, ma_IDirectSound* pDirectSound, const MA_DSBUFFERDESC* pDSBufferDesc) { return pThis->lpVtbl->Initialize(pThis, pDirectSound, pDSBufferDesc); }
+static MA_INLINE HRESULT ma_IDirectSoundBuffer_Lock(ma_IDirectSoundBuffer* pThis, DWORD dwOffset, DWORD dwBytes, void** ppAudioPtr1, DWORD* pAudioBytes1, void** ppAudioPtr2, DWORD* pAudioBytes2, DWORD dwFlags) { return pThis->lpVtbl->Lock(pThis, dwOffset, dwBytes, ppAudioPtr1, pAudioBytes1, ppAudioPtr2, pAudioBytes2, dwFlags); }
+static MA_INLINE HRESULT ma_IDirectSoundBuffer_Play(ma_IDirectSoundBuffer* pThis, DWORD dwReserved1, DWORD dwPriority, DWORD dwFlags) { return pThis->lpVtbl->Play(pThis, dwReserved1, dwPriority, dwFlags); }
+static MA_INLINE HRESULT ma_IDirectSoundBuffer_SetCurrentPosition(ma_IDirectSoundBuffer* pThis, DWORD dwNewPosition) { return pThis->lpVtbl->SetCurrentPosition(pThis, dwNewPosition); }
+static MA_INLINE HRESULT ma_IDirectSoundBuffer_SetFormat(ma_IDirectSoundBuffer* pThis, const WAVEFORMATEX* pFormat) { return pThis->lpVtbl->SetFormat(pThis, pFormat); }
+static MA_INLINE HRESULT ma_IDirectSoundBuffer_SetVolume(ma_IDirectSoundBuffer* pThis, LONG volume) { return pThis->lpVtbl->SetVolume(pThis, volume); }
+static MA_INLINE HRESULT ma_IDirectSoundBuffer_SetPan(ma_IDirectSoundBuffer* pThis, LONG pan) { return pThis->lpVtbl->SetPan(pThis, pan); }
+static MA_INLINE HRESULT ma_IDirectSoundBuffer_SetFrequency(ma_IDirectSoundBuffer* pThis, DWORD dwFrequency) { return pThis->lpVtbl->SetFrequency(pThis, dwFrequency); }
+static MA_INLINE HRESULT ma_IDirectSoundBuffer_Stop(ma_IDirectSoundBuffer* pThis) { return pThis->lpVtbl->Stop(pThis); }
+static MA_INLINE HRESULT ma_IDirectSoundBuffer_Unlock(ma_IDirectSoundBuffer* pThis, void* pAudioPtr1, DWORD dwAudioBytes1, void* pAudioPtr2, DWORD dwAudioBytes2) { return pThis->lpVtbl->Unlock(pThis, pAudioPtr1, dwAudioBytes1, pAudioPtr2, dwAudioBytes2); }
+static MA_INLINE HRESULT ma_IDirectSoundBuffer_Restore(ma_IDirectSoundBuffer* pThis) { return pThis->lpVtbl->Restore(pThis); }
+
+
+/* IDirectSoundCapture */
+typedef struct
+{
+ /* IUnknown */
+ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSoundCapture* pThis, const IID* const riid, void** ppObject);
+ ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSoundCapture* pThis);
+ ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSoundCapture* pThis);
+
+ /* IDirectSoundCapture */
+ HRESULT (STDMETHODCALLTYPE * CreateCaptureBuffer)(ma_IDirectSoundCapture* pThis, const MA_DSCBUFFERDESC* pDSCBufferDesc, ma_IDirectSoundCaptureBuffer** ppDSCBuffer, void* pUnkOuter);
+ HRESULT (STDMETHODCALLTYPE * GetCaps) (ma_IDirectSoundCapture* pThis, MA_DSCCAPS* pDSCCaps);
+ HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IDirectSoundCapture* pThis, const GUID* pGuidDevice);
+} ma_IDirectSoundCaptureVtbl;
+struct ma_IDirectSoundCapture
+{
+ ma_IDirectSoundCaptureVtbl* lpVtbl;
+};
+static MA_INLINE HRESULT ma_IDirectSoundCapture_QueryInterface (ma_IDirectSoundCapture* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
+static MA_INLINE ULONG ma_IDirectSoundCapture_AddRef (ma_IDirectSoundCapture* pThis) { return pThis->lpVtbl->AddRef(pThis); }
+static MA_INLINE ULONG ma_IDirectSoundCapture_Release (ma_IDirectSoundCapture* pThis) { return pThis->lpVtbl->Release(pThis); }
+static MA_INLINE HRESULT ma_IDirectSoundCapture_CreateCaptureBuffer(ma_IDirectSoundCapture* pThis, const MA_DSCBUFFERDESC* pDSCBufferDesc, ma_IDirectSoundCaptureBuffer** ppDSCBuffer, void* pUnkOuter) { return pThis->lpVtbl->CreateCaptureBuffer(pThis, pDSCBufferDesc, ppDSCBuffer, pUnkOuter); }
+static MA_INLINE HRESULT ma_IDirectSoundCapture_GetCaps (ma_IDirectSoundCapture* pThis, MA_DSCCAPS* pDSCCaps) { return pThis->lpVtbl->GetCaps(pThis, pDSCCaps); }
+static MA_INLINE HRESULT ma_IDirectSoundCapture_Initialize (ma_IDirectSoundCapture* pThis, const GUID* pGuidDevice) { return pThis->lpVtbl->Initialize(pThis, pGuidDevice); }
+
+
+/* IDirectSoundCaptureBuffer */
+typedef struct
+{
+ /* IUnknown */
+ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSoundCaptureBuffer* pThis, const IID* const riid, void** ppObject);
+ ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSoundCaptureBuffer* pThis);
+ ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSoundCaptureBuffer* pThis);
+
+ /* IDirectSoundCaptureBuffer */
+ HRESULT (STDMETHODCALLTYPE * GetCaps) (ma_IDirectSoundCaptureBuffer* pThis, MA_DSCBCAPS* pDSCBCaps);
+ HRESULT (STDMETHODCALLTYPE * GetCurrentPosition)(ma_IDirectSoundCaptureBuffer* pThis, DWORD* pCapturePosition, DWORD* pReadPosition);
+ HRESULT (STDMETHODCALLTYPE * GetFormat) (ma_IDirectSoundCaptureBuffer* pThis, WAVEFORMATEX* pFormat, DWORD dwSizeAllocated, DWORD* pSizeWritten);
+ HRESULT (STDMETHODCALLTYPE * GetStatus) (ma_IDirectSoundCaptureBuffer* pThis, DWORD* pStatus);
+ HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IDirectSoundCaptureBuffer* pThis, ma_IDirectSoundCapture* pDirectSoundCapture, const MA_DSCBUFFERDESC* pDSCBufferDesc);
+ HRESULT (STDMETHODCALLTYPE * Lock) (ma_IDirectSoundCaptureBuffer* pThis, DWORD dwOffset, DWORD dwBytes, void** ppAudioPtr1, DWORD* pAudioBytes1, void** ppAudioPtr2, DWORD* pAudioBytes2, DWORD dwFlags);
+ HRESULT (STDMETHODCALLTYPE * Start) (ma_IDirectSoundCaptureBuffer* pThis, DWORD dwFlags);
+ HRESULT (STDMETHODCALLTYPE * Stop) (ma_IDirectSoundCaptureBuffer* pThis);
+ HRESULT (STDMETHODCALLTYPE * Unlock) (ma_IDirectSoundCaptureBuffer* pThis, void* pAudioPtr1, DWORD dwAudioBytes1, void* pAudioPtr2, DWORD dwAudioBytes2);
+} ma_IDirectSoundCaptureBufferVtbl;
+struct ma_IDirectSoundCaptureBuffer
+{
+ ma_IDirectSoundCaptureBufferVtbl* lpVtbl;
+};
+static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_QueryInterface(ma_IDirectSoundCaptureBuffer* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
+static MA_INLINE ULONG ma_IDirectSoundCaptureBuffer_AddRef(ma_IDirectSoundCaptureBuffer* pThis) { return pThis->lpVtbl->AddRef(pThis); }
+static MA_INLINE ULONG ma_IDirectSoundCaptureBuffer_Release(ma_IDirectSoundCaptureBuffer* pThis) { return pThis->lpVtbl->Release(pThis); }
+static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_GetCaps(ma_IDirectSoundCaptureBuffer* pThis, MA_DSCBCAPS* pDSCBCaps) { return pThis->lpVtbl->GetCaps(pThis, pDSCBCaps); }
+static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_GetCurrentPosition(ma_IDirectSoundCaptureBuffer* pThis, DWORD* pCapturePosition, DWORD* pReadPosition) { return pThis->lpVtbl->GetCurrentPosition(pThis, pCapturePosition, pReadPosition); }
+static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_GetFormat(ma_IDirectSoundCaptureBuffer* pThis, WAVEFORMATEX* pFormat, DWORD dwSizeAllocated, DWORD* pSizeWritten) { return pThis->lpVtbl->GetFormat(pThis, pFormat, dwSizeAllocated, pSizeWritten); }
+static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_GetStatus(ma_IDirectSoundCaptureBuffer* pThis, DWORD* pStatus) { return pThis->lpVtbl->GetStatus(pThis, pStatus); }
+static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_Initialize(ma_IDirectSoundCaptureBuffer* pThis, ma_IDirectSoundCapture* pDirectSoundCapture, const MA_DSCBUFFERDESC* pDSCBufferDesc) { return pThis->lpVtbl->Initialize(pThis, pDirectSoundCapture, pDSCBufferDesc); }
+static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_Lock(ma_IDirectSoundCaptureBuffer* pThis, DWORD dwOffset, DWORD dwBytes, void** ppAudioPtr1, DWORD* pAudioBytes1, void** ppAudioPtr2, DWORD* pAudioBytes2, DWORD dwFlags) { return pThis->lpVtbl->Lock(pThis, dwOffset, dwBytes, ppAudioPtr1, pAudioBytes1, ppAudioPtr2, pAudioBytes2, dwFlags); }
+static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_Start(ma_IDirectSoundCaptureBuffer* pThis, DWORD dwFlags) { return pThis->lpVtbl->Start(pThis, dwFlags); }
+static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_Stop(ma_IDirectSoundCaptureBuffer* pThis) { return pThis->lpVtbl->Stop(pThis); }
+static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_Unlock(ma_IDirectSoundCaptureBuffer* pThis, void* pAudioPtr1, DWORD dwAudioBytes1, void* pAudioPtr2, DWORD dwAudioBytes2) { return pThis->lpVtbl->Unlock(pThis, pAudioPtr1, dwAudioBytes1, pAudioPtr2, dwAudioBytes2); }
+
+
+/* IDirectSoundNotify */
+typedef struct
+{
+ /* IUnknown */
+ HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSoundNotify* pThis, const IID* const riid, void** ppObject);
+ ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSoundNotify* pThis);
+ ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSoundNotify* pThis);
+
+ /* IDirectSoundNotify */
+ HRESULT (STDMETHODCALLTYPE * SetNotificationPositions)(ma_IDirectSoundNotify* pThis, DWORD dwPositionNotifies, const MA_DSBPOSITIONNOTIFY* pPositionNotifies);
+} ma_IDirectSoundNotifyVtbl;
+struct ma_IDirectSoundNotify
+{
+ ma_IDirectSoundNotifyVtbl* lpVtbl;
+};
+static MA_INLINE HRESULT ma_IDirectSoundNotify_QueryInterface(ma_IDirectSoundNotify* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
+static MA_INLINE ULONG ma_IDirectSoundNotify_AddRef(ma_IDirectSoundNotify* pThis) { return pThis->lpVtbl->AddRef(pThis); }
+static MA_INLINE ULONG ma_IDirectSoundNotify_Release(ma_IDirectSoundNotify* pThis) { return pThis->lpVtbl->Release(pThis); }
+static MA_INLINE HRESULT ma_IDirectSoundNotify_SetNotificationPositions(ma_IDirectSoundNotify* pThis, DWORD dwPositionNotifies, const MA_DSBPOSITIONNOTIFY* pPositionNotifies) { return pThis->lpVtbl->SetNotificationPositions(pThis, dwPositionNotifies, pPositionNotifies); }
+
+
+typedef BOOL (CALLBACK * ma_DSEnumCallbackAProc) (LPGUID pDeviceGUID, LPCSTR pDeviceDescription, LPCSTR pModule, LPVOID pContext);
+typedef HRESULT (WINAPI * ma_DirectSoundCreateProc) (const GUID* pcGuidDevice, ma_IDirectSound** ppDS8, LPUNKNOWN pUnkOuter);
+typedef HRESULT (WINAPI * ma_DirectSoundEnumerateAProc) (ma_DSEnumCallbackAProc pDSEnumCallback, LPVOID pContext);
+typedef HRESULT (WINAPI * ma_DirectSoundCaptureCreateProc) (const GUID* pcGuidDevice, ma_IDirectSoundCapture** ppDSC8, LPUNKNOWN pUnkOuter);
+typedef HRESULT (WINAPI * ma_DirectSoundCaptureEnumerateAProc)(ma_DSEnumCallbackAProc pDSEnumCallback, LPVOID pContext);
+
+static ma_uint32 ma_get_best_sample_rate_within_range(ma_uint32 sampleRateMin, ma_uint32 sampleRateMax)
+{
+ /* Normalize the range in case we were given something stupid. */
+ if (sampleRateMin < (ma_uint32)ma_standard_sample_rate_min) {
+ sampleRateMin = (ma_uint32)ma_standard_sample_rate_min;
+ }
+ if (sampleRateMax > (ma_uint32)ma_standard_sample_rate_max) {
+ sampleRateMax = (ma_uint32)ma_standard_sample_rate_max;
+ }
+ if (sampleRateMin > sampleRateMax) {
+ sampleRateMin = sampleRateMax;
+ }
+
+ if (sampleRateMin == sampleRateMax) {
+ return sampleRateMax;
+ } else {
+ size_t iStandardRate;
+ for (iStandardRate = 0; iStandardRate < ma_countof(g_maStandardSampleRatePriorities); ++iStandardRate) {
+ ma_uint32 standardRate = g_maStandardSampleRatePriorities[iStandardRate];
+ if (standardRate >= sampleRateMin && standardRate <= sampleRateMax) {
+ return standardRate;
+ }
+ }
+ }
+
+ /* Should never get here. */
+ MA_ASSERT(MA_FALSE);
+ return 0;
+}
+
+/*
+Retrieves the channel count and channel map for the given speaker configuration. If the speaker configuration is unknown,
+the channel count and channel map will be left unmodified.
+*/
+static void ma_get_channels_from_speaker_config__dsound(DWORD speakerConfig, WORD* pChannelsOut, DWORD* pChannelMapOut)
+{
+ WORD channels;
+ DWORD channelMap;
+
+ channels = 0;
+ if (pChannelsOut != NULL) {
+ channels = *pChannelsOut;
+ }
+
+ channelMap = 0;
+ if (pChannelMapOut != NULL) {
+ channelMap = *pChannelMapOut;
+ }
+
+ /*
+ The speaker configuration is a combination of speaker config and speaker geometry. The lower 8 bits is what we care about. The upper
+ 16 bits is for the geometry.
+ */
+ switch ((BYTE)(speakerConfig)) {
+ case 1 /*DSSPEAKER_HEADPHONE*/: channels = 2; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT; break;
+ case 2 /*DSSPEAKER_MONO*/: channels = 1; channelMap = SPEAKER_FRONT_CENTER; break;
+ case 3 /*DSSPEAKER_QUAD*/: channels = 4; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT; break;
+ case 4 /*DSSPEAKER_STEREO*/: channels = 2; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT; break;
+ case 5 /*DSSPEAKER_SURROUND*/: channels = 4; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_BACK_CENTER; break;
+ case 6 /*DSSPEAKER_5POINT1_BACK*/ /*DSSPEAKER_5POINT1*/: channels = 6; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT; break;
+ case 7 /*DSSPEAKER_7POINT1_WIDE*/ /*DSSPEAKER_7POINT1*/: channels = 8; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_FRONT_LEFT_OF_CENTER | SPEAKER_FRONT_RIGHT_OF_CENTER; break;
+ case 8 /*DSSPEAKER_7POINT1_SURROUND*/: channels = 8; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT; break;
+ case 9 /*DSSPEAKER_5POINT1_SURROUND*/: channels = 6; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT; break;
+ default: break;
+ }
+
+ if (pChannelsOut != NULL) {
+ *pChannelsOut = channels;
+ }
+
+ if (pChannelMapOut != NULL) {
+ *pChannelMapOut = channelMap;
+ }
+}
+
+
+static ma_result ma_context_create_IDirectSound__dsound(ma_context* pContext, ma_share_mode shareMode, const ma_device_id* pDeviceID, ma_IDirectSound** ppDirectSound)
+{
+ ma_IDirectSound* pDirectSound;
+ HWND hWnd;
+ HRESULT hr;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(ppDirectSound != NULL);
+
+ *ppDirectSound = NULL;
+ pDirectSound = NULL;
+
+ if (FAILED(((ma_DirectSoundCreateProc)pContext->dsound.DirectSoundCreate)((pDeviceID == NULL) ? NULL : (const GUID*)pDeviceID->dsound, &pDirectSound, NULL))) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[DirectSound] DirectSoundCreate() failed for playback device.");
+ return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
+ }
+
+ /* The cooperative level must be set before doing anything else. */
+ hWnd = ((MA_PFN_GetForegroundWindow)pContext->win32.GetForegroundWindow)();
+ if (hWnd == NULL) {
+ hWnd = ((MA_PFN_GetDesktopWindow)pContext->win32.GetDesktopWindow)();
+ }
+
+ hr = ma_IDirectSound_SetCooperativeLevel(pDirectSound, hWnd, (shareMode == ma_share_mode_exclusive) ? MA_DSSCL_EXCLUSIVE : MA_DSSCL_PRIORITY);
+ if (FAILED(hr)) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_SetCooperateiveLevel() failed for playback device.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ *ppDirectSound = pDirectSound;
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_create_IDirectSoundCapture__dsound(ma_context* pContext, ma_share_mode shareMode, const ma_device_id* pDeviceID, ma_IDirectSoundCapture** ppDirectSoundCapture)
+{
+ ma_IDirectSoundCapture* pDirectSoundCapture;
+ HRESULT hr;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(ppDirectSoundCapture != NULL);
+
+ /* DirectSound does not support exclusive mode for capture. */
+ if (shareMode == ma_share_mode_exclusive) {
+ return MA_SHARE_MODE_NOT_SUPPORTED;
+ }
+
+ *ppDirectSoundCapture = NULL;
+ pDirectSoundCapture = NULL;
+
+ hr = ((ma_DirectSoundCaptureCreateProc)pContext->dsound.DirectSoundCaptureCreate)((pDeviceID == NULL) ? NULL : (const GUID*)pDeviceID->dsound, &pDirectSoundCapture, NULL);
+ if (FAILED(hr)) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[DirectSound] DirectSoundCaptureCreate() failed for capture device.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ *ppDirectSoundCapture = pDirectSoundCapture;
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_get_format_info_for_IDirectSoundCapture__dsound(ma_context* pContext, ma_IDirectSoundCapture* pDirectSoundCapture, WORD* pChannels, WORD* pBitsPerSample, DWORD* pSampleRate)
+{
+ HRESULT hr;
+ MA_DSCCAPS caps;
+ WORD bitsPerSample;
+ DWORD sampleRate;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pDirectSoundCapture != NULL);
+
+ if (pChannels) {
+ *pChannels = 0;
+ }
+ if (pBitsPerSample) {
+ *pBitsPerSample = 0;
+ }
+ if (pSampleRate) {
+ *pSampleRate = 0;
+ }
+
+ MA_ZERO_OBJECT(&caps);
+ caps.dwSize = sizeof(caps);
+ hr = ma_IDirectSoundCapture_GetCaps(pDirectSoundCapture, &caps);
+ if (FAILED(hr)) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCapture_GetCaps() failed for capture device.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ if (pChannels) {
+ *pChannels = (WORD)caps.dwChannels;
+ }
+
+ /* The device can support multiple formats. We just go through the different formats in order of priority and pick the first one. This the same type of system as the WinMM backend. */
+ bitsPerSample = 16;
+ sampleRate = 48000;
+
+ if (caps.dwChannels == 1) {
+ if ((caps.dwFormats & WAVE_FORMAT_48M16) != 0) {
+ sampleRate = 48000;
+ } else if ((caps.dwFormats & WAVE_FORMAT_44M16) != 0) {
+ sampleRate = 44100;
+ } else if ((caps.dwFormats & WAVE_FORMAT_2M16) != 0) {
+ sampleRate = 22050;
+ } else if ((caps.dwFormats & WAVE_FORMAT_1M16) != 0) {
+ sampleRate = 11025;
+ } else if ((caps.dwFormats & WAVE_FORMAT_96M16) != 0) {
+ sampleRate = 96000;
+ } else {
+ bitsPerSample = 8;
+ if ((caps.dwFormats & WAVE_FORMAT_48M08) != 0) {
+ sampleRate = 48000;
+ } else if ((caps.dwFormats & WAVE_FORMAT_44M08) != 0) {
+ sampleRate = 44100;
+ } else if ((caps.dwFormats & WAVE_FORMAT_2M08) != 0) {
+ sampleRate = 22050;
+ } else if ((caps.dwFormats & WAVE_FORMAT_1M08) != 0) {
+ sampleRate = 11025;
+ } else if ((caps.dwFormats & WAVE_FORMAT_96M08) != 0) {
+ sampleRate = 96000;
+ } else {
+ bitsPerSample = 16; /* Didn't find it. Just fall back to 16-bit. */
+ }
+ }
+ } else if (caps.dwChannels == 2) {
+ if ((caps.dwFormats & WAVE_FORMAT_48S16) != 0) {
+ sampleRate = 48000;
+ } else if ((caps.dwFormats & WAVE_FORMAT_44S16) != 0) {
+ sampleRate = 44100;
+ } else if ((caps.dwFormats & WAVE_FORMAT_2S16) != 0) {
+ sampleRate = 22050;
+ } else if ((caps.dwFormats & WAVE_FORMAT_1S16) != 0) {
+ sampleRate = 11025;
+ } else if ((caps.dwFormats & WAVE_FORMAT_96S16) != 0) {
+ sampleRate = 96000;
+ } else {
+ bitsPerSample = 8;
+ if ((caps.dwFormats & WAVE_FORMAT_48S08) != 0) {
+ sampleRate = 48000;
+ } else if ((caps.dwFormats & WAVE_FORMAT_44S08) != 0) {
+ sampleRate = 44100;
+ } else if ((caps.dwFormats & WAVE_FORMAT_2S08) != 0) {
+ sampleRate = 22050;
+ } else if ((caps.dwFormats & WAVE_FORMAT_1S08) != 0) {
+ sampleRate = 11025;
+ } else if ((caps.dwFormats & WAVE_FORMAT_96S08) != 0) {
+ sampleRate = 96000;
+ } else {
+ bitsPerSample = 16; /* Didn't find it. Just fall back to 16-bit. */
+ }
+ }
+ }
+
+ if (pBitsPerSample) {
+ *pBitsPerSample = bitsPerSample;
+ }
+ if (pSampleRate) {
+ *pSampleRate = sampleRate;
+ }
+
+ return MA_SUCCESS;
+}
+
+
+typedef struct
+{
+ ma_context* pContext;
+ ma_device_type deviceType;
+ ma_enum_devices_callback_proc callback;
+ void* pUserData;
+ ma_bool32 terminated;
+} ma_context_enumerate_devices_callback_data__dsound;
+
+static BOOL CALLBACK ma_context_enumerate_devices_callback__dsound(LPGUID lpGuid, LPCSTR lpcstrDescription, LPCSTR lpcstrModule, LPVOID lpContext)
+{
+ ma_context_enumerate_devices_callback_data__dsound* pData = (ma_context_enumerate_devices_callback_data__dsound*)lpContext;
+ ma_device_info deviceInfo;
+
+ (void)lpcstrModule;
+
+ MA_ZERO_OBJECT(&deviceInfo);
+
+ /* ID. */
+ if (lpGuid != NULL) {
+ MA_COPY_MEMORY(deviceInfo.id.dsound, lpGuid, 16);
+ } else {
+ MA_ZERO_MEMORY(deviceInfo.id.dsound, 16);
+ deviceInfo.isDefault = MA_TRUE;
+ }
+
+ /* Name / Description */
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), lpcstrDescription, (size_t)-1);
+
+
+ /* Call the callback function, but make sure we stop enumerating if the callee requested so. */
+ MA_ASSERT(pData != NULL);
+ pData->terminated = !pData->callback(pData->pContext, pData->deviceType, &deviceInfo, pData->pUserData);
+ if (pData->terminated) {
+ return FALSE; /* Stop enumeration. */
+ } else {
+ return TRUE; /* Continue enumeration. */
+ }
+}
+
+static ma_result ma_context_enumerate_devices__dsound(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+{
+ ma_context_enumerate_devices_callback_data__dsound data;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(callback != NULL);
+
+ data.pContext = pContext;
+ data.callback = callback;
+ data.pUserData = pUserData;
+ data.terminated = MA_FALSE;
+
+ /* Playback. */
+ if (!data.terminated) {
+ data.deviceType = ma_device_type_playback;
+ ((ma_DirectSoundEnumerateAProc)pContext->dsound.DirectSoundEnumerateA)(ma_context_enumerate_devices_callback__dsound, &data);
+ }
+
+ /* Capture. */
+ if (!data.terminated) {
+ data.deviceType = ma_device_type_capture;
+ ((ma_DirectSoundCaptureEnumerateAProc)pContext->dsound.DirectSoundCaptureEnumerateA)(ma_context_enumerate_devices_callback__dsound, &data);
+ }
+
+ return MA_SUCCESS;
+}
+
+
+typedef struct
+{
+ const ma_device_id* pDeviceID;
+ ma_device_info* pDeviceInfo;
+ ma_bool32 found;
+} ma_context_get_device_info_callback_data__dsound;
+
+static BOOL CALLBACK ma_context_get_device_info_callback__dsound(LPGUID lpGuid, LPCSTR lpcstrDescription, LPCSTR lpcstrModule, LPVOID lpContext)
+{
+ ma_context_get_device_info_callback_data__dsound* pData = (ma_context_get_device_info_callback_data__dsound*)lpContext;
+ MA_ASSERT(pData != NULL);
+
+ if ((pData->pDeviceID == NULL || ma_is_guid_null(pData->pDeviceID->dsound)) && (lpGuid == NULL || ma_is_guid_null(lpGuid))) {
+ /* Default device. */
+ ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), lpcstrDescription, (size_t)-1);
+ pData->pDeviceInfo->isDefault = MA_TRUE;
+ pData->found = MA_TRUE;
+ return FALSE; /* Stop enumeration. */
+ } else {
+ /* Not the default device. */
+ if (lpGuid != NULL && pData->pDeviceID != NULL) {
+ if (memcmp(pData->pDeviceID->dsound, lpGuid, sizeof(pData->pDeviceID->dsound)) == 0) {
+ ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), lpcstrDescription, (size_t)-1);
+ pData->found = MA_TRUE;
+ return FALSE; /* Stop enumeration. */
+ }
+ }
+ }
+
+ (void)lpcstrModule;
+ return TRUE;
+}
+
+static ma_result ma_context_get_device_info__dsound(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+{
+ ma_result result;
+ HRESULT hr;
+
+ if (pDeviceID != NULL) {
+ ma_context_get_device_info_callback_data__dsound data;
+
+ /* ID. */
+ MA_COPY_MEMORY(pDeviceInfo->id.dsound, pDeviceID->dsound, 16);
+
+ /* Name / Description. This is retrieved by enumerating over each device until we find that one that matches the input ID. */
+ data.pDeviceID = pDeviceID;
+ data.pDeviceInfo = pDeviceInfo;
+ data.found = MA_FALSE;
+ if (deviceType == ma_device_type_playback) {
+ ((ma_DirectSoundEnumerateAProc)pContext->dsound.DirectSoundEnumerateA)(ma_context_get_device_info_callback__dsound, &data);
+ } else {
+ ((ma_DirectSoundCaptureEnumerateAProc)pContext->dsound.DirectSoundCaptureEnumerateA)(ma_context_get_device_info_callback__dsound, &data);
+ }
+
+ if (!data.found) {
+ return MA_NO_DEVICE;
+ }
+ } else {
+ /* I don't think there's a way to get the name of the default device with DirectSound. In this case we just need to use defaults. */
+
+ /* ID */
+ MA_ZERO_MEMORY(pDeviceInfo->id.dsound, 16);
+
+ /* Name / Description */
+ if (deviceType == ma_device_type_playback) {
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
+ } else {
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
+ }
+
+ pDeviceInfo->isDefault = MA_TRUE;
+ }
+
+ /* Retrieving detailed information is slightly different depending on the device type. */
+ if (deviceType == ma_device_type_playback) {
+ /* Playback. */
+ ma_IDirectSound* pDirectSound;
+ MA_DSCAPS caps;
+ WORD channels;
+
+ result = ma_context_create_IDirectSound__dsound(pContext, ma_share_mode_shared, pDeviceID, &pDirectSound);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ MA_ZERO_OBJECT(&caps);
+ caps.dwSize = sizeof(caps);
+ hr = ma_IDirectSound_GetCaps(pDirectSound, &caps);
+ if (FAILED(hr)) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_GetCaps() failed for playback device.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+
+ /* Channels. Only a single channel count is reported for DirectSound. */
+ if ((caps.dwFlags & MA_DSCAPS_PRIMARYSTEREO) != 0) {
+ /* It supports at least stereo, but could support more. */
+ DWORD speakerConfig;
+
+ channels = 2;
+
+ /* Look at the speaker configuration to get a better idea on the channel count. */
+ hr = ma_IDirectSound_GetSpeakerConfig(pDirectSound, &speakerConfig);
+ if (SUCCEEDED(hr)) {
+ ma_get_channels_from_speaker_config__dsound(speakerConfig, &channels, NULL);
+ }
+ } else {
+ /* It does not support stereo, which means we are stuck with mono. */
+ channels = 1;
+ }
+
+
+ /*
+ In DirectSound, our native formats are centered around sample rates. All formats are supported, and we're only reporting a single channel
+ count. However, DirectSound can report a range of supported sample rates. We're only going to include standard rates known by miniaudio
+ in order to keep the size of this within reason.
+ */
+ if ((caps.dwFlags & MA_DSCAPS_CONTINUOUSRATE) != 0) {
+ /* Multiple sample rates are supported. We'll report in order of our preferred sample rates. */
+ size_t iStandardSampleRate;
+ for (iStandardSampleRate = 0; iStandardSampleRate < ma_countof(g_maStandardSampleRatePriorities); iStandardSampleRate += 1) {
+ ma_uint32 sampleRate = g_maStandardSampleRatePriorities[iStandardSampleRate];
+ if (sampleRate >= caps.dwMinSecondarySampleRate && sampleRate <= caps.dwMaxSecondarySampleRate) {
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = ma_format_unknown;
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = channels;
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = sampleRate;
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = 0;
+ pDeviceInfo->nativeDataFormatCount += 1;
+ }
+ }
+ } else {
+ /* Only a single sample rate is supported. */
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = ma_format_unknown;
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = channels;
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = caps.dwMaxSecondarySampleRate;
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = 0;
+ pDeviceInfo->nativeDataFormatCount += 1;
+ }
+
+ ma_IDirectSound_Release(pDirectSound);
+ } else {
+ /*
+ Capture. This is a little different to playback due to the say the supported formats are reported. Technically capture
+ devices can support a number of different formats, but for simplicity and consistency with ma_device_init() I'm just
+ reporting the best format.
+ */
+ ma_IDirectSoundCapture* pDirectSoundCapture;
+ WORD channels;
+ WORD bitsPerSample;
+ DWORD sampleRate;
+
+ result = ma_context_create_IDirectSoundCapture__dsound(pContext, ma_share_mode_shared, pDeviceID, &pDirectSoundCapture);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ result = ma_context_get_format_info_for_IDirectSoundCapture__dsound(pContext, pDirectSoundCapture, &channels, &bitsPerSample, &sampleRate);
+ if (result != MA_SUCCESS) {
+ ma_IDirectSoundCapture_Release(pDirectSoundCapture);
+ return result;
+ }
+
+ ma_IDirectSoundCapture_Release(pDirectSoundCapture);
+
+ /* The format is always an integer format and is based on the bits per sample. */
+ if (bitsPerSample == 8) {
+ pDeviceInfo->nativeDataFormats[0].format = ma_format_u8;
+ } else if (bitsPerSample == 16) {
+ pDeviceInfo->nativeDataFormats[0].format = ma_format_s16;
+ } else if (bitsPerSample == 24) {
+ pDeviceInfo->nativeDataFormats[0].format = ma_format_s24;
+ } else if (bitsPerSample == 32) {
+ pDeviceInfo->nativeDataFormats[0].format = ma_format_s32;
+ } else {
+ return MA_FORMAT_NOT_SUPPORTED;
+ }
+
+ pDeviceInfo->nativeDataFormats[0].channels = channels;
+ pDeviceInfo->nativeDataFormats[0].sampleRate = sampleRate;
+ pDeviceInfo->nativeDataFormats[0].flags = 0;
+ pDeviceInfo->nativeDataFormatCount = 1;
+ }
+
+ return MA_SUCCESS;
+}
+
+
+
+static ma_result ma_device_uninit__dsound(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
+
+ if (pDevice->dsound.pCaptureBuffer != NULL) {
+ ma_IDirectSoundCaptureBuffer_Release((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer);
+ }
+ if (pDevice->dsound.pCapture != NULL) {
+ ma_IDirectSoundCapture_Release((ma_IDirectSoundCapture*)pDevice->dsound.pCapture);
+ }
+
+ if (pDevice->dsound.pPlaybackBuffer != NULL) {
+ ma_IDirectSoundBuffer_Release((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer);
+ }
+ if (pDevice->dsound.pPlaybackPrimaryBuffer != NULL) {
+ ma_IDirectSoundBuffer_Release((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackPrimaryBuffer);
+ }
+ if (pDevice->dsound.pPlayback != NULL) {
+ ma_IDirectSound_Release((ma_IDirectSound*)pDevice->dsound.pPlayback);
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_config_to_WAVEFORMATEXTENSIBLE(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, const ma_channel* pChannelMap, WAVEFORMATEXTENSIBLE* pWF)
+{
+ GUID subformat;
+
+ if (format == ma_format_unknown) {
+ format = MA_DEFAULT_FORMAT;
+ }
+
+ if (channels == 0) {
+ channels = MA_DEFAULT_CHANNELS;
+ }
+
+ if (sampleRate == 0) {
+ sampleRate = MA_DEFAULT_SAMPLE_RATE;
+ }
+
+ switch (format)
+ {
+ case ma_format_u8:
+ case ma_format_s16:
+ case ma_format_s24:
+ /*case ma_format_s24_32:*/
+ case ma_format_s32:
+ {
+ subformat = MA_GUID_KSDATAFORMAT_SUBTYPE_PCM;
+ } break;
+
+ case ma_format_f32:
+ {
+ subformat = MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
+ } break;
+
+ default:
+ return MA_FORMAT_NOT_SUPPORTED;
+ }
+
+ MA_ZERO_OBJECT(pWF);
+ pWF->Format.cbSize = sizeof(*pWF);
+ pWF->Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
+ pWF->Format.nChannels = (WORD)channels;
+ pWF->Format.nSamplesPerSec = (DWORD)sampleRate;
+ pWF->Format.wBitsPerSample = (WORD)(ma_get_bytes_per_sample(format)*8);
+ pWF->Format.nBlockAlign = (WORD)(pWF->Format.nChannels * pWF->Format.wBitsPerSample / 8);
+ pWF->Format.nAvgBytesPerSec = pWF->Format.nBlockAlign * pWF->Format.nSamplesPerSec;
+ pWF->Samples.wValidBitsPerSample = pWF->Format.wBitsPerSample;
+ pWF->dwChannelMask = ma_channel_map_to_channel_mask__win32(pChannelMap, channels);
+ pWF->SubFormat = subformat;
+
+ return MA_SUCCESS;
+}
+
+static ma_uint32 ma_calculate_period_size_in_frames_from_descriptor__dsound(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile)
+{
+ /*
+ DirectSound has a minimum period size of 20ms. In practice, this doesn't seem to be enough for
+ reliable glitch-free processing so going to use 30ms instead.
+ */
+ ma_uint32 minPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(30, nativeSampleRate);
+ ma_uint32 periodSizeInFrames;
+
+ periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, nativeSampleRate, performanceProfile);
+ if (periodSizeInFrames < minPeriodSizeInFrames) {
+ periodSizeInFrames = minPeriodSizeInFrames;
+ }
+
+ return periodSizeInFrames;
+}
+
+static ma_result ma_device_init__dsound(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+{
+ ma_result result;
+ HRESULT hr;
+
+ MA_ASSERT(pDevice != NULL);
+
+ MA_ZERO_OBJECT(&pDevice->dsound);
+
+ if (pConfig->deviceType == ma_device_type_loopback) {
+ return MA_DEVICE_TYPE_NOT_SUPPORTED;
+ }
+
+ /*
+ Unfortunately DirectSound uses different APIs and data structures for playback and catpure devices. We need to initialize
+ the capture device first because we'll want to match it's buffer size and period count on the playback side if we're using
+ full-duplex mode.
+ */
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
+ WAVEFORMATEXTENSIBLE wf;
+ MA_DSCBUFFERDESC descDS;
+ ma_uint32 periodSizeInFrames;
+ ma_uint32 periodCount;
+ char rawdata[1024]; /* <-- Ugly hack to avoid a malloc() due to a crappy DirectSound API. */
+ WAVEFORMATEXTENSIBLE* pActualFormat;
+
+ result = ma_config_to_WAVEFORMATEXTENSIBLE(pDescriptorCapture->format, pDescriptorCapture->channels, pDescriptorCapture->sampleRate, pDescriptorCapture->channelMap, &wf);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ result = ma_context_create_IDirectSoundCapture__dsound(pDevice->pContext, pDescriptorCapture->shareMode, pDescriptorCapture->pDeviceID, (ma_IDirectSoundCapture**)&pDevice->dsound.pCapture);
+ if (result != MA_SUCCESS) {
+ ma_device_uninit__dsound(pDevice);
+ return result;
+ }
+
+ result = ma_context_get_format_info_for_IDirectSoundCapture__dsound(pDevice->pContext, (ma_IDirectSoundCapture*)pDevice->dsound.pCapture, &wf.Format.nChannels, &wf.Format.wBitsPerSample, &wf.Format.nSamplesPerSec);
+ if (result != MA_SUCCESS) {
+ ma_device_uninit__dsound(pDevice);
+ return result;
+ }
+
+ wf.Format.nBlockAlign = (WORD)(wf.Format.nChannels * wf.Format.wBitsPerSample / 8);
+ wf.Format.nAvgBytesPerSec = wf.Format.nBlockAlign * wf.Format.nSamplesPerSec;
+ wf.Samples.wValidBitsPerSample = wf.Format.wBitsPerSample;
+ wf.SubFormat = MA_GUID_KSDATAFORMAT_SUBTYPE_PCM;
+
+ /* The size of the buffer must be a clean multiple of the period count. */
+ periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__dsound(pDescriptorCapture, wf.Format.nSamplesPerSec, pConfig->performanceProfile);
+ periodCount = (pDescriptorCapture->periodCount > 0) ? pDescriptorCapture->periodCount : MA_DEFAULT_PERIODS;
+
+ MA_ZERO_OBJECT(&descDS);
+ descDS.dwSize = sizeof(descDS);
+ descDS.dwFlags = 0;
+ descDS.dwBufferBytes = periodSizeInFrames * periodCount * wf.Format.nBlockAlign;
+ descDS.lpwfxFormat = (WAVEFORMATEX*)&wf;
+ hr = ma_IDirectSoundCapture_CreateCaptureBuffer((ma_IDirectSoundCapture*)pDevice->dsound.pCapture, &descDS, (ma_IDirectSoundCaptureBuffer**)&pDevice->dsound.pCaptureBuffer, NULL);
+ if (FAILED(hr)) {
+ ma_device_uninit__dsound(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCapture_CreateCaptureBuffer() failed for capture device.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ /* Get the _actual_ properties of the buffer. */
+ pActualFormat = (WAVEFORMATEXTENSIBLE*)rawdata;
+ hr = ma_IDirectSoundCaptureBuffer_GetFormat((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, (WAVEFORMATEX*)pActualFormat, sizeof(rawdata), NULL);
+ if (FAILED(hr)) {
+ ma_device_uninit__dsound(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to retrieve the actual format of the capture device's buffer.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ /* We can now start setting the output data formats. */
+ pDescriptorCapture->format = ma_format_from_WAVEFORMATEX((WAVEFORMATEX*)pActualFormat);
+ pDescriptorCapture->channels = pActualFormat->Format.nChannels;
+ pDescriptorCapture->sampleRate = pActualFormat->Format.nSamplesPerSec;
+
+ /* Get the native channel map based on the channel mask. */
+ if (pActualFormat->Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
+ ma_channel_mask_to_channel_map__win32(pActualFormat->dwChannelMask, pDescriptorCapture->channels, pDescriptorCapture->channelMap);
+ } else {
+ ma_channel_mask_to_channel_map__win32(wf.dwChannelMask, pDescriptorCapture->channels, pDescriptorCapture->channelMap);
+ }
+
+ /*
+ After getting the actual format the size of the buffer in frames may have actually changed. However, we want this to be as close to what the
+ user has asked for as possible, so let's go ahead and release the old capture buffer and create a new one in this case.
+ */
+ if (periodSizeInFrames != (descDS.dwBufferBytes / ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels) / periodCount)) {
+ descDS.dwBufferBytes = periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels) * periodCount;
+ ma_IDirectSoundCaptureBuffer_Release((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer);
+
+ hr = ma_IDirectSoundCapture_CreateCaptureBuffer((ma_IDirectSoundCapture*)pDevice->dsound.pCapture, &descDS, (ma_IDirectSoundCaptureBuffer**)&pDevice->dsound.pCaptureBuffer, NULL);
+ if (FAILED(hr)) {
+ ma_device_uninit__dsound(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Second attempt at IDirectSoundCapture_CreateCaptureBuffer() failed for capture device.");
+ return ma_result_from_HRESULT(hr);
+ }
+ }
+
+ /* DirectSound should give us a buffer exactly the size we asked for. */
+ pDescriptorCapture->periodSizeInFrames = periodSizeInFrames;
+ pDescriptorCapture->periodCount = periodCount;
+ }
+
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ WAVEFORMATEXTENSIBLE wf;
+ MA_DSBUFFERDESC descDSPrimary;
+ MA_DSCAPS caps;
+ char rawdata[1024]; /* <-- Ugly hack to avoid a malloc() due to a crappy DirectSound API. */
+ WAVEFORMATEXTENSIBLE* pActualFormat;
+ ma_uint32 periodSizeInFrames;
+ ma_uint32 periodCount;
+ MA_DSBUFFERDESC descDS;
+
+ result = ma_config_to_WAVEFORMATEXTENSIBLE(pDescriptorPlayback->format, pDescriptorPlayback->channels, pDescriptorPlayback->sampleRate, pDescriptorPlayback->channelMap, &wf);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ result = ma_context_create_IDirectSound__dsound(pDevice->pContext, pDescriptorPlayback->shareMode, pDescriptorPlayback->pDeviceID, (ma_IDirectSound**)&pDevice->dsound.pPlayback);
+ if (result != MA_SUCCESS) {
+ ma_device_uninit__dsound(pDevice);
+ return result;
+ }
+
+ MA_ZERO_OBJECT(&descDSPrimary);
+ descDSPrimary.dwSize = sizeof(MA_DSBUFFERDESC);
+ descDSPrimary.dwFlags = MA_DSBCAPS_PRIMARYBUFFER | MA_DSBCAPS_CTRLVOLUME;
+ hr = ma_IDirectSound_CreateSoundBuffer((ma_IDirectSound*)pDevice->dsound.pPlayback, &descDSPrimary, (ma_IDirectSoundBuffer**)&pDevice->dsound.pPlaybackPrimaryBuffer, NULL);
+ if (FAILED(hr)) {
+ ma_device_uninit__dsound(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_CreateSoundBuffer() failed for playback device's primary buffer.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+
+ /* We may want to make some adjustments to the format if we are using defaults. */
+ MA_ZERO_OBJECT(&caps);
+ caps.dwSize = sizeof(caps);
+ hr = ma_IDirectSound_GetCaps((ma_IDirectSound*)pDevice->dsound.pPlayback, &caps);
+ if (FAILED(hr)) {
+ ma_device_uninit__dsound(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_GetCaps() failed for playback device.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ if (pDescriptorPlayback->channels == 0) {
+ if ((caps.dwFlags & MA_DSCAPS_PRIMARYSTEREO) != 0) {
+ DWORD speakerConfig;
+
+ /* It supports at least stereo, but could support more. */
+ wf.Format.nChannels = 2;
+
+ /* Look at the speaker configuration to get a better idea on the channel count. */
+ if (SUCCEEDED(ma_IDirectSound_GetSpeakerConfig((ma_IDirectSound*)pDevice->dsound.pPlayback, &speakerConfig))) {
+ ma_get_channels_from_speaker_config__dsound(speakerConfig, &wf.Format.nChannels, &wf.dwChannelMask);
+ }
+ } else {
+ /* It does not support stereo, which means we are stuck with mono. */
+ wf.Format.nChannels = 1;
+ }
+ }
+
+ if (pDescriptorPlayback->sampleRate == 0) {
+ /* We base the sample rate on the values returned by GetCaps(). */
+ if ((caps.dwFlags & MA_DSCAPS_CONTINUOUSRATE) != 0) {
+ wf.Format.nSamplesPerSec = ma_get_best_sample_rate_within_range(caps.dwMinSecondarySampleRate, caps.dwMaxSecondarySampleRate);
+ } else {
+ wf.Format.nSamplesPerSec = caps.dwMaxSecondarySampleRate;
+ }
+ }
+
+ wf.Format.nBlockAlign = (WORD)(wf.Format.nChannels * wf.Format.wBitsPerSample / 8);
+ wf.Format.nAvgBytesPerSec = wf.Format.nBlockAlign * wf.Format.nSamplesPerSec;
+
+ /*
+ From MSDN:
+
+ The method succeeds even if the hardware does not support the requested format; DirectSound sets the buffer to the closest
+ supported format. To determine whether this has happened, an application can call the GetFormat method for the primary buffer
+ and compare the result with the format that was requested with the SetFormat method.
+ */
+ hr = ma_IDirectSoundBuffer_SetFormat((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackPrimaryBuffer, (WAVEFORMATEX*)&wf);
+ if (FAILED(hr)) {
+ ma_device_uninit__dsound(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to set format of playback device's primary buffer.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ /* Get the _actual_ properties of the buffer. */
+ pActualFormat = (WAVEFORMATEXTENSIBLE*)rawdata;
+ hr = ma_IDirectSoundBuffer_GetFormat((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackPrimaryBuffer, (WAVEFORMATEX*)pActualFormat, sizeof(rawdata), NULL);
+ if (FAILED(hr)) {
+ ma_device_uninit__dsound(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to retrieve the actual format of the playback device's primary buffer.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ /* We now have enough information to start setting some output properties. */
+ pDescriptorPlayback->format = ma_format_from_WAVEFORMATEX((WAVEFORMATEX*)pActualFormat);
+ pDescriptorPlayback->channels = pActualFormat->Format.nChannels;
+ pDescriptorPlayback->sampleRate = pActualFormat->Format.nSamplesPerSec;
+
+ /* Get the internal channel map based on the channel mask. */
+ if (pActualFormat->Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
+ ma_channel_mask_to_channel_map__win32(pActualFormat->dwChannelMask, pDescriptorPlayback->channels, pDescriptorPlayback->channelMap);
+ } else {
+ ma_channel_mask_to_channel_map__win32(wf.dwChannelMask, pDescriptorPlayback->channels, pDescriptorPlayback->channelMap);
+ }
+
+ /* The size of the buffer must be a clean multiple of the period count. */
+ periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__dsound(pDescriptorPlayback, pDescriptorPlayback->sampleRate, pConfig->performanceProfile);
+ periodCount = (pDescriptorPlayback->periodCount > 0) ? pDescriptorPlayback->periodCount : MA_DEFAULT_PERIODS;
+
+ /*
+ Meaning of dwFlags (from MSDN):
+
+ DSBCAPS_CTRLPOSITIONNOTIFY
+ The buffer has position notification capability.
+
+ DSBCAPS_GLOBALFOCUS
+ With this flag set, an application using DirectSound can continue to play its buffers if the user switches focus to
+ another application, even if the new application uses DirectSound.
+
+ DSBCAPS_GETCURRENTPOSITION2
+ In the first version of DirectSound, the play cursor was significantly ahead of the actual playing sound on emulated
+ sound cards; it was directly behind the write cursor. Now, if the DSBCAPS_GETCURRENTPOSITION2 flag is specified, the
+ application can get a more accurate play cursor.
+ */
+ MA_ZERO_OBJECT(&descDS);
+ descDS.dwSize = sizeof(descDS);
+ descDS.dwFlags = MA_DSBCAPS_CTRLPOSITIONNOTIFY | MA_DSBCAPS_GLOBALFOCUS | MA_DSBCAPS_GETCURRENTPOSITION2;
+ descDS.dwBufferBytes = periodSizeInFrames * periodCount * ma_get_bytes_per_frame(pDescriptorPlayback->format, pDescriptorPlayback->channels);
+ descDS.lpwfxFormat = (WAVEFORMATEX*)&wf;
+ hr = ma_IDirectSound_CreateSoundBuffer((ma_IDirectSound*)pDevice->dsound.pPlayback, &descDS, (ma_IDirectSoundBuffer**)&pDevice->dsound.pPlaybackBuffer, NULL);
+ if (FAILED(hr)) {
+ ma_device_uninit__dsound(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_CreateSoundBuffer() failed for playback device's secondary buffer.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ /* DirectSound should give us a buffer exactly the size we asked for. */
+ pDescriptorPlayback->periodSizeInFrames = periodSizeInFrames;
+ pDescriptorPlayback->periodCount = periodCount;
+ }
+
+ return MA_SUCCESS;
+}
+
+
+static ma_result ma_device_data_loop__dsound(ma_device* pDevice)
+{
+ ma_result result = MA_SUCCESS;
+ ma_uint32 bpfDeviceCapture = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
+ ma_uint32 bpfDevicePlayback = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
+ HRESULT hr;
+ DWORD lockOffsetInBytesCapture;
+ DWORD lockSizeInBytesCapture;
+ DWORD mappedSizeInBytesCapture;
+ DWORD mappedDeviceFramesProcessedCapture;
+ void* pMappedDeviceBufferCapture;
+ DWORD lockOffsetInBytesPlayback;
+ DWORD lockSizeInBytesPlayback;
+ DWORD mappedSizeInBytesPlayback;
+ void* pMappedDeviceBufferPlayback;
+ DWORD prevReadCursorInBytesCapture = 0;
+ DWORD prevPlayCursorInBytesPlayback = 0;
+ ma_bool32 physicalPlayCursorLoopFlagPlayback = 0;
+ DWORD virtualWriteCursorInBytesPlayback = 0;
+ ma_bool32 virtualWriteCursorLoopFlagPlayback = 0;
+ ma_bool32 isPlaybackDeviceStarted = MA_FALSE;
+ ma_uint32 framesWrittenToPlaybackDevice = 0; /* For knowing whether or not the playback device needs to be started. */
+ ma_uint32 waitTimeInMilliseconds = 1;
+
+ MA_ASSERT(pDevice != NULL);
+
+ /* The first thing to do is start the capture device. The playback device is only started after the first period is written. */
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ hr = ma_IDirectSoundCaptureBuffer_Start((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, MA_DSCBSTART_LOOPING);
+ if (FAILED(hr)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCaptureBuffer_Start() failed.");
+ return ma_result_from_HRESULT(hr);
+ }
+ }
+
+ while (ma_device_get_state(pDevice) == ma_device_state_started) {
+ switch (pDevice->type)
+ {
+ case ma_device_type_duplex:
+ {
+ DWORD physicalCaptureCursorInBytes;
+ DWORD physicalReadCursorInBytes;
+ hr = ma_IDirectSoundCaptureBuffer_GetCurrentPosition((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, &physicalCaptureCursorInBytes, &physicalReadCursorInBytes);
+ if (FAILED(hr)) {
+ return ma_result_from_HRESULT(hr);
+ }
+
+ /* If nothing is available we just sleep for a bit and return from this iteration. */
+ if (physicalReadCursorInBytes == prevReadCursorInBytesCapture) {
+ ma_sleep(waitTimeInMilliseconds);
+ continue; /* Nothing is available in the capture buffer. */
+ }
+
+ /*
+ The current position has moved. We need to map all of the captured samples and write them to the playback device, making sure
+ we don't return until every frame has been copied over.
+ */
+ if (prevReadCursorInBytesCapture < physicalReadCursorInBytes) {
+ /* The capture position has not looped. This is the simple case. */
+ lockOffsetInBytesCapture = prevReadCursorInBytesCapture;
+ lockSizeInBytesCapture = (physicalReadCursorInBytes - prevReadCursorInBytesCapture);
+ } else {
+ /*
+ The capture position has looped. This is the more complex case. Map to the end of the buffer. If this does not return anything,
+ do it again from the start.
+ */
+ if (prevReadCursorInBytesCapture < pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture) {
+ /* Lock up to the end of the buffer. */
+ lockOffsetInBytesCapture = prevReadCursorInBytesCapture;
+ lockSizeInBytesCapture = (pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture) - prevReadCursorInBytesCapture;
+ } else {
+ /* Lock starting from the start of the buffer. */
+ lockOffsetInBytesCapture = 0;
+ lockSizeInBytesCapture = physicalReadCursorInBytes;
+ }
+ }
+
+ if (lockSizeInBytesCapture == 0) {
+ ma_sleep(waitTimeInMilliseconds);
+ continue; /* Nothing is available in the capture buffer. */
+ }
+
+ hr = ma_IDirectSoundCaptureBuffer_Lock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, lockOffsetInBytesCapture, lockSizeInBytesCapture, &pMappedDeviceBufferCapture, &mappedSizeInBytesCapture, NULL, NULL, 0);
+ if (FAILED(hr)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from capture device in preparation for writing to the device.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+
+ /* At this point we have some input data that we need to output. We do not return until every mapped frame of the input data is written to the playback device. */
+ mappedDeviceFramesProcessedCapture = 0;
+
+ for (;;) { /* Keep writing to the playback device. */
+ ma_uint8 inputFramesInClientFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
+ ma_uint32 inputFramesInClientFormatCap = sizeof(inputFramesInClientFormat) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels);
+ ma_uint8 outputFramesInClientFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
+ ma_uint32 outputFramesInClientFormatCap = sizeof(outputFramesInClientFormat) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
+ ma_uint32 outputFramesInClientFormatCount;
+ ma_uint32 outputFramesInClientFormatConsumed = 0;
+ ma_uint64 clientCapturedFramesToProcess = ma_min(inputFramesInClientFormatCap, outputFramesInClientFormatCap);
+ ma_uint64 deviceCapturedFramesToProcess = (mappedSizeInBytesCapture / bpfDeviceCapture) - mappedDeviceFramesProcessedCapture;
+ void* pRunningMappedDeviceBufferCapture = ma_offset_ptr(pMappedDeviceBufferCapture, mappedDeviceFramesProcessedCapture * bpfDeviceCapture);
+
+ result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningMappedDeviceBufferCapture, &deviceCapturedFramesToProcess, inputFramesInClientFormat, &clientCapturedFramesToProcess);
+ if (result != MA_SUCCESS) {
+ break;
+ }
+
+ outputFramesInClientFormatCount = (ma_uint32)clientCapturedFramesToProcess;
+ mappedDeviceFramesProcessedCapture += (ma_uint32)deviceCapturedFramesToProcess;
+
+ ma_device__handle_data_callback(pDevice, outputFramesInClientFormat, inputFramesInClientFormat, (ma_uint32)clientCapturedFramesToProcess);
+
+ /* At this point we have input and output data in client format. All we need to do now is convert it to the output device format. This may take a few passes. */
+ for (;;) {
+ ma_uint32 framesWrittenThisIteration;
+ DWORD physicalPlayCursorInBytes;
+ DWORD physicalWriteCursorInBytes;
+ DWORD availableBytesPlayback;
+ DWORD silentPaddingInBytes = 0; /* <-- Must be initialized to 0. */
+
+ /* We need the physical play and write cursors. */
+ if (FAILED(ma_IDirectSoundBuffer_GetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &physicalPlayCursorInBytes, &physicalWriteCursorInBytes))) {
+ break;
+ }
+
+ if (physicalPlayCursorInBytes < prevPlayCursorInBytesPlayback) {
+ physicalPlayCursorLoopFlagPlayback = !physicalPlayCursorLoopFlagPlayback;
+ }
+ prevPlayCursorInBytesPlayback = physicalPlayCursorInBytes;
+
+ /* If there's any bytes available for writing we can do that now. The space between the virtual cursor position and play cursor. */
+ if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) {
+ /* Same loop iteration. The available bytes wraps all the way around from the virtual write cursor to the physical play cursor. */
+ if (physicalPlayCursorInBytes <= virtualWriteCursorInBytesPlayback) {
+ availableBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback;
+ availableBytesPlayback += physicalPlayCursorInBytes; /* Wrap around. */
+ } else {
+ /* This is an error. */
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[DirectSound] (Duplex/Playback): Play cursor has moved in front of the write cursor (same loop iteration). physicalPlayCursorInBytes=%ld, virtualWriteCursorInBytes=%ld.\n", physicalPlayCursorInBytes, virtualWriteCursorInBytesPlayback);
+ availableBytesPlayback = 0;
+ }
+ } else {
+ /* Different loop iterations. The available bytes only goes from the virtual write cursor to the physical play cursor. */
+ if (physicalPlayCursorInBytes >= virtualWriteCursorInBytesPlayback) {
+ availableBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback;
+ } else {
+ /* This is an error. */
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[DirectSound] (Duplex/Playback): Write cursor has moved behind the play cursor (different loop iterations). physicalPlayCursorInBytes=%ld, virtualWriteCursorInBytes=%ld.\n", physicalPlayCursorInBytes, virtualWriteCursorInBytesPlayback);
+ availableBytesPlayback = 0;
+ }
+ }
+
+ /* If there's no room available for writing we need to wait for more. */
+ if (availableBytesPlayback == 0) {
+ /* If we haven't started the device yet, this will never get beyond 0. In this case we need to get the device started. */
+ if (!isPlaybackDeviceStarted) {
+ hr = ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING);
+ if (FAILED(hr)) {
+ ma_IDirectSoundCaptureBuffer_Stop((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed.");
+ return ma_result_from_HRESULT(hr);
+ }
+ isPlaybackDeviceStarted = MA_TRUE;
+ } else {
+ ma_sleep(waitTimeInMilliseconds);
+ continue;
+ }
+ }
+
+
+ /* Getting here means there room available somewhere. We limit this to either the end of the buffer or the physical play cursor, whichever is closest. */
+ lockOffsetInBytesPlayback = virtualWriteCursorInBytesPlayback;
+ if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) {
+ /* Same loop iteration. Go up to the end of the buffer. */
+ lockSizeInBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback;
+ } else {
+ /* Different loop iterations. Go up to the physical play cursor. */
+ lockSizeInBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback;
+ }
+
+ hr = ma_IDirectSoundBuffer_Lock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, lockOffsetInBytesPlayback, lockSizeInBytesPlayback, &pMappedDeviceBufferPlayback, &mappedSizeInBytesPlayback, NULL, NULL, 0);
+ if (FAILED(hr)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from playback device in preparation for writing to the device.");
+ result = ma_result_from_HRESULT(hr);
+ break;
+ }
+
+ /*
+ Experiment: If the playback buffer is being starved, pad it with some silence to get it back in sync. This will cause a glitch, but it may prevent
+ endless glitching due to it constantly running out of data.
+ */
+ if (isPlaybackDeviceStarted) {
+ DWORD bytesQueuedForPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - availableBytesPlayback;
+ if (bytesQueuedForPlayback < (pDevice->playback.internalPeriodSizeInFrames*bpfDevicePlayback)) {
+ silentPaddingInBytes = (pDevice->playback.internalPeriodSizeInFrames*2*bpfDevicePlayback) - bytesQueuedForPlayback;
+ if (silentPaddingInBytes > lockSizeInBytesPlayback) {
+ silentPaddingInBytes = lockSizeInBytesPlayback;
+ }
+
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[DirectSound] (Duplex/Playback) Playback buffer starved. availableBytesPlayback=%ld, silentPaddingInBytes=%ld\n", availableBytesPlayback, silentPaddingInBytes);
+ }
+ }
+
+ /* At this point we have a buffer for output. */
+ if (silentPaddingInBytes > 0) {
+ MA_ZERO_MEMORY(pMappedDeviceBufferPlayback, silentPaddingInBytes);
+ framesWrittenThisIteration = silentPaddingInBytes/bpfDevicePlayback;
+ } else {
+ ma_uint64 convertedFrameCountIn = (outputFramesInClientFormatCount - outputFramesInClientFormatConsumed);
+ ma_uint64 convertedFrameCountOut = mappedSizeInBytesPlayback/bpfDevicePlayback;
+ void* pConvertedFramesIn = ma_offset_ptr(outputFramesInClientFormat, outputFramesInClientFormatConsumed * bpfDevicePlayback);
+ void* pConvertedFramesOut = pMappedDeviceBufferPlayback;
+
+ result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, pConvertedFramesIn, &convertedFrameCountIn, pConvertedFramesOut, &convertedFrameCountOut);
+ if (result != MA_SUCCESS) {
+ break;
+ }
+
+ outputFramesInClientFormatConsumed += (ma_uint32)convertedFrameCountOut;
+ framesWrittenThisIteration = (ma_uint32)convertedFrameCountOut;
+ }
+
+
+ hr = ma_IDirectSoundBuffer_Unlock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, pMappedDeviceBufferPlayback, framesWrittenThisIteration*bpfDevicePlayback, NULL, 0);
+ if (FAILED(hr)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from playback device after writing to the device.");
+ result = ma_result_from_HRESULT(hr);
+ break;
+ }
+
+ virtualWriteCursorInBytesPlayback += framesWrittenThisIteration*bpfDevicePlayback;
+ if ((virtualWriteCursorInBytesPlayback/bpfDevicePlayback) == pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods) {
+ virtualWriteCursorInBytesPlayback = 0;
+ virtualWriteCursorLoopFlagPlayback = !virtualWriteCursorLoopFlagPlayback;
+ }
+
+ /*
+ We may need to start the device. We want two full periods to be written before starting the playback device. Having an extra period adds
+ a bit of a buffer to prevent the playback buffer from getting starved.
+ */
+ framesWrittenToPlaybackDevice += framesWrittenThisIteration;
+ if (!isPlaybackDeviceStarted && framesWrittenToPlaybackDevice >= (pDevice->playback.internalPeriodSizeInFrames*2)) {
+ hr = ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING);
+ if (FAILED(hr)) {
+ ma_IDirectSoundCaptureBuffer_Stop((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed.");
+ return ma_result_from_HRESULT(hr);
+ }
+ isPlaybackDeviceStarted = MA_TRUE;
+ }
+
+ if (framesWrittenThisIteration < mappedSizeInBytesPlayback/bpfDevicePlayback) {
+ break; /* We're finished with the output data.*/
+ }
+ }
+
+ if (clientCapturedFramesToProcess == 0) {
+ break; /* We just consumed every input sample. */
+ }
+ }
+
+
+ /* At this point we're done with the mapped portion of the capture buffer. */
+ hr = ma_IDirectSoundCaptureBuffer_Unlock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, pMappedDeviceBufferCapture, mappedSizeInBytesCapture, NULL, 0);
+ if (FAILED(hr)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from capture device after reading from the device.");
+ return ma_result_from_HRESULT(hr);
+ }
+ prevReadCursorInBytesCapture = (lockOffsetInBytesCapture + mappedSizeInBytesCapture);
+ } break;
+
+
+
+ case ma_device_type_capture:
+ {
+ DWORD physicalCaptureCursorInBytes;
+ DWORD physicalReadCursorInBytes;
+ hr = ma_IDirectSoundCaptureBuffer_GetCurrentPosition((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, &physicalCaptureCursorInBytes, &physicalReadCursorInBytes);
+ if (FAILED(hr)) {
+ return MA_ERROR;
+ }
+
+ /* If the previous capture position is the same as the current position we need to wait a bit longer. */
+ if (prevReadCursorInBytesCapture == physicalReadCursorInBytes) {
+ ma_sleep(waitTimeInMilliseconds);
+ continue;
+ }
+
+ /* Getting here means we have capture data available. */
+ if (prevReadCursorInBytesCapture < physicalReadCursorInBytes) {
+ /* The capture position has not looped. This is the simple case. */
+ lockOffsetInBytesCapture = prevReadCursorInBytesCapture;
+ lockSizeInBytesCapture = (physicalReadCursorInBytes - prevReadCursorInBytesCapture);
+ } else {
+ /*
+ The capture position has looped. This is the more complex case. Map to the end of the buffer. If this does not return anything,
+ do it again from the start.
+ */
+ if (prevReadCursorInBytesCapture < pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture) {
+ /* Lock up to the end of the buffer. */
+ lockOffsetInBytesCapture = prevReadCursorInBytesCapture;
+ lockSizeInBytesCapture = (pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture) - prevReadCursorInBytesCapture;
+ } else {
+ /* Lock starting from the start of the buffer. */
+ lockOffsetInBytesCapture = 0;
+ lockSizeInBytesCapture = physicalReadCursorInBytes;
+ }
+ }
+
+ if (lockSizeInBytesCapture < pDevice->capture.internalPeriodSizeInFrames) {
+ ma_sleep(waitTimeInMilliseconds);
+ continue; /* Nothing is available in the capture buffer. */
+ }
+
+ hr = ma_IDirectSoundCaptureBuffer_Lock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, lockOffsetInBytesCapture, lockSizeInBytesCapture, &pMappedDeviceBufferCapture, &mappedSizeInBytesCapture, NULL, NULL, 0);
+ if (FAILED(hr)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from capture device in preparation for writing to the device.");
+ result = ma_result_from_HRESULT(hr);
+ }
+
+ if (lockSizeInBytesCapture != mappedSizeInBytesCapture) {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[DirectSound] (Capture) lockSizeInBytesCapture=%ld != mappedSizeInBytesCapture=%ld\n", lockSizeInBytesCapture, mappedSizeInBytesCapture);
+ }
+
+ ma_device__send_frames_to_client(pDevice, mappedSizeInBytesCapture/bpfDeviceCapture, pMappedDeviceBufferCapture);
+
+ hr = ma_IDirectSoundCaptureBuffer_Unlock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, pMappedDeviceBufferCapture, mappedSizeInBytesCapture, NULL, 0);
+ if (FAILED(hr)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from capture device after reading from the device.");
+ return ma_result_from_HRESULT(hr);
+ }
+ prevReadCursorInBytesCapture = lockOffsetInBytesCapture + mappedSizeInBytesCapture;
+
+ if (prevReadCursorInBytesCapture == (pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture)) {
+ prevReadCursorInBytesCapture = 0;
+ }
+ } break;
+
+
+
+ case ma_device_type_playback:
+ {
+ DWORD availableBytesPlayback;
+ DWORD physicalPlayCursorInBytes;
+ DWORD physicalWriteCursorInBytes;
+ hr = ma_IDirectSoundBuffer_GetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &physicalPlayCursorInBytes, &physicalWriteCursorInBytes);
+ if (FAILED(hr)) {
+ break;
+ }
+
+ if (physicalPlayCursorInBytes < prevPlayCursorInBytesPlayback) {
+ physicalPlayCursorLoopFlagPlayback = !physicalPlayCursorLoopFlagPlayback;
+ }
+ prevPlayCursorInBytesPlayback = physicalPlayCursorInBytes;
+
+ /* If there's any bytes available for writing we can do that now. The space between the virtual cursor position and play cursor. */
+ if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) {
+ /* Same loop iteration. The available bytes wraps all the way around from the virtual write cursor to the physical play cursor. */
+ if (physicalPlayCursorInBytes <= virtualWriteCursorInBytesPlayback) {
+ availableBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback;
+ availableBytesPlayback += physicalPlayCursorInBytes; /* Wrap around. */
+ } else {
+ /* This is an error. */
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[DirectSound] (Playback): Play cursor has moved in front of the write cursor (same loop iterations). physicalPlayCursorInBytes=%ld, virtualWriteCursorInBytes=%ld.\n", physicalPlayCursorInBytes, virtualWriteCursorInBytesPlayback);
+ availableBytesPlayback = 0;
+ }
+ } else {
+ /* Different loop iterations. The available bytes only goes from the virtual write cursor to the physical play cursor. */
+ if (physicalPlayCursorInBytes >= virtualWriteCursorInBytesPlayback) {
+ availableBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback;
+ } else {
+ /* This is an error. */
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[DirectSound] (Playback): Write cursor has moved behind the play cursor (different loop iterations). physicalPlayCursorInBytes=%ld, virtualWriteCursorInBytes=%ld.\n", physicalPlayCursorInBytes, virtualWriteCursorInBytesPlayback);
+ availableBytesPlayback = 0;
+ }
+ }
+
+ /* If there's no room available for writing we need to wait for more. */
+ if (availableBytesPlayback < pDevice->playback.internalPeriodSizeInFrames) {
+ /* If we haven't started the device yet, this will never get beyond 0. In this case we need to get the device started. */
+ if (availableBytesPlayback == 0 && !isPlaybackDeviceStarted) {
+ hr = ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING);
+ if (FAILED(hr)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed.");
+ return ma_result_from_HRESULT(hr);
+ }
+ isPlaybackDeviceStarted = MA_TRUE;
+ } else {
+ ma_sleep(waitTimeInMilliseconds);
+ continue;
+ }
+ }
+
+ /* Getting here means there room available somewhere. We limit this to either the end of the buffer or the physical play cursor, whichever is closest. */
+ lockOffsetInBytesPlayback = virtualWriteCursorInBytesPlayback;
+ if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) {
+ /* Same loop iteration. Go up to the end of the buffer. */
+ lockSizeInBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback;
+ } else {
+ /* Different loop iterations. Go up to the physical play cursor. */
+ lockSizeInBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback;
+ }
+
+ hr = ma_IDirectSoundBuffer_Lock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, lockOffsetInBytesPlayback, lockSizeInBytesPlayback, &pMappedDeviceBufferPlayback, &mappedSizeInBytesPlayback, NULL, NULL, 0);
+ if (FAILED(hr)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from playback device in preparation for writing to the device.");
+ result = ma_result_from_HRESULT(hr);
+ break;
+ }
+
+ /* At this point we have a buffer for output. */
+ ma_device__read_frames_from_client(pDevice, (mappedSizeInBytesPlayback/bpfDevicePlayback), pMappedDeviceBufferPlayback);
+
+ hr = ma_IDirectSoundBuffer_Unlock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, pMappedDeviceBufferPlayback, mappedSizeInBytesPlayback, NULL, 0);
+ if (FAILED(hr)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from playback device after writing to the device.");
+ result = ma_result_from_HRESULT(hr);
+ break;
+ }
+
+ virtualWriteCursorInBytesPlayback += mappedSizeInBytesPlayback;
+ if (virtualWriteCursorInBytesPlayback == pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) {
+ virtualWriteCursorInBytesPlayback = 0;
+ virtualWriteCursorLoopFlagPlayback = !virtualWriteCursorLoopFlagPlayback;
+ }
+
+ /*
+ We may need to start the device. We want two full periods to be written before starting the playback device. Having an extra period adds
+ a bit of a buffer to prevent the playback buffer from getting starved.
+ */
+ framesWrittenToPlaybackDevice += mappedSizeInBytesPlayback/bpfDevicePlayback;
+ if (!isPlaybackDeviceStarted && framesWrittenToPlaybackDevice >= pDevice->playback.internalPeriodSizeInFrames) {
+ hr = ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING);
+ if (FAILED(hr)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed.");
+ return ma_result_from_HRESULT(hr);
+ }
+ isPlaybackDeviceStarted = MA_TRUE;
+ }
+ } break;
+
+
+ default: return MA_INVALID_ARGS; /* Invalid device type. */
+ }
+
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ }
+
+ /* Getting here means the device is being stopped. */
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ hr = ma_IDirectSoundCaptureBuffer_Stop((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer);
+ if (FAILED(hr)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCaptureBuffer_Stop() failed.");
+ return ma_result_from_HRESULT(hr);
+ }
+ }
+
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ /* The playback device should be drained before stopping. All we do is wait until the available bytes is equal to the size of the buffer. */
+ if (isPlaybackDeviceStarted) {
+ for (;;) {
+ DWORD availableBytesPlayback = 0;
+ DWORD physicalPlayCursorInBytes;
+ DWORD physicalWriteCursorInBytes;
+ hr = ma_IDirectSoundBuffer_GetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &physicalPlayCursorInBytes, &physicalWriteCursorInBytes);
+ if (FAILED(hr)) {
+ break;
+ }
+
+ if (physicalPlayCursorInBytes < prevPlayCursorInBytesPlayback) {
+ physicalPlayCursorLoopFlagPlayback = !physicalPlayCursorLoopFlagPlayback;
+ }
+ prevPlayCursorInBytesPlayback = physicalPlayCursorInBytes;
+
+ if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) {
+ /* Same loop iteration. The available bytes wraps all the way around from the virtual write cursor to the physical play cursor. */
+ if (physicalPlayCursorInBytes <= virtualWriteCursorInBytesPlayback) {
+ availableBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback;
+ availableBytesPlayback += physicalPlayCursorInBytes; /* Wrap around. */
+ } else {
+ break;
+ }
+ } else {
+ /* Different loop iterations. The available bytes only goes from the virtual write cursor to the physical play cursor. */
+ if (physicalPlayCursorInBytes >= virtualWriteCursorInBytesPlayback) {
+ availableBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback;
+ } else {
+ break;
+ }
+ }
+
+ if (availableBytesPlayback >= (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback)) {
+ break;
+ }
+
+ ma_sleep(waitTimeInMilliseconds);
+ }
+ }
+
+ hr = ma_IDirectSoundBuffer_Stop((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer);
+ if (FAILED(hr)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Stop() failed.");
+ return ma_result_from_HRESULT(hr);
+ }
+
+ ma_IDirectSoundBuffer_SetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0);
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_uninit__dsound(ma_context* pContext)
+{
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pContext->backend == ma_backend_dsound);
+
+ ma_dlclose(pContext, pContext->dsound.hDSoundDLL);
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_init__dsound(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+{
+ MA_ASSERT(pContext != NULL);
+
+ (void)pConfig;
+
+ pContext->dsound.hDSoundDLL = ma_dlopen(pContext, "dsound.dll");
+ if (pContext->dsound.hDSoundDLL == NULL) {
+ return MA_API_NOT_FOUND;
+ }
+
+ pContext->dsound.DirectSoundCreate = ma_dlsym(pContext, pContext->dsound.hDSoundDLL, "DirectSoundCreate");
+ pContext->dsound.DirectSoundEnumerateA = ma_dlsym(pContext, pContext->dsound.hDSoundDLL, "DirectSoundEnumerateA");
+ pContext->dsound.DirectSoundCaptureCreate = ma_dlsym(pContext, pContext->dsound.hDSoundDLL, "DirectSoundCaptureCreate");
+ pContext->dsound.DirectSoundCaptureEnumerateA = ma_dlsym(pContext, pContext->dsound.hDSoundDLL, "DirectSoundCaptureEnumerateA");
+
+ pCallbacks->onContextInit = ma_context_init__dsound;
+ pCallbacks->onContextUninit = ma_context_uninit__dsound;
+ pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__dsound;
+ pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__dsound;
+ pCallbacks->onDeviceInit = ma_device_init__dsound;
+ pCallbacks->onDeviceUninit = ma_device_uninit__dsound;
+ pCallbacks->onDeviceStart = NULL; /* Not used. Started in onDeviceDataLoop. */
+ pCallbacks->onDeviceStop = NULL; /* Not used. Stopped in onDeviceDataLoop. */
+ pCallbacks->onDeviceRead = NULL; /* Not used. Data is read directly in onDeviceDataLoop. */
+ pCallbacks->onDeviceWrite = NULL; /* Not used. Data is written directly in onDeviceDataLoop. */
+ pCallbacks->onDeviceDataLoop = ma_device_data_loop__dsound;
+
+ return MA_SUCCESS;
+}
+#endif
+
+
+
+/******************************************************************************
+
+WinMM Backend
+
+******************************************************************************/
+#ifdef MA_HAS_WINMM
+
+/*
+Some older compilers don't have WAVEOUTCAPS2A and WAVEINCAPS2A, so we'll need to write this ourselves. These structures
+are exactly the same as the older ones but they have a few GUIDs for manufacturer/product/name identification. I'm keeping
+the names the same as the Win32 library for consistency, but namespaced to avoid naming conflicts with the Win32 version.
+*/
+typedef struct
+{
+ WORD wMid;
+ WORD wPid;
+ MMVERSION vDriverVersion;
+ CHAR szPname[MAXPNAMELEN];
+ DWORD dwFormats;
+ WORD wChannels;
+ WORD wReserved1;
+ DWORD dwSupport;
+ GUID ManufacturerGuid;
+ GUID ProductGuid;
+ GUID NameGuid;
+} MA_WAVEOUTCAPS2A;
+typedef struct
+{
+ WORD wMid;
+ WORD wPid;
+ MMVERSION vDriverVersion;
+ CHAR szPname[MAXPNAMELEN];
+ DWORD dwFormats;
+ WORD wChannels;
+ WORD wReserved1;
+ GUID ManufacturerGuid;
+ GUID ProductGuid;
+ GUID NameGuid;
+} MA_WAVEINCAPS2A;
+
+typedef UINT (WINAPI * MA_PFN_waveOutGetNumDevs)(void);
+typedef MMRESULT (WINAPI * MA_PFN_waveOutGetDevCapsA)(ma_uintptr uDeviceID, LPWAVEOUTCAPSA pwoc, UINT cbwoc);
+typedef MMRESULT (WINAPI * MA_PFN_waveOutOpen)(LPHWAVEOUT phwo, UINT uDeviceID, LPCWAVEFORMATEX pwfx, DWORD_PTR dwCallback, DWORD_PTR dwInstance, DWORD fdwOpen);
+typedef MMRESULT (WINAPI * MA_PFN_waveOutClose)(HWAVEOUT hwo);
+typedef MMRESULT (WINAPI * MA_PFN_waveOutPrepareHeader)(HWAVEOUT hwo, LPWAVEHDR pwh, UINT cbwh);
+typedef MMRESULT (WINAPI * MA_PFN_waveOutUnprepareHeader)(HWAVEOUT hwo, LPWAVEHDR pwh, UINT cbwh);
+typedef MMRESULT (WINAPI * MA_PFN_waveOutWrite)(HWAVEOUT hwo, LPWAVEHDR pwh, UINT cbwh);
+typedef MMRESULT (WINAPI * MA_PFN_waveOutReset)(HWAVEOUT hwo);
+typedef UINT (WINAPI * MA_PFN_waveInGetNumDevs)(void);
+typedef MMRESULT (WINAPI * MA_PFN_waveInGetDevCapsA)(ma_uintptr uDeviceID, LPWAVEINCAPSA pwic, UINT cbwic);
+typedef MMRESULT (WINAPI * MA_PFN_waveInOpen)(LPHWAVEIN phwi, UINT uDeviceID, LPCWAVEFORMATEX pwfx, DWORD_PTR dwCallback, DWORD_PTR dwInstance, DWORD fdwOpen);
+typedef MMRESULT (WINAPI * MA_PFN_waveInClose)(HWAVEIN hwi);
+typedef MMRESULT (WINAPI * MA_PFN_waveInPrepareHeader)(HWAVEIN hwi, LPWAVEHDR pwh, UINT cbwh);
+typedef MMRESULT (WINAPI * MA_PFN_waveInUnprepareHeader)(HWAVEIN hwi, LPWAVEHDR pwh, UINT cbwh);
+typedef MMRESULT (WINAPI * MA_PFN_waveInAddBuffer)(HWAVEIN hwi, LPWAVEHDR pwh, UINT cbwh);
+typedef MMRESULT (WINAPI * MA_PFN_waveInStart)(HWAVEIN hwi);
+typedef MMRESULT (WINAPI * MA_PFN_waveInReset)(HWAVEIN hwi);
+
+static ma_result ma_result_from_MMRESULT(MMRESULT resultMM)
+{
+ switch (resultMM) {
+ case MMSYSERR_NOERROR: return MA_SUCCESS;
+ case MMSYSERR_BADDEVICEID: return MA_INVALID_ARGS;
+ case MMSYSERR_INVALHANDLE: return MA_INVALID_ARGS;
+ case MMSYSERR_NOMEM: return MA_OUT_OF_MEMORY;
+ case MMSYSERR_INVALFLAG: return MA_INVALID_ARGS;
+ case MMSYSERR_INVALPARAM: return MA_INVALID_ARGS;
+ case MMSYSERR_HANDLEBUSY: return MA_BUSY;
+ case MMSYSERR_ERROR: return MA_ERROR;
+ default: return MA_ERROR;
+ }
+}
+
+static char* ma_find_last_character(char* str, char ch)
+{
+ char* last;
+
+ if (str == NULL) {
+ return NULL;
+ }
+
+ last = NULL;
+ while (*str != '\0') {
+ if (*str == ch) {
+ last = str;
+ }
+
+ str += 1;
+ }
+
+ return last;
+}
+
+static ma_uint32 ma_get_period_size_in_bytes(ma_uint32 periodSizeInFrames, ma_format format, ma_uint32 channels)
+{
+ return periodSizeInFrames * ma_get_bytes_per_frame(format, channels);
+}
+
+
+/*
+Our own "WAVECAPS" structure that contains generic information shared between WAVEOUTCAPS2 and WAVEINCAPS2 so
+we can do things generically and typesafely. Names are being kept the same for consistency.
+*/
+typedef struct
+{
+ CHAR szPname[MAXPNAMELEN];
+ DWORD dwFormats;
+ WORD wChannels;
+ GUID NameGuid;
+} MA_WAVECAPSA;
+
+static ma_result ma_get_best_info_from_formats_flags__winmm(DWORD dwFormats, WORD channels, WORD* pBitsPerSample, DWORD* pSampleRate)
+{
+ WORD bitsPerSample = 0;
+ DWORD sampleRate = 0;
+
+ if (pBitsPerSample) {
+ *pBitsPerSample = 0;
+ }
+ if (pSampleRate) {
+ *pSampleRate = 0;
+ }
+
+ if (channels == 1) {
+ bitsPerSample = 16;
+ if ((dwFormats & WAVE_FORMAT_48M16) != 0) {
+ sampleRate = 48000;
+ } else if ((dwFormats & WAVE_FORMAT_44M16) != 0) {
+ sampleRate = 44100;
+ } else if ((dwFormats & WAVE_FORMAT_2M16) != 0) {
+ sampleRate = 22050;
+ } else if ((dwFormats & WAVE_FORMAT_1M16) != 0) {
+ sampleRate = 11025;
+ } else if ((dwFormats & WAVE_FORMAT_96M16) != 0) {
+ sampleRate = 96000;
+ } else {
+ bitsPerSample = 8;
+ if ((dwFormats & WAVE_FORMAT_48M08) != 0) {
+ sampleRate = 48000;
+ } else if ((dwFormats & WAVE_FORMAT_44M08) != 0) {
+ sampleRate = 44100;
+ } else if ((dwFormats & WAVE_FORMAT_2M08) != 0) {
+ sampleRate = 22050;
+ } else if ((dwFormats & WAVE_FORMAT_1M08) != 0) {
+ sampleRate = 11025;
+ } else if ((dwFormats & WAVE_FORMAT_96M08) != 0) {
+ sampleRate = 96000;
+ } else {
+ return MA_FORMAT_NOT_SUPPORTED;
+ }
+ }
+ } else {
+ bitsPerSample = 16;
+ if ((dwFormats & WAVE_FORMAT_48S16) != 0) {
+ sampleRate = 48000;
+ } else if ((dwFormats & WAVE_FORMAT_44S16) != 0) {
+ sampleRate = 44100;
+ } else if ((dwFormats & WAVE_FORMAT_2S16) != 0) {
+ sampleRate = 22050;
+ } else if ((dwFormats & WAVE_FORMAT_1S16) != 0) {
+ sampleRate = 11025;
+ } else if ((dwFormats & WAVE_FORMAT_96S16) != 0) {
+ sampleRate = 96000;
+ } else {
+ bitsPerSample = 8;
+ if ((dwFormats & WAVE_FORMAT_48S08) != 0) {
+ sampleRate = 48000;
+ } else if ((dwFormats & WAVE_FORMAT_44S08) != 0) {
+ sampleRate = 44100;
+ } else if ((dwFormats & WAVE_FORMAT_2S08) != 0) {
+ sampleRate = 22050;
+ } else if ((dwFormats & WAVE_FORMAT_1S08) != 0) {
+ sampleRate = 11025;
+ } else if ((dwFormats & WAVE_FORMAT_96S08) != 0) {
+ sampleRate = 96000;
+ } else {
+ return MA_FORMAT_NOT_SUPPORTED;
+ }
+ }
+ }
+
+ if (pBitsPerSample) {
+ *pBitsPerSample = bitsPerSample;
+ }
+ if (pSampleRate) {
+ *pSampleRate = sampleRate;
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_formats_flags_to_WAVEFORMATEX__winmm(DWORD dwFormats, WORD channels, WAVEFORMATEX* pWF)
+{
+ ma_result result;
+
+ MA_ASSERT(pWF != NULL);
+
+ MA_ZERO_OBJECT(pWF);
+ pWF->cbSize = sizeof(*pWF);
+ pWF->wFormatTag = WAVE_FORMAT_PCM;
+ pWF->nChannels = (WORD)channels;
+ if (pWF->nChannels > 2) {
+ pWF->nChannels = 2;
+ }
+
+ result = ma_get_best_info_from_formats_flags__winmm(dwFormats, channels, &pWF->wBitsPerSample, &pWF->nSamplesPerSec);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ pWF->nBlockAlign = (WORD)(pWF->nChannels * pWF->wBitsPerSample / 8);
+ pWF->nAvgBytesPerSec = pWF->nBlockAlign * pWF->nSamplesPerSec;
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_get_device_info_from_WAVECAPS(ma_context* pContext, MA_WAVECAPSA* pCaps, ma_device_info* pDeviceInfo)
+{
+ WORD bitsPerSample;
+ DWORD sampleRate;
+ ma_result result;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pCaps != NULL);
+ MA_ASSERT(pDeviceInfo != NULL);
+
+ /*
+ Name / Description
+
+ Unfortunately the name specified in WAVE(OUT/IN)CAPS2 is limited to 31 characters. This results in an unprofessional looking
+ situation where the names of the devices are truncated. To help work around this, we need to look at the name GUID and try
+ looking in the registry for the full name. If we can't find it there, we need to just fall back to the default name.
+ */
+
+ /* Set the default to begin with. */
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), pCaps->szPname, (size_t)-1);
+
+ /*
+ Now try the registry. There's a few things to consider here:
+ - The name GUID can be null, in which we case we just need to stick to the original 31 characters.
+ - If the name GUID is not present in the registry we'll also need to stick to the original 31 characters.
+ - I like consistency, so I want the returned device names to be consistent with those returned by WASAPI and DirectSound. The
+ problem, however is that WASAPI and DirectSound use " ()" format (such as "Speakers (High Definition Audio)"),
+ but WinMM does not specificy the component name. From my admittedly limited testing, I've notice the component name seems to
+ usually fit within the 31 characters of the fixed sized buffer, so what I'm going to do is parse that string for the component
+ name, and then concatenate the name from the registry.
+ */
+ if (!ma_is_guid_null(&pCaps->NameGuid)) {
+ wchar_t guidStrW[256];
+ if (((MA_PFN_StringFromGUID2)pContext->win32.StringFromGUID2)(&pCaps->NameGuid, guidStrW, ma_countof(guidStrW)) > 0) {
+ char guidStr[256];
+ char keyStr[1024];
+ HKEY hKey;
+
+ WideCharToMultiByte(CP_UTF8, 0, guidStrW, -1, guidStr, sizeof(guidStr), 0, FALSE);
+
+ ma_strcpy_s(keyStr, sizeof(keyStr), "SYSTEM\\CurrentControlSet\\Control\\MediaCategories\\");
+ ma_strcat_s(keyStr, sizeof(keyStr), guidStr);
+
+ if (((MA_PFN_RegOpenKeyExA)pContext->win32.RegOpenKeyExA)(HKEY_LOCAL_MACHINE, keyStr, 0, KEY_READ, &hKey) == ERROR_SUCCESS) {
+ BYTE nameFromReg[512];
+ DWORD nameFromRegSize = sizeof(nameFromReg);
+ LONG resultWin32 = ((MA_PFN_RegQueryValueExA)pContext->win32.RegQueryValueExA)(hKey, "Name", 0, NULL, (LPBYTE)nameFromReg, (LPDWORD)&nameFromRegSize);
+ ((MA_PFN_RegCloseKey)pContext->win32.RegCloseKey)(hKey);
+
+ if (resultWin32 == ERROR_SUCCESS) {
+ /* We have the value from the registry, so now we need to construct the name string. */
+ char name[1024];
+ if (ma_strcpy_s(name, sizeof(name), pDeviceInfo->name) == 0) {
+ char* nameBeg = ma_find_last_character(name, '(');
+ if (nameBeg != NULL) {
+ size_t leadingLen = (nameBeg - name);
+ ma_strncpy_s(nameBeg + 1, sizeof(name) - leadingLen, (const char*)nameFromReg, (size_t)-1);
+
+ /* The closing ")", if it can fit. */
+ if (leadingLen + nameFromRegSize < sizeof(name)-1) {
+ ma_strcat_s(name, sizeof(name), ")");
+ }
+
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), name, (size_t)-1);
+ }
+ }
+ }
+ }
+ }
+ }
+
+
+ result = ma_get_best_info_from_formats_flags__winmm(pCaps->dwFormats, pCaps->wChannels, &bitsPerSample, &sampleRate);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ if (bitsPerSample == 8) {
+ pDeviceInfo->nativeDataFormats[0].format = ma_format_u8;
+ } else if (bitsPerSample == 16) {
+ pDeviceInfo->nativeDataFormats[0].format = ma_format_s16;
+ } else if (bitsPerSample == 24) {
+ pDeviceInfo->nativeDataFormats[0].format = ma_format_s24;
+ } else if (bitsPerSample == 32) {
+ pDeviceInfo->nativeDataFormats[0].format = ma_format_s32;
+ } else {
+ return MA_FORMAT_NOT_SUPPORTED;
+ }
+ pDeviceInfo->nativeDataFormats[0].channels = pCaps->wChannels;
+ pDeviceInfo->nativeDataFormats[0].sampleRate = sampleRate;
+ pDeviceInfo->nativeDataFormats[0].flags = 0;
+ pDeviceInfo->nativeDataFormatCount = 1;
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_get_device_info_from_WAVEOUTCAPS2(ma_context* pContext, MA_WAVEOUTCAPS2A* pCaps, ma_device_info* pDeviceInfo)
+{
+ MA_WAVECAPSA caps;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pCaps != NULL);
+ MA_ASSERT(pDeviceInfo != NULL);
+
+ MA_COPY_MEMORY(caps.szPname, pCaps->szPname, sizeof(caps.szPname));
+ caps.dwFormats = pCaps->dwFormats;
+ caps.wChannels = pCaps->wChannels;
+ caps.NameGuid = pCaps->NameGuid;
+ return ma_context_get_device_info_from_WAVECAPS(pContext, &caps, pDeviceInfo);
+}
+
+static ma_result ma_context_get_device_info_from_WAVEINCAPS2(ma_context* pContext, MA_WAVEINCAPS2A* pCaps, ma_device_info* pDeviceInfo)
+{
+ MA_WAVECAPSA caps;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pCaps != NULL);
+ MA_ASSERT(pDeviceInfo != NULL);
+
+ MA_COPY_MEMORY(caps.szPname, pCaps->szPname, sizeof(caps.szPname));
+ caps.dwFormats = pCaps->dwFormats;
+ caps.wChannels = pCaps->wChannels;
+ caps.NameGuid = pCaps->NameGuid;
+ return ma_context_get_device_info_from_WAVECAPS(pContext, &caps, pDeviceInfo);
+}
+
+
+static ma_result ma_context_enumerate_devices__winmm(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+{
+ UINT playbackDeviceCount;
+ UINT captureDeviceCount;
+ UINT iPlaybackDevice;
+ UINT iCaptureDevice;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(callback != NULL);
+
+ /* Playback. */
+ playbackDeviceCount = ((MA_PFN_waveOutGetNumDevs)pContext->winmm.waveOutGetNumDevs)();
+ for (iPlaybackDevice = 0; iPlaybackDevice < playbackDeviceCount; ++iPlaybackDevice) {
+ MMRESULT result;
+ MA_WAVEOUTCAPS2A caps;
+
+ MA_ZERO_OBJECT(&caps);
+
+ result = ((MA_PFN_waveOutGetDevCapsA)pContext->winmm.waveOutGetDevCapsA)(iPlaybackDevice, (WAVEOUTCAPSA*)&caps, sizeof(caps));
+ if (result == MMSYSERR_NOERROR) {
+ ma_device_info deviceInfo;
+
+ MA_ZERO_OBJECT(&deviceInfo);
+ deviceInfo.id.winmm = iPlaybackDevice;
+
+ /* The first enumerated device is the default device. */
+ if (iPlaybackDevice == 0) {
+ deviceInfo.isDefault = MA_TRUE;
+ }
+
+ if (ma_context_get_device_info_from_WAVEOUTCAPS2(pContext, &caps, &deviceInfo) == MA_SUCCESS) {
+ ma_bool32 cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
+ if (cbResult == MA_FALSE) {
+ return MA_SUCCESS; /* Enumeration was stopped. */
+ }
+ }
+ }
+ }
+
+ /* Capture. */
+ captureDeviceCount = ((MA_PFN_waveInGetNumDevs)pContext->winmm.waveInGetNumDevs)();
+ for (iCaptureDevice = 0; iCaptureDevice < captureDeviceCount; ++iCaptureDevice) {
+ MMRESULT result;
+ MA_WAVEINCAPS2A caps;
+
+ MA_ZERO_OBJECT(&caps);
+
+ result = ((MA_PFN_waveInGetDevCapsA)pContext->winmm.waveInGetDevCapsA)(iCaptureDevice, (WAVEINCAPSA*)&caps, sizeof(caps));
+ if (result == MMSYSERR_NOERROR) {
+ ma_device_info deviceInfo;
+
+ MA_ZERO_OBJECT(&deviceInfo);
+ deviceInfo.id.winmm = iCaptureDevice;
+
+ /* The first enumerated device is the default device. */
+ if (iCaptureDevice == 0) {
+ deviceInfo.isDefault = MA_TRUE;
+ }
+
+ if (ma_context_get_device_info_from_WAVEINCAPS2(pContext, &caps, &deviceInfo) == MA_SUCCESS) {
+ ma_bool32 cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
+ if (cbResult == MA_FALSE) {
+ return MA_SUCCESS; /* Enumeration was stopped. */
+ }
+ }
+ }
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_get_device_info__winmm(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+{
+ UINT winMMDeviceID;
+
+ MA_ASSERT(pContext != NULL);
+
+ winMMDeviceID = 0;
+ if (pDeviceID != NULL) {
+ winMMDeviceID = (UINT)pDeviceID->winmm;
+ }
+
+ pDeviceInfo->id.winmm = winMMDeviceID;
+
+ /* The first ID is the default device. */
+ if (winMMDeviceID == 0) {
+ pDeviceInfo->isDefault = MA_TRUE;
+ }
+
+ if (deviceType == ma_device_type_playback) {
+ MMRESULT result;
+ MA_WAVEOUTCAPS2A caps;
+
+ MA_ZERO_OBJECT(&caps);
+
+ result = ((MA_PFN_waveOutGetDevCapsA)pContext->winmm.waveOutGetDevCapsA)(winMMDeviceID, (WAVEOUTCAPSA*)&caps, sizeof(caps));
+ if (result == MMSYSERR_NOERROR) {
+ return ma_context_get_device_info_from_WAVEOUTCAPS2(pContext, &caps, pDeviceInfo);
+ }
+ } else {
+ MMRESULT result;
+ MA_WAVEINCAPS2A caps;
+
+ MA_ZERO_OBJECT(&caps);
+
+ result = ((MA_PFN_waveInGetDevCapsA)pContext->winmm.waveInGetDevCapsA)(winMMDeviceID, (WAVEINCAPSA*)&caps, sizeof(caps));
+ if (result == MMSYSERR_NOERROR) {
+ return ma_context_get_device_info_from_WAVEINCAPS2(pContext, &caps, pDeviceInfo);
+ }
+ }
+
+ return MA_NO_DEVICE;
+}
+
+
+static ma_result ma_device_uninit__winmm(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
+
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ ((MA_PFN_waveInClose)pDevice->pContext->winmm.waveInClose)((HWAVEIN)pDevice->winmm.hDeviceCapture);
+ CloseHandle((HANDLE)pDevice->winmm.hEventCapture);
+ }
+
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ ((MA_PFN_waveOutReset)pDevice->pContext->winmm.waveOutReset)((HWAVEOUT)pDevice->winmm.hDevicePlayback);
+ ((MA_PFN_waveOutClose)pDevice->pContext->winmm.waveOutClose)((HWAVEOUT)pDevice->winmm.hDevicePlayback);
+ CloseHandle((HANDLE)pDevice->winmm.hEventPlayback);
+ }
+
+ ma_free(pDevice->winmm._pHeapData, &pDevice->pContext->allocationCallbacks);
+
+ MA_ZERO_OBJECT(&pDevice->winmm); /* Safety. */
+
+ return MA_SUCCESS;
+}
+
+static ma_uint32 ma_calculate_period_size_in_frames_from_descriptor__winmm(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile)
+{
+ /* WinMM has a minimum period size of 40ms. */
+ ma_uint32 minPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(40, nativeSampleRate);
+ ma_uint32 periodSizeInFrames;
+
+ periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, nativeSampleRate, performanceProfile);
+ if (periodSizeInFrames < minPeriodSizeInFrames) {
+ periodSizeInFrames = minPeriodSizeInFrames;
+ }
+
+ return periodSizeInFrames;
+}
+
+static ma_result ma_device_init__winmm(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+{
+ const char* errorMsg = "";
+ ma_result errorCode = MA_ERROR;
+ ma_result result = MA_SUCCESS;
+ ma_uint32 heapSize;
+ UINT winMMDeviceIDPlayback = 0;
+ UINT winMMDeviceIDCapture = 0;
+
+ MA_ASSERT(pDevice != NULL);
+
+ MA_ZERO_OBJECT(&pDevice->winmm);
+
+ if (pConfig->deviceType == ma_device_type_loopback) {
+ return MA_DEVICE_TYPE_NOT_SUPPORTED;
+ }
+
+ /* No exlusive mode with WinMM. */
+ if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) ||
+ ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) {
+ return MA_SHARE_MODE_NOT_SUPPORTED;
+ }
+
+ if (pDescriptorPlayback->pDeviceID != NULL) {
+ winMMDeviceIDPlayback = (UINT)pDescriptorPlayback->pDeviceID->winmm;
+ }
+ if (pDescriptorCapture->pDeviceID != NULL) {
+ winMMDeviceIDCapture = (UINT)pDescriptorCapture->pDeviceID->winmm;
+ }
+
+ /* The capture device needs to be initialized first. */
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
+ WAVEINCAPSA caps;
+ WAVEFORMATEX wf;
+ MMRESULT resultMM;
+
+ /* We use an event to know when a new fragment needs to be enqueued. */
+ pDevice->winmm.hEventCapture = (ma_handle)CreateEventW(NULL, TRUE, TRUE, NULL);
+ if (pDevice->winmm.hEventCapture == NULL) {
+ errorMsg = "[WinMM] Failed to create event for fragment enqueing for the capture device.", errorCode = ma_result_from_GetLastError(GetLastError());
+ goto on_error;
+ }
+
+ /* The format should be based on the device's actual format. */
+ if (((MA_PFN_waveInGetDevCapsA)pDevice->pContext->winmm.waveInGetDevCapsA)(winMMDeviceIDCapture, &caps, sizeof(caps)) != MMSYSERR_NOERROR) {
+ errorMsg = "[WinMM] Failed to retrieve internal device caps.", errorCode = MA_FORMAT_NOT_SUPPORTED;
+ goto on_error;
+ }
+
+ result = ma_formats_flags_to_WAVEFORMATEX__winmm(caps.dwFormats, caps.wChannels, &wf);
+ if (result != MA_SUCCESS) {
+ errorMsg = "[WinMM] Could not find appropriate format for internal device.", errorCode = result;
+ goto on_error;
+ }
+
+ resultMM = ((MA_PFN_waveInOpen)pDevice->pContext->winmm.waveInOpen)((LPHWAVEIN)&pDevice->winmm.hDeviceCapture, winMMDeviceIDCapture, &wf, (DWORD_PTR)pDevice->winmm.hEventCapture, (DWORD_PTR)pDevice, CALLBACK_EVENT | WAVE_ALLOWSYNC);
+ if (resultMM != MMSYSERR_NOERROR) {
+ errorMsg = "[WinMM] Failed to open capture device.", errorCode = MA_FAILED_TO_OPEN_BACKEND_DEVICE;
+ goto on_error;
+ }
+
+ pDescriptorCapture->format = ma_format_from_WAVEFORMATEX(&wf);
+ pDescriptorCapture->channels = wf.nChannels;
+ pDescriptorCapture->sampleRate = wf.nSamplesPerSec;
+ ma_channel_map_init_standard(ma_standard_channel_map_microsoft, pDescriptorCapture->channelMap, ma_countof(pDescriptorCapture->channelMap), pDescriptorCapture->channels);
+ pDescriptorCapture->periodCount = pDescriptorCapture->periodCount;
+ pDescriptorCapture->periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__winmm(pDescriptorCapture, pDescriptorCapture->sampleRate, pConfig->performanceProfile);
+ }
+
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ WAVEOUTCAPSA caps;
+ WAVEFORMATEX wf;
+ MMRESULT resultMM;
+
+ /* We use an event to know when a new fragment needs to be enqueued. */
+ pDevice->winmm.hEventPlayback = (ma_handle)CreateEventW(NULL, TRUE, TRUE, NULL);
+ if (pDevice->winmm.hEventPlayback == NULL) {
+ errorMsg = "[WinMM] Failed to create event for fragment enqueing for the playback device.", errorCode = ma_result_from_GetLastError(GetLastError());
+ goto on_error;
+ }
+
+ /* The format should be based on the device's actual format. */
+ if (((MA_PFN_waveOutGetDevCapsA)pDevice->pContext->winmm.waveOutGetDevCapsA)(winMMDeviceIDPlayback, &caps, sizeof(caps)) != MMSYSERR_NOERROR) {
+ errorMsg = "[WinMM] Failed to retrieve internal device caps.", errorCode = MA_FORMAT_NOT_SUPPORTED;
+ goto on_error;
+ }
+
+ result = ma_formats_flags_to_WAVEFORMATEX__winmm(caps.dwFormats, caps.wChannels, &wf);
+ if (result != MA_SUCCESS) {
+ errorMsg = "[WinMM] Could not find appropriate format for internal device.", errorCode = result;
+ goto on_error;
+ }
+
+ resultMM = ((MA_PFN_waveOutOpen)pDevice->pContext->winmm.waveOutOpen)((LPHWAVEOUT)&pDevice->winmm.hDevicePlayback, winMMDeviceIDPlayback, &wf, (DWORD_PTR)pDevice->winmm.hEventPlayback, (DWORD_PTR)pDevice, CALLBACK_EVENT | WAVE_ALLOWSYNC);
+ if (resultMM != MMSYSERR_NOERROR) {
+ errorMsg = "[WinMM] Failed to open playback device.", errorCode = MA_FAILED_TO_OPEN_BACKEND_DEVICE;
+ goto on_error;
+ }
+
+ pDescriptorPlayback->format = ma_format_from_WAVEFORMATEX(&wf);
+ pDescriptorPlayback->channels = wf.nChannels;
+ pDescriptorPlayback->sampleRate = wf.nSamplesPerSec;
+ ma_channel_map_init_standard(ma_standard_channel_map_microsoft, pDescriptorPlayback->channelMap, ma_countof(pDescriptorPlayback->channelMap), pDescriptorPlayback->channels);
+ pDescriptorPlayback->periodCount = pDescriptorPlayback->periodCount;
+ pDescriptorPlayback->periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__winmm(pDescriptorPlayback, pDescriptorPlayback->sampleRate, pConfig->performanceProfile);
+ }
+
+ /*
+ The heap allocated data is allocated like so:
+
+ [Capture WAVEHDRs][Playback WAVEHDRs][Capture Intermediary Buffer][Playback Intermediary Buffer]
+ */
+ heapSize = 0;
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
+ heapSize += sizeof(WAVEHDR)*pDescriptorCapture->periodCount + (pDescriptorCapture->periodSizeInFrames * pDescriptorCapture->periodCount * ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels));
+ }
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ heapSize += sizeof(WAVEHDR)*pDescriptorPlayback->periodCount + (pDescriptorPlayback->periodSizeInFrames * pDescriptorPlayback->periodCount * ma_get_bytes_per_frame(pDescriptorPlayback->format, pDescriptorPlayback->channels));
+ }
+
+ pDevice->winmm._pHeapData = (ma_uint8*)ma_calloc(heapSize, &pDevice->pContext->allocationCallbacks);
+ if (pDevice->winmm._pHeapData == NULL) {
+ errorMsg = "[WinMM] Failed to allocate memory for the intermediary buffer.", errorCode = MA_OUT_OF_MEMORY;
+ goto on_error;
+ }
+
+ MA_ZERO_MEMORY(pDevice->winmm._pHeapData, heapSize);
+
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
+ ma_uint32 iPeriod;
+
+ if (pConfig->deviceType == ma_device_type_capture) {
+ pDevice->winmm.pWAVEHDRCapture = pDevice->winmm._pHeapData;
+ pDevice->winmm.pIntermediaryBufferCapture = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*(pDescriptorCapture->periodCount));
+ } else {
+ pDevice->winmm.pWAVEHDRCapture = pDevice->winmm._pHeapData;
+ pDevice->winmm.pIntermediaryBufferCapture = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*(pDescriptorCapture->periodCount + pDescriptorPlayback->periodCount));
+ }
+
+ /* Prepare headers. */
+ for (iPeriod = 0; iPeriod < pDescriptorCapture->periodCount; ++iPeriod) {
+ ma_uint32 periodSizeInBytes = ma_get_period_size_in_bytes(pDescriptorCapture->periodSizeInFrames, pDescriptorCapture->format, pDescriptorCapture->channels);
+
+ ((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].lpData = (LPSTR)(pDevice->winmm.pIntermediaryBufferCapture + (periodSizeInBytes*iPeriod));
+ ((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwBufferLength = periodSizeInBytes;
+ ((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwFlags = 0L;
+ ((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwLoops = 0L;
+ ((MA_PFN_waveInPrepareHeader)pDevice->pContext->winmm.waveInPrepareHeader)((HWAVEIN)pDevice->winmm.hDeviceCapture, &((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod], sizeof(WAVEHDR));
+
+ /*
+ The user data of the WAVEHDR structure is a single flag the controls whether or not it is ready for writing. Consider it to be named "isLocked". A value of 0 means
+ it's unlocked and available for writing. A value of 1 means it's locked.
+ */
+ ((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwUser = 0;
+ }
+ }
+
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ ma_uint32 iPeriod;
+
+ if (pConfig->deviceType == ma_device_type_playback) {
+ pDevice->winmm.pWAVEHDRPlayback = pDevice->winmm._pHeapData;
+ pDevice->winmm.pIntermediaryBufferPlayback = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*pDescriptorPlayback->periodCount);
+ } else {
+ pDevice->winmm.pWAVEHDRPlayback = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*(pDescriptorCapture->periodCount));
+ pDevice->winmm.pIntermediaryBufferPlayback = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*(pDescriptorCapture->periodCount + pDescriptorPlayback->periodCount)) + (pDescriptorCapture->periodSizeInFrames*pDescriptorCapture->periodCount*ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels));
+ }
+
+ /* Prepare headers. */
+ for (iPeriod = 0; iPeriod < pDescriptorPlayback->periodCount; ++iPeriod) {
+ ma_uint32 periodSizeInBytes = ma_get_period_size_in_bytes(pDescriptorPlayback->periodSizeInFrames, pDescriptorPlayback->format, pDescriptorPlayback->channels);
+
+ ((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].lpData = (LPSTR)(pDevice->winmm.pIntermediaryBufferPlayback + (periodSizeInBytes*iPeriod));
+ ((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwBufferLength = periodSizeInBytes;
+ ((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwFlags = 0L;
+ ((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwLoops = 0L;
+ ((MA_PFN_waveOutPrepareHeader)pDevice->pContext->winmm.waveOutPrepareHeader)((HWAVEOUT)pDevice->winmm.hDevicePlayback, &((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod], sizeof(WAVEHDR));
+
+ /*
+ The user data of the WAVEHDR structure is a single flag the controls whether or not it is ready for writing. Consider it to be named "isLocked". A value of 0 means
+ it's unlocked and available for writing. A value of 1 means it's locked.
+ */
+ ((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwUser = 0;
+ }
+ }
+
+ return MA_SUCCESS;
+
+on_error:
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ if (pDevice->winmm.pWAVEHDRCapture != NULL) {
+ ma_uint32 iPeriod;
+ for (iPeriod = 0; iPeriod < pDescriptorCapture->periodCount; ++iPeriod) {
+ ((MA_PFN_waveInUnprepareHeader)pDevice->pContext->winmm.waveInUnprepareHeader)((HWAVEIN)pDevice->winmm.hDeviceCapture, &((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod], sizeof(WAVEHDR));
+ }
+ }
+
+ ((MA_PFN_waveInClose)pDevice->pContext->winmm.waveInClose)((HWAVEIN)pDevice->winmm.hDeviceCapture);
+ }
+
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ if (pDevice->winmm.pWAVEHDRCapture != NULL) {
+ ma_uint32 iPeriod;
+ for (iPeriod = 0; iPeriod < pDescriptorPlayback->periodCount; ++iPeriod) {
+ ((MA_PFN_waveOutUnprepareHeader)pDevice->pContext->winmm.waveOutUnprepareHeader)((HWAVEOUT)pDevice->winmm.hDevicePlayback, &((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod], sizeof(WAVEHDR));
+ }
+ }
+
+ ((MA_PFN_waveOutClose)pDevice->pContext->winmm.waveOutClose)((HWAVEOUT)pDevice->winmm.hDevicePlayback);
+ }
+
+ ma_free(pDevice->winmm._pHeapData, &pDevice->pContext->allocationCallbacks);
+
+ if (errorMsg != NULL && errorMsg[0] != '\0') {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "%s", errorMsg);
+ }
+
+ return errorCode;
+}
+
+static ma_result ma_device_start__winmm(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
+
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ MMRESULT resultMM;
+ WAVEHDR* pWAVEHDR;
+ ma_uint32 iPeriod;
+
+ pWAVEHDR = (WAVEHDR*)pDevice->winmm.pWAVEHDRCapture;
+
+ /* Make sure the event is reset to a non-signaled state to ensure we don't prematurely return from WaitForSingleObject(). */
+ ResetEvent((HANDLE)pDevice->winmm.hEventCapture);
+
+ /* To start the device we attach all of the buffers and then start it. As the buffers are filled with data we will get notifications. */
+ for (iPeriod = 0; iPeriod < pDevice->capture.internalPeriods; ++iPeriod) {
+ resultMM = ((MA_PFN_waveInAddBuffer)pDevice->pContext->winmm.waveInAddBuffer)((HWAVEIN)pDevice->winmm.hDeviceCapture, &((LPWAVEHDR)pDevice->winmm.pWAVEHDRCapture)[iPeriod], sizeof(WAVEHDR));
+ if (resultMM != MMSYSERR_NOERROR) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WinMM] Failed to attach input buffers to capture device in preparation for capture.");
+ return ma_result_from_MMRESULT(resultMM);
+ }
+
+ /* Make sure all of the buffers start out locked. We don't want to access them until the backend tells us we can. */
+ pWAVEHDR[iPeriod].dwUser = 1; /* 1 = locked. */
+ }
+
+ /* Capture devices need to be explicitly started, unlike playback devices. */
+ resultMM = ((MA_PFN_waveInStart)pDevice->pContext->winmm.waveInStart)((HWAVEIN)pDevice->winmm.hDeviceCapture);
+ if (resultMM != MMSYSERR_NOERROR) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WinMM] Failed to start backend device.");
+ return ma_result_from_MMRESULT(resultMM);
+ }
+ }
+
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ /* Don't need to do anything for playback. It'll be started automatically in ma_device_start__winmm(). */
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_stop__winmm(ma_device* pDevice)
+{
+ MMRESULT resultMM;
+
+ MA_ASSERT(pDevice != NULL);
+
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ if (pDevice->winmm.hDeviceCapture == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ resultMM = ((MA_PFN_waveInReset)pDevice->pContext->winmm.waveInReset)((HWAVEIN)pDevice->winmm.hDeviceCapture);
+ if (resultMM != MMSYSERR_NOERROR) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[WinMM] WARNING: Failed to reset capture device.");
+ }
+ }
+
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ ma_uint32 iPeriod;
+ WAVEHDR* pWAVEHDR;
+
+ if (pDevice->winmm.hDevicePlayback == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ /* We need to drain the device. To do this we just loop over each header and if it's locked just wait for the event. */
+ pWAVEHDR = (WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback;
+ for (iPeriod = 0; iPeriod < pDevice->playback.internalPeriods; iPeriod += 1) {
+ if (pWAVEHDR[iPeriod].dwUser == 1) { /* 1 = locked. */
+ if (WaitForSingleObject((HANDLE)pDevice->winmm.hEventPlayback, INFINITE) != WAIT_OBJECT_0) {
+ break; /* An error occurred so just abandon ship and stop the device without draining. */
+ }
+
+ pWAVEHDR[iPeriod].dwUser = 0;
+ }
+ }
+
+ resultMM = ((MA_PFN_waveOutReset)pDevice->pContext->winmm.waveOutReset)((HWAVEOUT)pDevice->winmm.hDevicePlayback);
+ if (resultMM != MMSYSERR_NOERROR) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[WinMM] WARNING: Failed to reset playback device.");
+ }
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_write__winmm(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten)
+{
+ ma_result result = MA_SUCCESS;
+ MMRESULT resultMM;
+ ma_uint32 totalFramesWritten;
+ WAVEHDR* pWAVEHDR;
+
+ MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(pPCMFrames != NULL);
+
+ if (pFramesWritten != NULL) {
+ *pFramesWritten = 0;
+ }
+
+ pWAVEHDR = (WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback;
+
+ /* Keep processing as much data as possible. */
+ totalFramesWritten = 0;
+ while (totalFramesWritten < frameCount) {
+ /* If the current header has some space available we need to write part of it. */
+ if (pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwUser == 0) { /* 0 = unlocked. */
+ /*
+ This header has room in it. We copy as much of it as we can. If we end up fully consuming the buffer we need to
+ write it out and move on to the next iteration.
+ */
+ ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
+ ma_uint32 framesRemainingInHeader = (pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwBufferLength/bpf) - pDevice->winmm.headerFramesConsumedPlayback;
+
+ ma_uint32 framesToCopy = ma_min(framesRemainingInHeader, (frameCount - totalFramesWritten));
+ const void* pSrc = ma_offset_ptr(pPCMFrames, totalFramesWritten*bpf);
+ void* pDst = ma_offset_ptr(pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].lpData, pDevice->winmm.headerFramesConsumedPlayback*bpf);
+ MA_COPY_MEMORY(pDst, pSrc, framesToCopy*bpf);
+
+ pDevice->winmm.headerFramesConsumedPlayback += framesToCopy;
+ totalFramesWritten += framesToCopy;
+
+ /* If we've consumed the buffer entirely we need to write it out to the device. */
+ if (pDevice->winmm.headerFramesConsumedPlayback == (pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwBufferLength/bpf)) {
+ pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwUser = 1; /* 1 = locked. */
+ pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwFlags &= ~WHDR_DONE; /* <-- Need to make sure the WHDR_DONE flag is unset. */
+
+ /* Make sure the event is reset to a non-signaled state to ensure we don't prematurely return from WaitForSingleObject(). */
+ ResetEvent((HANDLE)pDevice->winmm.hEventPlayback);
+
+ /* The device will be started here. */
+ resultMM = ((MA_PFN_waveOutWrite)pDevice->pContext->winmm.waveOutWrite)((HWAVEOUT)pDevice->winmm.hDevicePlayback, &pWAVEHDR[pDevice->winmm.iNextHeaderPlayback], sizeof(WAVEHDR));
+ if (resultMM != MMSYSERR_NOERROR) {
+ result = ma_result_from_MMRESULT(resultMM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WinMM] waveOutWrite() failed.");
+ break;
+ }
+
+ /* Make sure we move to the next header. */
+ pDevice->winmm.iNextHeaderPlayback = (pDevice->winmm.iNextHeaderPlayback + 1) % pDevice->playback.internalPeriods;
+ pDevice->winmm.headerFramesConsumedPlayback = 0;
+ }
+
+ /* If at this point we have consumed the entire input buffer we can return. */
+ MA_ASSERT(totalFramesWritten <= frameCount);
+ if (totalFramesWritten == frameCount) {
+ break;
+ }
+
+ /* Getting here means there's more to process. */
+ continue;
+ }
+
+ /* Getting here means there isn't enough room in the buffer and we need to wait for one to become available. */
+ if (WaitForSingleObject((HANDLE)pDevice->winmm.hEventPlayback, INFINITE) != WAIT_OBJECT_0) {
+ result = MA_ERROR;
+ break;
+ }
+
+ /* Something happened. If the next buffer has been marked as done we need to reset a bit of state. */
+ if ((pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwFlags & WHDR_DONE) != 0) {
+ pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwUser = 0; /* 0 = unlocked (make it available for writing). */
+ pDevice->winmm.headerFramesConsumedPlayback = 0;
+ }
+
+ /* If the device has been stopped we need to break. */
+ if (ma_device_get_state(pDevice) != ma_device_state_started) {
+ break;
+ }
+ }
+
+ if (pFramesWritten != NULL) {
+ *pFramesWritten = totalFramesWritten;
+ }
+
+ return result;
+}
+
+static ma_result ma_device_read__winmm(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead)
+{
+ ma_result result = MA_SUCCESS;
+ MMRESULT resultMM;
+ ma_uint32 totalFramesRead;
+ WAVEHDR* pWAVEHDR;
+
+ MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(pPCMFrames != NULL);
+
+ if (pFramesRead != NULL) {
+ *pFramesRead = 0;
+ }
+
+ pWAVEHDR = (WAVEHDR*)pDevice->winmm.pWAVEHDRCapture;
+
+ /* Keep processing as much data as possible. */
+ totalFramesRead = 0;
+ while (totalFramesRead < frameCount) {
+ /* If the current header has some space available we need to write part of it. */
+ if (pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwUser == 0) { /* 0 = unlocked. */
+ /* The buffer is available for reading. If we fully consume it we need to add it back to the buffer. */
+ ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
+ ma_uint32 framesRemainingInHeader = (pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwBufferLength/bpf) - pDevice->winmm.headerFramesConsumedCapture;
+
+ ma_uint32 framesToCopy = ma_min(framesRemainingInHeader, (frameCount - totalFramesRead));
+ const void* pSrc = ma_offset_ptr(pWAVEHDR[pDevice->winmm.iNextHeaderCapture].lpData, pDevice->winmm.headerFramesConsumedCapture*bpf);
+ void* pDst = ma_offset_ptr(pPCMFrames, totalFramesRead*bpf);
+ MA_COPY_MEMORY(pDst, pSrc, framesToCopy*bpf);
+
+ pDevice->winmm.headerFramesConsumedCapture += framesToCopy;
+ totalFramesRead += framesToCopy;
+
+ /* If we've consumed the buffer entirely we need to add it back to the device. */
+ if (pDevice->winmm.headerFramesConsumedCapture == (pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwBufferLength/bpf)) {
+ pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwUser = 1; /* 1 = locked. */
+ pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwFlags &= ~WHDR_DONE; /* <-- Need to make sure the WHDR_DONE flag is unset. */
+
+ /* Make sure the event is reset to a non-signaled state to ensure we don't prematurely return from WaitForSingleObject(). */
+ ResetEvent((HANDLE)pDevice->winmm.hEventCapture);
+
+ /* The device will be started here. */
+ resultMM = ((MA_PFN_waveInAddBuffer)pDevice->pContext->winmm.waveInAddBuffer)((HWAVEIN)pDevice->winmm.hDeviceCapture, &((LPWAVEHDR)pDevice->winmm.pWAVEHDRCapture)[pDevice->winmm.iNextHeaderCapture], sizeof(WAVEHDR));
+ if (resultMM != MMSYSERR_NOERROR) {
+ result = ma_result_from_MMRESULT(resultMM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WinMM] waveInAddBuffer() failed.");
+ break;
+ }
+
+ /* Make sure we move to the next header. */
+ pDevice->winmm.iNextHeaderCapture = (pDevice->winmm.iNextHeaderCapture + 1) % pDevice->capture.internalPeriods;
+ pDevice->winmm.headerFramesConsumedCapture = 0;
+ }
+
+ /* If at this point we have filled the entire input buffer we can return. */
+ MA_ASSERT(totalFramesRead <= frameCount);
+ if (totalFramesRead == frameCount) {
+ break;
+ }
+
+ /* Getting here means there's more to process. */
+ continue;
+ }
+
+ /* Getting here means there isn't enough any data left to send to the client which means we need to wait for more. */
+ if (WaitForSingleObject((HANDLE)pDevice->winmm.hEventCapture, INFINITE) != WAIT_OBJECT_0) {
+ result = MA_ERROR;
+ break;
+ }
+
+ /* Something happened. If the next buffer has been marked as done we need to reset a bit of state. */
+ if ((pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwFlags & WHDR_DONE) != 0) {
+ pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwUser = 0; /* 0 = unlocked (make it available for reading). */
+ pDevice->winmm.headerFramesConsumedCapture = 0;
+ }
+
+ /* If the device has been stopped we need to break. */
+ if (ma_device_get_state(pDevice) != ma_device_state_started) {
+ break;
+ }
+ }
+
+ if (pFramesRead != NULL) {
+ *pFramesRead = totalFramesRead;
+ }
+
+ return result;
+}
+
+static ma_result ma_context_uninit__winmm(ma_context* pContext)
+{
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pContext->backend == ma_backend_winmm);
+
+ ma_dlclose(pContext, pContext->winmm.hWinMM);
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_init__winmm(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+{
+ MA_ASSERT(pContext != NULL);
+
+ (void)pConfig;
+
+ pContext->winmm.hWinMM = ma_dlopen(pContext, "winmm.dll");
+ if (pContext->winmm.hWinMM == NULL) {
+ return MA_NO_BACKEND;
+ }
+
+ pContext->winmm.waveOutGetNumDevs = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutGetNumDevs");
+ pContext->winmm.waveOutGetDevCapsA = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutGetDevCapsA");
+ pContext->winmm.waveOutOpen = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutOpen");
+ pContext->winmm.waveOutClose = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutClose");
+ pContext->winmm.waveOutPrepareHeader = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutPrepareHeader");
+ pContext->winmm.waveOutUnprepareHeader = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutUnprepareHeader");
+ pContext->winmm.waveOutWrite = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutWrite");
+ pContext->winmm.waveOutReset = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutReset");
+ pContext->winmm.waveInGetNumDevs = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInGetNumDevs");
+ pContext->winmm.waveInGetDevCapsA = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInGetDevCapsA");
+ pContext->winmm.waveInOpen = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInOpen");
+ pContext->winmm.waveInClose = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInClose");
+ pContext->winmm.waveInPrepareHeader = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInPrepareHeader");
+ pContext->winmm.waveInUnprepareHeader = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInUnprepareHeader");
+ pContext->winmm.waveInAddBuffer = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInAddBuffer");
+ pContext->winmm.waveInStart = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInStart");
+ pContext->winmm.waveInReset = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInReset");
+
+ pCallbacks->onContextInit = ma_context_init__winmm;
+ pCallbacks->onContextUninit = ma_context_uninit__winmm;
+ pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__winmm;
+ pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__winmm;
+ pCallbacks->onDeviceInit = ma_device_init__winmm;
+ pCallbacks->onDeviceUninit = ma_device_uninit__winmm;
+ pCallbacks->onDeviceStart = ma_device_start__winmm;
+ pCallbacks->onDeviceStop = ma_device_stop__winmm;
+ pCallbacks->onDeviceRead = ma_device_read__winmm;
+ pCallbacks->onDeviceWrite = ma_device_write__winmm;
+ pCallbacks->onDeviceDataLoop = NULL; /* This is a blocking read-write API, so this can be NULL since miniaudio will manage the audio thread for us. */
+
+ return MA_SUCCESS;
+}
+#endif
+
+
+
+
+/******************************************************************************
+
+ALSA Backend
+
+******************************************************************************/
+#ifdef MA_HAS_ALSA
+
+#include /* poll(), struct pollfd */
+#include /* eventfd() */
+
+#ifdef MA_NO_RUNTIME_LINKING
+
+/* asoundlib.h marks some functions with "inline" which isn't always supported. Need to emulate it. */
+#if !defined(__cplusplus)
+ #if defined(__STRICT_ANSI__)
+ #if !defined(inline)
+ #define inline __inline__ __attribute__((always_inline))
+ #define MA_INLINE_DEFINED
+ #endif
+ #endif
+#endif
+#include
+#if defined(MA_INLINE_DEFINED)
+ #undef inline
+ #undef MA_INLINE_DEFINED
+#endif
+
+typedef snd_pcm_uframes_t ma_snd_pcm_uframes_t;
+typedef snd_pcm_sframes_t ma_snd_pcm_sframes_t;
+typedef snd_pcm_stream_t ma_snd_pcm_stream_t;
+typedef snd_pcm_format_t ma_snd_pcm_format_t;
+typedef snd_pcm_access_t ma_snd_pcm_access_t;
+typedef snd_pcm_t ma_snd_pcm_t;
+typedef snd_pcm_hw_params_t ma_snd_pcm_hw_params_t;
+typedef snd_pcm_sw_params_t ma_snd_pcm_sw_params_t;
+typedef snd_pcm_format_mask_t ma_snd_pcm_format_mask_t;
+typedef snd_pcm_info_t ma_snd_pcm_info_t;
+typedef snd_pcm_channel_area_t ma_snd_pcm_channel_area_t;
+typedef snd_pcm_chmap_t ma_snd_pcm_chmap_t;
+typedef snd_pcm_state_t ma_snd_pcm_state_t;
+
+/* snd_pcm_stream_t */
+#define MA_SND_PCM_STREAM_PLAYBACK SND_PCM_STREAM_PLAYBACK
+#define MA_SND_PCM_STREAM_CAPTURE SND_PCM_STREAM_CAPTURE
+
+/* snd_pcm_format_t */
+#define MA_SND_PCM_FORMAT_UNKNOWN SND_PCM_FORMAT_UNKNOWN
+#define MA_SND_PCM_FORMAT_U8 SND_PCM_FORMAT_U8
+#define MA_SND_PCM_FORMAT_S16_LE SND_PCM_FORMAT_S16_LE
+#define MA_SND_PCM_FORMAT_S16_BE SND_PCM_FORMAT_S16_BE
+#define MA_SND_PCM_FORMAT_S24_LE SND_PCM_FORMAT_S24_LE
+#define MA_SND_PCM_FORMAT_S24_BE SND_PCM_FORMAT_S24_BE
+#define MA_SND_PCM_FORMAT_S32_LE SND_PCM_FORMAT_S32_LE
+#define MA_SND_PCM_FORMAT_S32_BE SND_PCM_FORMAT_S32_BE
+#define MA_SND_PCM_FORMAT_FLOAT_LE SND_PCM_FORMAT_FLOAT_LE
+#define MA_SND_PCM_FORMAT_FLOAT_BE SND_PCM_FORMAT_FLOAT_BE
+#define MA_SND_PCM_FORMAT_FLOAT64_LE SND_PCM_FORMAT_FLOAT64_LE
+#define MA_SND_PCM_FORMAT_FLOAT64_BE SND_PCM_FORMAT_FLOAT64_BE
+#define MA_SND_PCM_FORMAT_MU_LAW SND_PCM_FORMAT_MU_LAW
+#define MA_SND_PCM_FORMAT_A_LAW SND_PCM_FORMAT_A_LAW
+#define MA_SND_PCM_FORMAT_S24_3LE SND_PCM_FORMAT_S24_3LE
+#define MA_SND_PCM_FORMAT_S24_3BE SND_PCM_FORMAT_S24_3BE
+
+/* ma_snd_pcm_access_t */
+#define MA_SND_PCM_ACCESS_MMAP_INTERLEAVED SND_PCM_ACCESS_MMAP_INTERLEAVED
+#define MA_SND_PCM_ACCESS_MMAP_NONINTERLEAVED SND_PCM_ACCESS_MMAP_NONINTERLEAVED
+#define MA_SND_PCM_ACCESS_MMAP_COMPLEX SND_PCM_ACCESS_MMAP_COMPLEX
+#define MA_SND_PCM_ACCESS_RW_INTERLEAVED SND_PCM_ACCESS_RW_INTERLEAVED
+#define MA_SND_PCM_ACCESS_RW_NONINTERLEAVED SND_PCM_ACCESS_RW_NONINTERLEAVED
+
+/* Channel positions. */
+#define MA_SND_CHMAP_UNKNOWN SND_CHMAP_UNKNOWN
+#define MA_SND_CHMAP_NA SND_CHMAP_NA
+#define MA_SND_CHMAP_MONO SND_CHMAP_MONO
+#define MA_SND_CHMAP_FL SND_CHMAP_FL
+#define MA_SND_CHMAP_FR SND_CHMAP_FR
+#define MA_SND_CHMAP_RL SND_CHMAP_RL
+#define MA_SND_CHMAP_RR SND_CHMAP_RR
+#define MA_SND_CHMAP_FC SND_CHMAP_FC
+#define MA_SND_CHMAP_LFE SND_CHMAP_LFE
+#define MA_SND_CHMAP_SL SND_CHMAP_SL
+#define MA_SND_CHMAP_SR SND_CHMAP_SR
+#define MA_SND_CHMAP_RC SND_CHMAP_RC
+#define MA_SND_CHMAP_FLC SND_CHMAP_FLC
+#define MA_SND_CHMAP_FRC SND_CHMAP_FRC
+#define MA_SND_CHMAP_RLC SND_CHMAP_RLC
+#define MA_SND_CHMAP_RRC SND_CHMAP_RRC
+#define MA_SND_CHMAP_FLW SND_CHMAP_FLW
+#define MA_SND_CHMAP_FRW SND_CHMAP_FRW
+#define MA_SND_CHMAP_FLH SND_CHMAP_FLH
+#define MA_SND_CHMAP_FCH SND_CHMAP_FCH
+#define MA_SND_CHMAP_FRH SND_CHMAP_FRH
+#define MA_SND_CHMAP_TC SND_CHMAP_TC
+#define MA_SND_CHMAP_TFL SND_CHMAP_TFL
+#define MA_SND_CHMAP_TFR SND_CHMAP_TFR
+#define MA_SND_CHMAP_TFC SND_CHMAP_TFC
+#define MA_SND_CHMAP_TRL SND_CHMAP_TRL
+#define MA_SND_CHMAP_TRR SND_CHMAP_TRR
+#define MA_SND_CHMAP_TRC SND_CHMAP_TRC
+#define MA_SND_CHMAP_TFLC SND_CHMAP_TFLC
+#define MA_SND_CHMAP_TFRC SND_CHMAP_TFRC
+#define MA_SND_CHMAP_TSL SND_CHMAP_TSL
+#define MA_SND_CHMAP_TSR SND_CHMAP_TSR
+#define MA_SND_CHMAP_LLFE SND_CHMAP_LLFE
+#define MA_SND_CHMAP_RLFE SND_CHMAP_RLFE
+#define MA_SND_CHMAP_BC SND_CHMAP_BC
+#define MA_SND_CHMAP_BLC SND_CHMAP_BLC
+#define MA_SND_CHMAP_BRC SND_CHMAP_BRC
+
+/* Open mode flags. */
+#define MA_SND_PCM_NO_AUTO_RESAMPLE SND_PCM_NO_AUTO_RESAMPLE
+#define MA_SND_PCM_NO_AUTO_CHANNELS SND_PCM_NO_AUTO_CHANNELS
+#define MA_SND_PCM_NO_AUTO_FORMAT SND_PCM_NO_AUTO_FORMAT
#else
-#define C89ATOMIC_32BIT
+#include /* For EPIPE, etc. */
+typedef unsigned long ma_snd_pcm_uframes_t;
+typedef long ma_snd_pcm_sframes_t;
+typedef int ma_snd_pcm_stream_t;
+typedef int ma_snd_pcm_format_t;
+typedef int ma_snd_pcm_access_t;
+typedef int ma_snd_pcm_state_t;
+typedef struct ma_snd_pcm_t ma_snd_pcm_t;
+typedef struct ma_snd_pcm_hw_params_t ma_snd_pcm_hw_params_t;
+typedef struct ma_snd_pcm_sw_params_t ma_snd_pcm_sw_params_t;
+typedef struct ma_snd_pcm_format_mask_t ma_snd_pcm_format_mask_t;
+typedef struct ma_snd_pcm_info_t ma_snd_pcm_info_t;
+typedef struct
+{
+ void* addr;
+ unsigned int first;
+ unsigned int step;
+} ma_snd_pcm_channel_area_t;
+typedef struct
+{
+ unsigned int channels;
+ unsigned int pos[1];
+} ma_snd_pcm_chmap_t;
+
+/* snd_pcm_state_t */
+#define MA_SND_PCM_STATE_OPEN 0
+#define MA_SND_PCM_STATE_SETUP 1
+#define MA_SND_PCM_STATE_PREPARED 2
+#define MA_SND_PCM_STATE_RUNNING 3
+#define MA_SND_PCM_STATE_XRUN 4
+#define MA_SND_PCM_STATE_DRAINING 5
+#define MA_SND_PCM_STATE_PAUSED 6
+#define MA_SND_PCM_STATE_SUSPENDED 7
+#define MA_SND_PCM_STATE_DISCONNECTED 8
+
+/* snd_pcm_stream_t */
+#define MA_SND_PCM_STREAM_PLAYBACK 0
+#define MA_SND_PCM_STREAM_CAPTURE 1
+
+/* snd_pcm_format_t */
+#define MA_SND_PCM_FORMAT_UNKNOWN -1
+#define MA_SND_PCM_FORMAT_U8 1
+#define MA_SND_PCM_FORMAT_S16_LE 2
+#define MA_SND_PCM_FORMAT_S16_BE 3
+#define MA_SND_PCM_FORMAT_S24_LE 6
+#define MA_SND_PCM_FORMAT_S24_BE 7
+#define MA_SND_PCM_FORMAT_S32_LE 10
+#define MA_SND_PCM_FORMAT_S32_BE 11
+#define MA_SND_PCM_FORMAT_FLOAT_LE 14
+#define MA_SND_PCM_FORMAT_FLOAT_BE 15
+#define MA_SND_PCM_FORMAT_FLOAT64_LE 16
+#define MA_SND_PCM_FORMAT_FLOAT64_BE 17
+#define MA_SND_PCM_FORMAT_MU_LAW 20
+#define MA_SND_PCM_FORMAT_A_LAW 21
+#define MA_SND_PCM_FORMAT_S24_3LE 32
+#define MA_SND_PCM_FORMAT_S24_3BE 33
+
+/* snd_pcm_access_t */
+#define MA_SND_PCM_ACCESS_MMAP_INTERLEAVED 0
+#define MA_SND_PCM_ACCESS_MMAP_NONINTERLEAVED 1
+#define MA_SND_PCM_ACCESS_MMAP_COMPLEX 2
+#define MA_SND_PCM_ACCESS_RW_INTERLEAVED 3
+#define MA_SND_PCM_ACCESS_RW_NONINTERLEAVED 4
+
+/* Channel positions. */
+#define MA_SND_CHMAP_UNKNOWN 0
+#define MA_SND_CHMAP_NA 1
+#define MA_SND_CHMAP_MONO 2
+#define MA_SND_CHMAP_FL 3
+#define MA_SND_CHMAP_FR 4
+#define MA_SND_CHMAP_RL 5
+#define MA_SND_CHMAP_RR 6
+#define MA_SND_CHMAP_FC 7
+#define MA_SND_CHMAP_LFE 8
+#define MA_SND_CHMAP_SL 9
+#define MA_SND_CHMAP_SR 10
+#define MA_SND_CHMAP_RC 11
+#define MA_SND_CHMAP_FLC 12
+#define MA_SND_CHMAP_FRC 13
+#define MA_SND_CHMAP_RLC 14
+#define MA_SND_CHMAP_RRC 15
+#define MA_SND_CHMAP_FLW 16
+#define MA_SND_CHMAP_FRW 17
+#define MA_SND_CHMAP_FLH 18
+#define MA_SND_CHMAP_FCH 19
+#define MA_SND_CHMAP_FRH 20
+#define MA_SND_CHMAP_TC 21
+#define MA_SND_CHMAP_TFL 22
+#define MA_SND_CHMAP_TFR 23
+#define MA_SND_CHMAP_TFC 24
+#define MA_SND_CHMAP_TRL 25
+#define MA_SND_CHMAP_TRR 26
+#define MA_SND_CHMAP_TRC 27
+#define MA_SND_CHMAP_TFLC 28
+#define MA_SND_CHMAP_TFRC 29
+#define MA_SND_CHMAP_TSL 30
+#define MA_SND_CHMAP_TSR 31
+#define MA_SND_CHMAP_LLFE 32
+#define MA_SND_CHMAP_RLFE 33
+#define MA_SND_CHMAP_BC 34
+#define MA_SND_CHMAP_BLC 35
+#define MA_SND_CHMAP_BRC 36
+
+/* Open mode flags. */
+#define MA_SND_PCM_NO_AUTO_RESAMPLE 0x00010000
+#define MA_SND_PCM_NO_AUTO_CHANNELS 0x00020000
+#define MA_SND_PCM_NO_AUTO_FORMAT 0x00040000
#endif
+
+typedef int (* ma_snd_pcm_open_proc) (ma_snd_pcm_t **pcm, const char *name, ma_snd_pcm_stream_t stream, int mode);
+typedef int (* ma_snd_pcm_close_proc) (ma_snd_pcm_t *pcm);
+typedef size_t (* ma_snd_pcm_hw_params_sizeof_proc) (void);
+typedef int (* ma_snd_pcm_hw_params_any_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params);
+typedef int (* ma_snd_pcm_hw_params_set_format_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t val);
+typedef int (* ma_snd_pcm_hw_params_set_format_first_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t *format);
+typedef void (* ma_snd_pcm_hw_params_get_format_mask_proc) (ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_mask_t *mask);
+typedef int (* ma_snd_pcm_hw_params_set_channels_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val);
+typedef int (* ma_snd_pcm_hw_params_set_channels_near_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *val);
+typedef int (* ma_snd_pcm_hw_params_set_channels_minmax_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *minimum, unsigned int *maximum);
+typedef int (* ma_snd_pcm_hw_params_set_rate_resample_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val);
+typedef int (* ma_snd_pcm_hw_params_set_rate_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val, int dir);
+typedef int (* ma_snd_pcm_hw_params_set_rate_near_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *val, int *dir);
+typedef int (* ma_snd_pcm_hw_params_set_buffer_size_near_proc)(ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_uframes_t *val);
+typedef int (* ma_snd_pcm_hw_params_set_periods_near_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *val, int *dir);
+typedef int (* ma_snd_pcm_hw_params_set_access_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_access_t _access);
+typedef int (* ma_snd_pcm_hw_params_get_format_proc) (const ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t *format);
+typedef int (* ma_snd_pcm_hw_params_get_channels_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val);
+typedef int (* ma_snd_pcm_hw_params_get_channels_min_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val);
+typedef int (* ma_snd_pcm_hw_params_get_channels_max_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val);
+typedef int (* ma_snd_pcm_hw_params_get_rate_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *rate, int *dir);
+typedef int (* ma_snd_pcm_hw_params_get_rate_min_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *rate, int *dir);
+typedef int (* ma_snd_pcm_hw_params_get_rate_max_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *rate, int *dir);
+typedef int (* ma_snd_pcm_hw_params_get_buffer_size_proc) (const ma_snd_pcm_hw_params_t *params, ma_snd_pcm_uframes_t *val);
+typedef int (* ma_snd_pcm_hw_params_get_periods_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val, int *dir);
+typedef int (* ma_snd_pcm_hw_params_get_access_proc) (const ma_snd_pcm_hw_params_t *params, ma_snd_pcm_access_t *_access);
+typedef int (* ma_snd_pcm_hw_params_test_format_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t val);
+typedef int (* ma_snd_pcm_hw_params_test_channels_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val);
+typedef int (* ma_snd_pcm_hw_params_test_rate_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val, int dir);
+typedef int (* ma_snd_pcm_hw_params_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params);
+typedef size_t (* ma_snd_pcm_sw_params_sizeof_proc) (void);
+typedef int (* ma_snd_pcm_sw_params_current_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params);
+typedef int (* ma_snd_pcm_sw_params_get_boundary_proc) (const ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t* val);
+typedef int (* ma_snd_pcm_sw_params_set_avail_min_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t val);
+typedef int (* ma_snd_pcm_sw_params_set_start_threshold_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t val);
+typedef int (* ma_snd_pcm_sw_params_set_stop_threshold_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t val);
+typedef int (* ma_snd_pcm_sw_params_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params);
+typedef size_t (* ma_snd_pcm_format_mask_sizeof_proc) (void);
+typedef int (* ma_snd_pcm_format_mask_test_proc) (const ma_snd_pcm_format_mask_t *mask, ma_snd_pcm_format_t val);
+typedef ma_snd_pcm_chmap_t * (* ma_snd_pcm_get_chmap_proc) (ma_snd_pcm_t *pcm);
+typedef ma_snd_pcm_state_t (* ma_snd_pcm_state_proc) (ma_snd_pcm_t *pcm);
+typedef int (* ma_snd_pcm_prepare_proc) (ma_snd_pcm_t *pcm);
+typedef int (* ma_snd_pcm_start_proc) (ma_snd_pcm_t *pcm);
+typedef int (* ma_snd_pcm_drop_proc) (ma_snd_pcm_t *pcm);
+typedef int (* ma_snd_pcm_drain_proc) (ma_snd_pcm_t *pcm);
+typedef int (* ma_snd_pcm_reset_proc) (ma_snd_pcm_t *pcm);
+typedef int (* ma_snd_device_name_hint_proc) (int card, const char *iface, void ***hints);
+typedef char * (* ma_snd_device_name_get_hint_proc) (const void *hint, const char *id);
+typedef int (* ma_snd_card_get_index_proc) (const char *name);
+typedef int (* ma_snd_device_name_free_hint_proc) (void **hints);
+typedef int (* ma_snd_pcm_mmap_begin_proc) (ma_snd_pcm_t *pcm, const ma_snd_pcm_channel_area_t **areas, ma_snd_pcm_uframes_t *offset, ma_snd_pcm_uframes_t *frames);
+typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_mmap_commit_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_uframes_t offset, ma_snd_pcm_uframes_t frames);
+typedef int (* ma_snd_pcm_recover_proc) (ma_snd_pcm_t *pcm, int err, int silent);
+typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_readi_proc) (ma_snd_pcm_t *pcm, void *buffer, ma_snd_pcm_uframes_t size);
+typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_writei_proc) (ma_snd_pcm_t *pcm, const void *buffer, ma_snd_pcm_uframes_t size);
+typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_avail_proc) (ma_snd_pcm_t *pcm);
+typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_avail_update_proc) (ma_snd_pcm_t *pcm);
+typedef int (* ma_snd_pcm_wait_proc) (ma_snd_pcm_t *pcm, int timeout);
+typedef int (* ma_snd_pcm_nonblock_proc) (ma_snd_pcm_t *pcm, int nonblock);
+typedef int (* ma_snd_pcm_info_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_info_t* info);
+typedef size_t (* ma_snd_pcm_info_sizeof_proc) (void);
+typedef const char* (* ma_snd_pcm_info_get_name_proc) (const ma_snd_pcm_info_t* info);
+typedef int (* ma_snd_pcm_poll_descriptors_proc) (ma_snd_pcm_t *pcm, struct pollfd *pfds, unsigned int space);
+typedef int (* ma_snd_pcm_poll_descriptors_count_proc) (ma_snd_pcm_t *pcm);
+typedef int (* ma_snd_pcm_poll_descriptors_revents_proc) (ma_snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents);
+typedef int (* ma_snd_config_update_free_global_proc) (void);
+
+/* This array specifies each of the common devices that can be used for both playback and capture. */
+static const char* g_maCommonDeviceNamesALSA[] = {
+ "default",
+ "null",
+ "pulse",
+ "jack"
+};
+
+/* This array allows us to blacklist specific playback devices. */
+static const char* g_maBlacklistedPlaybackDeviceNamesALSA[] = {
+ ""
+};
+
+/* This array allows us to blacklist specific capture devices. */
+static const char* g_maBlacklistedCaptureDeviceNamesALSA[] = {
+ ""
+};
+
+
+static ma_snd_pcm_format_t ma_convert_ma_format_to_alsa_format(ma_format format)
+{
+ ma_snd_pcm_format_t ALSAFormats[] = {
+ MA_SND_PCM_FORMAT_UNKNOWN, /* ma_format_unknown */
+ MA_SND_PCM_FORMAT_U8, /* ma_format_u8 */
+ MA_SND_PCM_FORMAT_S16_LE, /* ma_format_s16 */
+ MA_SND_PCM_FORMAT_S24_3LE, /* ma_format_s24 */
+ MA_SND_PCM_FORMAT_S32_LE, /* ma_format_s32 */
+ MA_SND_PCM_FORMAT_FLOAT_LE /* ma_format_f32 */
+ };
+
+ if (ma_is_big_endian()) {
+ ALSAFormats[0] = MA_SND_PCM_FORMAT_UNKNOWN;
+ ALSAFormats[1] = MA_SND_PCM_FORMAT_U8;
+ ALSAFormats[2] = MA_SND_PCM_FORMAT_S16_BE;
+ ALSAFormats[3] = MA_SND_PCM_FORMAT_S24_3BE;
+ ALSAFormats[4] = MA_SND_PCM_FORMAT_S32_BE;
+ ALSAFormats[5] = MA_SND_PCM_FORMAT_FLOAT_BE;
+ }
+
+ return ALSAFormats[format];
+}
+
+static ma_format ma_format_from_alsa(ma_snd_pcm_format_t formatALSA)
+{
+ if (ma_is_little_endian()) {
+ switch (formatALSA) {
+ case MA_SND_PCM_FORMAT_S16_LE: return ma_format_s16;
+ case MA_SND_PCM_FORMAT_S24_3LE: return ma_format_s24;
+ case MA_SND_PCM_FORMAT_S32_LE: return ma_format_s32;
+ case MA_SND_PCM_FORMAT_FLOAT_LE: return ma_format_f32;
+ default: break;
+ }
+ } else {
+ switch (formatALSA) {
+ case MA_SND_PCM_FORMAT_S16_BE: return ma_format_s16;
+ case MA_SND_PCM_FORMAT_S24_3BE: return ma_format_s24;
+ case MA_SND_PCM_FORMAT_S32_BE: return ma_format_s32;
+ case MA_SND_PCM_FORMAT_FLOAT_BE: return ma_format_f32;
+ default: break;
+ }
+ }
+
+ /* Endian agnostic. */
+ switch (formatALSA) {
+ case MA_SND_PCM_FORMAT_U8: return ma_format_u8;
+ default: return ma_format_unknown;
+ }
+}
+
+static ma_channel ma_convert_alsa_channel_position_to_ma_channel(unsigned int alsaChannelPos)
+{
+ switch (alsaChannelPos)
+ {
+ case MA_SND_CHMAP_MONO: return MA_CHANNEL_MONO;
+ case MA_SND_CHMAP_FL: return MA_CHANNEL_FRONT_LEFT;
+ case MA_SND_CHMAP_FR: return MA_CHANNEL_FRONT_RIGHT;
+ case MA_SND_CHMAP_RL: return MA_CHANNEL_BACK_LEFT;
+ case MA_SND_CHMAP_RR: return MA_CHANNEL_BACK_RIGHT;
+ case MA_SND_CHMAP_FC: return MA_CHANNEL_FRONT_CENTER;
+ case MA_SND_CHMAP_LFE: return MA_CHANNEL_LFE;
+ case MA_SND_CHMAP_SL: return MA_CHANNEL_SIDE_LEFT;
+ case MA_SND_CHMAP_SR: return MA_CHANNEL_SIDE_RIGHT;
+ case MA_SND_CHMAP_RC: return MA_CHANNEL_BACK_CENTER;
+ case MA_SND_CHMAP_FLC: return MA_CHANNEL_FRONT_LEFT_CENTER;
+ case MA_SND_CHMAP_FRC: return MA_CHANNEL_FRONT_RIGHT_CENTER;
+ case MA_SND_CHMAP_RLC: return 0;
+ case MA_SND_CHMAP_RRC: return 0;
+ case MA_SND_CHMAP_FLW: return 0;
+ case MA_SND_CHMAP_FRW: return 0;
+ case MA_SND_CHMAP_FLH: return 0;
+ case MA_SND_CHMAP_FCH: return 0;
+ case MA_SND_CHMAP_FRH: return 0;
+ case MA_SND_CHMAP_TC: return MA_CHANNEL_TOP_CENTER;
+ case MA_SND_CHMAP_TFL: return MA_CHANNEL_TOP_FRONT_LEFT;
+ case MA_SND_CHMAP_TFR: return MA_CHANNEL_TOP_FRONT_RIGHT;
+ case MA_SND_CHMAP_TFC: return MA_CHANNEL_TOP_FRONT_CENTER;
+ case MA_SND_CHMAP_TRL: return MA_CHANNEL_TOP_BACK_LEFT;
+ case MA_SND_CHMAP_TRR: return MA_CHANNEL_TOP_BACK_RIGHT;
+ case MA_SND_CHMAP_TRC: return MA_CHANNEL_TOP_BACK_CENTER;
+ default: break;
+ }
+
+ return 0;
+}
+
+static ma_bool32 ma_is_common_device_name__alsa(const char* name)
+{
+ size_t iName;
+ for (iName = 0; iName < ma_countof(g_maCommonDeviceNamesALSA); ++iName) {
+ if (ma_strcmp(name, g_maCommonDeviceNamesALSA[iName]) == 0) {
+ return MA_TRUE;
+ }
+ }
+
+ return MA_FALSE;
+}
+
+
+static ma_bool32 ma_is_playback_device_blacklisted__alsa(const char* name)
+{
+ size_t iName;
+ for (iName = 0; iName < ma_countof(g_maBlacklistedPlaybackDeviceNamesALSA); ++iName) {
+ if (ma_strcmp(name, g_maBlacklistedPlaybackDeviceNamesALSA[iName]) == 0) {
+ return MA_TRUE;
+ }
+ }
+
+ return MA_FALSE;
+}
+
+static ma_bool32 ma_is_capture_device_blacklisted__alsa(const char* name)
+{
+ size_t iName;
+ for (iName = 0; iName < ma_countof(g_maBlacklistedCaptureDeviceNamesALSA); ++iName) {
+ if (ma_strcmp(name, g_maBlacklistedCaptureDeviceNamesALSA[iName]) == 0) {
+ return MA_TRUE;
+ }
+ }
+
+ return MA_FALSE;
+}
+
+static ma_bool32 ma_is_device_blacklisted__alsa(ma_device_type deviceType, const char* name)
+{
+ if (deviceType == ma_device_type_playback) {
+ return ma_is_playback_device_blacklisted__alsa(name);
+ } else {
+ return ma_is_capture_device_blacklisted__alsa(name);
+ }
+}
+
+
+static const char* ma_find_char(const char* str, char c, int* index)
+{
+ int i = 0;
+ for (;;) {
+ if (str[i] == '\0') {
+ if (index) *index = -1;
+ return NULL;
+ }
+
+ if (str[i] == c) {
+ if (index) *index = i;
+ return str + i;
+ }
+
+ i += 1;
+ }
+
+ /* Should never get here, but treat it as though the character was not found to make me feel better inside. */
+ if (index) *index = -1;
+ return NULL;
+}
+
+static ma_bool32 ma_is_device_name_in_hw_format__alsa(const char* hwid)
+{
+ /* This function is just checking whether or not hwid is in "hw:%d,%d" format. */
+
+ int commaPos;
+ const char* dev;
+ int i;
+
+ if (hwid == NULL) {
+ return MA_FALSE;
+ }
+
+ if (hwid[0] != 'h' || hwid[1] != 'w' || hwid[2] != ':') {
+ return MA_FALSE;
+ }
+
+ hwid += 3;
+
+ dev = ma_find_char(hwid, ',', &commaPos);
+ if (dev == NULL) {
+ return MA_FALSE;
+ } else {
+ dev += 1; /* Skip past the ",". */
+ }
+
+ /* Check if the part between the ":" and the "," contains only numbers. If not, return false. */
+ for (i = 0; i < commaPos; ++i) {
+ if (hwid[i] < '0' || hwid[i] > '9') {
+ return MA_FALSE;
+ }
+ }
+
+ /* Check if everything after the "," is numeric. If not, return false. */
+ i = 0;
+ while (dev[i] != '\0') {
+ if (dev[i] < '0' || dev[i] > '9') {
+ return MA_FALSE;
+ }
+ i += 1;
+ }
+
+ return MA_TRUE;
+}
+
+static int ma_convert_device_name_to_hw_format__alsa(ma_context* pContext, char* dst, size_t dstSize, const char* src) /* Returns 0 on success, non-0 on error. */
+{
+ /* src should look something like this: "hw:CARD=I82801AAICH,DEV=0" */
+
+ int colonPos;
+ int commaPos;
+ char card[256];
+ const char* dev;
+ int cardIndex;
+
+ if (dst == NULL) {
+ return -1;
+ }
+ if (dstSize < 7) {
+ return -1; /* Absolute minimum size of the output buffer is 7 bytes. */
+ }
+
+ *dst = '\0'; /* Safety. */
+ if (src == NULL) {
+ return -1;
+ }
+
+ /* If the input name is already in "hw:%d,%d" format, just return that verbatim. */
+ if (ma_is_device_name_in_hw_format__alsa(src)) {
+ return ma_strcpy_s(dst, dstSize, src);
+ }
+
+ src = ma_find_char(src, ':', &colonPos);
+ if (src == NULL) {
+ return -1; /* Couldn't find a colon */
+ }
+
+ dev = ma_find_char(src, ',', &commaPos);
+ if (dev == NULL) {
+ dev = "0";
+ ma_strncpy_s(card, sizeof(card), src+6, (size_t)-1); /* +6 = ":CARD=" */
+ } else {
+ dev = dev + 5; /* +5 = ",DEV=" */
+ ma_strncpy_s(card, sizeof(card), src+6, commaPos-6); /* +6 = ":CARD=" */
+ }
+
+ cardIndex = ((ma_snd_card_get_index_proc)pContext->alsa.snd_card_get_index)(card);
+ if (cardIndex < 0) {
+ return -2; /* Failed to retrieve the card index. */
+ }
+
+
+ /* Construction. */
+ dst[0] = 'h'; dst[1] = 'w'; dst[2] = ':';
+ if (ma_itoa_s(cardIndex, dst+3, dstSize-3, 10) != 0) {
+ return -3;
+ }
+ if (ma_strcat_s(dst, dstSize, ",") != 0) {
+ return -3;
+ }
+ if (ma_strcat_s(dst, dstSize, dev) != 0) {
+ return -3;
+ }
+
+ return 0;
+}
+
+static ma_bool32 ma_does_id_exist_in_list__alsa(ma_device_id* pUniqueIDs, ma_uint32 count, const char* pHWID)
+{
+ ma_uint32 i;
+
+ MA_ASSERT(pHWID != NULL);
+
+ for (i = 0; i < count; ++i) {
+ if (ma_strcmp(pUniqueIDs[i].alsa, pHWID) == 0) {
+ return MA_TRUE;
+ }
+ }
+
+ return MA_FALSE;
+}
+
+
+static ma_result ma_context_open_pcm__alsa(ma_context* pContext, ma_share_mode shareMode, ma_device_type deviceType, const ma_device_id* pDeviceID, int openMode, ma_snd_pcm_t** ppPCM)
+{
+ ma_snd_pcm_t* pPCM;
+ ma_snd_pcm_stream_t stream;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(ppPCM != NULL);
+
+ *ppPCM = NULL;
+ pPCM = NULL;
+
+ stream = (deviceType == ma_device_type_playback) ? MA_SND_PCM_STREAM_PLAYBACK : MA_SND_PCM_STREAM_CAPTURE;
+
+ if (pDeviceID == NULL) {
+ ma_bool32 isDeviceOpen;
+ size_t i;
+
+ /*
+ We're opening the default device. I don't know if trying anything other than "default" is necessary, but it makes
+ me feel better to try as hard as we can get to get _something_ working.
+ */
+ const char* defaultDeviceNames[] = {
+ "default",
+ NULL,
+ NULL,
+ NULL,
+ NULL,
+ NULL,
+ NULL
+ };
+
+ if (shareMode == ma_share_mode_exclusive) {
+ defaultDeviceNames[1] = "hw";
+ defaultDeviceNames[2] = "hw:0";
+ defaultDeviceNames[3] = "hw:0,0";
+ } else {
+ if (deviceType == ma_device_type_playback) {
+ defaultDeviceNames[1] = "dmix";
+ defaultDeviceNames[2] = "dmix:0";
+ defaultDeviceNames[3] = "dmix:0,0";
+ } else {
+ defaultDeviceNames[1] = "dsnoop";
+ defaultDeviceNames[2] = "dsnoop:0";
+ defaultDeviceNames[3] = "dsnoop:0,0";
+ }
+ defaultDeviceNames[4] = "hw";
+ defaultDeviceNames[5] = "hw:0";
+ defaultDeviceNames[6] = "hw:0,0";
+ }
+
+ isDeviceOpen = MA_FALSE;
+ for (i = 0; i < ma_countof(defaultDeviceNames); ++i) {
+ if (defaultDeviceNames[i] != NULL && defaultDeviceNames[i][0] != '\0') {
+ if (((ma_snd_pcm_open_proc)pContext->alsa.snd_pcm_open)(&pPCM, defaultDeviceNames[i], stream, openMode) == 0) {
+ isDeviceOpen = MA_TRUE;
+ break;
+ }
+ }
+ }
+
+ if (!isDeviceOpen) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_open() failed when trying to open an appropriate default device.");
+ return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
+ }
+ } else {
+ /*
+ We're trying to open a specific device. There's a few things to consider here:
+
+ miniaudio recongnizes a special format of device id that excludes the "hw", "dmix", etc. prefix. It looks like this: ":0,0", ":0,1", etc. When
+ an ID of this format is specified, it indicates to miniaudio that it can try different combinations of plugins ("hw", "dmix", etc.) until it
+ finds an appropriate one that works. This comes in very handy when trying to open a device in shared mode ("dmix"), vs exclusive mode ("hw").
+ */
+
+ /* May end up needing to make small adjustments to the ID, so make a copy. */
+ ma_device_id deviceID = *pDeviceID;
+ int resultALSA = -ENODEV;
+
+ if (deviceID.alsa[0] != ':') {
+ /* The ID is not in ":0,0" format. Use the ID exactly as-is. */
+ resultALSA = ((ma_snd_pcm_open_proc)pContext->alsa.snd_pcm_open)(&pPCM, deviceID.alsa, stream, openMode);
+ } else {
+ char hwid[256];
+
+ /* The ID is in ":0,0" format. Try different plugins depending on the shared mode. */
+ if (deviceID.alsa[1] == '\0') {
+ deviceID.alsa[0] = '\0'; /* An ID of ":" should be converted to "". */
+ }
+
+ if (shareMode == ma_share_mode_shared) {
+ if (deviceType == ma_device_type_playback) {
+ ma_strcpy_s(hwid, sizeof(hwid), "dmix");
+ } else {
+ ma_strcpy_s(hwid, sizeof(hwid), "dsnoop");
+ }
+
+ if (ma_strcat_s(hwid, sizeof(hwid), deviceID.alsa) == 0) {
+ resultALSA = ((ma_snd_pcm_open_proc)pContext->alsa.snd_pcm_open)(&pPCM, hwid, stream, openMode);
+ }
+ }
+
+ /* If at this point we still don't have an open device it means we're either preferencing exclusive mode or opening with "dmix"/"dsnoop" failed. */
+ if (resultALSA != 0) {
+ ma_strcpy_s(hwid, sizeof(hwid), "hw");
+ if (ma_strcat_s(hwid, sizeof(hwid), deviceID.alsa) == 0) {
+ resultALSA = ((ma_snd_pcm_open_proc)pContext->alsa.snd_pcm_open)(&pPCM, hwid, stream, openMode);
+ }
+ }
+ }
+
+ if (resultALSA < 0) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_open() failed.");
+ return ma_result_from_errno(-resultALSA);
+ }
+ }
+
+ *ppPCM = pPCM;
+ return MA_SUCCESS;
+}
+
+
+static ma_result ma_context_enumerate_devices__alsa(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+{
+ int resultALSA;
+ ma_bool32 cbResult = MA_TRUE;
+ char** ppDeviceHints;
+ ma_device_id* pUniqueIDs = NULL;
+ ma_uint32 uniqueIDCount = 0;
+ char** ppNextDeviceHint;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(callback != NULL);
+
+ ma_mutex_lock(&pContext->alsa.internalDeviceEnumLock);
+
+ resultALSA = ((ma_snd_device_name_hint_proc)pContext->alsa.snd_device_name_hint)(-1, "pcm", (void***)&ppDeviceHints);
+ if (resultALSA < 0) {
+ ma_mutex_unlock(&pContext->alsa.internalDeviceEnumLock);
+ return ma_result_from_errno(-resultALSA);
+ }
+
+ ppNextDeviceHint = ppDeviceHints;
+ while (*ppNextDeviceHint != NULL) {
+ char* NAME = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "NAME");
+ char* DESC = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "DESC");
+ char* IOID = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "IOID");
+ ma_device_type deviceType = ma_device_type_playback;
+ ma_bool32 stopEnumeration = MA_FALSE;
+ char hwid[sizeof(pUniqueIDs->alsa)];
+ ma_device_info deviceInfo;
+
+ if ((IOID == NULL || ma_strcmp(IOID, "Output") == 0)) {
+ deviceType = ma_device_type_playback;
+ }
+ if ((IOID != NULL && ma_strcmp(IOID, "Input" ) == 0)) {
+ deviceType = ma_device_type_capture;
+ }
+
+ if (NAME != NULL) {
+ if (pContext->alsa.useVerboseDeviceEnumeration) {
+ /* Verbose mode. Use the name exactly as-is. */
+ ma_strncpy_s(hwid, sizeof(hwid), NAME, (size_t)-1);
+ } else {
+ /* Simplified mode. Use ":%d,%d" format. */
+ if (ma_convert_device_name_to_hw_format__alsa(pContext, hwid, sizeof(hwid), NAME) == 0) {
+ /*
+ At this point, hwid looks like "hw:0,0". In simplified enumeration mode, we actually want to strip off the
+ plugin name so it looks like ":0,0". The reason for this is that this special format is detected at device
+ initialization time and is used as an indicator to try and use the most appropriate plugin depending on the
+ device type and sharing mode.
+ */
+ char* dst = hwid;
+ char* src = hwid+2;
+ while ((*dst++ = *src++));
+ } else {
+ /* Conversion to "hw:%d,%d" failed. Just use the name as-is. */
+ ma_strncpy_s(hwid, sizeof(hwid), NAME, (size_t)-1);
+ }
+
+ if (ma_does_id_exist_in_list__alsa(pUniqueIDs, uniqueIDCount, hwid)) {
+ goto next_device; /* The device has already been enumerated. Move on to the next one. */
+ } else {
+ /* The device has not yet been enumerated. Make sure it's added to our list so that it's not enumerated again. */
+ size_t newCapacity = sizeof(*pUniqueIDs) * (uniqueIDCount + 1);
+ ma_device_id* pNewUniqueIDs = (ma_device_id*)ma_realloc(pUniqueIDs, newCapacity, &pContext->allocationCallbacks);
+ if (pNewUniqueIDs == NULL) {
+ goto next_device; /* Failed to allocate memory. */
+ }
+
+ pUniqueIDs = pNewUniqueIDs;
+ MA_COPY_MEMORY(pUniqueIDs[uniqueIDCount].alsa, hwid, sizeof(hwid));
+ uniqueIDCount += 1;
+ }
+ }
+ } else {
+ MA_ZERO_MEMORY(hwid, sizeof(hwid));
+ }
+
+ MA_ZERO_OBJECT(&deviceInfo);
+ ma_strncpy_s(deviceInfo.id.alsa, sizeof(deviceInfo.id.alsa), hwid, (size_t)-1);
+
+ /*
+ There's no good way to determine whether or not a device is the default on Linux. We're just going to do something simple and
+ just use the name of "default" as the indicator.
+ */
+ if (ma_strcmp(deviceInfo.id.alsa, "default") == 0) {
+ deviceInfo.isDefault = MA_TRUE;
+ }
+
+
+ /*
+ DESC is the friendly name. We treat this slightly differently depending on whether or not we are using verbose
+ device enumeration. In verbose mode we want to take the entire description so that the end-user can distinguish
+ between the subdevices of each card/dev pair. In simplified mode, however, we only want the first part of the
+ description.
+
+ The value in DESC seems to be split into two lines, with the first line being the name of the device and the
+ second line being a description of the device. I don't like having the description be across two lines because
+ it makes formatting ugly and annoying. I'm therefore deciding to put it all on a single line with the second line
+ being put into parentheses. In simplified mode I'm just stripping the second line entirely.
+ */
+ if (DESC != NULL) {
+ int lfPos;
+ const char* line2 = ma_find_char(DESC, '\n', &lfPos);
+ if (line2 != NULL) {
+ line2 += 1; /* Skip past the new-line character. */
+
+ if (pContext->alsa.useVerboseDeviceEnumeration) {
+ /* Verbose mode. Put the second line in brackets. */
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), DESC, lfPos);
+ ma_strcat_s (deviceInfo.name, sizeof(deviceInfo.name), " (");
+ ma_strcat_s (deviceInfo.name, sizeof(deviceInfo.name), line2);
+ ma_strcat_s (deviceInfo.name, sizeof(deviceInfo.name), ")");
+ } else {
+ /* Simplified mode. Strip the second line entirely. */
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), DESC, lfPos);
+ }
+ } else {
+ /* There's no second line. Just copy the whole description. */
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), DESC, (size_t)-1);
+ }
+ }
+
+ if (!ma_is_device_blacklisted__alsa(deviceType, NAME)) {
+ cbResult = callback(pContext, deviceType, &deviceInfo, pUserData);
+ }
+
+ /*
+ Some devices are both playback and capture, but they are only enumerated by ALSA once. We need to fire the callback
+ again for the other device type in this case. We do this for known devices and where the IOID hint is NULL, which
+ means both Input and Output.
+ */
+ if (cbResult) {
+ if (ma_is_common_device_name__alsa(NAME) || IOID == NULL) {
+ if (deviceType == ma_device_type_playback) {
+ if (!ma_is_capture_device_blacklisted__alsa(NAME)) {
+ cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
+ }
+ } else {
+ if (!ma_is_playback_device_blacklisted__alsa(NAME)) {
+ cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
+ }
+ }
+ }
+ }
+
+ if (cbResult == MA_FALSE) {
+ stopEnumeration = MA_TRUE;
+ }
+
+ next_device:
+ free(NAME);
+ free(DESC);
+ free(IOID);
+ ppNextDeviceHint += 1;
+
+ /* We need to stop enumeration if the callback returned false. */
+ if (stopEnumeration) {
+ break;
+ }
+ }
+
+ ma_free(pUniqueIDs, &pContext->allocationCallbacks);
+ ((ma_snd_device_name_free_hint_proc)pContext->alsa.snd_device_name_free_hint)((void**)ppDeviceHints);
+
+ ma_mutex_unlock(&pContext->alsa.internalDeviceEnumLock);
+
+ return MA_SUCCESS;
+}
+
+
+typedef struct
+{
+ ma_device_type deviceType;
+ const ma_device_id* pDeviceID;
+ ma_share_mode shareMode;
+ ma_device_info* pDeviceInfo;
+ ma_bool32 foundDevice;
+} ma_context_get_device_info_enum_callback_data__alsa;
+
+static ma_bool32 ma_context_get_device_info_enum_callback__alsa(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pDeviceInfo, void* pUserData)
+{
+ ma_context_get_device_info_enum_callback_data__alsa* pData = (ma_context_get_device_info_enum_callback_data__alsa*)pUserData;
+ MA_ASSERT(pData != NULL);
+
+ (void)pContext;
+
+ if (pData->pDeviceID == NULL && ma_strcmp(pDeviceInfo->id.alsa, "default") == 0) {
+ ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), pDeviceInfo->name, (size_t)-1);
+ pData->foundDevice = MA_TRUE;
+ } else {
+ if (pData->deviceType == deviceType && (pData->pDeviceID != NULL && ma_strcmp(pData->pDeviceID->alsa, pDeviceInfo->id.alsa) == 0)) {
+ ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), pDeviceInfo->name, (size_t)-1);
+ pData->foundDevice = MA_TRUE;
+ }
+ }
+
+ /* Keep enumerating until we have found the device. */
+ return !pData->foundDevice;
+}
+
+static void ma_context_test_rate_and_add_native_data_format__alsa(ma_context* pContext, ma_snd_pcm_t* pPCM, ma_snd_pcm_hw_params_t* pHWParams, ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 flags, ma_device_info* pDeviceInfo)
+{
+ MA_ASSERT(pPCM != NULL);
+ MA_ASSERT(pHWParams != NULL);
+ MA_ASSERT(pDeviceInfo != NULL);
+
+ if (pDeviceInfo->nativeDataFormatCount < ma_countof(pDeviceInfo->nativeDataFormats) && ((ma_snd_pcm_hw_params_test_rate_proc)pContext->alsa.snd_pcm_hw_params_test_rate)(pPCM, pHWParams, sampleRate, 0) == 0) {
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = format;
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = channels;
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = sampleRate;
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = flags;
+ pDeviceInfo->nativeDataFormatCount += 1;
+ }
+}
+
+static void ma_context_iterate_rates_and_add_native_data_format__alsa(ma_context* pContext, ma_snd_pcm_t* pPCM, ma_snd_pcm_hw_params_t* pHWParams, ma_format format, ma_uint32 channels, ma_uint32 flags, ma_device_info* pDeviceInfo)
+{
+ ma_uint32 iSampleRate;
+ unsigned int minSampleRate;
+ unsigned int maxSampleRate;
+ int sampleRateDir; /* Not used. Just passed into snd_pcm_hw_params_get_rate_min/max(). */
+
+ /* There could be a range. */
+ ((ma_snd_pcm_hw_params_get_rate_min_proc)pContext->alsa.snd_pcm_hw_params_get_rate_min)(pHWParams, &minSampleRate, &sampleRateDir);
+ ((ma_snd_pcm_hw_params_get_rate_max_proc)pContext->alsa.snd_pcm_hw_params_get_rate_max)(pHWParams, &maxSampleRate, &sampleRateDir);
+
+ /* Make sure our sample rates are clamped to sane values. Stupid devices like "pulse" will reports rates like "1" which is ridiculus. */
+ minSampleRate = ma_clamp(minSampleRate, (unsigned int)ma_standard_sample_rate_min, (unsigned int)ma_standard_sample_rate_max);
+ maxSampleRate = ma_clamp(maxSampleRate, (unsigned int)ma_standard_sample_rate_min, (unsigned int)ma_standard_sample_rate_max);
+
+ for (iSampleRate = 0; iSampleRate < ma_countof(g_maStandardSampleRatePriorities); iSampleRate += 1) {
+ ma_uint32 standardSampleRate = g_maStandardSampleRatePriorities[iSampleRate];
+
+ if (standardSampleRate >= minSampleRate && standardSampleRate <= maxSampleRate) {
+ ma_context_test_rate_and_add_native_data_format__alsa(pContext, pPCM, pHWParams, format, channels, standardSampleRate, flags, pDeviceInfo);
+ }
+ }
+
+ /* Now make sure our min and max rates are included just in case they aren't in the range of our standard rates. */
+ if (!ma_is_standard_sample_rate(minSampleRate)) {
+ ma_context_test_rate_and_add_native_data_format__alsa(pContext, pPCM, pHWParams, format, channels, minSampleRate, flags, pDeviceInfo);
+ }
+
+ if (!ma_is_standard_sample_rate(maxSampleRate) && maxSampleRate != minSampleRate) {
+ ma_context_test_rate_and_add_native_data_format__alsa(pContext, pPCM, pHWParams, format, channels, maxSampleRate, flags, pDeviceInfo);
+ }
+}
+
+static ma_result ma_context_get_device_info__alsa(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+{
+ ma_context_get_device_info_enum_callback_data__alsa data;
+ ma_result result;
+ int resultALSA;
+ ma_snd_pcm_t* pPCM;
+ ma_snd_pcm_hw_params_t* pHWParams;
+ ma_uint32 iFormat;
+ ma_uint32 iChannel;
+
+ MA_ASSERT(pContext != NULL);
+
+ /* We just enumerate to find basic information about the device. */
+ data.deviceType = deviceType;
+ data.pDeviceID = pDeviceID;
+ data.pDeviceInfo = pDeviceInfo;
+ data.foundDevice = MA_FALSE;
+ result = ma_context_enumerate_devices__alsa(pContext, ma_context_get_device_info_enum_callback__alsa, &data);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ if (!data.foundDevice) {
+ return MA_NO_DEVICE;
+ }
+
+ if (ma_strcmp(pDeviceInfo->id.alsa, "default") == 0) {
+ pDeviceInfo->isDefault = MA_TRUE;
+ }
+
+ /* For detailed info we need to open the device. */
+ result = ma_context_open_pcm__alsa(pContext, ma_share_mode_shared, deviceType, pDeviceID, 0, &pPCM);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ /* We need to initialize a HW parameters object in order to know what formats are supported. */
+ pHWParams = (ma_snd_pcm_hw_params_t*)ma_calloc(((ma_snd_pcm_hw_params_sizeof_proc)pContext->alsa.snd_pcm_hw_params_sizeof)(), &pContext->allocationCallbacks);
+ if (pHWParams == NULL) {
+ ((ma_snd_pcm_close_proc)pContext->alsa.snd_pcm_close)(pPCM);
+ return MA_OUT_OF_MEMORY;
+ }
+
+ resultALSA = ((ma_snd_pcm_hw_params_any_proc)pContext->alsa.snd_pcm_hw_params_any)(pPCM, pHWParams);
+ if (resultALSA < 0) {
+ ma_free(pHWParams, &pContext->allocationCallbacks);
+ ((ma_snd_pcm_close_proc)pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to initialize hardware parameters. snd_pcm_hw_params_any() failed.");
+ return ma_result_from_errno(-resultALSA);
+ }
+
+ /*
+ Some ALSA devices can support many permutations of formats, channels and rates. We only support
+ a fixed number of permutations which means we need to employ some strategies to ensure the best
+ combinations are returned. An example is the "pulse" device which can do it's own data conversion
+ in software and as a result can support any combination of format, channels and rate.
+
+ We want to ensure the the first data formats are the best. We have a list of favored sample
+ formats and sample rates, so these will be the basis of our iteration.
+ */
+
+ /* Formats. We just iterate over our standard formats and test them, making sure we reset the configuration space each iteration. */
+ for (iFormat = 0; iFormat < ma_countof(g_maFormatPriorities); iFormat += 1) {
+ ma_format format = g_maFormatPriorities[iFormat];
+
+ /*
+ For each format we need to make sure we reset the configuration space so we don't return
+ channel counts and rates that aren't compatible with a format.
+ */
+ ((ma_snd_pcm_hw_params_any_proc)pContext->alsa.snd_pcm_hw_params_any)(pPCM, pHWParams);
+
+ /* Test the format first. If this fails it means the format is not supported and we can skip it. */
+ if (((ma_snd_pcm_hw_params_test_format_proc)pContext->alsa.snd_pcm_hw_params_test_format)(pPCM, pHWParams, ma_convert_ma_format_to_alsa_format(format)) == 0) {
+ /* The format is supported. */
+ unsigned int minChannels;
+ unsigned int maxChannels;
+
+ /*
+ The configuration space needs to be restricted to this format so we can get an accurate
+ picture of which sample rates and channel counts are support with this format.
+ */
+ ((ma_snd_pcm_hw_params_set_format_proc)pContext->alsa.snd_pcm_hw_params_set_format)(pPCM, pHWParams, ma_convert_ma_format_to_alsa_format(format));
+
+ /* Now we need to check for supported channels. */
+ ((ma_snd_pcm_hw_params_get_channels_min_proc)pContext->alsa.snd_pcm_hw_params_get_channels_min)(pHWParams, &minChannels);
+ ((ma_snd_pcm_hw_params_get_channels_max_proc)pContext->alsa.snd_pcm_hw_params_get_channels_max)(pHWParams, &maxChannels);
+
+ if (minChannels > MA_MAX_CHANNELS) {
+ continue; /* Too many channels. */
+ }
+ if (maxChannels < MA_MIN_CHANNELS) {
+ continue; /* Not enough channels. */
+ }
+
+ /*
+ Make sure the channel count is clamped. This is mainly intended for the max channels
+ because some devices can report an unbound maximum.
+ */
+ minChannels = ma_clamp(minChannels, MA_MIN_CHANNELS, MA_MAX_CHANNELS);
+ maxChannels = ma_clamp(maxChannels, MA_MIN_CHANNELS, MA_MAX_CHANNELS);
+
+ if (minChannels == MA_MIN_CHANNELS && maxChannels == MA_MAX_CHANNELS) {
+ /* The device supports all channels. Don't iterate over every single one. Instead just set the channels to 0 which means all channels are supported. */
+ ma_context_iterate_rates_and_add_native_data_format__alsa(pContext, pPCM, pHWParams, format, 0, 0, pDeviceInfo); /* Intentionally setting the channel count to 0 as that means all channels are supported. */
+ } else {
+ /* The device only supports a specific set of channels. We need to iterate over all of them. */
+ for (iChannel = minChannels; iChannel <= maxChannels; iChannel += 1) {
+ /* Test the channel before applying it to the configuration space. */
+ unsigned int channels = iChannel;
+
+ /* Make sure our channel range is reset before testing again or else we'll always fail the test. */
+ ((ma_snd_pcm_hw_params_any_proc)pContext->alsa.snd_pcm_hw_params_any)(pPCM, pHWParams);
+ ((ma_snd_pcm_hw_params_set_format_proc)pContext->alsa.snd_pcm_hw_params_set_format)(pPCM, pHWParams, ma_convert_ma_format_to_alsa_format(format));
+
+ if (((ma_snd_pcm_hw_params_test_channels_proc)pContext->alsa.snd_pcm_hw_params_test_channels)(pPCM, pHWParams, channels) == 0) {
+ /* The channel count is supported. */
+
+ /* The configuration space now needs to be restricted to the channel count before extracting the sample rate. */
+ ((ma_snd_pcm_hw_params_set_channels_proc)pContext->alsa.snd_pcm_hw_params_set_channels)(pPCM, pHWParams, channels);
+
+ /* Only after the configuration space has been restricted to the specific channel count should we iterate over our sample rates. */
+ ma_context_iterate_rates_and_add_native_data_format__alsa(pContext, pPCM, pHWParams, format, channels, 0, pDeviceInfo);
+ } else {
+ /* The channel count is not supported. Skip. */
+ }
+ }
+ }
+ } else {
+ /* The format is not supported. Skip. */
+ }
+ }
+
+ ma_free(pHWParams, &pContext->allocationCallbacks);
+
+ ((ma_snd_pcm_close_proc)pContext->alsa.snd_pcm_close)(pPCM);
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_uninit__alsa(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
+
+ if ((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture) {
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture);
+ close(pDevice->alsa.wakeupfdCapture);
+ ma_free(pDevice->alsa.pPollDescriptorsCapture, &pDevice->pContext->allocationCallbacks);
+ }
+
+ if ((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback) {
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback);
+ close(pDevice->alsa.wakeupfdPlayback);
+ ma_free(pDevice->alsa.pPollDescriptorsPlayback, &pDevice->pContext->allocationCallbacks);
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_init_by_type__alsa(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptor, ma_device_type deviceType)
+{
+ ma_result result;
+ int resultALSA;
+ ma_snd_pcm_t* pPCM;
+ ma_bool32 isUsingMMap;
+ ma_snd_pcm_format_t formatALSA;
+ ma_format internalFormat;
+ ma_uint32 internalChannels;
+ ma_uint32 internalSampleRate;
+ ma_channel internalChannelMap[MA_MAX_CHANNELS];
+ ma_uint32 internalPeriodSizeInFrames;
+ ma_uint32 internalPeriods;
+ int openMode;
+ ma_snd_pcm_hw_params_t* pHWParams;
+ ma_snd_pcm_sw_params_t* pSWParams;
+ ma_snd_pcm_uframes_t bufferBoundary;
+ int pollDescriptorCount;
+ struct pollfd* pPollDescriptors;
+ int wakeupfd;
+
+ MA_ASSERT(pConfig != NULL);
+ MA_ASSERT(deviceType != ma_device_type_duplex); /* This function should only be called for playback _or_ capture, never duplex. */
+ MA_ASSERT(pDevice != NULL);
+
+ formatALSA = ma_convert_ma_format_to_alsa_format(pDescriptor->format);
+
+ openMode = 0;
+ if (pConfig->alsa.noAutoResample) {
+ openMode |= MA_SND_PCM_NO_AUTO_RESAMPLE;
+ }
+ if (pConfig->alsa.noAutoChannels) {
+ openMode |= MA_SND_PCM_NO_AUTO_CHANNELS;
+ }
+ if (pConfig->alsa.noAutoFormat) {
+ openMode |= MA_SND_PCM_NO_AUTO_FORMAT;
+ }
+
+ result = ma_context_open_pcm__alsa(pDevice->pContext, pDescriptor->shareMode, deviceType, pDescriptor->pDeviceID, openMode, &pPCM);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+
+ /* Hardware parameters. */
+ pHWParams = (ma_snd_pcm_hw_params_t*)ma_calloc(((ma_snd_pcm_hw_params_sizeof_proc)pDevice->pContext->alsa.snd_pcm_hw_params_sizeof)(), &pDevice->pContext->allocationCallbacks);
+ if (pHWParams == NULL) {
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to allocate memory for hardware parameters.");
+ return MA_OUT_OF_MEMORY;
+ }
+
+ resultALSA = ((ma_snd_pcm_hw_params_any_proc)pDevice->pContext->alsa.snd_pcm_hw_params_any)(pPCM, pHWParams);
+ if (resultALSA < 0) {
+ ma_free(pHWParams, &pDevice->pContext->allocationCallbacks);
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to initialize hardware parameters. snd_pcm_hw_params_any() failed.");
+ return ma_result_from_errno(-resultALSA);
+ }
+
+ /* MMAP Mode. Try using interleaved MMAP access. If this fails, fall back to standard readi/writei. */
+ isUsingMMap = MA_FALSE;
+#if 0 /* NOTE: MMAP mode temporarily disabled. */
+ if (deviceType != ma_device_type_capture) { /* <-- Disabling MMAP mode for capture devices because I apparently do not have a device that supports it which means I can't test it... Contributions welcome. */
+ if (!pConfig->alsa.noMMap && ma_device__is_async(pDevice)) {
+ if (((ma_snd_pcm_hw_params_set_access_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_access)(pPCM, pHWParams, MA_SND_PCM_ACCESS_MMAP_INTERLEAVED) == 0) {
+ pDevice->alsa.isUsingMMap = MA_TRUE;
+ }
+ }
+ }
#endif
+
+ if (!isUsingMMap) {
+ resultALSA = ((ma_snd_pcm_hw_params_set_access_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_access)(pPCM, pHWParams, MA_SND_PCM_ACCESS_RW_INTERLEAVED);
+ if (resultALSA < 0) {
+ ma_free(pHWParams, &pDevice->pContext->allocationCallbacks);
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set access mode to neither SND_PCM_ACCESS_MMAP_INTERLEAVED nor SND_PCM_ACCESS_RW_INTERLEAVED. snd_pcm_hw_params_set_access() failed.");
+ return ma_result_from_errno(-resultALSA);
+ }
+ }
+
+ /*
+ Most important properties first. The documentation for OSS (yes, I know this is ALSA!) recommends format, channels, then sample rate. I can't
+ find any documentation for ALSA specifically, so I'm going to copy the recommendation for OSS.
+ */
+
+ /* Format. */
+ {
+ /*
+ At this point we should have a list of supported formats, so now we need to find the best one. We first check if the requested format is
+ supported, and if so, use that one. If it's not supported, we just run though a list of formats and try to find the best one.
+ */
+ if (formatALSA == MA_SND_PCM_FORMAT_UNKNOWN || ((ma_snd_pcm_hw_params_test_format_proc)pDevice->pContext->alsa.snd_pcm_hw_params_test_format)(pPCM, pHWParams, formatALSA) != 0) {
+ /* We're either requesting the native format or the specified format is not supported. */
+ size_t iFormat;
+
+ formatALSA = MA_SND_PCM_FORMAT_UNKNOWN;
+ for (iFormat = 0; iFormat < ma_countof(g_maFormatPriorities); ++iFormat) {
+ if (((ma_snd_pcm_hw_params_test_format_proc)pDevice->pContext->alsa.snd_pcm_hw_params_test_format)(pPCM, pHWParams, ma_convert_ma_format_to_alsa_format(g_maFormatPriorities[iFormat])) == 0) {
+ formatALSA = ma_convert_ma_format_to_alsa_format(g_maFormatPriorities[iFormat]);
+ break;
+ }
+ }
+
+ if (formatALSA == MA_SND_PCM_FORMAT_UNKNOWN) {
+ ma_free(pHWParams, &pDevice->pContext->allocationCallbacks);
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Format not supported. The device does not support any miniaudio formats.");
+ return MA_FORMAT_NOT_SUPPORTED;
+ }
+ }
+
+ resultALSA = ((ma_snd_pcm_hw_params_set_format_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_format)(pPCM, pHWParams, formatALSA);
+ if (resultALSA < 0) {
+ ma_free(pHWParams, &pDevice->pContext->allocationCallbacks);
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Format not supported. snd_pcm_hw_params_set_format() failed.");
+ return ma_result_from_errno(-resultALSA);
+ }
+
+ internalFormat = ma_format_from_alsa(formatALSA);
+ if (internalFormat == ma_format_unknown) {
+ ma_free(pHWParams, &pDevice->pContext->allocationCallbacks);
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] The chosen format is not supported by miniaudio.");
+ return MA_FORMAT_NOT_SUPPORTED;
+ }
+ }
+
+ /* Channels. */
+ {
+ unsigned int channels = pDescriptor->channels;
+ if (channels == 0) {
+ channels = MA_DEFAULT_CHANNELS;
+ }
+
+ resultALSA = ((ma_snd_pcm_hw_params_set_channels_near_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_channels_near)(pPCM, pHWParams, &channels);
+ if (resultALSA < 0) {
+ ma_free(pHWParams, &pDevice->pContext->allocationCallbacks);
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set channel count. snd_pcm_hw_params_set_channels_near() failed.");
+ return ma_result_from_errno(-resultALSA);
+ }
+
+ internalChannels = (ma_uint32)channels;
+ }
+
+ /* Sample Rate */
+ {
+ unsigned int sampleRate;
+
+ /*
+ It appears there's either a bug in ALSA, a bug in some drivers, or I'm doing something silly; but having resampling enabled causes
+ problems with some device configurations when used in conjunction with MMAP access mode. To fix this problem we need to disable
+ resampling.
+
+ To reproduce this problem, open the "plug:dmix" device, and set the sample rate to 44100. Internally, it looks like dmix uses a
+ sample rate of 48000. The hardware parameters will get set correctly with no errors, but it looks like the 44100 -> 48000 resampling
+ doesn't work properly - but only with MMAP access mode. You will notice skipping/crackling in the audio, and it'll run at a slightly
+ faster rate.
+
+ miniaudio has built-in support for sample rate conversion (albeit low quality at the moment), so disabling resampling should be fine
+ for us. The only problem is that it won't be taking advantage of any kind of hardware-accelerated resampling and it won't be very
+ good quality until I get a chance to improve the quality of miniaudio's software sample rate conversion.
+
+ I don't currently know if the dmix plugin is the only one with this error. Indeed, this is the only one I've been able to reproduce
+ this error with. In the future, we may want to restrict the disabling of resampling to only known bad plugins.
+ */
+ ((ma_snd_pcm_hw_params_set_rate_resample_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_rate_resample)(pPCM, pHWParams, 0);
+
+ sampleRate = pDescriptor->sampleRate;
+ if (sampleRate == 0) {
+ sampleRate = MA_DEFAULT_SAMPLE_RATE;
+ }
+
+ resultALSA = ((ma_snd_pcm_hw_params_set_rate_near_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_rate_near)(pPCM, pHWParams, &sampleRate, 0);
+ if (resultALSA < 0) {
+ ma_free(pHWParams, &pDevice->pContext->allocationCallbacks);
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Sample rate not supported. snd_pcm_hw_params_set_rate_near() failed.");
+ return ma_result_from_errno(-resultALSA);
+ }
+
+ internalSampleRate = (ma_uint32)sampleRate;
+ }
+
+ /* Periods. */
+ {
+ ma_uint32 periods = pDescriptor->periodCount;
+
+ resultALSA = ((ma_snd_pcm_hw_params_set_periods_near_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_periods_near)(pPCM, pHWParams, &periods, NULL);
+ if (resultALSA < 0) {
+ ma_free(pHWParams, &pDevice->pContext->allocationCallbacks);
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set period count. snd_pcm_hw_params_set_periods_near() failed.");
+ return ma_result_from_errno(-resultALSA);
+ }
+
+ internalPeriods = periods;
+ }
+
+ /* Buffer Size */
+ {
+ ma_snd_pcm_uframes_t actualBufferSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, internalSampleRate, pConfig->performanceProfile) * internalPeriods;
+
+ resultALSA = ((ma_snd_pcm_hw_params_set_buffer_size_near_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_buffer_size_near)(pPCM, pHWParams, &actualBufferSizeInFrames);
+ if (resultALSA < 0) {
+ ma_free(pHWParams, &pDevice->pContext->allocationCallbacks);
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set buffer size for device. snd_pcm_hw_params_set_buffer_size() failed.");
+ return ma_result_from_errno(-resultALSA);
+ }
+
+ internalPeriodSizeInFrames = actualBufferSizeInFrames / internalPeriods;
+ }
+
+ /* Apply hardware parameters. */
+ resultALSA = ((ma_snd_pcm_hw_params_proc)pDevice->pContext->alsa.snd_pcm_hw_params)(pPCM, pHWParams);
+ if (resultALSA < 0) {
+ ma_free(pHWParams, &pDevice->pContext->allocationCallbacks);
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set hardware parameters. snd_pcm_hw_params() failed.");
+ return ma_result_from_errno(-resultALSA);
+ }
+
+ ma_free(pHWParams, &pDevice->pContext->allocationCallbacks);
+ pHWParams = NULL;
+
+
+ /* Software parameters. */
+ pSWParams = (ma_snd_pcm_sw_params_t*)ma_calloc(((ma_snd_pcm_sw_params_sizeof_proc)pDevice->pContext->alsa.snd_pcm_sw_params_sizeof)(), &pDevice->pContext->allocationCallbacks);
+ if (pSWParams == NULL) {
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to allocate memory for software parameters.");
+ return MA_OUT_OF_MEMORY;
+ }
+
+ resultALSA = ((ma_snd_pcm_sw_params_current_proc)pDevice->pContext->alsa.snd_pcm_sw_params_current)(pPCM, pSWParams);
+ if (resultALSA < 0) {
+ ma_free(pSWParams, &pDevice->pContext->allocationCallbacks);
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to initialize software parameters. snd_pcm_sw_params_current() failed.");
+ return ma_result_from_errno(-resultALSA);
+ }
+
+ resultALSA = ((ma_snd_pcm_sw_params_set_avail_min_proc)pDevice->pContext->alsa.snd_pcm_sw_params_set_avail_min)(pPCM, pSWParams, ma_prev_power_of_2(internalPeriodSizeInFrames));
+ if (resultALSA < 0) {
+ ma_free(pSWParams, &pDevice->pContext->allocationCallbacks);
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_sw_params_set_avail_min() failed.");
+ return ma_result_from_errno(-resultALSA);
+ }
+
+ resultALSA = ((ma_snd_pcm_sw_params_get_boundary_proc)pDevice->pContext->alsa.snd_pcm_sw_params_get_boundary)(pSWParams, &bufferBoundary);
+ if (resultALSA < 0) {
+ bufferBoundary = internalPeriodSizeInFrames * internalPeriods;
+ }
+
+ if (deviceType == ma_device_type_playback && !isUsingMMap) { /* Only playback devices in writei/readi mode need a start threshold. */
+ /*
+ Subtle detail here with the start threshold. When in playback-only mode (no full-duplex) we can set the start threshold to
+ the size of a period. But for full-duplex we need to set it such that it is at least two periods.
+ */
+ resultALSA = ((ma_snd_pcm_sw_params_set_start_threshold_proc)pDevice->pContext->alsa.snd_pcm_sw_params_set_start_threshold)(pPCM, pSWParams, internalPeriodSizeInFrames*2);
+ if (resultALSA < 0) {
+ ma_free(pSWParams, &pDevice->pContext->allocationCallbacks);
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set start threshold for playback device. snd_pcm_sw_params_set_start_threshold() failed.");
+ return ma_result_from_errno(-resultALSA);
+ }
+
+ resultALSA = ((ma_snd_pcm_sw_params_set_stop_threshold_proc)pDevice->pContext->alsa.snd_pcm_sw_params_set_stop_threshold)(pPCM, pSWParams, bufferBoundary);
+ if (resultALSA < 0) { /* Set to boundary to loop instead of stop in the event of an xrun. */
+ ma_free(pSWParams, &pDevice->pContext->allocationCallbacks);
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set stop threshold for playback device. snd_pcm_sw_params_set_stop_threshold() failed.");
+ return ma_result_from_errno(-resultALSA);
+ }
+ }
+
+ resultALSA = ((ma_snd_pcm_sw_params_proc)pDevice->pContext->alsa.snd_pcm_sw_params)(pPCM, pSWParams);
+ if (resultALSA < 0) {
+ ma_free(pSWParams, &pDevice->pContext->allocationCallbacks);
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set software parameters. snd_pcm_sw_params() failed.");
+ return ma_result_from_errno(-resultALSA);
+ }
+
+ ma_free(pSWParams, &pDevice->pContext->allocationCallbacks);
+ pSWParams = NULL;
+
+
+ /* Grab the internal channel map. For now we're not going to bother trying to change the channel map and instead just do it ourselves. */
+ {
+ ma_snd_pcm_chmap_t* pChmap = ((ma_snd_pcm_get_chmap_proc)pDevice->pContext->alsa.snd_pcm_get_chmap)(pPCM);
+ if (pChmap != NULL) {
+ ma_uint32 iChannel;
+
+ /* There are cases where the returned channel map can have a different channel count than was returned by snd_pcm_hw_params_set_channels_near(). */
+ if (pChmap->channels >= internalChannels) {
+ /* Drop excess channels. */
+ for (iChannel = 0; iChannel < internalChannels; ++iChannel) {
+ internalChannelMap[iChannel] = ma_convert_alsa_channel_position_to_ma_channel(pChmap->pos[iChannel]);
+ }
+ } else {
+ ma_uint32 i;
+
+ /*
+ Excess channels use defaults. Do an initial fill with defaults, overwrite the first pChmap->channels, validate to ensure there are no duplicate
+ channels. If validation fails, fall back to defaults.
+ */
+ ma_bool32 isValid = MA_TRUE;
+
+ /* Fill with defaults. */
+ ma_channel_map_init_standard(ma_standard_channel_map_alsa, internalChannelMap, ma_countof(internalChannelMap), internalChannels);
+
+ /* Overwrite first pChmap->channels channels. */
+ for (iChannel = 0; iChannel < pChmap->channels; ++iChannel) {
+ internalChannelMap[iChannel] = ma_convert_alsa_channel_position_to_ma_channel(pChmap->pos[iChannel]);
+ }
+
+ /* Validate. */
+ for (i = 0; i < internalChannels && isValid; ++i) {
+ ma_uint32 j;
+ for (j = i+1; j < internalChannels; ++j) {
+ if (internalChannelMap[i] == internalChannelMap[j]) {
+ isValid = MA_FALSE;
+ break;
+ }
+ }
+ }
+
+ /* If our channel map is invalid, fall back to defaults. */
+ if (!isValid) {
+ ma_channel_map_init_standard(ma_standard_channel_map_alsa, internalChannelMap, ma_countof(internalChannelMap), internalChannels);
+ }
+ }
+
+ free(pChmap);
+ pChmap = NULL;
+ } else {
+ /* Could not retrieve the channel map. Fall back to a hard-coded assumption. */
+ ma_channel_map_init_standard(ma_standard_channel_map_alsa, internalChannelMap, ma_countof(internalChannelMap), internalChannels);
+ }
+ }
+
+
+ /*
+ We need to retrieve the poll descriptors so we can use poll() to wait for data to become
+ available for reading or writing. There's no well defined maximum for this so we're just going
+ to allocate this on the heap.
+ */
+ pollDescriptorCount = ((ma_snd_pcm_poll_descriptors_count_proc)pDevice->pContext->alsa.snd_pcm_poll_descriptors_count)(pPCM);
+ if (pollDescriptorCount <= 0) {
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to retrieve poll descriptors count.");
+ return MA_ERROR;
+ }
+
+ pPollDescriptors = (struct pollfd*)ma_malloc(sizeof(*pPollDescriptors) * (pollDescriptorCount + 1), &pDevice->pContext->allocationCallbacks); /* +1 because we want room for the wakeup descriptor. */
+ if (pPollDescriptors == NULL) {
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to allocate memory for poll descriptors.");
+ return MA_OUT_OF_MEMORY;
+ }
+
+ /*
+ We need an eventfd to wakeup from poll() and avoid a deadlock in situations where the driver
+ never returns from writei() and readi(). This has been observed with the "pulse" device.
+ */
+ wakeupfd = eventfd(0, 0);
+ if (wakeupfd < 0) {
+ ma_free(pPollDescriptors, &pDevice->pContext->allocationCallbacks);
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to create eventfd for poll wakeup.");
+ return ma_result_from_errno(errno);
+ }
+
+ /* We'll place the wakeup fd at the start of the buffer. */
+ pPollDescriptors[0].fd = wakeupfd;
+ pPollDescriptors[0].events = POLLIN; /* We only care about waiting to read from the wakeup file descriptor. */
+ pPollDescriptors[0].revents = 0;
+
+ /* We can now extract the PCM poll descriptors which we place after the wakeup descriptor. */
+ pollDescriptorCount = ((ma_snd_pcm_poll_descriptors_proc)pDevice->pContext->alsa.snd_pcm_poll_descriptors)(pPCM, pPollDescriptors + 1, pollDescriptorCount); /* +1 because we want to place these descriptors after the wakeup descriptor. */
+ if (pollDescriptorCount <= 0) {
+ close(wakeupfd);
+ ma_free(pPollDescriptors, &pDevice->pContext->allocationCallbacks);
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to retrieve poll descriptors.");
+ return MA_ERROR;
+ }
+
+ if (deviceType == ma_device_type_capture) {
+ pDevice->alsa.pollDescriptorCountCapture = pollDescriptorCount;
+ pDevice->alsa.pPollDescriptorsCapture = pPollDescriptors;
+ pDevice->alsa.wakeupfdCapture = wakeupfd;
+ } else {
+ pDevice->alsa.pollDescriptorCountPlayback = pollDescriptorCount;
+ pDevice->alsa.pPollDescriptorsPlayback = pPollDescriptors;
+ pDevice->alsa.wakeupfdPlayback = wakeupfd;
+ }
+
+
+ /* We're done. Prepare the device. */
+ resultALSA = ((ma_snd_pcm_prepare_proc)pDevice->pContext->alsa.snd_pcm_prepare)(pPCM);
+ if (resultALSA < 0) {
+ close(wakeupfd);
+ ma_free(pPollDescriptors, &pDevice->pContext->allocationCallbacks);
+ ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to prepare device.");
+ return ma_result_from_errno(-resultALSA);
+ }
+
+
+ if (deviceType == ma_device_type_capture) {
+ pDevice->alsa.pPCMCapture = (ma_ptr)pPCM;
+ pDevice->alsa.isUsingMMapCapture = isUsingMMap;
+ } else {
+ pDevice->alsa.pPCMPlayback = (ma_ptr)pPCM;
+ pDevice->alsa.isUsingMMapPlayback = isUsingMMap;
+ }
+
+ pDescriptor->format = internalFormat;
+ pDescriptor->channels = internalChannels;
+ pDescriptor->sampleRate = internalSampleRate;
+ ma_channel_map_copy(pDescriptor->channelMap, internalChannelMap, ma_min(internalChannels, MA_MAX_CHANNELS));
+ pDescriptor->periodSizeInFrames = internalPeriodSizeInFrames;
+ pDescriptor->periodCount = internalPeriods;
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_init__alsa(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+{
+ MA_ASSERT(pDevice != NULL);
+
+ MA_ZERO_OBJECT(&pDevice->alsa);
+
+ if (pConfig->deviceType == ma_device_type_loopback) {
+ return MA_DEVICE_TYPE_NOT_SUPPORTED;
+ }
+
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
+ ma_result result = ma_device_init_by_type__alsa(pDevice, pConfig, pDescriptorCapture, ma_device_type_capture);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ }
+
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ ma_result result = ma_device_init_by_type__alsa(pDevice, pConfig, pDescriptorPlayback, ma_device_type_playback);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_start__alsa(ma_device* pDevice)
+{
+ int resultALSA;
+
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ resultALSA = ((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture);
+ if (resultALSA < 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to start capture device.");
+ return ma_result_from_errno(-resultALSA);
+ }
+ }
+
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ /* Don't need to do anything for playback because it'll be started automatically when enough data has been written. */
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_stop__alsa(ma_device* pDevice)
+{
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Dropping capture device...\n");
+ ((ma_snd_pcm_drop_proc)pDevice->pContext->alsa.snd_pcm_drop)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture);
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Dropping capture device successful.\n");
+
+ /* We need to prepare the device again, otherwise we won't be able to restart the device. */
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Preparing capture device...\n");
+ if (((ma_snd_pcm_prepare_proc)pDevice->pContext->alsa.snd_pcm_prepare)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture) < 0) {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Preparing capture device failed.\n");
+ } else {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Preparing capture device successful.\n");
+ }
+ }
+
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Dropping playback device...\n");
+ ((ma_snd_pcm_drop_proc)pDevice->pContext->alsa.snd_pcm_drop)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback);
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Dropping playback device successful.\n");
+
+ /* We need to prepare the device again, otherwise we won't be able to restart the device. */
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Preparing playback device...\n");
+ if (((ma_snd_pcm_prepare_proc)pDevice->pContext->alsa.snd_pcm_prepare)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback) < 0) {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Preparing playback device failed.\n");
+ } else {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Preparing playback device successful.\n");
+ }
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_wait__alsa(ma_device* pDevice, ma_snd_pcm_t* pPCM, struct pollfd* pPollDescriptors, int pollDescriptorCount, short requiredEvent)
+{
+ for (;;) {
+ unsigned short revents;
+ int resultALSA;
+ int resultPoll = poll(pPollDescriptors, pollDescriptorCount, -1);
+ if (resultPoll < 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] poll() failed.");
+ return ma_result_from_errno(errno);
+ }
+
+ /*
+ Before checking the ALSA poll descriptor flag we need to check if the wakeup descriptor
+ has had it's POLLIN flag set. If so, we need to actually read the data and then exit
+ function. The wakeup descriptor will be the first item in the descriptors buffer.
+ */
+ if ((pPollDescriptors[0].revents & POLLIN) != 0) {
+ ma_uint64 t;
+ int resultRead = read(pPollDescriptors[0].fd, &t, sizeof(t)); /* <-- Important that we read here so that the next write() does not block. */
+ if (resultRead < 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] read() failed.");
+ return ma_result_from_errno(errno);
+ }
+
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] POLLIN set for wakeupfd\n");
+ return MA_DEVICE_NOT_STARTED;
+ }
+
+ /*
+ Getting here means that some data should be able to be read. We need to use ALSA to
+ translate the revents flags for us.
+ */
+ resultALSA = ((ma_snd_pcm_poll_descriptors_revents_proc)pDevice->pContext->alsa.snd_pcm_poll_descriptors_revents)(pPCM, pPollDescriptors + 1, pollDescriptorCount - 1, &revents); /* +1, -1 to ignore the wakeup descriptor. */
+ if (resultALSA < 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_poll_descriptors_revents() failed.");
+ return ma_result_from_errno(-resultALSA);
+ }
+
+ if ((revents & POLLERR) != 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] POLLERR detected.");
+ return ma_result_from_errno(errno);
+ }
+
+ if ((revents & requiredEvent) == requiredEvent) {
+ break; /* We're done. Data available for reading or writing. */
+ }
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_wait_read__alsa(ma_device* pDevice)
+{
+ return ma_device_wait__alsa(pDevice, (ma_snd_pcm_t*)pDevice->alsa.pPCMCapture, (struct pollfd*)pDevice->alsa.pPollDescriptorsCapture, pDevice->alsa.pollDescriptorCountCapture + 1, POLLIN); /* +1 to account for the wakeup descriptor. */
+}
+
+static ma_result ma_device_wait_write__alsa(ma_device* pDevice)
+{
+ return ma_device_wait__alsa(pDevice, (ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback, (struct pollfd*)pDevice->alsa.pPollDescriptorsPlayback, pDevice->alsa.pollDescriptorCountPlayback + 1, POLLOUT); /* +1 to account for the wakeup descriptor. */
+}
+
+static ma_result ma_device_read__alsa(ma_device* pDevice, void* pFramesOut, ma_uint32 frameCount, ma_uint32* pFramesRead)
+{
+ ma_snd_pcm_sframes_t resultALSA = 0;
+
+ MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(pFramesOut != NULL);
+
+ if (pFramesRead != NULL) {
+ *pFramesRead = 0;
+ }
+
+ while (ma_device_get_state(pDevice) == ma_device_state_started) {
+ ma_result result;
+
+ /* The first thing to do is wait for data to become available for reading. This will return an error code if the device has been stopped. */
+ result = ma_device_wait_read__alsa(pDevice);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ /* Getting here means we should have data available. */
+ resultALSA = ((ma_snd_pcm_readi_proc)pDevice->pContext->alsa.snd_pcm_readi)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture, pFramesOut, frameCount);
+ if (resultALSA >= 0) {
+ break; /* Success. */
+ } else {
+ if (resultALSA == -EAGAIN) {
+ /*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "EGAIN (read)\n");*/
+ continue; /* Try again. */
+ } else if (resultALSA == -EPIPE) {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "EPIPE (read)\n");
+
+ /* Overrun. Recover and try again. If this fails we need to return an error. */
+ resultALSA = ((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture, resultALSA, MA_TRUE);
+ if (resultALSA < 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to recover device after overrun.");
+ return ma_result_from_errno((int)-resultALSA);
+ }
+
+ resultALSA = ((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture);
+ if (resultALSA < 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to start device after underrun.");
+ return ma_result_from_errno((int)-resultALSA);
+ }
+
+ continue; /* Try reading again. */
+ }
+ }
+ }
+
+ if (pFramesRead != NULL) {
+ *pFramesRead = resultALSA;
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_write__alsa(ma_device* pDevice, const void* pFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten)
+{
+ ma_snd_pcm_sframes_t resultALSA = 0;
+
+ MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(pFrames != NULL);
+
+ if (pFramesWritten != NULL) {
+ *pFramesWritten = 0;
+ }
+
+ while (ma_device_get_state(pDevice) == ma_device_state_started) {
+ ma_result result;
+
+ /* The first thing to do is wait for space to become available for writing. This will return an error code if the device has been stopped. */
+ result = ma_device_wait_write__alsa(pDevice);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ resultALSA = ((ma_snd_pcm_writei_proc)pDevice->pContext->alsa.snd_pcm_writei)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback, pFrames, frameCount);
+ if (resultALSA >= 0) {
+ break; /* Success. */
+ } else {
+ if (resultALSA == -EAGAIN) {
+ /*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "EGAIN (write)\n");*/
+ continue; /* Try again. */
+ } else if (resultALSA == -EPIPE) {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "EPIPE (write)\n");
+
+ /* Underrun. Recover and try again. If this fails we need to return an error. */
+ resultALSA = ((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback, resultALSA, MA_TRUE); /* MA_TRUE=silent (don't print anything on error). */
+ if (resultALSA < 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to recover device after underrun.");
+ return ma_result_from_errno((int)-resultALSA);
+ }
+
+ /*
+ In my testing I have had a situation where writei() does not automatically restart the device even though I've set it
+ up as such in the software parameters. What will happen is writei() will block indefinitely even though the number of
+ frames is well beyond the auto-start threshold. To work around this I've needed to add an explicit start here. Not sure
+ if this is me just being stupid and not recovering the device properly, but this definitely feels like something isn't
+ quite right here.
+ */
+ resultALSA = ((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback);
+ if (resultALSA < 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to start device after underrun.");
+ return ma_result_from_errno((int)-resultALSA);
+ }
+
+ continue; /* Try writing again. */
+ }
+ }
+ }
+
+ if (pFramesWritten != NULL) {
+ *pFramesWritten = resultALSA;
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_data_loop_wakeup__alsa(ma_device* pDevice)
+{
+ ma_uint64 t = 1;
+ int resultWrite = 0;
+
+ MA_ASSERT(pDevice != NULL);
+
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Waking up...\n");
+
+ /* Write to an eventfd to trigger a wakeup from poll() and abort any reading or writing. */
+ if (pDevice->alsa.pPollDescriptorsCapture != NULL) {
+ resultWrite = write(pDevice->alsa.wakeupfdCapture, &t, sizeof(t));
+ }
+ if (pDevice->alsa.pPollDescriptorsPlayback != NULL) {
+ resultWrite = write(pDevice->alsa.wakeupfdPlayback, &t, sizeof(t));
+ }
+
+ if (resultWrite < 0) {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] write() failed.\n");
+ return ma_result_from_errno(errno);
+ }
+
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Waking up completed successfully.\n");
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_uninit__alsa(ma_context* pContext)
+{
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pContext->backend == ma_backend_alsa);
+
+ /* Clean up memory for memory leak checkers. */
+ ((ma_snd_config_update_free_global_proc)pContext->alsa.snd_config_update_free_global)();
+
+#ifndef MA_NO_RUNTIME_LINKING
+ ma_dlclose(pContext, pContext->alsa.asoundSO);
#endif
-#if !defined(C89ATOMIC_64BIT) && !defined(C89ATOMIC_32BIT)
-#ifdef __GNUC__
-#ifdef __LP64__
-#define C89ATOMIC_64BIT
+
+ ma_mutex_uninit(&pContext->alsa.internalDeviceEnumLock);
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_init__alsa(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+{
+ ma_result result;
+#ifndef MA_NO_RUNTIME_LINKING
+ const char* libasoundNames[] = {
+ "libasound.so.2",
+ "libasound.so"
+ };
+ size_t i;
+
+ for (i = 0; i < ma_countof(libasoundNames); ++i) {
+ pContext->alsa.asoundSO = ma_dlopen(pContext, libasoundNames[i]);
+ if (pContext->alsa.asoundSO != NULL) {
+ break;
+ }
+ }
+
+ if (pContext->alsa.asoundSO == NULL) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[ALSA] Failed to open shared object.\n");
+ return MA_NO_BACKEND;
+ }
+
+ pContext->alsa.snd_pcm_open = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_open");
+ pContext->alsa.snd_pcm_close = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_close");
+ pContext->alsa.snd_pcm_hw_params_sizeof = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_sizeof");
+ pContext->alsa.snd_pcm_hw_params_any = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_any");
+ pContext->alsa.snd_pcm_hw_params_set_format = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_format");
+ pContext->alsa.snd_pcm_hw_params_set_format_first = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_format_first");
+ pContext->alsa.snd_pcm_hw_params_get_format_mask = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_format_mask");
+ pContext->alsa.snd_pcm_hw_params_set_channels = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_channels");
+ pContext->alsa.snd_pcm_hw_params_set_channels_near = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_channels_near");
+ pContext->alsa.snd_pcm_hw_params_set_channels_minmax = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_channels_minmax");
+ pContext->alsa.snd_pcm_hw_params_set_rate_resample = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_rate_resample");
+ pContext->alsa.snd_pcm_hw_params_set_rate = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_rate");
+ pContext->alsa.snd_pcm_hw_params_set_rate_near = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_rate_near");
+ pContext->alsa.snd_pcm_hw_params_set_buffer_size_near = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_buffer_size_near");
+ pContext->alsa.snd_pcm_hw_params_set_periods_near = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_periods_near");
+ pContext->alsa.snd_pcm_hw_params_set_access = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_access");
+ pContext->alsa.snd_pcm_hw_params_get_format = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_format");
+ pContext->alsa.snd_pcm_hw_params_get_channels = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_channels");
+ pContext->alsa.snd_pcm_hw_params_get_channels_min = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_channels_min");
+ pContext->alsa.snd_pcm_hw_params_get_channels_max = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_channels_max");
+ pContext->alsa.snd_pcm_hw_params_get_rate = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_rate");
+ pContext->alsa.snd_pcm_hw_params_get_rate_min = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_rate_min");
+ pContext->alsa.snd_pcm_hw_params_get_rate_max = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_rate_max");
+ pContext->alsa.snd_pcm_hw_params_get_buffer_size = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_buffer_size");
+ pContext->alsa.snd_pcm_hw_params_get_periods = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_periods");
+ pContext->alsa.snd_pcm_hw_params_get_access = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_access");
+ pContext->alsa.snd_pcm_hw_params_test_format = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_test_format");
+ pContext->alsa.snd_pcm_hw_params_test_channels = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_test_channels");
+ pContext->alsa.snd_pcm_hw_params_test_rate = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_test_rate");
+ pContext->alsa.snd_pcm_hw_params = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params");
+ pContext->alsa.snd_pcm_sw_params_sizeof = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_sizeof");
+ pContext->alsa.snd_pcm_sw_params_current = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_current");
+ pContext->alsa.snd_pcm_sw_params_get_boundary = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_get_boundary");
+ pContext->alsa.snd_pcm_sw_params_set_avail_min = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_set_avail_min");
+ pContext->alsa.snd_pcm_sw_params_set_start_threshold = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_set_start_threshold");
+ pContext->alsa.snd_pcm_sw_params_set_stop_threshold = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_set_stop_threshold");
+ pContext->alsa.snd_pcm_sw_params = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params");
+ pContext->alsa.snd_pcm_format_mask_sizeof = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_format_mask_sizeof");
+ pContext->alsa.snd_pcm_format_mask_test = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_format_mask_test");
+ pContext->alsa.snd_pcm_get_chmap = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_get_chmap");
+ pContext->alsa.snd_pcm_state = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_state");
+ pContext->alsa.snd_pcm_prepare = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_prepare");
+ pContext->alsa.snd_pcm_start = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_start");
+ pContext->alsa.snd_pcm_drop = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_drop");
+ pContext->alsa.snd_pcm_drain = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_drain");
+ pContext->alsa.snd_pcm_reset = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_reset");
+ pContext->alsa.snd_device_name_hint = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_device_name_hint");
+ pContext->alsa.snd_device_name_get_hint = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_device_name_get_hint");
+ pContext->alsa.snd_card_get_index = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_card_get_index");
+ pContext->alsa.snd_device_name_free_hint = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_device_name_free_hint");
+ pContext->alsa.snd_pcm_mmap_begin = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_mmap_begin");
+ pContext->alsa.snd_pcm_mmap_commit = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_mmap_commit");
+ pContext->alsa.snd_pcm_recover = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_recover");
+ pContext->alsa.snd_pcm_readi = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_readi");
+ pContext->alsa.snd_pcm_writei = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_writei");
+ pContext->alsa.snd_pcm_avail = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_avail");
+ pContext->alsa.snd_pcm_avail_update = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_avail_update");
+ pContext->alsa.snd_pcm_wait = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_wait");
+ pContext->alsa.snd_pcm_nonblock = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_nonblock");
+ pContext->alsa.snd_pcm_info = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_info");
+ pContext->alsa.snd_pcm_info_sizeof = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_info_sizeof");
+ pContext->alsa.snd_pcm_info_get_name = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_info_get_name");
+ pContext->alsa.snd_pcm_poll_descriptors = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_poll_descriptors");
+ pContext->alsa.snd_pcm_poll_descriptors_count = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_poll_descriptors_count");
+ pContext->alsa.snd_pcm_poll_descriptors_revents = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_poll_descriptors_revents");
+ pContext->alsa.snd_config_update_free_global = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_config_update_free_global");
#else
-#define C89ATOMIC_32BIT
+ /* The system below is just for type safety. */
+ ma_snd_pcm_open_proc _snd_pcm_open = snd_pcm_open;
+ ma_snd_pcm_close_proc _snd_pcm_close = snd_pcm_close;
+ ma_snd_pcm_hw_params_sizeof_proc _snd_pcm_hw_params_sizeof = snd_pcm_hw_params_sizeof;
+ ma_snd_pcm_hw_params_any_proc _snd_pcm_hw_params_any = snd_pcm_hw_params_any;
+ ma_snd_pcm_hw_params_set_format_proc _snd_pcm_hw_params_set_format = snd_pcm_hw_params_set_format;
+ ma_snd_pcm_hw_params_set_format_first_proc _snd_pcm_hw_params_set_format_first = snd_pcm_hw_params_set_format_first;
+ ma_snd_pcm_hw_params_get_format_mask_proc _snd_pcm_hw_params_get_format_mask = snd_pcm_hw_params_get_format_mask;
+ ma_snd_pcm_hw_params_set_channels_proc _snd_pcm_hw_params_set_channels = snd_pcm_hw_params_set_channels;
+ ma_snd_pcm_hw_params_set_channels_near_proc _snd_pcm_hw_params_set_channels_near = snd_pcm_hw_params_set_channels_near;
+ ma_snd_pcm_hw_params_set_rate_resample_proc _snd_pcm_hw_params_set_rate_resample = snd_pcm_hw_params_set_rate_resample;
+ ma_snd_pcm_hw_params_set_rate_near _snd_pcm_hw_params_set_rate = snd_pcm_hw_params_set_rate;
+ ma_snd_pcm_hw_params_set_rate_near_proc _snd_pcm_hw_params_set_rate_near = snd_pcm_hw_params_set_rate_near;
+ ma_snd_pcm_hw_params_set_rate_minmax_proc _snd_pcm_hw_params_set_rate_minmax = snd_pcm_hw_params_set_rate_minmax;
+ ma_snd_pcm_hw_params_set_buffer_size_near_proc _snd_pcm_hw_params_set_buffer_size_near = snd_pcm_hw_params_set_buffer_size_near;
+ ma_snd_pcm_hw_params_set_periods_near_proc _snd_pcm_hw_params_set_periods_near = snd_pcm_hw_params_set_periods_near;
+ ma_snd_pcm_hw_params_set_access_proc _snd_pcm_hw_params_set_access = snd_pcm_hw_params_set_access;
+ ma_snd_pcm_hw_params_get_format_proc _snd_pcm_hw_params_get_format = snd_pcm_hw_params_get_format;
+ ma_snd_pcm_hw_params_get_channels_proc _snd_pcm_hw_params_get_channels = snd_pcm_hw_params_get_channels;
+ ma_snd_pcm_hw_params_get_channels_min_proc _snd_pcm_hw_params_get_channels_min = snd_pcm_hw_params_get_channels_min;
+ ma_snd_pcm_hw_params_get_channels_max_proc _snd_pcm_hw_params_get_channels_max = snd_pcm_hw_params_get_channels_max;
+ ma_snd_pcm_hw_params_get_rate_proc _snd_pcm_hw_params_get_rate = snd_pcm_hw_params_get_rate;
+ ma_snd_pcm_hw_params_get_rate_min_proc _snd_pcm_hw_params_get_rate_min = snd_pcm_hw_params_get_rate_min;
+ ma_snd_pcm_hw_params_get_rate_max_proc _snd_pcm_hw_params_get_rate_max = snd_pcm_hw_params_get_rate_max;
+ ma_snd_pcm_hw_params_get_buffer_size_proc _snd_pcm_hw_params_get_buffer_size = snd_pcm_hw_params_get_buffer_size;
+ ma_snd_pcm_hw_params_get_periods_proc _snd_pcm_hw_params_get_periods = snd_pcm_hw_params_get_periods;
+ ma_snd_pcm_hw_params_get_access_proc _snd_pcm_hw_params_get_access = snd_pcm_hw_params_get_access;
+ ma_snd_pcm_hw_params_test_format_proc _snd_pcm_hw_params_test_format = snd_pcm_hw_params_test_format;
+ ma_snd_pcm_hw_params_test_channels_proc _snd_pcm_hw_params_test_channels = snd_pcm_hw_params_test_channels;
+ ma_snd_pcm_hw_params_test_rate_proc _snd_pcm_hw_params_test_rate = snd_pcm_hw_params_test_rate;
+ ma_snd_pcm_hw_params_proc _snd_pcm_hw_params = snd_pcm_hw_params;
+ ma_snd_pcm_sw_params_sizeof_proc _snd_pcm_sw_params_sizeof = snd_pcm_sw_params_sizeof;
+ ma_snd_pcm_sw_params_current_proc _snd_pcm_sw_params_current = snd_pcm_sw_params_current;
+ ma_snd_pcm_sw_params_get_boundary_proc _snd_pcm_sw_params_get_boundary = snd_pcm_sw_params_get_boundary;
+ ma_snd_pcm_sw_params_set_avail_min_proc _snd_pcm_sw_params_set_avail_min = snd_pcm_sw_params_set_avail_min;
+ ma_snd_pcm_sw_params_set_start_threshold_proc _snd_pcm_sw_params_set_start_threshold = snd_pcm_sw_params_set_start_threshold;
+ ma_snd_pcm_sw_params_set_stop_threshold_proc _snd_pcm_sw_params_set_stop_threshold = snd_pcm_sw_params_set_stop_threshold;
+ ma_snd_pcm_sw_params_proc _snd_pcm_sw_params = snd_pcm_sw_params;
+ ma_snd_pcm_format_mask_sizeof_proc _snd_pcm_format_mask_sizeof = snd_pcm_format_mask_sizeof;
+ ma_snd_pcm_format_mask_test_proc _snd_pcm_format_mask_test = snd_pcm_format_mask_test;
+ ma_snd_pcm_get_chmap_proc _snd_pcm_get_chmap = snd_pcm_get_chmap;
+ ma_snd_pcm_state_proc _snd_pcm_state = snd_pcm_state;
+ ma_snd_pcm_prepare_proc _snd_pcm_prepare = snd_pcm_prepare;
+ ma_snd_pcm_start_proc _snd_pcm_start = snd_pcm_start;
+ ma_snd_pcm_drop_proc _snd_pcm_drop = snd_pcm_drop;
+ ma_snd_pcm_drain_proc _snd_pcm_drain = snd_pcm_drain;
+ ma_snd_pcm_reset_proc _snd_pcm_reset = snd_pcm_reset;
+ ma_snd_device_name_hint_proc _snd_device_name_hint = snd_device_name_hint;
+ ma_snd_device_name_get_hint_proc _snd_device_name_get_hint = snd_device_name_get_hint;
+ ma_snd_card_get_index_proc _snd_card_get_index = snd_card_get_index;
+ ma_snd_device_name_free_hint_proc _snd_device_name_free_hint = snd_device_name_free_hint;
+ ma_snd_pcm_mmap_begin_proc _snd_pcm_mmap_begin = snd_pcm_mmap_begin;
+ ma_snd_pcm_mmap_commit_proc _snd_pcm_mmap_commit = snd_pcm_mmap_commit;
+ ma_snd_pcm_recover_proc _snd_pcm_recover = snd_pcm_recover;
+ ma_snd_pcm_readi_proc _snd_pcm_readi = snd_pcm_readi;
+ ma_snd_pcm_writei_proc _snd_pcm_writei = snd_pcm_writei;
+ ma_snd_pcm_avail_proc _snd_pcm_avail = snd_pcm_avail;
+ ma_snd_pcm_avail_update_proc _snd_pcm_avail_update = snd_pcm_avail_update;
+ ma_snd_pcm_wait_proc _snd_pcm_wait = snd_pcm_wait;
+ ma_snd_pcm_nonblock_proc _snd_pcm_nonblock = snd_pcm_nonblock;
+ ma_snd_pcm_info_proc _snd_pcm_info = snd_pcm_info;
+ ma_snd_pcm_info_sizeof_proc _snd_pcm_info_sizeof = snd_pcm_info_sizeof;
+ ma_snd_pcm_info_get_name_proc _snd_pcm_info_get_name = snd_pcm_info_get_name;
+ ma_snd_pcm_poll_descriptors _snd_pcm_poll_descriptors = snd_pcm_poll_descriptors;
+ ma_snd_pcm_poll_descriptors_count _snd_pcm_poll_descriptors_count = snd_pcm_poll_descriptors_count;
+ ma_snd_pcm_poll_descriptors_revents _snd_pcm_poll_descriptors_revents = snd_pcm_poll_descriptors_revents;
+ ma_snd_config_update_free_global_proc _snd_config_update_free_global = snd_config_update_free_global;
+
+ pContext->alsa.snd_pcm_open = (ma_proc)_snd_pcm_open;
+ pContext->alsa.snd_pcm_close = (ma_proc)_snd_pcm_close;
+ pContext->alsa.snd_pcm_hw_params_sizeof = (ma_proc)_snd_pcm_hw_params_sizeof;
+ pContext->alsa.snd_pcm_hw_params_any = (ma_proc)_snd_pcm_hw_params_any;
+ pContext->alsa.snd_pcm_hw_params_set_format = (ma_proc)_snd_pcm_hw_params_set_format;
+ pContext->alsa.snd_pcm_hw_params_set_format_first = (ma_proc)_snd_pcm_hw_params_set_format_first;
+ pContext->alsa.snd_pcm_hw_params_get_format_mask = (ma_proc)_snd_pcm_hw_params_get_format_mask;
+ pContext->alsa.snd_pcm_hw_params_set_channels = (ma_proc)_snd_pcm_hw_params_set_channels;
+ pContext->alsa.snd_pcm_hw_params_set_channels_near = (ma_proc)_snd_pcm_hw_params_set_channels_near;
+ pContext->alsa.snd_pcm_hw_params_set_channels_minmax = (ma_proc)_snd_pcm_hw_params_set_channels_minmax;
+ pContext->alsa.snd_pcm_hw_params_set_rate_resample = (ma_proc)_snd_pcm_hw_params_set_rate_resample;
+ pContext->alsa.snd_pcm_hw_params_set_rate = (ma_proc)_snd_pcm_hw_params_set_rate;
+ pContext->alsa.snd_pcm_hw_params_set_rate_near = (ma_proc)_snd_pcm_hw_params_set_rate_near;
+ pContext->alsa.snd_pcm_hw_params_set_buffer_size_near = (ma_proc)_snd_pcm_hw_params_set_buffer_size_near;
+ pContext->alsa.snd_pcm_hw_params_set_periods_near = (ma_proc)_snd_pcm_hw_params_set_periods_near;
+ pContext->alsa.snd_pcm_hw_params_set_access = (ma_proc)_snd_pcm_hw_params_set_access;
+ pContext->alsa.snd_pcm_hw_params_get_format = (ma_proc)_snd_pcm_hw_params_get_format;
+ pContext->alsa.snd_pcm_hw_params_get_channels = (ma_proc)_snd_pcm_hw_params_get_channels;
+ pContext->alsa.snd_pcm_hw_params_get_channels_min = (ma_proc)_snd_pcm_hw_params_get_channels_min;
+ pContext->alsa.snd_pcm_hw_params_get_channels_max = (ma_proc)_snd_pcm_hw_params_get_channels_max;
+ pContext->alsa.snd_pcm_hw_params_get_rate = (ma_proc)_snd_pcm_hw_params_get_rate;
+ pContext->alsa.snd_pcm_hw_params_get_rate_min = (ma_proc)_snd_pcm_hw_params_get_rate_min;
+ pContext->alsa.snd_pcm_hw_params_get_rate_max = (ma_proc)_snd_pcm_hw_params_get_rate_max;
+ pContext->alsa.snd_pcm_hw_params_get_buffer_size = (ma_proc)_snd_pcm_hw_params_get_buffer_size;
+ pContext->alsa.snd_pcm_hw_params_get_periods = (ma_proc)_snd_pcm_hw_params_get_periods;
+ pContext->alsa.snd_pcm_hw_params_get_access = (ma_proc)_snd_pcm_hw_params_get_access;
+ pContext->alsa.snd_pcm_hw_params_test_format = (ma_proc)_snd_pcm_hw_params_test_format;
+ pContext->alsa.snd_pcm_hw_params_test_channels = (ma_proc)_snd_pcm_hw_params_test_channels;
+ pContext->alsa.snd_pcm_hw_params_test_rate = (ma_proc)_snd_pcm_hw_params_test_rate;
+ pContext->alsa.snd_pcm_hw_params = (ma_proc)_snd_pcm_hw_params;
+ pContext->alsa.snd_pcm_sw_params_sizeof = (ma_proc)_snd_pcm_sw_params_sizeof;
+ pContext->alsa.snd_pcm_sw_params_current = (ma_proc)_snd_pcm_sw_params_current;
+ pContext->alsa.snd_pcm_sw_params_get_boundary = (ma_proc)_snd_pcm_sw_params_get_boundary;
+ pContext->alsa.snd_pcm_sw_params_set_avail_min = (ma_proc)_snd_pcm_sw_params_set_avail_min;
+ pContext->alsa.snd_pcm_sw_params_set_start_threshold = (ma_proc)_snd_pcm_sw_params_set_start_threshold;
+ pContext->alsa.snd_pcm_sw_params_set_stop_threshold = (ma_proc)_snd_pcm_sw_params_set_stop_threshold;
+ pContext->alsa.snd_pcm_sw_params = (ma_proc)_snd_pcm_sw_params;
+ pContext->alsa.snd_pcm_format_mask_sizeof = (ma_proc)_snd_pcm_format_mask_sizeof;
+ pContext->alsa.snd_pcm_format_mask_test = (ma_proc)_snd_pcm_format_mask_test;
+ pContext->alsa.snd_pcm_get_chmap = (ma_proc)_snd_pcm_get_chmap;
+ pContext->alsa.snd_pcm_state = (ma_proc)_snd_pcm_state;
+ pContext->alsa.snd_pcm_prepare = (ma_proc)_snd_pcm_prepare;
+ pContext->alsa.snd_pcm_start = (ma_proc)_snd_pcm_start;
+ pContext->alsa.snd_pcm_drop = (ma_proc)_snd_pcm_drop;
+ pContext->alsa.snd_pcm_drain = (ma_proc)_snd_pcm_drain;
+ pContext->alsa.snd_pcm_reset = (ma_proc)_snd_pcm_reset;
+ pContext->alsa.snd_device_name_hint = (ma_proc)_snd_device_name_hint;
+ pContext->alsa.snd_device_name_get_hint = (ma_proc)_snd_device_name_get_hint;
+ pContext->alsa.snd_card_get_index = (ma_proc)_snd_card_get_index;
+ pContext->alsa.snd_device_name_free_hint = (ma_proc)_snd_device_name_free_hint;
+ pContext->alsa.snd_pcm_mmap_begin = (ma_proc)_snd_pcm_mmap_begin;
+ pContext->alsa.snd_pcm_mmap_commit = (ma_proc)_snd_pcm_mmap_commit;
+ pContext->alsa.snd_pcm_recover = (ma_proc)_snd_pcm_recover;
+ pContext->alsa.snd_pcm_readi = (ma_proc)_snd_pcm_readi;
+ pContext->alsa.snd_pcm_writei = (ma_proc)_snd_pcm_writei;
+ pContext->alsa.snd_pcm_avail = (ma_proc)_snd_pcm_avail;
+ pContext->alsa.snd_pcm_avail_update = (ma_proc)_snd_pcm_avail_update;
+ pContext->alsa.snd_pcm_wait = (ma_proc)_snd_pcm_wait;
+ pContext->alsa.snd_pcm_nonblock = (ma_proc)_snd_pcm_nonblock;
+ pContext->alsa.snd_pcm_info = (ma_proc)_snd_pcm_info;
+ pContext->alsa.snd_pcm_info_sizeof = (ma_proc)_snd_pcm_info_sizeof;
+ pContext->alsa.snd_pcm_info_get_name = (ma_proc)_snd_pcm_info_get_name;
+ pContext->alsa.snd_pcm_poll_descriptors = (ma_proc)_snd_pcm_poll_descriptors;
+ pContext->alsa.snd_pcm_poll_descriptors_count = (ma_proc)_snd_pcm_poll_descriptors_count;
+ pContext->alsa.snd_pcm_poll_descriptors_revents = (ma_proc)_snd_pcm_poll_descriptors_revents;
+ pContext->alsa.snd_config_update_free_global = (ma_proc)_snd_config_update_free_global;
#endif
+
+ pContext->alsa.useVerboseDeviceEnumeration = pConfig->alsa.useVerboseDeviceEnumeration;
+
+ result = ma_mutex_init(&pContext->alsa.internalDeviceEnumLock);
+ if (result != MA_SUCCESS) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[ALSA] WARNING: Failed to initialize mutex for internal device enumeration.");
+ return result;
+ }
+
+ pCallbacks->onContextInit = ma_context_init__alsa;
+ pCallbacks->onContextUninit = ma_context_uninit__alsa;
+ pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__alsa;
+ pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__alsa;
+ pCallbacks->onDeviceInit = ma_device_init__alsa;
+ pCallbacks->onDeviceUninit = ma_device_uninit__alsa;
+ pCallbacks->onDeviceStart = ma_device_start__alsa;
+ pCallbacks->onDeviceStop = ma_device_stop__alsa;
+ pCallbacks->onDeviceRead = ma_device_read__alsa;
+ pCallbacks->onDeviceWrite = ma_device_write__alsa;
+ pCallbacks->onDeviceDataLoop = NULL;
+ pCallbacks->onDeviceDataLoopWakeup = ma_device_data_loop_wakeup__alsa;
+
+ return MA_SUCCESS;
+}
+#endif /* ALSA */
+
+
+
+/******************************************************************************
+
+PulseAudio Backend
+
+******************************************************************************/
+#ifdef MA_HAS_PULSEAUDIO
+/*
+The PulseAudio API, along with Apple's Core Audio, is the worst of the maintream audio APIs. This is a brief description of what's going on
+in the PulseAudio backend. I apologize if this gets a bit ranty for your liking - you might want to skip this discussion.
+
+PulseAudio has something they call the "Simple API", which unfortunately isn't suitable for miniaudio. I've not seen anywhere where it
+allows you to enumerate over devices, nor does it seem to support the ability to stop and start streams. Looking at the documentation, it
+appears as though the stream is constantly running and you prevent sound from being emitted or captured by simply not calling the read or
+write functions. This is not a professional solution as it would be much better to *actually* stop the underlying stream. Perhaps the
+simple API has some smarts to do this automatically, but I'm not sure. Another limitation with the simple API is that it seems inefficient
+when you want to have multiple streams to a single context. For these reasons, miniaudio is not using the simple API.
+
+Since we're not using the simple API, we're left with the asynchronous API as our only other option. And boy, is this where it starts to
+get fun, and I don't mean that in a good way...
+
+The problems start with the very name of the API - "asynchronous". Yes, this is an asynchronous oriented API which means your commands
+don't immediately take effect. You instead need to issue your commands, and then wait for them to complete. The waiting mechanism is
+enabled through the use of a "main loop". In the asychronous API you cannot get away from the main loop, and the main loop is where almost
+all of PulseAudio's problems stem from.
+
+When you first initialize PulseAudio you need an object referred to as "main loop". You can implement this yourself by defining your own
+vtable, but it's much easier to just use one of the built-in main loop implementations. There's two generic implementations called
+pa_mainloop and pa_threaded_mainloop, and another implementation specific to GLib called pa_glib_mainloop. We're using pa_threaded_mainloop
+because it simplifies management of the worker thread. The idea of the main loop object is pretty self explanatory - you're supposed to use
+it to implement a worker thread which runs in a loop. The main loop is where operations are actually executed.
+
+To initialize the main loop, you just use `pa_threaded_mainloop_new()`. This is the first function you'll call. You can then get a pointer
+to the vtable with `pa_threaded_mainloop_get_api()` (the main loop vtable is called `pa_mainloop_api`). Again, you can bypass the threaded
+main loop object entirely and just implement `pa_mainloop_api` directly, but there's no need for it unless you're doing something extremely
+specialized such as if you want to integrate it into your application's existing main loop infrastructure.
+
+(EDIT 2021-01-26: miniaudio is no longer using `pa_threaded_mainloop` due to this issue: https://github.com/mackron/miniaudio/issues/262.
+It is now using `pa_mainloop` which turns out to be a simpler solution anyway. The rest of this rant still applies, however.)
+
+Once you have your main loop vtable (the `pa_mainloop_api` object) you can create the PulseAudio context. This is very similar to
+miniaudio's context and they map to each other quite well. You have one context to many streams, which is basically the same as miniaudio's
+one `ma_context` to many `ma_device`s. Here's where it starts to get annoying, however. When you first create the PulseAudio context, which
+is done with `pa_context_new()`, it's not actually connected to anything. When you connect, you call `pa_context_connect()`. However, if
+you remember, PulseAudio is an asynchronous API. That means you cannot just assume the context is connected after `pa_context_context()`
+has returned. You instead need to wait for it to connect. To do this, you need to either wait for a callback to get fired, which you can
+set with `pa_context_set_state_callback()`, or you can continuously poll the context's state. Either way, you need to run this in a loop.
+All objects from here out are created from the context, and, I believe, you can't be creating these objects until the context is connected.
+This waiting loop is therefore unavoidable. In order for the waiting to ever complete, however, the main loop needs to be running. Before
+attempting to connect the context, the main loop needs to be started with `pa_threaded_mainloop_start()`.
+
+The reason for this asynchronous design is to support cases where you're connecting to a remote server, say through a local network or an
+internet connection. However, the *VAST* majority of cases don't involve this at all - they just connect to a local "server" running on the
+host machine. The fact that this would be the default rather than making `pa_context_connect()` synchronous tends to boggle the mind.
+
+Once the context has been created and connected you can start creating a stream. A PulseAudio stream is analogous to miniaudio's device.
+The initialization of a stream is fairly standard - you configure some attributes (analogous to miniaudio's device config) and then call
+`pa_stream_new()` to actually create it. Here is where we start to get into "operations". When configuring the stream, you can get
+information about the source (such as sample format, sample rate, etc.), however it's not synchronous. Instead, a `pa_operation` object
+is returned from `pa_context_get_source_info_by_name()` (capture) or `pa_context_get_sink_info_by_name()` (playback). Then, you need to
+run a loop (again!) to wait for the operation to complete which you can determine via a callback or polling, just like we did with the
+context. Then, as an added bonus, you need to decrement the reference counter of the `pa_operation` object to ensure memory is cleaned up.
+All of that just to retrieve basic information about a device!
+
+Once the basic information about the device has been retrieved, miniaudio can now create the stream with `ma_stream_new()`. Like the
+context, this needs to be connected. But we need to be careful here, because we're now about to introduce one of the most horrific design
+choices in PulseAudio.
+
+PulseAudio allows you to specify a callback that is fired when data can be written to or read from a stream. The language is important here
+because PulseAudio takes it literally, specifically the "can be". You would think these callbacks would be appropriate as the place for
+writing and reading data to and from the stream, and that would be right, except when it's not. When you initialize the stream, you can
+set a flag that tells PulseAudio to not start the stream automatically. This is required because miniaudio does not auto-start devices
+straight after initialization - you need to call `ma_device_start()` manually. The problem is that even when this flag is specified,
+PulseAudio will immediately fire it's write or read callback. This is *technically* correct (based on the wording in the documentation)
+because indeed, data *can* be written at this point. The problem is that it's not *practical*. It makes sense that the write/read callback
+would be where a program will want to write or read data to or from the stream, but when it's called before the application has even
+requested that the stream be started, it's just not practical because the program probably isn't ready for any kind of data delivery at
+that point (it may still need to load files or whatnot). Instead, this callback should only be fired when the application requests the
+stream be started which is how it works with literally *every* other callback-based audio API. Since miniaudio forbids firing of the data
+callback until the device has been started (as it should be with *all* callback based APIs), logic needs to be added to ensure miniaudio
+doesn't just blindly fire the application-defined data callback from within the PulseAudio callback before the stream has actually been
+started. The device state is used for this - if the state is anything other than `ma_device_state_starting` or `ma_device_state_started`, the main data
+callback is not fired.
+
+This, unfortunately, is not the end of the problems with the PulseAudio write callback. Any normal callback based audio API will
+continuously fire the callback at regular intervals based on the size of the internal buffer. This will only ever be fired when the device
+is running, and will be fired regardless of whether or not the user actually wrote anything to the device/stream. This not the case in
+PulseAudio. In PulseAudio, the data callback will *only* be called if you wrote something to it previously. That means, if you don't call
+`pa_stream_write()`, the callback will not get fired. On the surface you wouldn't think this would matter because you should be always
+writing data, and if you don't have anything to write, just write silence. That's fine until you want to drain the stream. You see, if
+you're continuously writing data to the stream, the stream will never get drained! That means in order to drain the stream, you need to
+*not* write data to it! But remember, when you don't write data to the stream, the callback won't get fired again! Why is draining
+important? Because that's how we've defined stopping to work in miniaudio. In miniaudio, stopping the device requires it to be drained
+before returning from ma_device_stop(). So we've stopped the device, which requires us to drain, but draining requires us to *not* write
+data to the stream (or else it won't ever complete draining), but not writing to the stream means the callback won't get fired again!
+
+This becomes a problem when stopping and then restarting the device. When the device is stopped, it's drained, which requires us to *not*
+write anything to the stream. But then, since we didn't write anything to it, the write callback will *never* get called again if we just
+resume the stream naively. This means that starting the stream requires us to write data to the stream from outside the callback. This
+disconnect is something PulseAudio has got seriously wrong - there should only ever be a single source of data delivery, that being the
+callback. (I have tried using `pa_stream_flush()` to trigger the write callback to fire, but this just doesn't work for some reason.)
+
+Once you've created the stream, you need to connect it which involves the whole waiting procedure. This is the same process as the context,
+only this time you'll poll for the state with `pa_stream_get_status()`. The starting and stopping of a streaming is referred to as
+"corking" in PulseAudio. The analogy is corking a barrel. To start the stream, you uncork it, to stop it you cork it. Personally I think
+it's silly - why would you not just call it "starting" and "stopping" like any other normal audio API? Anyway, the act of corking is, you
+guessed it, asynchronous. This means you'll need our waiting loop as usual. Again, why this asynchronous design is the default is
+absolutely beyond me. Would it really be that hard to just make it run synchronously?
+
+Teardown is pretty simple (what?!). It's just a matter of calling the relevant `_unref()` function on each object in reverse order that
+they were initialized in.
+
+That's about it from the PulseAudio side. A bit ranty, I know, but they really need to fix that main loop and callback system. They're
+embarrassingly unpractical. The main loop thing is an easy fix - have synchronous versions of all APIs. If an application wants these to
+run asynchronously, they can execute them in a separate thread themselves. The desire to run these asynchronously is such a niche
+requirement - it makes no sense to make it the default. The stream write callback needs to be change, or an alternative provided, that is
+constantly fired, regardless of whether or not `pa_stream_write()` has been called, and it needs to take a pointer to a buffer as a
+parameter which the program just writes to directly rather than having to call `pa_stream_writable_size()` and `pa_stream_write()`. These
+changes alone will change PulseAudio from one of the worst audio APIs to one of the best.
+*/
+
+
+/*
+It is assumed pulseaudio.h is available when linking at compile time. When linking at compile time, we use the declarations in the header
+to check for type safety. We cannot do this when linking at run time because the header might not be available.
+*/
+#ifdef MA_NO_RUNTIME_LINKING
+
+/* pulseaudio.h marks some functions with "inline" which isn't always supported. Need to emulate it. */
+#if !defined(__cplusplus)
+ #if defined(__STRICT_ANSI__)
+ #if !defined(inline)
+ #define inline __inline__ __attribute__((always_inline))
+ #define MA_INLINE_DEFINED
+ #endif
+ #endif
#endif
+#include
+#if defined(MA_INLINE_DEFINED)
+ #undef inline
+ #undef MA_INLINE_DEFINED
#endif
-#if !defined(C89ATOMIC_64BIT) && !defined(C89ATOMIC_32BIT)
-#include
-#if INTPTR_MAX == INT64_MAX
-#define C89ATOMIC_64BIT
+
+#define MA_PA_OK PA_OK
+#define MA_PA_ERR_ACCESS PA_ERR_ACCESS
+#define MA_PA_ERR_INVALID PA_ERR_INVALID
+#define MA_PA_ERR_NOENTITY PA_ERR_NOENTITY
+#define MA_PA_ERR_NOTSUPPORTED PA_ERR_NOTSUPPORTED
+
+#define MA_PA_CHANNELS_MAX PA_CHANNELS_MAX
+#define MA_PA_RATE_MAX PA_RATE_MAX
+
+typedef pa_context_flags_t ma_pa_context_flags_t;
+#define MA_PA_CONTEXT_NOFLAGS PA_CONTEXT_NOFLAGS
+#define MA_PA_CONTEXT_NOAUTOSPAWN PA_CONTEXT_NOAUTOSPAWN
+#define MA_PA_CONTEXT_NOFAIL PA_CONTEXT_NOFAIL
+
+typedef pa_stream_flags_t ma_pa_stream_flags_t;
+#define MA_PA_STREAM_NOFLAGS PA_STREAM_NOFLAGS
+#define MA_PA_STREAM_START_CORKED PA_STREAM_START_CORKED
+#define MA_PA_STREAM_INTERPOLATE_TIMING PA_STREAM_INTERPOLATE_TIMING
+#define MA_PA_STREAM_NOT_MONOTONIC PA_STREAM_NOT_MONOTONIC
+#define MA_PA_STREAM_AUTO_TIMING_UPDATE PA_STREAM_AUTO_TIMING_UPDATE
+#define MA_PA_STREAM_NO_REMAP_CHANNELS PA_STREAM_NO_REMAP_CHANNELS
+#define MA_PA_STREAM_NO_REMIX_CHANNELS PA_STREAM_NO_REMIX_CHANNELS
+#define MA_PA_STREAM_FIX_FORMAT PA_STREAM_FIX_FORMAT
+#define MA_PA_STREAM_FIX_RATE PA_STREAM_FIX_RATE
+#define MA_PA_STREAM_FIX_CHANNELS PA_STREAM_FIX_CHANNELS
+#define MA_PA_STREAM_DONT_MOVE PA_STREAM_DONT_MOVE
+#define MA_PA_STREAM_VARIABLE_RATE PA_STREAM_VARIABLE_RATE
+#define MA_PA_STREAM_PEAK_DETECT PA_STREAM_PEAK_DETECT
+#define MA_PA_STREAM_START_MUTED PA_STREAM_START_MUTED
+#define MA_PA_STREAM_ADJUST_LATENCY PA_STREAM_ADJUST_LATENCY
+#define MA_PA_STREAM_EARLY_REQUESTS PA_STREAM_EARLY_REQUESTS
+#define MA_PA_STREAM_DONT_INHIBIT_AUTO_SUSPEND PA_STREAM_DONT_INHIBIT_AUTO_SUSPEND
+#define MA_PA_STREAM_START_UNMUTED PA_STREAM_START_UNMUTED
+#define MA_PA_STREAM_FAIL_ON_SUSPEND PA_STREAM_FAIL_ON_SUSPEND
+#define MA_PA_STREAM_RELATIVE_VOLUME PA_STREAM_RELATIVE_VOLUME
+#define MA_PA_STREAM_PASSTHROUGH PA_STREAM_PASSTHROUGH
+
+typedef pa_sink_flags_t ma_pa_sink_flags_t;
+#define MA_PA_SINK_NOFLAGS PA_SINK_NOFLAGS
+#define MA_PA_SINK_HW_VOLUME_CTRL PA_SINK_HW_VOLUME_CTRL
+#define MA_PA_SINK_LATENCY PA_SINK_LATENCY
+#define MA_PA_SINK_HARDWARE PA_SINK_HARDWARE
+#define MA_PA_SINK_NETWORK PA_SINK_NETWORK
+#define MA_PA_SINK_HW_MUTE_CTRL PA_SINK_HW_MUTE_CTRL
+#define MA_PA_SINK_DECIBEL_VOLUME PA_SINK_DECIBEL_VOLUME
+#define MA_PA_SINK_FLAT_VOLUME PA_SINK_FLAT_VOLUME
+#define MA_PA_SINK_DYNAMIC_LATENCY PA_SINK_DYNAMIC_LATENCY
+#define MA_PA_SINK_SET_FORMATS PA_SINK_SET_FORMATS
+
+typedef pa_source_flags_t ma_pa_source_flags_t;
+#define MA_PA_SOURCE_NOFLAGS PA_SOURCE_NOFLAGS
+#define MA_PA_SOURCE_HW_VOLUME_CTRL PA_SOURCE_HW_VOLUME_CTRL
+#define MA_PA_SOURCE_LATENCY PA_SOURCE_LATENCY
+#define MA_PA_SOURCE_HARDWARE PA_SOURCE_HARDWARE
+#define MA_PA_SOURCE_NETWORK PA_SOURCE_NETWORK
+#define MA_PA_SOURCE_HW_MUTE_CTRL PA_SOURCE_HW_MUTE_CTRL
+#define MA_PA_SOURCE_DECIBEL_VOLUME PA_SOURCE_DECIBEL_VOLUME
+#define MA_PA_SOURCE_DYNAMIC_LATENCY PA_SOURCE_DYNAMIC_LATENCY
+#define MA_PA_SOURCE_FLAT_VOLUME PA_SOURCE_FLAT_VOLUME
+
+typedef pa_context_state_t ma_pa_context_state_t;
+#define MA_PA_CONTEXT_UNCONNECTED PA_CONTEXT_UNCONNECTED
+#define MA_PA_CONTEXT_CONNECTING PA_CONTEXT_CONNECTING
+#define MA_PA_CONTEXT_AUTHORIZING PA_CONTEXT_AUTHORIZING
+#define MA_PA_CONTEXT_SETTING_NAME PA_CONTEXT_SETTING_NAME
+#define MA_PA_CONTEXT_READY PA_CONTEXT_READY
+#define MA_PA_CONTEXT_FAILED PA_CONTEXT_FAILED
+#define MA_PA_CONTEXT_TERMINATED PA_CONTEXT_TERMINATED
+
+typedef pa_stream_state_t ma_pa_stream_state_t;
+#define MA_PA_STREAM_UNCONNECTED PA_STREAM_UNCONNECTED
+#define MA_PA_STREAM_CREATING PA_STREAM_CREATING
+#define MA_PA_STREAM_READY PA_STREAM_READY
+#define MA_PA_STREAM_FAILED PA_STREAM_FAILED
+#define MA_PA_STREAM_TERMINATED PA_STREAM_TERMINATED
+
+typedef pa_operation_state_t ma_pa_operation_state_t;
+#define MA_PA_OPERATION_RUNNING PA_OPERATION_RUNNING
+#define MA_PA_OPERATION_DONE PA_OPERATION_DONE
+#define MA_PA_OPERATION_CANCELLED PA_OPERATION_CANCELLED
+
+typedef pa_sink_state_t ma_pa_sink_state_t;
+#define MA_PA_SINK_INVALID_STATE PA_SINK_INVALID_STATE
+#define MA_PA_SINK_RUNNING PA_SINK_RUNNING
+#define MA_PA_SINK_IDLE PA_SINK_IDLE
+#define MA_PA_SINK_SUSPENDED PA_SINK_SUSPENDED
+
+typedef pa_source_state_t ma_pa_source_state_t;
+#define MA_PA_SOURCE_INVALID_STATE PA_SOURCE_INVALID_STATE
+#define MA_PA_SOURCE_RUNNING PA_SOURCE_RUNNING
+#define MA_PA_SOURCE_IDLE PA_SOURCE_IDLE
+#define MA_PA_SOURCE_SUSPENDED PA_SOURCE_SUSPENDED
+
+typedef pa_seek_mode_t ma_pa_seek_mode_t;
+#define MA_PA_SEEK_RELATIVE PA_SEEK_RELATIVE
+#define MA_PA_SEEK_ABSOLUTE PA_SEEK_ABSOLUTE
+#define MA_PA_SEEK_RELATIVE_ON_READ PA_SEEK_RELATIVE_ON_READ
+#define MA_PA_SEEK_RELATIVE_END PA_SEEK_RELATIVE_END
+
+typedef pa_channel_position_t ma_pa_channel_position_t;
+#define MA_PA_CHANNEL_POSITION_INVALID PA_CHANNEL_POSITION_INVALID
+#define MA_PA_CHANNEL_POSITION_MONO PA_CHANNEL_POSITION_MONO
+#define MA_PA_CHANNEL_POSITION_FRONT_LEFT PA_CHANNEL_POSITION_FRONT_LEFT
+#define MA_PA_CHANNEL_POSITION_FRONT_RIGHT PA_CHANNEL_POSITION_FRONT_RIGHT
+#define MA_PA_CHANNEL_POSITION_FRONT_CENTER PA_CHANNEL_POSITION_FRONT_CENTER
+#define MA_PA_CHANNEL_POSITION_REAR_CENTER PA_CHANNEL_POSITION_REAR_CENTER
+#define MA_PA_CHANNEL_POSITION_REAR_LEFT PA_CHANNEL_POSITION_REAR_LEFT
+#define MA_PA_CHANNEL_POSITION_REAR_RIGHT PA_CHANNEL_POSITION_REAR_RIGHT
+#define MA_PA_CHANNEL_POSITION_LFE PA_CHANNEL_POSITION_LFE
+#define MA_PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER
+#define MA_PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER
+#define MA_PA_CHANNEL_POSITION_SIDE_LEFT PA_CHANNEL_POSITION_SIDE_LEFT
+#define MA_PA_CHANNEL_POSITION_SIDE_RIGHT PA_CHANNEL_POSITION_SIDE_RIGHT
+#define MA_PA_CHANNEL_POSITION_AUX0 PA_CHANNEL_POSITION_AUX0
+#define MA_PA_CHANNEL_POSITION_AUX1 PA_CHANNEL_POSITION_AUX1
+#define MA_PA_CHANNEL_POSITION_AUX2 PA_CHANNEL_POSITION_AUX2
+#define MA_PA_CHANNEL_POSITION_AUX3 PA_CHANNEL_POSITION_AUX3
+#define MA_PA_CHANNEL_POSITION_AUX4 PA_CHANNEL_POSITION_AUX4
+#define MA_PA_CHANNEL_POSITION_AUX5 PA_CHANNEL_POSITION_AUX5
+#define MA_PA_CHANNEL_POSITION_AUX6 PA_CHANNEL_POSITION_AUX6
+#define MA_PA_CHANNEL_POSITION_AUX7 PA_CHANNEL_POSITION_AUX7
+#define MA_PA_CHANNEL_POSITION_AUX8 PA_CHANNEL_POSITION_AUX8
+#define MA_PA_CHANNEL_POSITION_AUX9 PA_CHANNEL_POSITION_AUX9
+#define MA_PA_CHANNEL_POSITION_AUX10 PA_CHANNEL_POSITION_AUX10
+#define MA_PA_CHANNEL_POSITION_AUX11 PA_CHANNEL_POSITION_AUX11
+#define MA_PA_CHANNEL_POSITION_AUX12 PA_CHANNEL_POSITION_AUX12
+#define MA_PA_CHANNEL_POSITION_AUX13 PA_CHANNEL_POSITION_AUX13
+#define MA_PA_CHANNEL_POSITION_AUX14 PA_CHANNEL_POSITION_AUX14
+#define MA_PA_CHANNEL_POSITION_AUX15 PA_CHANNEL_POSITION_AUX15
+#define MA_PA_CHANNEL_POSITION_AUX16 PA_CHANNEL_POSITION_AUX16
+#define MA_PA_CHANNEL_POSITION_AUX17 PA_CHANNEL_POSITION_AUX17
+#define MA_PA_CHANNEL_POSITION_AUX18 PA_CHANNEL_POSITION_AUX18
+#define MA_PA_CHANNEL_POSITION_AUX19 PA_CHANNEL_POSITION_AUX19
+#define MA_PA_CHANNEL_POSITION_AUX20 PA_CHANNEL_POSITION_AUX20
+#define MA_PA_CHANNEL_POSITION_AUX21 PA_CHANNEL_POSITION_AUX21
+#define MA_PA_CHANNEL_POSITION_AUX22 PA_CHANNEL_POSITION_AUX22
+#define MA_PA_CHANNEL_POSITION_AUX23 PA_CHANNEL_POSITION_AUX23
+#define MA_PA_CHANNEL_POSITION_AUX24 PA_CHANNEL_POSITION_AUX24
+#define MA_PA_CHANNEL_POSITION_AUX25 PA_CHANNEL_POSITION_AUX25
+#define MA_PA_CHANNEL_POSITION_AUX26 PA_CHANNEL_POSITION_AUX26
+#define MA_PA_CHANNEL_POSITION_AUX27 PA_CHANNEL_POSITION_AUX27
+#define MA_PA_CHANNEL_POSITION_AUX28 PA_CHANNEL_POSITION_AUX28
+#define MA_PA_CHANNEL_POSITION_AUX29 PA_CHANNEL_POSITION_AUX29
+#define MA_PA_CHANNEL_POSITION_AUX30 PA_CHANNEL_POSITION_AUX30
+#define MA_PA_CHANNEL_POSITION_AUX31 PA_CHANNEL_POSITION_AUX31
+#define MA_PA_CHANNEL_POSITION_TOP_CENTER PA_CHANNEL_POSITION_TOP_CENTER
+#define MA_PA_CHANNEL_POSITION_TOP_FRONT_LEFT PA_CHANNEL_POSITION_TOP_FRONT_LEFT
+#define MA_PA_CHANNEL_POSITION_TOP_FRONT_RIGHT PA_CHANNEL_POSITION_TOP_FRONT_RIGHT
+#define MA_PA_CHANNEL_POSITION_TOP_FRONT_CENTER PA_CHANNEL_POSITION_TOP_FRONT_CENTER
+#define MA_PA_CHANNEL_POSITION_TOP_REAR_LEFT PA_CHANNEL_POSITION_TOP_REAR_LEFT
+#define MA_PA_CHANNEL_POSITION_TOP_REAR_RIGHT PA_CHANNEL_POSITION_TOP_REAR_RIGHT
+#define MA_PA_CHANNEL_POSITION_TOP_REAR_CENTER PA_CHANNEL_POSITION_TOP_REAR_CENTER
+#define MA_PA_CHANNEL_POSITION_LEFT PA_CHANNEL_POSITION_LEFT
+#define MA_PA_CHANNEL_POSITION_RIGHT PA_CHANNEL_POSITION_RIGHT
+#define MA_PA_CHANNEL_POSITION_CENTER PA_CHANNEL_POSITION_CENTER
+#define MA_PA_CHANNEL_POSITION_SUBWOOFER PA_CHANNEL_POSITION_SUBWOOFER
+
+typedef pa_channel_map_def_t ma_pa_channel_map_def_t;
+#define MA_PA_CHANNEL_MAP_AIFF PA_CHANNEL_MAP_AIFF
+#define MA_PA_CHANNEL_MAP_ALSA PA_CHANNEL_MAP_ALSA
+#define MA_PA_CHANNEL_MAP_AUX PA_CHANNEL_MAP_AUX
+#define MA_PA_CHANNEL_MAP_WAVEEX PA_CHANNEL_MAP_WAVEEX
+#define MA_PA_CHANNEL_MAP_OSS PA_CHANNEL_MAP_OSS
+#define MA_PA_CHANNEL_MAP_DEFAULT PA_CHANNEL_MAP_DEFAULT
+
+typedef pa_sample_format_t ma_pa_sample_format_t;
+#define MA_PA_SAMPLE_INVALID PA_SAMPLE_INVALID
+#define MA_PA_SAMPLE_U8 PA_SAMPLE_U8
+#define MA_PA_SAMPLE_ALAW PA_SAMPLE_ALAW
+#define MA_PA_SAMPLE_ULAW PA_SAMPLE_ULAW
+#define MA_PA_SAMPLE_S16LE PA_SAMPLE_S16LE
+#define MA_PA_SAMPLE_S16BE PA_SAMPLE_S16BE
+#define MA_PA_SAMPLE_FLOAT32LE PA_SAMPLE_FLOAT32LE
+#define MA_PA_SAMPLE_FLOAT32BE PA_SAMPLE_FLOAT32BE
+#define MA_PA_SAMPLE_S32LE PA_SAMPLE_S32LE
+#define MA_PA_SAMPLE_S32BE PA_SAMPLE_S32BE
+#define MA_PA_SAMPLE_S24LE PA_SAMPLE_S24LE
+#define MA_PA_SAMPLE_S24BE PA_SAMPLE_S24BE
+#define MA_PA_SAMPLE_S24_32LE PA_SAMPLE_S24_32LE
+#define MA_PA_SAMPLE_S24_32BE PA_SAMPLE_S24_32BE
+
+typedef pa_mainloop ma_pa_mainloop;
+typedef pa_threaded_mainloop ma_pa_threaded_mainloop;
+typedef pa_mainloop_api ma_pa_mainloop_api;
+typedef pa_context ma_pa_context;
+typedef pa_operation ma_pa_operation;
+typedef pa_stream ma_pa_stream;
+typedef pa_spawn_api ma_pa_spawn_api;
+typedef pa_buffer_attr ma_pa_buffer_attr;
+typedef pa_channel_map ma_pa_channel_map;
+typedef pa_cvolume ma_pa_cvolume;
+typedef pa_sample_spec ma_pa_sample_spec;
+typedef pa_sink_info ma_pa_sink_info;
+typedef pa_source_info ma_pa_source_info;
+
+typedef pa_context_notify_cb_t ma_pa_context_notify_cb_t;
+typedef pa_sink_info_cb_t ma_pa_sink_info_cb_t;
+typedef pa_source_info_cb_t ma_pa_source_info_cb_t;
+typedef pa_stream_success_cb_t ma_pa_stream_success_cb_t;
+typedef pa_stream_request_cb_t ma_pa_stream_request_cb_t;
+typedef pa_stream_notify_cb_t ma_pa_stream_notify_cb_t;
+typedef pa_free_cb_t ma_pa_free_cb_t;
#else
-#define C89ATOMIC_32BIT
-#endif
+#define MA_PA_OK 0
+#define MA_PA_ERR_ACCESS 1
+#define MA_PA_ERR_INVALID 2
+#define MA_PA_ERR_NOENTITY 5
+#define MA_PA_ERR_NOTSUPPORTED 19
+
+#define MA_PA_CHANNELS_MAX 32
+#define MA_PA_RATE_MAX 384000
+
+typedef int ma_pa_context_flags_t;
+#define MA_PA_CONTEXT_NOFLAGS 0x00000000
+#define MA_PA_CONTEXT_NOAUTOSPAWN 0x00000001
+#define MA_PA_CONTEXT_NOFAIL 0x00000002
+
+typedef int ma_pa_stream_flags_t;
+#define MA_PA_STREAM_NOFLAGS 0x00000000
+#define MA_PA_STREAM_START_CORKED 0x00000001
+#define MA_PA_STREAM_INTERPOLATE_TIMING 0x00000002
+#define MA_PA_STREAM_NOT_MONOTONIC 0x00000004
+#define MA_PA_STREAM_AUTO_TIMING_UPDATE 0x00000008
+#define MA_PA_STREAM_NO_REMAP_CHANNELS 0x00000010
+#define MA_PA_STREAM_NO_REMIX_CHANNELS 0x00000020
+#define MA_PA_STREAM_FIX_FORMAT 0x00000040
+#define MA_PA_STREAM_FIX_RATE 0x00000080
+#define MA_PA_STREAM_FIX_CHANNELS 0x00000100
+#define MA_PA_STREAM_DONT_MOVE 0x00000200
+#define MA_PA_STREAM_VARIABLE_RATE 0x00000400
+#define MA_PA_STREAM_PEAK_DETECT 0x00000800
+#define MA_PA_STREAM_START_MUTED 0x00001000
+#define MA_PA_STREAM_ADJUST_LATENCY 0x00002000
+#define MA_PA_STREAM_EARLY_REQUESTS 0x00004000
+#define MA_PA_STREAM_DONT_INHIBIT_AUTO_SUSPEND 0x00008000
+#define MA_PA_STREAM_START_UNMUTED 0x00010000
+#define MA_PA_STREAM_FAIL_ON_SUSPEND 0x00020000
+#define MA_PA_STREAM_RELATIVE_VOLUME 0x00040000
+#define MA_PA_STREAM_PASSTHROUGH 0x00080000
+
+typedef int ma_pa_sink_flags_t;
+#define MA_PA_SINK_NOFLAGS 0x00000000
+#define MA_PA_SINK_HW_VOLUME_CTRL 0x00000001
+#define MA_PA_SINK_LATENCY 0x00000002
+#define MA_PA_SINK_HARDWARE 0x00000004
+#define MA_PA_SINK_NETWORK 0x00000008
+#define MA_PA_SINK_HW_MUTE_CTRL 0x00000010
+#define MA_PA_SINK_DECIBEL_VOLUME 0x00000020
+#define MA_PA_SINK_FLAT_VOLUME 0x00000040
+#define MA_PA_SINK_DYNAMIC_LATENCY 0x00000080
+#define MA_PA_SINK_SET_FORMATS 0x00000100
+
+typedef int ma_pa_source_flags_t;
+#define MA_PA_SOURCE_NOFLAGS 0x00000000
+#define MA_PA_SOURCE_HW_VOLUME_CTRL 0x00000001
+#define MA_PA_SOURCE_LATENCY 0x00000002
+#define MA_PA_SOURCE_HARDWARE 0x00000004
+#define MA_PA_SOURCE_NETWORK 0x00000008
+#define MA_PA_SOURCE_HW_MUTE_CTRL 0x00000010
+#define MA_PA_SOURCE_DECIBEL_VOLUME 0x00000020
+#define MA_PA_SOURCE_DYNAMIC_LATENCY 0x00000040
+#define MA_PA_SOURCE_FLAT_VOLUME 0x00000080
+
+typedef int ma_pa_context_state_t;
+#define MA_PA_CONTEXT_UNCONNECTED 0
+#define MA_PA_CONTEXT_CONNECTING 1
+#define MA_PA_CONTEXT_AUTHORIZING 2
+#define MA_PA_CONTEXT_SETTING_NAME 3
+#define MA_PA_CONTEXT_READY 4
+#define MA_PA_CONTEXT_FAILED 5
+#define MA_PA_CONTEXT_TERMINATED 6
+
+typedef int ma_pa_stream_state_t;
+#define MA_PA_STREAM_UNCONNECTED 0
+#define MA_PA_STREAM_CREATING 1
+#define MA_PA_STREAM_READY 2
+#define MA_PA_STREAM_FAILED 3
+#define MA_PA_STREAM_TERMINATED 4
+
+typedef int ma_pa_operation_state_t;
+#define MA_PA_OPERATION_RUNNING 0
+#define MA_PA_OPERATION_DONE 1
+#define MA_PA_OPERATION_CANCELLED 2
+
+typedef int ma_pa_sink_state_t;
+#define MA_PA_SINK_INVALID_STATE -1
+#define MA_PA_SINK_RUNNING 0
+#define MA_PA_SINK_IDLE 1
+#define MA_PA_SINK_SUSPENDED 2
+
+typedef int ma_pa_source_state_t;
+#define MA_PA_SOURCE_INVALID_STATE -1
+#define MA_PA_SOURCE_RUNNING 0
+#define MA_PA_SOURCE_IDLE 1
+#define MA_PA_SOURCE_SUSPENDED 2
+
+typedef int ma_pa_seek_mode_t;
+#define MA_PA_SEEK_RELATIVE 0
+#define MA_PA_SEEK_ABSOLUTE 1
+#define MA_PA_SEEK_RELATIVE_ON_READ 2
+#define MA_PA_SEEK_RELATIVE_END 3
+
+typedef int ma_pa_channel_position_t;
+#define MA_PA_CHANNEL_POSITION_INVALID -1
+#define MA_PA_CHANNEL_POSITION_MONO 0
+#define MA_PA_CHANNEL_POSITION_FRONT_LEFT 1
+#define MA_PA_CHANNEL_POSITION_FRONT_RIGHT 2
+#define MA_PA_CHANNEL_POSITION_FRONT_CENTER 3
+#define MA_PA_CHANNEL_POSITION_REAR_CENTER 4
+#define MA_PA_CHANNEL_POSITION_REAR_LEFT 5
+#define MA_PA_CHANNEL_POSITION_REAR_RIGHT 6
+#define MA_PA_CHANNEL_POSITION_LFE 7
+#define MA_PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER 8
+#define MA_PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER 9
+#define MA_PA_CHANNEL_POSITION_SIDE_LEFT 10
+#define MA_PA_CHANNEL_POSITION_SIDE_RIGHT 11
+#define MA_PA_CHANNEL_POSITION_AUX0 12
+#define MA_PA_CHANNEL_POSITION_AUX1 13
+#define MA_PA_CHANNEL_POSITION_AUX2 14
+#define MA_PA_CHANNEL_POSITION_AUX3 15
+#define MA_PA_CHANNEL_POSITION_AUX4 16
+#define MA_PA_CHANNEL_POSITION_AUX5 17
+#define MA_PA_CHANNEL_POSITION_AUX6 18
+#define MA_PA_CHANNEL_POSITION_AUX7 19
+#define MA_PA_CHANNEL_POSITION_AUX8 20
+#define MA_PA_CHANNEL_POSITION_AUX9 21
+#define MA_PA_CHANNEL_POSITION_AUX10 22
+#define MA_PA_CHANNEL_POSITION_AUX11 23
+#define MA_PA_CHANNEL_POSITION_AUX12 24
+#define MA_PA_CHANNEL_POSITION_AUX13 25
+#define MA_PA_CHANNEL_POSITION_AUX14 26
+#define MA_PA_CHANNEL_POSITION_AUX15 27
+#define MA_PA_CHANNEL_POSITION_AUX16 28
+#define MA_PA_CHANNEL_POSITION_AUX17 29
+#define MA_PA_CHANNEL_POSITION_AUX18 30
+#define MA_PA_CHANNEL_POSITION_AUX19 31
+#define MA_PA_CHANNEL_POSITION_AUX20 32
+#define MA_PA_CHANNEL_POSITION_AUX21 33
+#define MA_PA_CHANNEL_POSITION_AUX22 34
+#define MA_PA_CHANNEL_POSITION_AUX23 35
+#define MA_PA_CHANNEL_POSITION_AUX24 36
+#define MA_PA_CHANNEL_POSITION_AUX25 37
+#define MA_PA_CHANNEL_POSITION_AUX26 38
+#define MA_PA_CHANNEL_POSITION_AUX27 39
+#define MA_PA_CHANNEL_POSITION_AUX28 40
+#define MA_PA_CHANNEL_POSITION_AUX29 41
+#define MA_PA_CHANNEL_POSITION_AUX30 42
+#define MA_PA_CHANNEL_POSITION_AUX31 43
+#define MA_PA_CHANNEL_POSITION_TOP_CENTER 44
+#define MA_PA_CHANNEL_POSITION_TOP_FRONT_LEFT 45
+#define MA_PA_CHANNEL_POSITION_TOP_FRONT_RIGHT 46
+#define MA_PA_CHANNEL_POSITION_TOP_FRONT_CENTER 47
+#define MA_PA_CHANNEL_POSITION_TOP_REAR_LEFT 48
+#define MA_PA_CHANNEL_POSITION_TOP_REAR_RIGHT 49
+#define MA_PA_CHANNEL_POSITION_TOP_REAR_CENTER 50
+#define MA_PA_CHANNEL_POSITION_LEFT MA_PA_CHANNEL_POSITION_FRONT_LEFT
+#define MA_PA_CHANNEL_POSITION_RIGHT MA_PA_CHANNEL_POSITION_FRONT_RIGHT
+#define MA_PA_CHANNEL_POSITION_CENTER MA_PA_CHANNEL_POSITION_FRONT_CENTER
+#define MA_PA_CHANNEL_POSITION_SUBWOOFER MA_PA_CHANNEL_POSITION_LFE
+
+typedef int ma_pa_channel_map_def_t;
+#define MA_PA_CHANNEL_MAP_AIFF 0
+#define MA_PA_CHANNEL_MAP_ALSA 1
+#define MA_PA_CHANNEL_MAP_AUX 2
+#define MA_PA_CHANNEL_MAP_WAVEEX 3
+#define MA_PA_CHANNEL_MAP_OSS 4
+#define MA_PA_CHANNEL_MAP_DEFAULT MA_PA_CHANNEL_MAP_AIFF
+
+typedef int ma_pa_sample_format_t;
+#define MA_PA_SAMPLE_INVALID -1
+#define MA_PA_SAMPLE_U8 0
+#define MA_PA_SAMPLE_ALAW 1
+#define MA_PA_SAMPLE_ULAW 2
+#define MA_PA_SAMPLE_S16LE 3
+#define MA_PA_SAMPLE_S16BE 4
+#define MA_PA_SAMPLE_FLOAT32LE 5
+#define MA_PA_SAMPLE_FLOAT32BE 6
+#define MA_PA_SAMPLE_S32LE 7
+#define MA_PA_SAMPLE_S32BE 8
+#define MA_PA_SAMPLE_S24LE 9
+#define MA_PA_SAMPLE_S24BE 10
+#define MA_PA_SAMPLE_S24_32LE 11
+#define MA_PA_SAMPLE_S24_32BE 12
+
+typedef struct ma_pa_mainloop ma_pa_mainloop;
+typedef struct ma_pa_threaded_mainloop ma_pa_threaded_mainloop;
+typedef struct ma_pa_mainloop_api ma_pa_mainloop_api;
+typedef struct ma_pa_context ma_pa_context;
+typedef struct ma_pa_operation ma_pa_operation;
+typedef struct ma_pa_stream ma_pa_stream;
+typedef struct ma_pa_spawn_api ma_pa_spawn_api;
+
+typedef struct
+{
+ ma_uint32 maxlength;
+ ma_uint32 tlength;
+ ma_uint32 prebuf;
+ ma_uint32 minreq;
+ ma_uint32 fragsize;
+} ma_pa_buffer_attr;
+
+typedef struct
+{
+ ma_uint8 channels;
+ ma_pa_channel_position_t map[MA_PA_CHANNELS_MAX];
+} ma_pa_channel_map;
+
+typedef struct
+{
+ ma_uint8 channels;
+ ma_uint32 values[MA_PA_CHANNELS_MAX];
+} ma_pa_cvolume;
+
+typedef struct
+{
+ ma_pa_sample_format_t format;
+ ma_uint32 rate;
+ ma_uint8 channels;
+} ma_pa_sample_spec;
+
+typedef struct
+{
+ const char* name;
+ ma_uint32 index;
+ const char* description;
+ ma_pa_sample_spec sample_spec;
+ ma_pa_channel_map channel_map;
+ ma_uint32 owner_module;
+ ma_pa_cvolume volume;
+ int mute;
+ ma_uint32 monitor_source;
+ const char* monitor_source_name;
+ ma_uint64 latency;
+ const char* driver;
+ ma_pa_sink_flags_t flags;
+ void* proplist;
+ ma_uint64 configured_latency;
+ ma_uint32 base_volume;
+ ma_pa_sink_state_t state;
+ ma_uint32 n_volume_steps;
+ ma_uint32 card;
+ ma_uint32 n_ports;
+ void** ports;
+ void* active_port;
+ ma_uint8 n_formats;
+ void** formats;
+} ma_pa_sink_info;
+
+typedef struct
+{
+ const char *name;
+ ma_uint32 index;
+ const char *description;
+ ma_pa_sample_spec sample_spec;
+ ma_pa_channel_map channel_map;
+ ma_uint32 owner_module;
+ ma_pa_cvolume volume;
+ int mute;
+ ma_uint32 monitor_of_sink;
+ const char *monitor_of_sink_name;
+ ma_uint64 latency;
+ const char *driver;
+ ma_pa_source_flags_t flags;
+ void* proplist;
+ ma_uint64 configured_latency;
+ ma_uint32 base_volume;
+ ma_pa_source_state_t state;
+ ma_uint32 n_volume_steps;
+ ma_uint32 card;
+ ma_uint32 n_ports;
+ void** ports;
+ void* active_port;
+ ma_uint8 n_formats;
+ void** formats;
+} ma_pa_source_info;
+
+typedef void (* ma_pa_context_notify_cb_t)(ma_pa_context* c, void* userdata);
+typedef void (* ma_pa_sink_info_cb_t) (ma_pa_context* c, const ma_pa_sink_info* i, int eol, void* userdata);
+typedef void (* ma_pa_source_info_cb_t) (ma_pa_context* c, const ma_pa_source_info* i, int eol, void* userdata);
+typedef void (* ma_pa_stream_success_cb_t)(ma_pa_stream* s, int success, void* userdata);
+typedef void (* ma_pa_stream_request_cb_t)(ma_pa_stream* s, size_t nbytes, void* userdata);
+typedef void (* ma_pa_stream_notify_cb_t) (ma_pa_stream* s, void* userdata);
+typedef void (* ma_pa_free_cb_t) (void* p);
#endif
-#if defined(__x86_64__) || defined(_M_X64)
-#define C89ATOMIC_X64
-#elif defined(__i386) || defined(_M_IX86)
-#define C89ATOMIC_X86
-#elif defined(__arm__) || defined(_M_ARM)
-#define C89ATOMIC_ARM
+
+
+typedef ma_pa_mainloop* (* ma_pa_mainloop_new_proc) (void);
+typedef void (* ma_pa_mainloop_free_proc) (ma_pa_mainloop* m);
+typedef void (* ma_pa_mainloop_quit_proc) (ma_pa_mainloop* m, int retval);
+typedef ma_pa_mainloop_api* (* ma_pa_mainloop_get_api_proc) (ma_pa_mainloop* m);
+typedef int (* ma_pa_mainloop_iterate_proc) (ma_pa_mainloop* m, int block, int* retval);
+typedef void (* ma_pa_mainloop_wakeup_proc) (ma_pa_mainloop* m);
+typedef ma_pa_threaded_mainloop* (* ma_pa_threaded_mainloop_new_proc) (void);
+typedef void (* ma_pa_threaded_mainloop_free_proc) (ma_pa_threaded_mainloop* m);
+typedef int (* ma_pa_threaded_mainloop_start_proc) (ma_pa_threaded_mainloop* m);
+typedef void (* ma_pa_threaded_mainloop_stop_proc) (ma_pa_threaded_mainloop* m);
+typedef void (* ma_pa_threaded_mainloop_lock_proc) (ma_pa_threaded_mainloop* m);
+typedef void (* ma_pa_threaded_mainloop_unlock_proc) (ma_pa_threaded_mainloop* m);
+typedef void (* ma_pa_threaded_mainloop_wait_proc) (ma_pa_threaded_mainloop* m);
+typedef void (* ma_pa_threaded_mainloop_signal_proc) (ma_pa_threaded_mainloop* m, int wait_for_accept);
+typedef void (* ma_pa_threaded_mainloop_accept_proc) (ma_pa_threaded_mainloop* m);
+typedef int (* ma_pa_threaded_mainloop_get_retval_proc) (ma_pa_threaded_mainloop* m);
+typedef ma_pa_mainloop_api* (* ma_pa_threaded_mainloop_get_api_proc) (ma_pa_threaded_mainloop* m);
+typedef int (* ma_pa_threaded_mainloop_in_thread_proc) (ma_pa_threaded_mainloop* m);
+typedef void (* ma_pa_threaded_mainloop_set_name_proc) (ma_pa_threaded_mainloop* m, const char* name);
+typedef ma_pa_context* (* ma_pa_context_new_proc) (ma_pa_mainloop_api* mainloop, const char* name);
+typedef void (* ma_pa_context_unref_proc) (ma_pa_context* c);
+typedef int (* ma_pa_context_connect_proc) (ma_pa_context* c, const char* server, ma_pa_context_flags_t flags, const ma_pa_spawn_api* api);
+typedef void (* ma_pa_context_disconnect_proc) (ma_pa_context* c);
+typedef void (* ma_pa_context_set_state_callback_proc) (ma_pa_context* c, ma_pa_context_notify_cb_t cb, void* userdata);
+typedef ma_pa_context_state_t (* ma_pa_context_get_state_proc) (ma_pa_context* c);
+typedef ma_pa_operation* (* ma_pa_context_get_sink_info_list_proc) (ma_pa_context* c, ma_pa_sink_info_cb_t cb, void* userdata);
+typedef ma_pa_operation* (* ma_pa_context_get_source_info_list_proc) (ma_pa_context* c, ma_pa_source_info_cb_t cb, void* userdata);
+typedef ma_pa_operation* (* ma_pa_context_get_sink_info_by_name_proc) (ma_pa_context* c, const char* name, ma_pa_sink_info_cb_t cb, void* userdata);
+typedef ma_pa_operation* (* ma_pa_context_get_source_info_by_name_proc)(ma_pa_context* c, const char* name, ma_pa_source_info_cb_t cb, void* userdata);
+typedef void (* ma_pa_operation_unref_proc) (ma_pa_operation* o);
+typedef ma_pa_operation_state_t (* ma_pa_operation_get_state_proc) (ma_pa_operation* o);
+typedef ma_pa_channel_map* (* ma_pa_channel_map_init_extend_proc) (ma_pa_channel_map* m, unsigned channels, ma_pa_channel_map_def_t def);
+typedef int (* ma_pa_channel_map_valid_proc) (const ma_pa_channel_map* m);
+typedef int (* ma_pa_channel_map_compatible_proc) (const ma_pa_channel_map* m, const ma_pa_sample_spec* ss);
+typedef ma_pa_stream* (* ma_pa_stream_new_proc) (ma_pa_context* c, const char* name, const ma_pa_sample_spec* ss, const ma_pa_channel_map* map);
+typedef void (* ma_pa_stream_unref_proc) (ma_pa_stream* s);
+typedef int (* ma_pa_stream_connect_playback_proc) (ma_pa_stream* s, const char* dev, const ma_pa_buffer_attr* attr, ma_pa_stream_flags_t flags, const ma_pa_cvolume* volume, ma_pa_stream* sync_stream);
+typedef int (* ma_pa_stream_connect_record_proc) (ma_pa_stream* s, const char* dev, const ma_pa_buffer_attr* attr, ma_pa_stream_flags_t flags);
+typedef int (* ma_pa_stream_disconnect_proc) (ma_pa_stream* s);
+typedef ma_pa_stream_state_t (* ma_pa_stream_get_state_proc) (ma_pa_stream* s);
+typedef const ma_pa_sample_spec* (* ma_pa_stream_get_sample_spec_proc) (ma_pa_stream* s);
+typedef const ma_pa_channel_map* (* ma_pa_stream_get_channel_map_proc) (ma_pa_stream* s);
+typedef const ma_pa_buffer_attr* (* ma_pa_stream_get_buffer_attr_proc) (ma_pa_stream* s);
+typedef ma_pa_operation* (* ma_pa_stream_set_buffer_attr_proc) (ma_pa_stream* s, const ma_pa_buffer_attr* attr, ma_pa_stream_success_cb_t cb, void* userdata);
+typedef const char* (* ma_pa_stream_get_device_name_proc) (ma_pa_stream* s);
+typedef void (* ma_pa_stream_set_write_callback_proc) (ma_pa_stream* s, ma_pa_stream_request_cb_t cb, void* userdata);
+typedef void (* ma_pa_stream_set_read_callback_proc) (ma_pa_stream* s, ma_pa_stream_request_cb_t cb, void* userdata);
+typedef void (* ma_pa_stream_set_suspended_callback_proc) (ma_pa_stream* s, ma_pa_stream_notify_cb_t cb, void* userdata);
+typedef void (* ma_pa_stream_set_moved_callback_proc) (ma_pa_stream* s, ma_pa_stream_notify_cb_t cb, void* userdata);
+typedef int (* ma_pa_stream_is_suspended_proc) (const ma_pa_stream* s);
+typedef ma_pa_operation* (* ma_pa_stream_flush_proc) (ma_pa_stream* s, ma_pa_stream_success_cb_t cb, void* userdata);
+typedef ma_pa_operation* (* ma_pa_stream_drain_proc) (ma_pa_stream* s, ma_pa_stream_success_cb_t cb, void* userdata);
+typedef int (* ma_pa_stream_is_corked_proc) (ma_pa_stream* s);
+typedef ma_pa_operation* (* ma_pa_stream_cork_proc) (ma_pa_stream* s, int b, ma_pa_stream_success_cb_t cb, void* userdata);
+typedef ma_pa_operation* (* ma_pa_stream_trigger_proc) (ma_pa_stream* s, ma_pa_stream_success_cb_t cb, void* userdata);
+typedef int (* ma_pa_stream_begin_write_proc) (ma_pa_stream* s, void** data, size_t* nbytes);
+typedef int (* ma_pa_stream_write_proc) (ma_pa_stream* s, const void* data, size_t nbytes, ma_pa_free_cb_t free_cb, int64_t offset, ma_pa_seek_mode_t seek);
+typedef int (* ma_pa_stream_peek_proc) (ma_pa_stream* s, const void** data, size_t* nbytes);
+typedef int (* ma_pa_stream_drop_proc) (ma_pa_stream* s);
+typedef size_t (* ma_pa_stream_writable_size_proc) (ma_pa_stream* s);
+typedef size_t (* ma_pa_stream_readable_size_proc) (ma_pa_stream* s);
+
+typedef struct
+{
+ ma_uint32 count;
+ ma_uint32 capacity;
+ ma_device_info* pInfo;
+} ma_pulse_device_enum_data;
+
+static ma_result ma_result_from_pulse(int result)
+{
+ if (result < 0) {
+ return MA_ERROR;
+ }
+
+ switch (result) {
+ case MA_PA_OK: return MA_SUCCESS;
+ case MA_PA_ERR_ACCESS: return MA_ACCESS_DENIED;
+ case MA_PA_ERR_INVALID: return MA_INVALID_ARGS;
+ case MA_PA_ERR_NOENTITY: return MA_NO_DEVICE;
+ default: return MA_ERROR;
+ }
+}
+
+#if 0
+static ma_pa_sample_format_t ma_format_to_pulse(ma_format format)
+{
+ if (ma_is_little_endian()) {
+ switch (format) {
+ case ma_format_s16: return MA_PA_SAMPLE_S16LE;
+ case ma_format_s24: return MA_PA_SAMPLE_S24LE;
+ case ma_format_s32: return MA_PA_SAMPLE_S32LE;
+ case ma_format_f32: return MA_PA_SAMPLE_FLOAT32LE;
+ default: break;
+ }
+ } else {
+ switch (format) {
+ case ma_format_s16: return MA_PA_SAMPLE_S16BE;
+ case ma_format_s24: return MA_PA_SAMPLE_S24BE;
+ case ma_format_s32: return MA_PA_SAMPLE_S32BE;
+ case ma_format_f32: return MA_PA_SAMPLE_FLOAT32BE;
+ default: break;
+ }
+ }
+
+ /* Endian agnostic. */
+ switch (format) {
+ case ma_format_u8: return MA_PA_SAMPLE_U8;
+ default: return MA_PA_SAMPLE_INVALID;
+ }
+}
#endif
-#if defined(_MSC_VER)
- #define C89ATOMIC_INLINE __forceinline
-#elif defined(__GNUC__)
- #if defined(__STRICT_ANSI__)
- #define C89ATOMIC_INLINE __inline__ __attribute__((always_inline))
- #else
- #define C89ATOMIC_INLINE inline __attribute__((always_inline))
- #endif
-#elif defined(__WATCOMC__) || defined(__DMC__)
- #define C89ATOMIC_INLINE __inline
-#else
- #define C89ATOMIC_INLINE
+
+static ma_format ma_format_from_pulse(ma_pa_sample_format_t format)
+{
+ if (ma_is_little_endian()) {
+ switch (format) {
+ case MA_PA_SAMPLE_S16LE: return ma_format_s16;
+ case MA_PA_SAMPLE_S24LE: return ma_format_s24;
+ case MA_PA_SAMPLE_S32LE: return ma_format_s32;
+ case MA_PA_SAMPLE_FLOAT32LE: return ma_format_f32;
+ default: break;
+ }
+ } else {
+ switch (format) {
+ case MA_PA_SAMPLE_S16BE: return ma_format_s16;
+ case MA_PA_SAMPLE_S24BE: return ma_format_s24;
+ case MA_PA_SAMPLE_S32BE: return ma_format_s32;
+ case MA_PA_SAMPLE_FLOAT32BE: return ma_format_f32;
+ default: break;
+ }
+ }
+
+ /* Endian agnostic. */
+ switch (format) {
+ case MA_PA_SAMPLE_U8: return ma_format_u8;
+ default: return ma_format_unknown;
+ }
+}
+
+static ma_channel ma_channel_position_from_pulse(ma_pa_channel_position_t position)
+{
+ switch (position)
+ {
+ case MA_PA_CHANNEL_POSITION_INVALID: return MA_CHANNEL_NONE;
+ case MA_PA_CHANNEL_POSITION_MONO: return MA_CHANNEL_MONO;
+ case MA_PA_CHANNEL_POSITION_FRONT_LEFT: return MA_CHANNEL_FRONT_LEFT;
+ case MA_PA_CHANNEL_POSITION_FRONT_RIGHT: return MA_CHANNEL_FRONT_RIGHT;
+ case MA_PA_CHANNEL_POSITION_FRONT_CENTER: return MA_CHANNEL_FRONT_CENTER;
+ case MA_PA_CHANNEL_POSITION_REAR_CENTER: return MA_CHANNEL_BACK_CENTER;
+ case MA_PA_CHANNEL_POSITION_REAR_LEFT: return MA_CHANNEL_BACK_LEFT;
+ case MA_PA_CHANNEL_POSITION_REAR_RIGHT: return MA_CHANNEL_BACK_RIGHT;
+ case MA_PA_CHANNEL_POSITION_LFE: return MA_CHANNEL_LFE;
+ case MA_PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER: return MA_CHANNEL_FRONT_LEFT_CENTER;
+ case MA_PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER: return MA_CHANNEL_FRONT_RIGHT_CENTER;
+ case MA_PA_CHANNEL_POSITION_SIDE_LEFT: return MA_CHANNEL_SIDE_LEFT;
+ case MA_PA_CHANNEL_POSITION_SIDE_RIGHT: return MA_CHANNEL_SIDE_RIGHT;
+ case MA_PA_CHANNEL_POSITION_AUX0: return MA_CHANNEL_AUX_0;
+ case MA_PA_CHANNEL_POSITION_AUX1: return MA_CHANNEL_AUX_1;
+ case MA_PA_CHANNEL_POSITION_AUX2: return MA_CHANNEL_AUX_2;
+ case MA_PA_CHANNEL_POSITION_AUX3: return MA_CHANNEL_AUX_3;
+ case MA_PA_CHANNEL_POSITION_AUX4: return MA_CHANNEL_AUX_4;
+ case MA_PA_CHANNEL_POSITION_AUX5: return MA_CHANNEL_AUX_5;
+ case MA_PA_CHANNEL_POSITION_AUX6: return MA_CHANNEL_AUX_6;
+ case MA_PA_CHANNEL_POSITION_AUX7: return MA_CHANNEL_AUX_7;
+ case MA_PA_CHANNEL_POSITION_AUX8: return MA_CHANNEL_AUX_8;
+ case MA_PA_CHANNEL_POSITION_AUX9: return MA_CHANNEL_AUX_9;
+ case MA_PA_CHANNEL_POSITION_AUX10: return MA_CHANNEL_AUX_10;
+ case MA_PA_CHANNEL_POSITION_AUX11: return MA_CHANNEL_AUX_11;
+ case MA_PA_CHANNEL_POSITION_AUX12: return MA_CHANNEL_AUX_12;
+ case MA_PA_CHANNEL_POSITION_AUX13: return MA_CHANNEL_AUX_13;
+ case MA_PA_CHANNEL_POSITION_AUX14: return MA_CHANNEL_AUX_14;
+ case MA_PA_CHANNEL_POSITION_AUX15: return MA_CHANNEL_AUX_15;
+ case MA_PA_CHANNEL_POSITION_AUX16: return MA_CHANNEL_AUX_16;
+ case MA_PA_CHANNEL_POSITION_AUX17: return MA_CHANNEL_AUX_17;
+ case MA_PA_CHANNEL_POSITION_AUX18: return MA_CHANNEL_AUX_18;
+ case MA_PA_CHANNEL_POSITION_AUX19: return MA_CHANNEL_AUX_19;
+ case MA_PA_CHANNEL_POSITION_AUX20: return MA_CHANNEL_AUX_20;
+ case MA_PA_CHANNEL_POSITION_AUX21: return MA_CHANNEL_AUX_21;
+ case MA_PA_CHANNEL_POSITION_AUX22: return MA_CHANNEL_AUX_22;
+ case MA_PA_CHANNEL_POSITION_AUX23: return MA_CHANNEL_AUX_23;
+ case MA_PA_CHANNEL_POSITION_AUX24: return MA_CHANNEL_AUX_24;
+ case MA_PA_CHANNEL_POSITION_AUX25: return MA_CHANNEL_AUX_25;
+ case MA_PA_CHANNEL_POSITION_AUX26: return MA_CHANNEL_AUX_26;
+ case MA_PA_CHANNEL_POSITION_AUX27: return MA_CHANNEL_AUX_27;
+ case MA_PA_CHANNEL_POSITION_AUX28: return MA_CHANNEL_AUX_28;
+ case MA_PA_CHANNEL_POSITION_AUX29: return MA_CHANNEL_AUX_29;
+ case MA_PA_CHANNEL_POSITION_AUX30: return MA_CHANNEL_AUX_30;
+ case MA_PA_CHANNEL_POSITION_AUX31: return MA_CHANNEL_AUX_31;
+ case MA_PA_CHANNEL_POSITION_TOP_CENTER: return MA_CHANNEL_TOP_CENTER;
+ case MA_PA_CHANNEL_POSITION_TOP_FRONT_LEFT: return MA_CHANNEL_TOP_FRONT_LEFT;
+ case MA_PA_CHANNEL_POSITION_TOP_FRONT_RIGHT: return MA_CHANNEL_TOP_FRONT_RIGHT;
+ case MA_PA_CHANNEL_POSITION_TOP_FRONT_CENTER: return MA_CHANNEL_TOP_FRONT_CENTER;
+ case MA_PA_CHANNEL_POSITION_TOP_REAR_LEFT: return MA_CHANNEL_TOP_BACK_LEFT;
+ case MA_PA_CHANNEL_POSITION_TOP_REAR_RIGHT: return MA_CHANNEL_TOP_BACK_RIGHT;
+ case MA_PA_CHANNEL_POSITION_TOP_REAR_CENTER: return MA_CHANNEL_TOP_BACK_CENTER;
+ default: return MA_CHANNEL_NONE;
+ }
+}
+
+#if 0
+static ma_pa_channel_position_t ma_channel_position_to_pulse(ma_channel position)
+{
+ switch (position)
+ {
+ case MA_CHANNEL_NONE: return MA_PA_CHANNEL_POSITION_INVALID;
+ case MA_CHANNEL_FRONT_LEFT: return MA_PA_CHANNEL_POSITION_FRONT_LEFT;
+ case MA_CHANNEL_FRONT_RIGHT: return MA_PA_CHANNEL_POSITION_FRONT_RIGHT;
+ case MA_CHANNEL_FRONT_CENTER: return MA_PA_CHANNEL_POSITION_FRONT_CENTER;
+ case MA_CHANNEL_LFE: return MA_PA_CHANNEL_POSITION_LFE;
+ case MA_CHANNEL_BACK_LEFT: return MA_PA_CHANNEL_POSITION_REAR_LEFT;
+ case MA_CHANNEL_BACK_RIGHT: return MA_PA_CHANNEL_POSITION_REAR_RIGHT;
+ case MA_CHANNEL_FRONT_LEFT_CENTER: return MA_PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER;
+ case MA_CHANNEL_FRONT_RIGHT_CENTER: return MA_PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER;
+ case MA_CHANNEL_BACK_CENTER: return MA_PA_CHANNEL_POSITION_REAR_CENTER;
+ case MA_CHANNEL_SIDE_LEFT: return MA_PA_CHANNEL_POSITION_SIDE_LEFT;
+ case MA_CHANNEL_SIDE_RIGHT: return MA_PA_CHANNEL_POSITION_SIDE_RIGHT;
+ case MA_CHANNEL_TOP_CENTER: return MA_PA_CHANNEL_POSITION_TOP_CENTER;
+ case MA_CHANNEL_TOP_FRONT_LEFT: return MA_PA_CHANNEL_POSITION_TOP_FRONT_LEFT;
+ case MA_CHANNEL_TOP_FRONT_CENTER: return MA_PA_CHANNEL_POSITION_TOP_FRONT_CENTER;
+ case MA_CHANNEL_TOP_FRONT_RIGHT: return MA_PA_CHANNEL_POSITION_TOP_FRONT_RIGHT;
+ case MA_CHANNEL_TOP_BACK_LEFT: return MA_PA_CHANNEL_POSITION_TOP_REAR_LEFT;
+ case MA_CHANNEL_TOP_BACK_CENTER: return MA_PA_CHANNEL_POSITION_TOP_REAR_CENTER;
+ case MA_CHANNEL_TOP_BACK_RIGHT: return MA_PA_CHANNEL_POSITION_TOP_REAR_RIGHT;
+ case MA_CHANNEL_19: return MA_PA_CHANNEL_POSITION_AUX18;
+ case MA_CHANNEL_20: return MA_PA_CHANNEL_POSITION_AUX19;
+ case MA_CHANNEL_21: return MA_PA_CHANNEL_POSITION_AUX20;
+ case MA_CHANNEL_22: return MA_PA_CHANNEL_POSITION_AUX21;
+ case MA_CHANNEL_23: return MA_PA_CHANNEL_POSITION_AUX22;
+ case MA_CHANNEL_24: return MA_PA_CHANNEL_POSITION_AUX23;
+ case MA_CHANNEL_25: return MA_PA_CHANNEL_POSITION_AUX24;
+ case MA_CHANNEL_26: return MA_PA_CHANNEL_POSITION_AUX25;
+ case MA_CHANNEL_27: return MA_PA_CHANNEL_POSITION_AUX26;
+ case MA_CHANNEL_28: return MA_PA_CHANNEL_POSITION_AUX27;
+ case MA_CHANNEL_29: return MA_PA_CHANNEL_POSITION_AUX28;
+ case MA_CHANNEL_30: return MA_PA_CHANNEL_POSITION_AUX29;
+ case MA_CHANNEL_31: return MA_PA_CHANNEL_POSITION_AUX30;
+ case MA_CHANNEL_32: return MA_PA_CHANNEL_POSITION_AUX31;
+ default: return (ma_pa_channel_position_t)position;
+ }
+}
#endif
-#define C89ATOMIC_HAS_8
-#define C89ATOMIC_HAS_16
-#define C89ATOMIC_HAS_32
-#define C89ATOMIC_HAS_64
-#if (defined(_MSC_VER) ) || defined(__WATCOMC__) || defined(__DMC__)
- #define c89atomic_memory_order_relaxed 0
- #define c89atomic_memory_order_consume 1
- #define c89atomic_memory_order_acquire 2
- #define c89atomic_memory_order_release 3
- #define c89atomic_memory_order_acq_rel 4
- #define c89atomic_memory_order_seq_cst 5
- #if _MSC_VER < 1600 && defined(C89ATOMIC_X86)
- #define C89ATOMIC_MSVC_USE_INLINED_ASSEMBLY
- #endif
- #if _MSC_VER < 1600
- #undef C89ATOMIC_HAS_8
- #undef C89ATOMIC_HAS_16
- #endif
- #if !defined(C89ATOMIC_MSVC_USE_INLINED_ASSEMBLY)
- #include
- #endif
- #if defined(C89ATOMIC_MSVC_USE_INLINED_ASSEMBLY)
- #if defined(C89ATOMIC_HAS_8)
- static C89ATOMIC_INLINE c89atomic_uint8 __stdcall c89atomic_compare_and_swap_8(volatile c89atomic_uint8* dst, c89atomic_uint8 expected, c89atomic_uint8 desired)
- {
- c89atomic_uint8 result = 0;
- __asm {
- mov ecx, dst
- mov al, expected
- mov dl, desired
- lock cmpxchg [ecx], dl
- mov result, al
- }
- return result;
- }
- #endif
- #if defined(C89ATOMIC_HAS_16)
- static C89ATOMIC_INLINE c89atomic_uint16 __stdcall c89atomic_compare_and_swap_16(volatile c89atomic_uint16* dst, c89atomic_uint16 expected, c89atomic_uint16 desired)
- {
- c89atomic_uint16 result = 0;
- __asm {
- mov ecx, dst
- mov ax, expected
- mov dx, desired
- lock cmpxchg [ecx], dx
- mov result, ax
- }
- return result;
- }
- #endif
- #if defined(C89ATOMIC_HAS_32)
- static C89ATOMIC_INLINE c89atomic_uint32 __stdcall c89atomic_compare_and_swap_32(volatile c89atomic_uint32* dst, c89atomic_uint32 expected, c89atomic_uint32 desired)
- {
- c89atomic_uint32 result = 0;
- __asm {
- mov ecx, dst
- mov eax, expected
- mov edx, desired
- lock cmpxchg [ecx], edx
- mov result, eax
- }
- return result;
- }
- #endif
- #if defined(C89ATOMIC_HAS_64)
- static C89ATOMIC_INLINE c89atomic_uint64 __stdcall c89atomic_compare_and_swap_64(volatile c89atomic_uint64* dst, c89atomic_uint64 expected, c89atomic_uint64 desired)
- {
- c89atomic_uint32 resultEAX = 0;
- c89atomic_uint32 resultEDX = 0;
- __asm {
- mov esi, dst
- mov eax, dword ptr expected
- mov edx, dword ptr expected + 4
- mov ebx, dword ptr desired
- mov ecx, dword ptr desired + 4
- lock cmpxchg8b qword ptr [esi]
- mov resultEAX, eax
- mov resultEDX, edx
- }
- return ((c89atomic_uint64)resultEDX << 32) | resultEAX;
- }
- #endif
- #else
- #if defined(C89ATOMIC_HAS_8)
- #define c89atomic_compare_and_swap_8( dst, expected, desired) (c89atomic_uint8 )_InterlockedCompareExchange8((volatile char*)dst, (char)desired, (char)expected)
- #endif
- #if defined(C89ATOMIC_HAS_16)
- #define c89atomic_compare_and_swap_16(dst, expected, desired) (c89atomic_uint16)_InterlockedCompareExchange16((volatile short*)dst, (short)desired, (short)expected)
- #endif
- #if defined(C89ATOMIC_HAS_32)
- #define c89atomic_compare_and_swap_32(dst, expected, desired) (c89atomic_uint32)_InterlockedCompareExchange((volatile long*)dst, (long)desired, (long)expected)
- #endif
- #if defined(C89ATOMIC_HAS_64)
- #define c89atomic_compare_and_swap_64(dst, expected, desired) (c89atomic_uint64)_InterlockedCompareExchange64((volatile long long*)dst, (long long)desired, (long long)expected)
- #endif
- #endif
- #if defined(C89ATOMIC_MSVC_USE_INLINED_ASSEMBLY)
- #if defined(C89ATOMIC_HAS_8)
- static C89ATOMIC_INLINE c89atomic_uint8 __stdcall c89atomic_exchange_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
- {
- c89atomic_uint8 result = 0;
- (void)order;
- __asm {
- mov ecx, dst
- mov al, src
- lock xchg [ecx], al
- mov result, al
- }
- return result;
- }
- #endif
- #if defined(C89ATOMIC_HAS_16)
- static C89ATOMIC_INLINE c89atomic_uint16 __stdcall c89atomic_exchange_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
- {
- c89atomic_uint16 result = 0;
- (void)order;
- __asm {
- mov ecx, dst
- mov ax, src
- lock xchg [ecx], ax
- mov result, ax
- }
- return result;
- }
- #endif
- #if defined(C89ATOMIC_HAS_32)
- static C89ATOMIC_INLINE c89atomic_uint32 __stdcall c89atomic_exchange_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
- {
- c89atomic_uint32 result = 0;
- (void)order;
- __asm {
- mov ecx, dst
- mov eax, src
- lock xchg [ecx], eax
- mov result, eax
- }
- return result;
- }
- #endif
- #else
- #if defined(C89ATOMIC_HAS_8)
- static C89ATOMIC_INLINE c89atomic_uint8 __stdcall c89atomic_exchange_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
- {
- (void)order;
- return (c89atomic_uint8)_InterlockedExchange8((volatile char*)dst, (char)src);
- }
- #endif
- #if defined(C89ATOMIC_HAS_16)
- static C89ATOMIC_INLINE c89atomic_uint16 __stdcall c89atomic_exchange_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
- {
- (void)order;
- return (c89atomic_uint16)_InterlockedExchange16((volatile short*)dst, (short)src);
- }
- #endif
- #if defined(C89ATOMIC_HAS_32)
- static C89ATOMIC_INLINE c89atomic_uint32 __stdcall c89atomic_exchange_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
- {
- (void)order;
- return (c89atomic_uint32)_InterlockedExchange((volatile long*)dst, (long)src);
- }
- #endif
- #if defined(C89ATOMIC_HAS_64) && defined(C89ATOMIC_64BIT)
- static C89ATOMIC_INLINE c89atomic_uint64 __stdcall c89atomic_exchange_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
- {
- (void)order;
- return (c89atomic_uint64)_InterlockedExchange64((volatile long long*)dst, (long long)src);
- }
- #else
- #endif
- #endif
- #if defined(C89ATOMIC_HAS_64) && !defined(C89ATOMIC_64BIT)
- static C89ATOMIC_INLINE c89atomic_uint64 __stdcall c89atomic_exchange_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
- {
- c89atomic_uint64 oldValue;
- do {
- oldValue = *dst;
- } while (c89atomic_compare_and_swap_64(dst, oldValue, src) != oldValue);
- (void)order;
- return oldValue;
+
+static ma_result ma_wait_for_operation__pulse(ma_context* pContext, ma_ptr pMainLoop, ma_pa_operation* pOP)
+{
+ int resultPA;
+ ma_pa_operation_state_t state;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pOP != NULL);
+
+ for (;;) {
+ state = ((ma_pa_operation_get_state_proc)pContext->pulse.pa_operation_get_state)(pOP);
+ if (state != MA_PA_OPERATION_RUNNING) {
+ break; /* Done. */
}
- #endif
- #if defined(C89ATOMIC_MSVC_USE_INLINED_ASSEMBLY)
- #if defined(C89ATOMIC_HAS_8)
- static C89ATOMIC_INLINE c89atomic_uint8 __stdcall c89atomic_fetch_add_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
- {
- c89atomic_uint8 result = 0;
- (void)order;
- __asm {
- mov ecx, dst
- mov al, src
- lock xadd [ecx], al
- mov result, al
- }
- return result;
- }
- #endif
- #if defined(C89ATOMIC_HAS_16)
- static C89ATOMIC_INLINE c89atomic_uint16 __stdcall c89atomic_fetch_add_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
- {
- c89atomic_uint16 result = 0;
- (void)order;
- __asm {
- mov ecx, dst
- mov ax, src
- lock xadd [ecx], ax
- mov result, ax
- }
- return result;
- }
- #endif
- #if defined(C89ATOMIC_HAS_32)
- static C89ATOMIC_INLINE c89atomic_uint32 __stdcall c89atomic_fetch_add_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
- {
- c89atomic_uint32 result = 0;
- (void)order;
- __asm {
- mov ecx, dst
- mov eax, src
- lock xadd [ecx], eax
- mov result, eax
- }
- return result;
- }
- #endif
- #else
- #if defined(C89ATOMIC_HAS_8)
- static C89ATOMIC_INLINE c89atomic_uint8 __stdcall c89atomic_fetch_add_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
- {
- (void)order;
- return (c89atomic_uint8)_InterlockedExchangeAdd8((volatile char*)dst, (char)src);
- }
- #endif
- #if defined(C89ATOMIC_HAS_16)
- static C89ATOMIC_INLINE c89atomic_uint16 __stdcall c89atomic_fetch_add_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
- {
- (void)order;
- return (c89atomic_uint16)_InterlockedExchangeAdd16((volatile short*)dst, (short)src);
- }
- #endif
- #if defined(C89ATOMIC_HAS_32)
- static C89ATOMIC_INLINE c89atomic_uint32 __stdcall c89atomic_fetch_add_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
- {
- (void)order;
- return (c89atomic_uint32)_InterlockedExchangeAdd((volatile long*)dst, (long)src);
- }
- #endif
- #if defined(C89ATOMIC_HAS_64) && defined(C89ATOMIC_64BIT)
- static C89ATOMIC_INLINE c89atomic_uint64 __stdcall c89atomic_fetch_add_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
- {
- (void)order;
- return (c89atomic_uint64)_InterlockedExchangeAdd64((volatile long long*)dst, (long long)src);
- }
- #else
- #endif
- #endif
- #if defined(C89ATOMIC_HAS_64) && !defined(C89ATOMIC_64BIT)
- static C89ATOMIC_INLINE c89atomic_uint64 __stdcall c89atomic_fetch_add_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
- {
- c89atomic_uint64 oldValue;
- c89atomic_uint64 newValue;
- do {
- oldValue = *dst;
- newValue = oldValue + src;
- } while (c89atomic_compare_and_swap_64(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ resultPA = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pMainLoop, 1, NULL);
+ if (resultPA < 0) {
+ return ma_result_from_pulse(resultPA);
}
- #endif
- #if defined(C89ATOMIC_MSVC_USE_INLINED_ASSEMBLY)
- static C89ATOMIC_INLINE void __stdcall c89atomic_thread_fence(c89atomic_memory_order order)
- {
- (void)order;
- __asm {
- lock add [esp], 0
- }
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_wait_for_operation_and_unref__pulse(ma_context* pContext, ma_ptr pMainLoop, ma_pa_operation* pOP)
+{
+ ma_result result;
+
+ if (pOP == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ result = ma_wait_for_operation__pulse(pContext, pMainLoop, pOP);
+ ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP);
+
+ return result;
+}
+
+static ma_result ma_wait_for_pa_context_to_connect__pulse(ma_context* pContext, ma_ptr pMainLoop, ma_ptr pPulseContext)
+{
+ int resultPA;
+ ma_pa_context_state_t state;
+
+ for (;;) {
+ state = ((ma_pa_context_get_state_proc)pContext->pulse.pa_context_get_state)((ma_pa_context*)pPulseContext);
+ if (state == MA_PA_CONTEXT_READY) {
+ break; /* Done. */
}
- #else
- #if defined(C89ATOMIC_X64)
- #define c89atomic_thread_fence(order) __faststorefence(), (void)order
- #else
- static C89ATOMIC_INLINE void c89atomic_thread_fence(c89atomic_memory_order order)
- {
- volatile c89atomic_uint32 barrier = 0;
- c89atomic_fetch_add_explicit_32(&barrier, 0, order);
- }
- #endif
- #endif
- #define c89atomic_compiler_fence() c89atomic_thread_fence(c89atomic_memory_order_seq_cst)
- #define c89atomic_signal_fence(order) c89atomic_thread_fence(order)
- #if defined(C89ATOMIC_HAS_8)
- static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_load_explicit_8(volatile const c89atomic_uint8* ptr, c89atomic_memory_order order)
- {
- (void)order;
- return c89atomic_compare_and_swap_8((volatile c89atomic_uint8*)ptr, 0, 0);
+
+ if (state == MA_PA_CONTEXT_FAILED || state == MA_PA_CONTEXT_TERMINATED) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[PulseAudio] An error occurred while connecting the PulseAudio context.");
+ return MA_ERROR;
}
- #endif
- #if defined(C89ATOMIC_HAS_16)
- static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_load_explicit_16(volatile const c89atomic_uint16* ptr, c89atomic_memory_order order)
- {
- (void)order;
- return c89atomic_compare_and_swap_16((volatile c89atomic_uint16*)ptr, 0, 0);
+
+ resultPA = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pMainLoop, 1, NULL);
+ if (resultPA < 0) {
+ return ma_result_from_pulse(resultPA);
}
- #endif
- #if defined(C89ATOMIC_HAS_32)
- static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_load_explicit_32(volatile const c89atomic_uint32* ptr, c89atomic_memory_order order)
- {
- (void)order;
- return c89atomic_compare_and_swap_32((volatile c89atomic_uint32*)ptr, 0, 0);
+ }
+
+ /* Should never get here. */
+ return MA_SUCCESS;
+}
+
+static ma_result ma_wait_for_pa_stream_to_connect__pulse(ma_context* pContext, ma_ptr pMainLoop, ma_ptr pStream)
+{
+ int resultPA;
+ ma_pa_stream_state_t state;
+
+ for (;;) {
+ state = ((ma_pa_stream_get_state_proc)pContext->pulse.pa_stream_get_state)((ma_pa_stream*)pStream);
+ if (state == MA_PA_STREAM_READY) {
+ break; /* Done. */
}
- #endif
- #if defined(C89ATOMIC_HAS_64)
- static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_load_explicit_64(volatile const c89atomic_uint64* ptr, c89atomic_memory_order order)
- {
- (void)order;
- return c89atomic_compare_and_swap_64((volatile c89atomic_uint64*)ptr, 0, 0);
+
+ if (state == MA_PA_STREAM_FAILED || state == MA_PA_STREAM_TERMINATED) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[PulseAudio] An error occurred while connecting the PulseAudio stream.");
+ return MA_ERROR;
+ }
+
+ resultPA = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pMainLoop, 1, NULL);
+ if (resultPA < 0) {
+ return ma_result_from_pulse(resultPA);
+ }
+ }
+
+ return MA_SUCCESS;
+}
+
+
+static ma_result ma_init_pa_mainloop_and_pa_context__pulse(ma_context* pContext, const char* pApplicationName, const char* pServerName, ma_bool32 tryAutoSpawn, ma_ptr* ppMainLoop, ma_ptr* ppPulseContext)
+{
+ ma_result result;
+ ma_ptr pMainLoop;
+ ma_ptr pPulseContext;
+
+ MA_ASSERT(ppMainLoop != NULL);
+ MA_ASSERT(ppPulseContext != NULL);
+
+ /* The PulseAudio context maps well to miniaudio's notion of a context. The pa_context object will be initialized as part of the ma_context. */
+ pMainLoop = ((ma_pa_mainloop_new_proc)pContext->pulse.pa_mainloop_new)();
+ if (pMainLoop == NULL) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create mainloop.");
+ return MA_FAILED_TO_INIT_BACKEND;
+ }
+
+ pPulseContext = ((ma_pa_context_new_proc)pContext->pulse.pa_context_new)(((ma_pa_mainloop_get_api_proc)pContext->pulse.pa_mainloop_get_api)((ma_pa_mainloop*)pMainLoop), pApplicationName);
+ if (pPulseContext == NULL) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create PulseAudio context.");
+ ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)((ma_pa_mainloop*)(pMainLoop));
+ return MA_FAILED_TO_INIT_BACKEND;
+ }
+
+ /* Now we need to connect to the context. Everything is asynchronous so we need to wait for it to connect before returning. */
+ result = ma_result_from_pulse(((ma_pa_context_connect_proc)pContext->pulse.pa_context_connect)((ma_pa_context*)pPulseContext, pServerName, (tryAutoSpawn) ? 0 : MA_PA_CONTEXT_NOAUTOSPAWN, NULL));
+ if (result != MA_SUCCESS) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to connect PulseAudio context.");
+ ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)((ma_pa_mainloop*)(pMainLoop));
+ return result;
+ }
+
+ /* Since ma_context_init() runs synchronously we need to wait for the PulseAudio context to connect before we return. */
+ result = ma_wait_for_pa_context_to_connect__pulse(pContext, pMainLoop, pPulseContext);
+ if (result != MA_SUCCESS) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[PulseAudio] Waiting for connection failed.");
+ ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)((ma_pa_mainloop*)(pMainLoop));
+ return result;
+ }
+
+ *ppMainLoop = pMainLoop;
+ *ppPulseContext = pPulseContext;
+
+ return MA_SUCCESS;
+}
+
+
+static void ma_device_sink_info_callback(ma_pa_context* pPulseContext, const ma_pa_sink_info* pInfo, int endOfList, void* pUserData)
+{
+ ma_pa_sink_info* pInfoOut;
+
+ if (endOfList > 0) {
+ return;
+ }
+
+ pInfoOut = (ma_pa_sink_info*)pUserData;
+ MA_ASSERT(pInfoOut != NULL);
+
+ *pInfoOut = *pInfo;
+
+ (void)pPulseContext; /* Unused. */
+}
+
+static void ma_device_source_info_callback(ma_pa_context* pPulseContext, const ma_pa_source_info* pInfo, int endOfList, void* pUserData)
+{
+ ma_pa_source_info* pInfoOut;
+
+ if (endOfList > 0) {
+ return;
+ }
+
+ pInfoOut = (ma_pa_source_info*)pUserData;
+ MA_ASSERT(pInfoOut != NULL);
+
+ *pInfoOut = *pInfo;
+
+ (void)pPulseContext; /* Unused. */
+}
+
+#if 0
+static void ma_device_sink_name_callback(ma_pa_context* pPulseContext, const ma_pa_sink_info* pInfo, int endOfList, void* pUserData)
+{
+ ma_device* pDevice;
+
+ if (endOfList > 0) {
+ return;
+ }
+
+ pDevice = (ma_device*)pUserData;
+ MA_ASSERT(pDevice != NULL);
+
+ ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), pInfo->description, (size_t)-1);
+
+ (void)pPulseContext; /* Unused. */
+}
+
+static void ma_device_source_name_callback(ma_pa_context* pPulseContext, const ma_pa_source_info* pInfo, int endOfList, void* pUserData)
+{
+ ma_device* pDevice;
+
+ if (endOfList > 0) {
+ return;
+ }
+
+ pDevice = (ma_device*)pUserData;
+ MA_ASSERT(pDevice != NULL);
+
+ ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), pInfo->description, (size_t)-1);
+
+ (void)pPulseContext; /* Unused. */
+}
+#endif
+
+static ma_result ma_context_get_sink_info__pulse(ma_context* pContext, const char* pDeviceName, ma_pa_sink_info* pSinkInfo)
+{
+ ma_pa_operation* pOP;
+
+ pOP = ((ma_pa_context_get_sink_info_by_name_proc)pContext->pulse.pa_context_get_sink_info_by_name)((ma_pa_context*)pContext->pulse.pPulseContext, pDeviceName, ma_device_sink_info_callback, pSinkInfo);
+ if (pOP == NULL) {
+ return MA_ERROR;
+ }
+
+ return ma_wait_for_operation_and_unref__pulse(pContext, pContext->pulse.pMainLoop, pOP);
+}
+
+static ma_result ma_context_get_source_info__pulse(ma_context* pContext, const char* pDeviceName, ma_pa_source_info* pSourceInfo)
+{
+ ma_pa_operation* pOP;
+
+ pOP = ((ma_pa_context_get_source_info_by_name_proc)pContext->pulse.pa_context_get_source_info_by_name)((ma_pa_context*)pContext->pulse.pPulseContext, pDeviceName, ma_device_source_info_callback, pSourceInfo);
+ if (pOP == NULL) {
+ return MA_ERROR;
+ }
+
+ return ma_wait_for_operation_and_unref__pulse(pContext, pContext->pulse.pMainLoop, pOP);
+}
+
+static ma_result ma_context_get_default_device_index__pulse(ma_context* pContext, ma_device_type deviceType, ma_uint32* pIndex)
+{
+ ma_result result;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pIndex != NULL);
+
+ if (pIndex != NULL) {
+ *pIndex = (ma_uint32)-1;
+ }
+
+ if (deviceType == ma_device_type_playback) {
+ ma_pa_sink_info sinkInfo;
+ result = ma_context_get_sink_info__pulse(pContext, NULL, &sinkInfo);
+ if (result != MA_SUCCESS) {
+ return result;
}
- #endif
- #if defined(C89ATOMIC_HAS_8)
- #define c89atomic_store_explicit_8( dst, src, order) (void)c89atomic_exchange_explicit_8 (dst, src, order)
- #endif
- #if defined(C89ATOMIC_HAS_16)
- #define c89atomic_store_explicit_16(dst, src, order) (void)c89atomic_exchange_explicit_16(dst, src, order)
- #endif
- #if defined(C89ATOMIC_HAS_32)
- #define c89atomic_store_explicit_32(dst, src, order) (void)c89atomic_exchange_explicit_32(dst, src, order)
- #endif
- #if defined(C89ATOMIC_HAS_64)
- #define c89atomic_store_explicit_64(dst, src, order) (void)c89atomic_exchange_explicit_64(dst, src, order)
- #endif
- #if defined(C89ATOMIC_HAS_8)
- static C89ATOMIC_INLINE c89atomic_uint8 __stdcall c89atomic_fetch_sub_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
- {
- c89atomic_uint8 oldValue;
- c89atomic_uint8 newValue;
- do {
- oldValue = *dst;
- newValue = (c89atomic_uint8)(oldValue - src);
- } while (c89atomic_compare_and_swap_8(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ if (pIndex != NULL) {
+ *pIndex = sinkInfo.index;
}
- #endif
- #if defined(C89ATOMIC_HAS_16)
- static C89ATOMIC_INLINE c89atomic_uint16 __stdcall c89atomic_fetch_sub_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
- {
- c89atomic_uint16 oldValue;
- c89atomic_uint16 newValue;
- do {
- oldValue = *dst;
- newValue = (c89atomic_uint16)(oldValue - src);
- } while (c89atomic_compare_and_swap_16(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+ }
+
+ if (deviceType == ma_device_type_capture) {
+ ma_pa_source_info sourceInfo;
+ result = ma_context_get_source_info__pulse(pContext, NULL, &sourceInfo);
+ if (result != MA_SUCCESS) {
+ return result;
}
- #endif
- #if defined(C89ATOMIC_HAS_32)
- static C89ATOMIC_INLINE c89atomic_uint32 __stdcall c89atomic_fetch_sub_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
- {
- c89atomic_uint32 oldValue;
- c89atomic_uint32 newValue;
- do {
- oldValue = *dst;
- newValue = oldValue - src;
- } while (c89atomic_compare_and_swap_32(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ if (pIndex != NULL) {
+ *pIndex = sourceInfo.index;
}
- #endif
- #if defined(C89ATOMIC_HAS_64)
- static C89ATOMIC_INLINE c89atomic_uint64 __stdcall c89atomic_fetch_sub_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
- {
- c89atomic_uint64 oldValue;
- c89atomic_uint64 newValue;
- do {
- oldValue = *dst;
- newValue = oldValue - src;
- } while (c89atomic_compare_and_swap_64(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+ }
+
+ return MA_SUCCESS;
+}
+
+
+typedef struct
+{
+ ma_context* pContext;
+ ma_enum_devices_callback_proc callback;
+ void* pUserData;
+ ma_bool32 isTerminated;
+ ma_uint32 defaultDeviceIndexPlayback;
+ ma_uint32 defaultDeviceIndexCapture;
+} ma_context_enumerate_devices_callback_data__pulse;
+
+static void ma_context_enumerate_devices_sink_callback__pulse(ma_pa_context* pPulseContext, const ma_pa_sink_info* pSinkInfo, int endOfList, void* pUserData)
+{
+ ma_context_enumerate_devices_callback_data__pulse* pData = (ma_context_enumerate_devices_callback_data__pulse*)pUserData;
+ ma_device_info deviceInfo;
+
+ MA_ASSERT(pData != NULL);
+
+ if (endOfList || pData->isTerminated) {
+ return;
+ }
+
+ MA_ZERO_OBJECT(&deviceInfo);
+
+ /* The name from PulseAudio is the ID for miniaudio. */
+ if (pSinkInfo->name != NULL) {
+ ma_strncpy_s(deviceInfo.id.pulse, sizeof(deviceInfo.id.pulse), pSinkInfo->name, (size_t)-1);
+ }
+
+ /* The description from PulseAudio is the name for miniaudio. */
+ if (pSinkInfo->description != NULL) {
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), pSinkInfo->description, (size_t)-1);
+ }
+
+ if (pSinkInfo->index == pData->defaultDeviceIndexPlayback) {
+ deviceInfo.isDefault = MA_TRUE;
+ }
+
+ pData->isTerminated = !pData->callback(pData->pContext, ma_device_type_playback, &deviceInfo, pData->pUserData);
+
+ (void)pPulseContext; /* Unused. */
+}
+
+static void ma_context_enumerate_devices_source_callback__pulse(ma_pa_context* pPulseContext, const ma_pa_source_info* pSourceInfo, int endOfList, void* pUserData)
+{
+ ma_context_enumerate_devices_callback_data__pulse* pData = (ma_context_enumerate_devices_callback_data__pulse*)pUserData;
+ ma_device_info deviceInfo;
+
+ MA_ASSERT(pData != NULL);
+
+ if (endOfList || pData->isTerminated) {
+ return;
+ }
+
+ MA_ZERO_OBJECT(&deviceInfo);
+
+ /* The name from PulseAudio is the ID for miniaudio. */
+ if (pSourceInfo->name != NULL) {
+ ma_strncpy_s(deviceInfo.id.pulse, sizeof(deviceInfo.id.pulse), pSourceInfo->name, (size_t)-1);
+ }
+
+ /* The description from PulseAudio is the name for miniaudio. */
+ if (pSourceInfo->description != NULL) {
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), pSourceInfo->description, (size_t)-1);
+ }
+
+ if (pSourceInfo->index == pData->defaultDeviceIndexCapture) {
+ deviceInfo.isDefault = MA_TRUE;
+ }
+
+ pData->isTerminated = !pData->callback(pData->pContext, ma_device_type_capture, &deviceInfo, pData->pUserData);
+
+ (void)pPulseContext; /* Unused. */
+}
+
+static ma_result ma_context_enumerate_devices__pulse(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+{
+ ma_result result = MA_SUCCESS;
+ ma_context_enumerate_devices_callback_data__pulse callbackData;
+ ma_pa_operation* pOP = NULL;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(callback != NULL);
+
+ callbackData.pContext = pContext;
+ callbackData.callback = callback;
+ callbackData.pUserData = pUserData;
+ callbackData.isTerminated = MA_FALSE;
+ callbackData.defaultDeviceIndexPlayback = (ma_uint32)-1;
+ callbackData.defaultDeviceIndexCapture = (ma_uint32)-1;
+
+ /* We need to get the index of the default devices. */
+ ma_context_get_default_device_index__pulse(pContext, ma_device_type_playback, &callbackData.defaultDeviceIndexPlayback);
+ ma_context_get_default_device_index__pulse(pContext, ma_device_type_capture, &callbackData.defaultDeviceIndexCapture);
+
+ /* Playback. */
+ if (!callbackData.isTerminated) {
+ pOP = ((ma_pa_context_get_sink_info_list_proc)pContext->pulse.pa_context_get_sink_info_list)((ma_pa_context*)(pContext->pulse.pPulseContext), ma_context_enumerate_devices_sink_callback__pulse, &callbackData);
+ if (pOP == NULL) {
+ result = MA_ERROR;
+ goto done;
}
- #endif
- #if defined(C89ATOMIC_HAS_8)
- static C89ATOMIC_INLINE c89atomic_uint8 __stdcall c89atomic_fetch_and_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
- {
- c89atomic_uint8 oldValue;
- c89atomic_uint8 newValue;
- do {
- oldValue = *dst;
- newValue = (c89atomic_uint8)(oldValue & src);
- } while (c89atomic_compare_and_swap_8(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ result = ma_wait_for_operation__pulse(pContext, pContext->pulse.pMainLoop, pOP);
+ ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP);
+
+ if (result != MA_SUCCESS) {
+ goto done;
}
- #endif
- #if defined(C89ATOMIC_HAS_16)
- static C89ATOMIC_INLINE c89atomic_uint16 __stdcall c89atomic_fetch_and_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
- {
- c89atomic_uint16 oldValue;
- c89atomic_uint16 newValue;
- do {
- oldValue = *dst;
- newValue = (c89atomic_uint16)(oldValue & src);
- } while (c89atomic_compare_and_swap_16(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+ }
+
+
+ /* Capture. */
+ if (!callbackData.isTerminated) {
+ pOP = ((ma_pa_context_get_source_info_list_proc)pContext->pulse.pa_context_get_source_info_list)((ma_pa_context*)(pContext->pulse.pPulseContext), ma_context_enumerate_devices_source_callback__pulse, &callbackData);
+ if (pOP == NULL) {
+ result = MA_ERROR;
+ goto done;
}
- #endif
- #if defined(C89ATOMIC_HAS_32)
- static C89ATOMIC_INLINE c89atomic_uint32 __stdcall c89atomic_fetch_and_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
- {
- c89atomic_uint32 oldValue;
- c89atomic_uint32 newValue;
- do {
- oldValue = *dst;
- newValue = oldValue & src;
- } while (c89atomic_compare_and_swap_32(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ result = ma_wait_for_operation__pulse(pContext, pContext->pulse.pMainLoop, pOP);
+ ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP);
+
+ if (result != MA_SUCCESS) {
+ goto done;
}
- #endif
- #if defined(C89ATOMIC_HAS_64)
- static C89ATOMIC_INLINE c89atomic_uint64 __stdcall c89atomic_fetch_and_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
- {
- c89atomic_uint64 oldValue;
- c89atomic_uint64 newValue;
- do {
- oldValue = *dst;
- newValue = oldValue & src;
- } while (c89atomic_compare_and_swap_64(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+ }
+
+done:
+ return result;
+}
+
+
+typedef struct
+{
+ ma_device_info* pDeviceInfo;
+ ma_uint32 defaultDeviceIndex;
+ ma_bool32 foundDevice;
+} ma_context_get_device_info_callback_data__pulse;
+
+static void ma_context_get_device_info_sink_callback__pulse(ma_pa_context* pPulseContext, const ma_pa_sink_info* pInfo, int endOfList, void* pUserData)
+{
+ ma_context_get_device_info_callback_data__pulse* pData = (ma_context_get_device_info_callback_data__pulse*)pUserData;
+
+ if (endOfList > 0) {
+ return;
+ }
+
+ MA_ASSERT(pData != NULL);
+ pData->foundDevice = MA_TRUE;
+
+ if (pInfo->name != NULL) {
+ ma_strncpy_s(pData->pDeviceInfo->id.pulse, sizeof(pData->pDeviceInfo->id.pulse), pInfo->name, (size_t)-1);
+ }
+
+ if (pInfo->description != NULL) {
+ ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), pInfo->description, (size_t)-1);
+ }
+
+ /*
+ We're just reporting a single data format here. I think technically PulseAudio might support
+ all formats, but I don't trust that PulseAudio will do *anything* right, so I'm just going to
+ report the "native" device format.
+ */
+ pData->pDeviceInfo->nativeDataFormats[0].format = ma_format_from_pulse(pInfo->sample_spec.format);
+ pData->pDeviceInfo->nativeDataFormats[0].channels = pInfo->sample_spec.channels;
+ pData->pDeviceInfo->nativeDataFormats[0].sampleRate = pInfo->sample_spec.rate;
+ pData->pDeviceInfo->nativeDataFormats[0].flags = 0;
+ pData->pDeviceInfo->nativeDataFormatCount = 1;
+
+ if (pData->defaultDeviceIndex == pInfo->index) {
+ pData->pDeviceInfo->isDefault = MA_TRUE;
+ }
+
+ (void)pPulseContext; /* Unused. */
+}
+
+static void ma_context_get_device_info_source_callback__pulse(ma_pa_context* pPulseContext, const ma_pa_source_info* pInfo, int endOfList, void* pUserData)
+{
+ ma_context_get_device_info_callback_data__pulse* pData = (ma_context_get_device_info_callback_data__pulse*)pUserData;
+
+ if (endOfList > 0) {
+ return;
+ }
+
+ MA_ASSERT(pData != NULL);
+ pData->foundDevice = MA_TRUE;
+
+ if (pInfo->name != NULL) {
+ ma_strncpy_s(pData->pDeviceInfo->id.pulse, sizeof(pData->pDeviceInfo->id.pulse), pInfo->name, (size_t)-1);
+ }
+
+ if (pInfo->description != NULL) {
+ ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), pInfo->description, (size_t)-1);
+ }
+
+ /*
+ We're just reporting a single data format here. I think technically PulseAudio might support
+ all formats, but I don't trust that PulseAudio will do *anything* right, so I'm just going to
+ report the "native" device format.
+ */
+ pData->pDeviceInfo->nativeDataFormats[0].format = ma_format_from_pulse(pInfo->sample_spec.format);
+ pData->pDeviceInfo->nativeDataFormats[0].channels = pInfo->sample_spec.channels;
+ pData->pDeviceInfo->nativeDataFormats[0].sampleRate = pInfo->sample_spec.rate;
+ pData->pDeviceInfo->nativeDataFormats[0].flags = 0;
+ pData->pDeviceInfo->nativeDataFormatCount = 1;
+
+ if (pData->defaultDeviceIndex == pInfo->index) {
+ pData->pDeviceInfo->isDefault = MA_TRUE;
+ }
+
+ (void)pPulseContext; /* Unused. */
+}
+
+static ma_result ma_context_get_device_info__pulse(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+{
+ ma_result result = MA_SUCCESS;
+ ma_context_get_device_info_callback_data__pulse callbackData;
+ ma_pa_operation* pOP = NULL;
+ const char* pDeviceName = NULL;
+
+ MA_ASSERT(pContext != NULL);
+
+ callbackData.pDeviceInfo = pDeviceInfo;
+ callbackData.foundDevice = MA_FALSE;
+
+ if (pDeviceID != NULL) {
+ pDeviceName = pDeviceID->pulse;
+ } else {
+ pDeviceName = NULL;
+ }
+
+ result = ma_context_get_default_device_index__pulse(pContext, deviceType, &callbackData.defaultDeviceIndex);
+
+ if (deviceType == ma_device_type_playback) {
+ pOP = ((ma_pa_context_get_sink_info_by_name_proc)pContext->pulse.pa_context_get_sink_info_by_name)((ma_pa_context*)(pContext->pulse.pPulseContext), pDeviceName, ma_context_get_device_info_sink_callback__pulse, &callbackData);
+ } else {
+ pOP = ((ma_pa_context_get_source_info_by_name_proc)pContext->pulse.pa_context_get_source_info_by_name)((ma_pa_context*)(pContext->pulse.pPulseContext), pDeviceName, ma_context_get_device_info_source_callback__pulse, &callbackData);
+ }
+
+ if (pOP != NULL) {
+ ma_wait_for_operation_and_unref__pulse(pContext, pContext->pulse.pMainLoop, pOP);
+ } else {
+ result = MA_ERROR;
+ goto done;
+ }
+
+ if (!callbackData.foundDevice) {
+ result = MA_NO_DEVICE;
+ goto done;
+ }
+
+done:
+ return result;
+}
+
+static ma_result ma_device_uninit__pulse(ma_device* pDevice)
+{
+ ma_context* pContext;
+
+ MA_ASSERT(pDevice != NULL);
+
+ pContext = pDevice->pContext;
+ MA_ASSERT(pContext != NULL);
+
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ ((ma_pa_stream_disconnect_proc)pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
+ ((ma_pa_stream_unref_proc)pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
+ }
+
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ ((ma_pa_stream_disconnect_proc)pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
+ ((ma_pa_stream_unref_proc)pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
+ }
+
+ if (pDevice->type == ma_device_type_duplex) {
+ ma_duplex_rb_uninit(&pDevice->duplexRB);
+ }
+
+ ((ma_pa_context_disconnect_proc)pContext->pulse.pa_context_disconnect)((ma_pa_context*)pDevice->pulse.pPulseContext);
+ ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)((ma_pa_context*)pDevice->pulse.pPulseContext);
+ ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)((ma_pa_mainloop*)pDevice->pulse.pMainLoop);
+
+ return MA_SUCCESS;
+}
+
+static ma_pa_buffer_attr ma_device__pa_buffer_attr_new(ma_uint32 periodSizeInFrames, ma_uint32 periods, const ma_pa_sample_spec* ss)
+{
+ ma_pa_buffer_attr attr;
+ attr.maxlength = periodSizeInFrames * periods * ma_get_bytes_per_frame(ma_format_from_pulse(ss->format), ss->channels);
+ attr.tlength = attr.maxlength / periods;
+ attr.prebuf = (ma_uint32)-1;
+ attr.minreq = (ma_uint32)-1;
+ attr.fragsize = attr.maxlength / periods;
+
+ return attr;
+}
+
+static ma_pa_stream* ma_device__pa_stream_new__pulse(ma_device* pDevice, const char* pStreamName, const ma_pa_sample_spec* ss, const ma_pa_channel_map* cmap)
+{
+ static int g_StreamCounter = 0;
+ char actualStreamName[256];
+
+ if (pStreamName != NULL) {
+ ma_strncpy_s(actualStreamName, sizeof(actualStreamName), pStreamName, (size_t)-1);
+ } else {
+ ma_strcpy_s(actualStreamName, sizeof(actualStreamName), "miniaudio:");
+ ma_itoa_s(g_StreamCounter, actualStreamName + 8, sizeof(actualStreamName)-8, 10); /* 8 = strlen("miniaudio:") */
+ }
+ g_StreamCounter += 1;
+
+ return ((ma_pa_stream_new_proc)pDevice->pContext->pulse.pa_stream_new)((ma_pa_context*)pDevice->pulse.pPulseContext, actualStreamName, ss, cmap);
+}
+
+
+static void ma_device_on_read__pulse(ma_pa_stream* pStream, size_t byteCount, void* pUserData)
+{
+ ma_device* pDevice = (ma_device*)pUserData;
+ ma_uint32 bpf;
+ ma_uint32 deviceState;
+ ma_uint64 frameCount;
+ ma_uint64 framesProcessed;
+
+ MA_ASSERT(pDevice != NULL);
+
+ /*
+ Don't do anything if the device isn't initialized yet. Yes, this can happen because PulseAudio
+ can fire this callback before the stream has even started. Ridiculous.
+ */
+ deviceState = ma_device_get_state(pDevice);
+ if (deviceState != ma_device_state_starting && deviceState != ma_device_state_started) {
+ return;
+ }
+
+ bpf = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
+ MA_ASSERT(bpf > 0);
+
+ frameCount = byteCount / bpf;
+ framesProcessed = 0;
+
+ while (ma_device_get_state(pDevice) == ma_device_state_started && framesProcessed < frameCount) {
+ const void* pMappedPCMFrames;
+ size_t bytesMapped;
+ ma_uint64 framesMapped;
+
+ int pulseResult = ((ma_pa_stream_peek_proc)pDevice->pContext->pulse.pa_stream_peek)(pStream, &pMappedPCMFrames, &bytesMapped);
+ if (pulseResult < 0) {
+ break; /* Failed to map. Abort. */
}
- #endif
- #if defined(C89ATOMIC_HAS_8)
- static C89ATOMIC_INLINE c89atomic_uint8 __stdcall c89atomic_fetch_xor_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
- {
- c89atomic_uint8 oldValue;
- c89atomic_uint8 newValue;
- do {
- oldValue = *dst;
- newValue = (c89atomic_uint8)(oldValue ^ src);
- } while (c89atomic_compare_and_swap_8(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ framesMapped = bytesMapped / bpf;
+ if (framesMapped > 0) {
+ if (pMappedPCMFrames != NULL) {
+ ma_device_handle_backend_data_callback(pDevice, NULL, pMappedPCMFrames, framesMapped);
+ } else {
+ /* It's a hole. */
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[PulseAudio] ma_device_on_read__pulse: Hole.\n");
+ }
+
+ pulseResult = ((ma_pa_stream_drop_proc)pDevice->pContext->pulse.pa_stream_drop)(pStream);
+ if (pulseResult < 0) {
+ break; /* Failed to drop the buffer. */
+ }
+
+ framesProcessed += framesMapped;
+
+ } else {
+ /* Nothing was mapped. Just abort. */
+ break;
}
- #endif
- #if defined(C89ATOMIC_HAS_16)
- static C89ATOMIC_INLINE c89atomic_uint16 __stdcall c89atomic_fetch_xor_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
- {
- c89atomic_uint16 oldValue;
- c89atomic_uint16 newValue;
- do {
- oldValue = *dst;
- newValue = (c89atomic_uint16)(oldValue ^ src);
- } while (c89atomic_compare_and_swap_16(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+ }
+}
+
+static ma_result ma_device_write_to_stream__pulse(ma_device* pDevice, ma_pa_stream* pStream, ma_uint64* pFramesProcessed)
+{
+ ma_result result = MA_SUCCESS;
+ ma_uint64 framesProcessed = 0;
+ size_t bytesMapped;
+ ma_uint32 bpf;
+ ma_uint32 deviceState;
+
+ MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(pStream != NULL);
+
+ bpf = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
+ MA_ASSERT(bpf > 0);
+
+ deviceState = ma_device_get_state(pDevice);
+
+ bytesMapped = ((ma_pa_stream_writable_size_proc)pDevice->pContext->pulse.pa_stream_writable_size)(pStream);
+ if (bytesMapped != (size_t)-1) {
+ if (bytesMapped > 0) {
+ ma_uint64 framesMapped;
+ void* pMappedPCMFrames;
+ int pulseResult = ((ma_pa_stream_begin_write_proc)pDevice->pContext->pulse.pa_stream_begin_write)(pStream, &pMappedPCMFrames, &bytesMapped);
+ if (pulseResult < 0) {
+ result = ma_result_from_pulse(pulseResult);
+ goto done;
+ }
+
+ framesMapped = bytesMapped / bpf;
+
+ if (deviceState == ma_device_state_started || deviceState == ma_device_state_starting) { /* Check for starting state just in case this is being used to do the initial fill. */
+ ma_device_handle_backend_data_callback(pDevice, pMappedPCMFrames, NULL, framesMapped);
+ } else {
+ /* Device is not started. Write silence. */
+ ma_silence_pcm_frames(pMappedPCMFrames, framesMapped, pDevice->playback.format, pDevice->playback.channels);
+ }
+
+ pulseResult = ((ma_pa_stream_write_proc)pDevice->pContext->pulse.pa_stream_write)(pStream, pMappedPCMFrames, bytesMapped, NULL, 0, MA_PA_SEEK_RELATIVE);
+ if (pulseResult < 0) {
+ result = ma_result_from_pulse(pulseResult);
+ goto done; /* Failed to write data to stream. */
+ }
+
+ framesProcessed += framesMapped;
+ } else {
+ result = MA_SUCCESS; /* No data available for writing. */
+ goto done;
}
- #endif
- #if defined(C89ATOMIC_HAS_32)
- static C89ATOMIC_INLINE c89atomic_uint32 __stdcall c89atomic_fetch_xor_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
- {
- c89atomic_uint32 oldValue;
- c89atomic_uint32 newValue;
- do {
- oldValue = *dst;
- newValue = oldValue ^ src;
- } while (c89atomic_compare_and_swap_32(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+ } else {
+ result = MA_ERROR; /* Failed to retrieve the writable size. Abort. */
+ goto done;
+ }
+
+done:
+ if (pFramesProcessed != NULL) {
+ *pFramesProcessed = framesProcessed;
+ }
+
+ return result;
+}
+
+static void ma_device_on_write__pulse(ma_pa_stream* pStream, size_t byteCount, void* pUserData)
+{
+ ma_device* pDevice = (ma_device*)pUserData;
+ ma_uint32 bpf;
+ ma_uint64 frameCount;
+ ma_uint64 framesProcessed;
+ ma_uint32 deviceState;
+ ma_result result;
+
+ MA_ASSERT(pDevice != NULL);
+
+ /*
+ Don't do anything if the device isn't initialized yet. Yes, this can happen because PulseAudio
+ can fire this callback before the stream has even started. Ridiculous.
+ */
+ deviceState = ma_device_get_state(pDevice);
+ if (deviceState != ma_device_state_starting && deviceState != ma_device_state_started) {
+ return;
+ }
+
+ bpf = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
+ MA_ASSERT(bpf > 0);
+
+ frameCount = byteCount / bpf;
+ framesProcessed = 0;
+
+ while (framesProcessed < frameCount) {
+ ma_uint64 framesProcessedThisIteration;
+
+ /* Don't keep trying to process frames if the device isn't started. */
+ deviceState = ma_device_get_state(pDevice);
+ if (deviceState != ma_device_state_starting && deviceState != ma_device_state_started) {
+ break;
}
- #endif
- #if defined(C89ATOMIC_HAS_64)
- static C89ATOMIC_INLINE c89atomic_uint64 __stdcall c89atomic_fetch_xor_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
- {
- c89atomic_uint64 oldValue;
- c89atomic_uint64 newValue;
- do {
- oldValue = *dst;
- newValue = oldValue ^ src;
- } while (c89atomic_compare_and_swap_64(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ result = ma_device_write_to_stream__pulse(pDevice, pStream, &framesProcessedThisIteration);
+ if (result != MA_SUCCESS) {
+ break;
}
- #endif
- #if defined(C89ATOMIC_HAS_8)
- static C89ATOMIC_INLINE c89atomic_uint8 __stdcall c89atomic_fetch_or_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
- {
- c89atomic_uint8 oldValue;
- c89atomic_uint8 newValue;
- do {
- oldValue = *dst;
- newValue = (c89atomic_uint8)(oldValue | src);
- } while (c89atomic_compare_and_swap_8(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ framesProcessed += framesProcessedThisIteration;
+ }
+}
+
+static void ma_device_on_suspended__pulse(ma_pa_stream* pStream, void* pUserData)
+{
+ ma_device* pDevice = (ma_device*)pUserData;
+ int suspended;
+
+ (void)pStream;
+
+ suspended = ((ma_pa_stream_is_suspended_proc)pDevice->pContext->pulse.pa_stream_is_suspended)(pStream);
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[Pulse] Device suspended state changed. pa_stream_is_suspended() returned %d.\n", suspended);
+
+ if (suspended < 0) {
+ return;
+ }
+
+ if (suspended == 1) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[Pulse] Device suspended state changed. Suspended.\n");
+ ma_device__on_notification_stopped(pDevice);
+ } else {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[Pulse] Device suspended state changed. Resumed.\n");
+ ma_device__on_notification_started(pDevice);
+ }
+}
+
+static void ma_device_on_rerouted__pulse(ma_pa_stream* pStream, void* pUserData)
+{
+ ma_device* pDevice = (ma_device*)pUserData;
+
+ (void)pStream;
+ (void)pUserData;
+
+ ma_device__on_notification_rerouted(pDevice);
+}
+
+static ma_uint32 ma_calculate_period_size_in_frames_from_descriptor__pulse(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile)
+{
+ /*
+ There have been reports from users where buffers of < ~20ms result glitches when running through
+ PipeWire. To work around this we're going to have to use a different default buffer size.
+ */
+ const ma_uint32 defaultPeriodSizeInMilliseconds_LowLatency = 25;
+ const ma_uint32 defaultPeriodSizeInMilliseconds_Conservative = MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE;
+
+ MA_ASSERT(nativeSampleRate != 0);
+
+ if (pDescriptor->periodSizeInFrames == 0) {
+ if (pDescriptor->periodSizeInMilliseconds == 0) {
+ if (performanceProfile == ma_performance_profile_low_latency) {
+ return ma_calculate_buffer_size_in_frames_from_milliseconds(defaultPeriodSizeInMilliseconds_LowLatency, nativeSampleRate);
+ } else {
+ return ma_calculate_buffer_size_in_frames_from_milliseconds(defaultPeriodSizeInMilliseconds_Conservative, nativeSampleRate);
+ }
+ } else {
+ return ma_calculate_buffer_size_in_frames_from_milliseconds(pDescriptor->periodSizeInMilliseconds, nativeSampleRate);
}
- #endif
- #if defined(C89ATOMIC_HAS_16)
- static C89ATOMIC_INLINE c89atomic_uint16 __stdcall c89atomic_fetch_or_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
- {
- c89atomic_uint16 oldValue;
- c89atomic_uint16 newValue;
- do {
- oldValue = *dst;
- newValue = (c89atomic_uint16)(oldValue | src);
- } while (c89atomic_compare_and_swap_16(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+ } else {
+ return pDescriptor->periodSizeInFrames;
+ }
+}
+
+static ma_result ma_device_init__pulse(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+{
+ /*
+ Notes for PulseAudio:
+
+ - We're always using native format/channels/rate regardless of whether or not PulseAudio
+ supports the format directly through their own data conversion system. I'm doing this to
+ reduce as much variability from the PulseAudio side as possible because it's seems to be
+ extremely unreliable at everything it does.
+
+ - When both the period size in frames and milliseconds are 0, we default to miniaudio's
+ default buffer sizes rather than leaving it up to PulseAudio because I don't trust
+ PulseAudio to give us any kind of reasonable latency by default.
+
+ - Do not ever, *ever* forget to use MA_PA_STREAM_ADJUST_LATENCY. If you don't specify this
+ flag, capture mode will just not work properly until you open another PulseAudio app.
+ */
+
+ ma_result result = MA_SUCCESS;
+ int error = 0;
+ const char* devPlayback = NULL;
+ const char* devCapture = NULL;
+ ma_format format = ma_format_unknown;
+ ma_uint32 channels = 0;
+ ma_uint32 sampleRate = 0;
+ ma_pa_sink_info sinkInfo;
+ ma_pa_source_info sourceInfo;
+ ma_pa_sample_spec ss;
+ ma_pa_channel_map cmap;
+ ma_pa_buffer_attr attr;
+ const ma_pa_sample_spec* pActualSS = NULL;
+ const ma_pa_channel_map* pActualCMap = NULL;
+ const ma_pa_buffer_attr* pActualAttr = NULL;
+ ma_uint32 iChannel;
+ ma_pa_stream_flags_t streamFlags;
+
+ MA_ASSERT(pDevice != NULL);
+ MA_ZERO_OBJECT(&pDevice->pulse);
+
+ if (pConfig->deviceType == ma_device_type_loopback) {
+ return MA_DEVICE_TYPE_NOT_SUPPORTED;
+ }
+
+ /* No exclusive mode with the PulseAudio backend. */
+ if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive) ||
+ ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive)) {
+ return MA_SHARE_MODE_NOT_SUPPORTED;
+ }
+
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ if (pDescriptorPlayback->pDeviceID != NULL) {
+ devPlayback = pDescriptorPlayback->pDeviceID->pulse;
}
- #endif
- #if defined(C89ATOMIC_HAS_32)
- static C89ATOMIC_INLINE c89atomic_uint32 __stdcall c89atomic_fetch_or_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
- {
- c89atomic_uint32 oldValue;
- c89atomic_uint32 newValue;
- do {
- oldValue = *dst;
- newValue = oldValue | src;
- } while (c89atomic_compare_and_swap_32(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ format = pDescriptorPlayback->format;
+ channels = pDescriptorPlayback->channels;
+ sampleRate = pDescriptorPlayback->sampleRate;
+ }
+
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
+ if (pDescriptorCapture->pDeviceID != NULL) {
+ devCapture = pDescriptorCapture->pDeviceID->pulse;
}
- #endif
- #if defined(C89ATOMIC_HAS_64)
- static C89ATOMIC_INLINE c89atomic_uint64 __stdcall c89atomic_fetch_or_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
- {
- c89atomic_uint64 oldValue;
- c89atomic_uint64 newValue;
- do {
- oldValue = *dst;
- newValue = oldValue | src;
- } while (c89atomic_compare_and_swap_64(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ format = pDescriptorCapture->format;
+ channels = pDescriptorCapture->channels;
+ sampleRate = pDescriptorCapture->sampleRate;
+ }
+
+
+
+ result = ma_init_pa_mainloop_and_pa_context__pulse(pDevice->pContext, pDevice->pContext->pulse.pApplicationName, pDevice->pContext->pulse.pServerName, MA_FALSE, &pDevice->pulse.pMainLoop, &pDevice->pulse.pPulseContext);
+ if (result != MA_SUCCESS) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to initialize PA mainloop and context for device.\n");
+ return result;
+ }
+
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
+ result = ma_context_get_source_info__pulse(pDevice->pContext, devCapture, &sourceInfo);
+ if (result != MA_SUCCESS) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to retrieve source info for capture device.");
+ goto on_error0;
}
- #endif
- #if defined(C89ATOMIC_HAS_8)
- #define c89atomic_test_and_set_explicit_8( dst, order) c89atomic_exchange_explicit_8 (dst, 1, order)
- #endif
- #if defined(C89ATOMIC_HAS_16)
- #define c89atomic_test_and_set_explicit_16(dst, order) c89atomic_exchange_explicit_16(dst, 1, order)
- #endif
- #if defined(C89ATOMIC_HAS_32)
- #define c89atomic_test_and_set_explicit_32(dst, order) c89atomic_exchange_explicit_32(dst, 1, order)
- #endif
- #if defined(C89ATOMIC_HAS_64)
- #define c89atomic_test_and_set_explicit_64(dst, order) c89atomic_exchange_explicit_64(dst, 1, order)
- #endif
- #if defined(C89ATOMIC_HAS_8)
- #define c89atomic_clear_explicit_8( dst, order) c89atomic_store_explicit_8 (dst, 0, order)
- #endif
- #if defined(C89ATOMIC_HAS_16)
- #define c89atomic_clear_explicit_16(dst, order) c89atomic_store_explicit_16(dst, 0, order)
- #endif
- #if defined(C89ATOMIC_HAS_32)
- #define c89atomic_clear_explicit_32(dst, order) c89atomic_store_explicit_32(dst, 0, order)
- #endif
- #if defined(C89ATOMIC_HAS_64)
- #define c89atomic_clear_explicit_64(dst, order) c89atomic_store_explicit_64(dst, 0, order)
- #endif
- #if defined(C89ATOMIC_HAS_8)
- typedef c89atomic_uint8 c89atomic_flag;
- #define c89atomic_flag_test_and_set_explicit(ptr, order) (c89atomic_bool)c89atomic_test_and_set_explicit_8(ptr, order)
- #define c89atomic_flag_clear_explicit(ptr, order) c89atomic_clear_explicit_8(ptr, order)
- #define c89atoimc_flag_load_explicit(ptr, order) c89atomic_load_explicit_8(ptr, order)
- #else
- typedef c89atomic_uint32 c89atomic_flag;
- #define c89atomic_flag_test_and_set_explicit(ptr, order) (c89atomic_bool)c89atomic_test_and_set_explicit_32(ptr, order)
- #define c89atomic_flag_clear_explicit(ptr, order) c89atomic_clear_explicit_32(ptr, order)
- #define c89atoimc_flag_load_explicit(ptr, order) c89atomic_load_explicit_32(ptr, order)
- #endif
-#elif defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 7)))
- #define C89ATOMIC_HAS_NATIVE_COMPARE_EXCHANGE
- #define C89ATOMIC_HAS_NATIVE_IS_LOCK_FREE
- #define c89atomic_memory_order_relaxed __ATOMIC_RELAXED
- #define c89atomic_memory_order_consume __ATOMIC_CONSUME
- #define c89atomic_memory_order_acquire __ATOMIC_ACQUIRE
- #define c89atomic_memory_order_release __ATOMIC_RELEASE
- #define c89atomic_memory_order_acq_rel __ATOMIC_ACQ_REL
- #define c89atomic_memory_order_seq_cst __ATOMIC_SEQ_CST
- #define c89atomic_compiler_fence() __asm__ __volatile__("":::"memory")
- #define c89atomic_thread_fence(order) __atomic_thread_fence(order)
- #define c89atomic_signal_fence(order) __atomic_signal_fence(order)
- #define c89atomic_is_lock_free_8(ptr) __atomic_is_lock_free(1, ptr)
- #define c89atomic_is_lock_free_16(ptr) __atomic_is_lock_free(2, ptr)
- #define c89atomic_is_lock_free_32(ptr) __atomic_is_lock_free(4, ptr)
- #define c89atomic_is_lock_free_64(ptr) __atomic_is_lock_free(8, ptr)
- #define c89atomic_test_and_set_explicit_8( dst, order) __atomic_exchange_n(dst, 1, order)
- #define c89atomic_test_and_set_explicit_16(dst, order) __atomic_exchange_n(dst, 1, order)
- #define c89atomic_test_and_set_explicit_32(dst, order) __atomic_exchange_n(dst, 1, order)
- #define c89atomic_test_and_set_explicit_64(dst, order) __atomic_exchange_n(dst, 1, order)
- #define c89atomic_clear_explicit_8( dst, order) __atomic_store_n(dst, 0, order)
- #define c89atomic_clear_explicit_16(dst, order) __atomic_store_n(dst, 0, order)
- #define c89atomic_clear_explicit_32(dst, order) __atomic_store_n(dst, 0, order)
- #define c89atomic_clear_explicit_64(dst, order) __atomic_store_n(dst, 0, order)
- #define c89atomic_store_explicit_8( dst, src, order) __atomic_store_n(dst, src, order)
- #define c89atomic_store_explicit_16(dst, src, order) __atomic_store_n(dst, src, order)
- #define c89atomic_store_explicit_32(dst, src, order) __atomic_store_n(dst, src, order)
- #define c89atomic_store_explicit_64(dst, src, order) __atomic_store_n(dst, src, order)
- #define c89atomic_load_explicit_8( dst, order) __atomic_load_n(dst, order)
- #define c89atomic_load_explicit_16(dst, order) __atomic_load_n(dst, order)
- #define c89atomic_load_explicit_32(dst, order) __atomic_load_n(dst, order)
- #define c89atomic_load_explicit_64(dst, order) __atomic_load_n(dst, order)
- #define c89atomic_exchange_explicit_8( dst, src, order) __atomic_exchange_n(dst, src, order)
- #define c89atomic_exchange_explicit_16(dst, src, order) __atomic_exchange_n(dst, src, order)
- #define c89atomic_exchange_explicit_32(dst, src, order) __atomic_exchange_n(dst, src, order)
- #define c89atomic_exchange_explicit_64(dst, src, order) __atomic_exchange_n(dst, src, order)
- #define c89atomic_compare_exchange_strong_explicit_8( dst, expected, desired, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, desired, 0, successOrder, failureOrder)
- #define c89atomic_compare_exchange_strong_explicit_16(dst, expected, desired, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, desired, 0, successOrder, failureOrder)
- #define c89atomic_compare_exchange_strong_explicit_32(dst, expected, desired, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, desired, 0, successOrder, failureOrder)
- #define c89atomic_compare_exchange_strong_explicit_64(dst, expected, desired, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, desired, 0, successOrder, failureOrder)
- #define c89atomic_compare_exchange_weak_explicit_8( dst, expected, desired, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, desired, 1, successOrder, failureOrder)
- #define c89atomic_compare_exchange_weak_explicit_16(dst, expected, desired, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, desired, 1, successOrder, failureOrder)
- #define c89atomic_compare_exchange_weak_explicit_32(dst, expected, desired, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, desired, 1, successOrder, failureOrder)
- #define c89atomic_compare_exchange_weak_explicit_64(dst, expected, desired, successOrder, failureOrder) __atomic_compare_exchange_n(dst, expected, desired, 1, successOrder, failureOrder)
- #define c89atomic_fetch_add_explicit_8( dst, src, order) __atomic_fetch_add(dst, src, order)
- #define c89atomic_fetch_add_explicit_16(dst, src, order) __atomic_fetch_add(dst, src, order)
- #define c89atomic_fetch_add_explicit_32(dst, src, order) __atomic_fetch_add(dst, src, order)
- #define c89atomic_fetch_add_explicit_64(dst, src, order) __atomic_fetch_add(dst, src, order)
- #define c89atomic_fetch_sub_explicit_8( dst, src, order) __atomic_fetch_sub(dst, src, order)
- #define c89atomic_fetch_sub_explicit_16(dst, src, order) __atomic_fetch_sub(dst, src, order)
- #define c89atomic_fetch_sub_explicit_32(dst, src, order) __atomic_fetch_sub(dst, src, order)
- #define c89atomic_fetch_sub_explicit_64(dst, src, order) __atomic_fetch_sub(dst, src, order)
- #define c89atomic_fetch_or_explicit_8( dst, src, order) __atomic_fetch_or(dst, src, order)
- #define c89atomic_fetch_or_explicit_16(dst, src, order) __atomic_fetch_or(dst, src, order)
- #define c89atomic_fetch_or_explicit_32(dst, src, order) __atomic_fetch_or(dst, src, order)
- #define c89atomic_fetch_or_explicit_64(dst, src, order) __atomic_fetch_or(dst, src, order)
- #define c89atomic_fetch_xor_explicit_8( dst, src, order) __atomic_fetch_xor(dst, src, order)
- #define c89atomic_fetch_xor_explicit_16(dst, src, order) __atomic_fetch_xor(dst, src, order)
- #define c89atomic_fetch_xor_explicit_32(dst, src, order) __atomic_fetch_xor(dst, src, order)
- #define c89atomic_fetch_xor_explicit_64(dst, src, order) __atomic_fetch_xor(dst, src, order)
- #define c89atomic_fetch_and_explicit_8( dst, src, order) __atomic_fetch_and(dst, src, order)
- #define c89atomic_fetch_and_explicit_16(dst, src, order) __atomic_fetch_and(dst, src, order)
- #define c89atomic_fetch_and_explicit_32(dst, src, order) __atomic_fetch_and(dst, src, order)
- #define c89atomic_fetch_and_explicit_64(dst, src, order) __atomic_fetch_and(dst, src, order)
- #define c89atomic_compare_and_swap_8 (dst, expected, desired) __sync_val_compare_and_swap(dst, expected, desired)
- #define c89atomic_compare_and_swap_16(dst, expected, desired) __sync_val_compare_and_swap(dst, expected, desired)
- #define c89atomic_compare_and_swap_32(dst, expected, desired) __sync_val_compare_and_swap(dst, expected, desired)
- #define c89atomic_compare_and_swap_64(dst, expected, desired) __sync_val_compare_and_swap(dst, expected, desired)
- typedef c89atomic_uint8 c89atomic_flag;
- #define c89atomic_flag_test_and_set_explicit(dst, order) (c89atomic_bool)__atomic_test_and_set(dst, order)
- #define c89atomic_flag_clear_explicit(dst, order) __atomic_clear(dst, order)
- #define c89atoimc_flag_load_explicit(ptr, order) c89atomic_load_explicit_8(ptr, order)
-#else
- #define c89atomic_memory_order_relaxed 1
- #define c89atomic_memory_order_consume 2
- #define c89atomic_memory_order_acquire 3
- #define c89atomic_memory_order_release 4
- #define c89atomic_memory_order_acq_rel 5
- #define c89atomic_memory_order_seq_cst 6
- #define c89atomic_compiler_fence() __asm__ __volatile__("":::"memory")
- #if defined(__GNUC__)
- #define c89atomic_thread_fence(order) __sync_synchronize(), (void)order
- static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_exchange_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
- {
- if (order > c89atomic_memory_order_acquire) {
- __sync_synchronize();
+
+ ss = sourceInfo.sample_spec;
+ cmap = sourceInfo.channel_map;
+
+ if (ma_format_from_pulse(ss.format) == ma_format_unknown) {
+ if (ma_is_little_endian()) {
+ ss.format = MA_PA_SAMPLE_FLOAT32LE;
+ } else {
+ ss.format = MA_PA_SAMPLE_FLOAT32BE;
}
- return __sync_lock_test_and_set(dst, src);
- }
- static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_exchange_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
- {
- c89atomic_uint16 oldValue;
- do {
- oldValue = *dst;
- } while (__sync_val_compare_and_swap(dst, oldValue, src) != oldValue);
- (void)order;
- return oldValue;
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] sample_spec.format not supported by miniaudio. Defaulting to PA_SAMPLE_FLOAT32.\n");
}
- static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_exchange_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
- {
- c89atomic_uint32 oldValue;
- do {
- oldValue = *dst;
- } while (__sync_val_compare_and_swap(dst, oldValue, src) != oldValue);
- (void)order;
- return oldValue;
+ if (ss.rate == 0) {
+ ss.rate = MA_DEFAULT_SAMPLE_RATE;
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] sample_spec.rate = 0. Defaulting to %d.\n", ss.rate);
}
- static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_exchange_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
- {
- c89atomic_uint64 oldValue;
- do {
- oldValue = *dst;
- } while (__sync_val_compare_and_swap(dst, oldValue, src) != oldValue);
- (void)order;
- return oldValue;
+ if (ss.channels == 0) {
+ ss.channels = MA_DEFAULT_CHANNELS;
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] sample_spec.channels = 0. Defaulting to %d.\n", ss.channels);
}
- static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_fetch_add_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
- {
- (void)order;
- return __sync_fetch_and_add(dst, src);
+
+ /* We now have enough information to calculate our actual period size in frames. */
+ pDescriptorCapture->periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__pulse(pDescriptorCapture, ss.rate, pConfig->performanceProfile);
+
+ attr = ma_device__pa_buffer_attr_new(pDescriptorCapture->periodSizeInFrames, pDescriptorCapture->periodCount, &ss);
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Capture attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; periodSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsize, pDescriptorCapture->periodSizeInFrames);
+
+ pDevice->pulse.pStreamCapture = ma_device__pa_stream_new__pulse(pDevice, pConfig->pulse.pStreamNameCapture, &ss, &cmap);
+ if (pDevice->pulse.pStreamCapture == NULL) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create PulseAudio capture stream.\n");
+ result = MA_ERROR;
+ goto on_error0;
}
- static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_fetch_add_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
- {
- (void)order;
- return __sync_fetch_and_add(dst, src);
+
+
+ /* The callback needs to be set before connecting the stream. */
+ ((ma_pa_stream_set_read_callback_proc)pDevice->pContext->pulse.pa_stream_set_read_callback)((ma_pa_stream*)pDevice->pulse.pStreamCapture, ma_device_on_read__pulse, pDevice);
+
+ /* State callback for checking when the device has been corked. */
+ ((ma_pa_stream_set_suspended_callback_proc)pDevice->pContext->pulse.pa_stream_set_suspended_callback)((ma_pa_stream*)pDevice->pulse.pStreamCapture, ma_device_on_suspended__pulse, pDevice);
+
+ /* Rerouting notification. */
+ ((ma_pa_stream_set_moved_callback_proc)pDevice->pContext->pulse.pa_stream_set_moved_callback)((ma_pa_stream*)pDevice->pulse.pStreamCapture, ma_device_on_rerouted__pulse, pDevice);
+
+
+ /* Connect after we've got all of our internal state set up. */
+ streamFlags = MA_PA_STREAM_START_CORKED | MA_PA_STREAM_ADJUST_LATENCY | MA_PA_STREAM_FIX_FORMAT | MA_PA_STREAM_FIX_RATE | MA_PA_STREAM_FIX_CHANNELS;
+ if (devCapture != NULL) {
+ streamFlags |= MA_PA_STREAM_DONT_MOVE;
}
- static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_fetch_add_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
- {
- (void)order;
- return __sync_fetch_and_add(dst, src);
+
+ error = ((ma_pa_stream_connect_record_proc)pDevice->pContext->pulse.pa_stream_connect_record)((ma_pa_stream*)pDevice->pulse.pStreamCapture, devCapture, &attr, streamFlags);
+ if (error != MA_PA_OK) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to connect PulseAudio capture stream.");
+ result = ma_result_from_pulse(error);
+ goto on_error1;
}
- static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_fetch_add_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
- {
- (void)order;
- return __sync_fetch_and_add(dst, src);
+
+ result = ma_wait_for_pa_stream_to_connect__pulse(pDevice->pContext, pDevice->pulse.pMainLoop, (ma_pa_stream*)pDevice->pulse.pStreamCapture);
+ if (result != MA_SUCCESS) {
+ goto on_error2;
}
- static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_fetch_sub_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
- {
- (void)order;
- return __sync_fetch_and_sub(dst, src);
+
+
+ /* Internal format. */
+ pActualSS = ((ma_pa_stream_get_sample_spec_proc)pDevice->pContext->pulse.pa_stream_get_sample_spec)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
+ if (pActualSS != NULL) {
+ ss = *pActualSS;
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Capture sample spec: format=%s, channels=%d, rate=%d\n", ma_get_format_name(ma_format_from_pulse(ss.format)), ss.channels, ss.rate);
+ } else {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Failed to retrieve capture sample spec.\n");
}
- static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_fetch_sub_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
- {
- (void)order;
- return __sync_fetch_and_sub(dst, src);
+
+ pDescriptorCapture->format = ma_format_from_pulse(ss.format);
+ pDescriptorCapture->channels = ss.channels;
+ pDescriptorCapture->sampleRate = ss.rate;
+
+ if (pDescriptorCapture->format == ma_format_unknown || pDescriptorCapture->channels == 0 || pDescriptorCapture->sampleRate == 0) {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Capture sample spec is invalid. Device unusable by miniaudio. format=%s, channels=%d, sampleRate=%d.\n", ma_get_format_name(pDescriptorCapture->format), pDescriptorCapture->channels, pDescriptorCapture->sampleRate);
+ result = MA_ERROR;
+ goto on_error4;
}
- static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_fetch_sub_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
- {
- (void)order;
- return __sync_fetch_and_sub(dst, src);
+
+ /* Internal channel map. */
+
+ /*
+ Bug in PipeWire. There have been reports that PipeWire is returning AUX channels when reporting
+ the channel map. To somewhat workaround this, I'm hacking in a hard coded channel map for mono
+ and stereo. In this case it should be safe to assume mono = MONO and stereo = LEFT/RIGHT. For
+ all other channel counts we need to just put up with whatever PipeWire reports and hope it gets
+ fixed sooner than later. I might remove this hack later.
+ */
+ if (pDescriptorCapture->channels > 2) {
+ pActualCMap = ((ma_pa_stream_get_channel_map_proc)pDevice->pContext->pulse.pa_stream_get_channel_map)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
+ if (pActualCMap != NULL) {
+ cmap = *pActualCMap;
+ }
+
+ for (iChannel = 0; iChannel < pDescriptorCapture->channels; ++iChannel) {
+ pDescriptorCapture->channelMap[iChannel] = ma_channel_position_from_pulse(cmap.map[iChannel]);
+ }
+ } else {
+ /* Hack for mono and stereo. */
+ if (pDescriptorCapture->channels == 1) {
+ pDescriptorCapture->channelMap[0] = MA_CHANNEL_MONO;
+ } else if (pDescriptorCapture->channels == 2) {
+ pDescriptorCapture->channelMap[0] = MA_CHANNEL_FRONT_LEFT;
+ pDescriptorCapture->channelMap[1] = MA_CHANNEL_FRONT_RIGHT;
+ } else {
+ MA_ASSERT(MA_FALSE); /* Should never hit this. */
+ }
}
- static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_fetch_sub_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
- {
- (void)order;
- return __sync_fetch_and_sub(dst, src);
+
+
+ /* Buffer. */
+ pActualAttr = ((ma_pa_stream_get_buffer_attr_proc)pDevice->pContext->pulse.pa_stream_get_buffer_attr)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
+ if (pActualAttr != NULL) {
+ attr = *pActualAttr;
}
- static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_fetch_or_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
- {
- (void)order;
- return __sync_fetch_and_or(dst, src);
+
+ if (attr.fragsize > 0) {
+ pDescriptorCapture->periodCount = ma_max(attr.maxlength / attr.fragsize, 1);
+ } else {
+ pDescriptorCapture->periodCount = 1;
}
- static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_fetch_or_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
- {
- (void)order;
- return __sync_fetch_and_or(dst, src);
+
+ pDescriptorCapture->periodSizeInFrames = attr.maxlength / ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels) / pDescriptorCapture->periodCount;
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Capture actual attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; periodSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsize, pDescriptorCapture->periodSizeInFrames);
+ }
+
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ result = ma_context_get_sink_info__pulse(pDevice->pContext, devPlayback, &sinkInfo);
+ if (result != MA_SUCCESS) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to retrieve sink info for playback device.\n");
+ goto on_error2;
}
- static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_fetch_or_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
- {
- (void)order;
- return __sync_fetch_and_or(dst, src);
+
+ ss = sinkInfo.sample_spec;
+ cmap = sinkInfo.channel_map;
+
+ if (ma_format_from_pulse(ss.format) == ma_format_unknown) {
+ if (ma_is_little_endian()) {
+ ss.format = MA_PA_SAMPLE_FLOAT32LE;
+ } else {
+ ss.format = MA_PA_SAMPLE_FLOAT32BE;
+ }
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] sample_spec.format not supported by miniaudio. Defaulting to PA_SAMPLE_FLOAT32.\n");
}
- static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_fetch_or_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
- {
- (void)order;
- return __sync_fetch_and_or(dst, src);
+ if (ss.rate == 0) {
+ ss.rate = MA_DEFAULT_SAMPLE_RATE;
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] sample_spec.rate = 0. Defaulting to %d.\n", ss.rate);
}
- static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_fetch_xor_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
- {
- (void)order;
- return __sync_fetch_and_xor(dst, src);
+ if (ss.channels == 0) {
+ ss.channels = MA_DEFAULT_CHANNELS;
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] sample_spec.channels = 0. Defaulting to %d.\n", ss.channels);
}
- static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_fetch_xor_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
- {
- (void)order;
- return __sync_fetch_and_xor(dst, src);
+
+ /* We now have enough information to calculate the actual buffer size in frames. */
+ pDescriptorPlayback->periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__pulse(pDescriptorPlayback, ss.rate, pConfig->performanceProfile);
+
+ attr = ma_device__pa_buffer_attr_new(pDescriptorPlayback->periodSizeInFrames, pDescriptorPlayback->periodCount, &ss);
+
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Playback attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; periodSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsize, pDescriptorPlayback->periodSizeInFrames);
+
+ pDevice->pulse.pStreamPlayback = ma_device__pa_stream_new__pulse(pDevice, pConfig->pulse.pStreamNamePlayback, &ss, &cmap);
+ if (pDevice->pulse.pStreamPlayback == NULL) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create PulseAudio playback stream.\n");
+ result = MA_ERROR;
+ goto on_error2;
}
- static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_fetch_xor_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
- {
- (void)order;
- return __sync_fetch_and_xor(dst, src);
+
+
+ /*
+ Note that this callback will be fired as soon as the stream is connected, even though it's started as corked. The callback needs to handle a
+ device state of ma_device_state_uninitialized.
+ */
+ ((ma_pa_stream_set_write_callback_proc)pDevice->pContext->pulse.pa_stream_set_write_callback)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, ma_device_on_write__pulse, pDevice);
+
+ /* State callback for checking when the device has been corked. */
+ ((ma_pa_stream_set_suspended_callback_proc)pDevice->pContext->pulse.pa_stream_set_suspended_callback)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, ma_device_on_suspended__pulse, pDevice);
+
+ /* Rerouting notification. */
+ ((ma_pa_stream_set_moved_callback_proc)pDevice->pContext->pulse.pa_stream_set_moved_callback)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, ma_device_on_rerouted__pulse, pDevice);
+
+
+ /* Connect after we've got all of our internal state set up. */
+ streamFlags = MA_PA_STREAM_START_CORKED | MA_PA_STREAM_ADJUST_LATENCY | MA_PA_STREAM_FIX_FORMAT | MA_PA_STREAM_FIX_RATE | MA_PA_STREAM_FIX_CHANNELS;
+ if (devPlayback != NULL) {
+ streamFlags |= MA_PA_STREAM_DONT_MOVE;
}
- static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_fetch_xor_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
- {
- (void)order;
- return __sync_fetch_and_xor(dst, src);
+
+ error = ((ma_pa_stream_connect_playback_proc)pDevice->pContext->pulse.pa_stream_connect_playback)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, devPlayback, &attr, streamFlags, NULL, NULL);
+ if (error != MA_PA_OK) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to connect PulseAudio playback stream.");
+ result = ma_result_from_pulse(error);
+ goto on_error3;
}
- static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_fetch_and_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
- {
- (void)order;
- return __sync_fetch_and_and(dst, src);
+
+ result = ma_wait_for_pa_stream_to_connect__pulse(pDevice->pContext, pDevice->pulse.pMainLoop, (ma_pa_stream*)pDevice->pulse.pStreamPlayback);
+ if (result != MA_SUCCESS) {
+ goto on_error3;
}
- static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_fetch_and_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
- {
- (void)order;
- return __sync_fetch_and_and(dst, src);
+
+
+ /* Internal format. */
+ pActualSS = ((ma_pa_stream_get_sample_spec_proc)pDevice->pContext->pulse.pa_stream_get_sample_spec)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
+ if (pActualSS != NULL) {
+ ss = *pActualSS;
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Playback sample spec: format=%s, channels=%d, rate=%d\n", ma_get_format_name(ma_format_from_pulse(ss.format)), ss.channels, ss.rate);
+ } else {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Failed to retrieve playback sample spec.\n");
}
- static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_fetch_and_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
- {
- (void)order;
- return __sync_fetch_and_and(dst, src);
+
+ pDescriptorPlayback->format = ma_format_from_pulse(ss.format);
+ pDescriptorPlayback->channels = ss.channels;
+ pDescriptorPlayback->sampleRate = ss.rate;
+
+ if (pDescriptorPlayback->format == ma_format_unknown || pDescriptorPlayback->channels == 0 || pDescriptorPlayback->sampleRate == 0) {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Playback sample spec is invalid. Device unusable by miniaudio. format=%s, channels=%d, sampleRate=%d.\n", ma_get_format_name(pDescriptorPlayback->format), pDescriptorPlayback->channels, pDescriptorPlayback->sampleRate);
+ result = MA_ERROR;
+ goto on_error4;
}
- static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_fetch_and_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
- {
- (void)order;
- return __sync_fetch_and_and(dst, src);
+
+ /* Internal channel map. */
+
+ /*
+ Bug in PipeWire. There have been reports that PipeWire is returning AUX channels when reporting
+ the channel map. To somewhat workaround this, I'm hacking in a hard coded channel map for mono
+ and stereo. In this case it should be safe to assume mono = MONO and stereo = LEFT/RIGHT. For
+ all other channel counts we need to just put up with whatever PipeWire reports and hope it gets
+ fixed sooner than later. I might remove this hack later.
+ */
+ if (pDescriptorPlayback->channels > 2) {
+ pActualCMap = ((ma_pa_stream_get_channel_map_proc)pDevice->pContext->pulse.pa_stream_get_channel_map)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
+ if (pActualCMap != NULL) {
+ cmap = *pActualCMap;
+ }
+
+ for (iChannel = 0; iChannel < pDescriptorPlayback->channels; ++iChannel) {
+ pDescriptorPlayback->channelMap[iChannel] = ma_channel_position_from_pulse(cmap.map[iChannel]);
+ }
+ } else {
+ /* Hack for mono and stereo. */
+ if (pDescriptorPlayback->channels == 1) {
+ pDescriptorPlayback->channelMap[0] = MA_CHANNEL_MONO;
+ } else if (pDescriptorPlayback->channels == 2) {
+ pDescriptorPlayback->channelMap[0] = MA_CHANNEL_FRONT_LEFT;
+ pDescriptorPlayback->channelMap[1] = MA_CHANNEL_FRONT_RIGHT;
+ } else {
+ MA_ASSERT(MA_FALSE); /* Should never hit this. */
+ }
}
- #define c89atomic_compare_and_swap_8( dst, expected, desired) __sync_val_compare_and_swap(dst, expected, desired)
- #define c89atomic_compare_and_swap_16(dst, expected, desired) __sync_val_compare_and_swap(dst, expected, desired)
- #define c89atomic_compare_and_swap_32(dst, expected, desired) __sync_val_compare_and_swap(dst, expected, desired)
- #define c89atomic_compare_and_swap_64(dst, expected, desired) __sync_val_compare_and_swap(dst, expected, desired)
- #else
- #if defined(C89ATOMIC_X86)
- #define c89atomic_thread_fence(order) __asm__ __volatile__("lock; addl $0, (%%esp)" ::: "memory", "cc")
- #elif defined(C89ATOMIC_X64)
- #define c89atomic_thread_fence(order) __asm__ __volatile__("lock; addq $0, (%%rsp)" ::: "memory", "cc")
- #else
- #error Unsupported architecture. Please submit a feature request.
- #endif
- static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_compare_and_swap_8(volatile c89atomic_uint8* dst, c89atomic_uint8 expected, c89atomic_uint8 desired)
- {
- c89atomic_uint8 result;
- #if defined(C89ATOMIC_X86) || defined(C89ATOMIC_X64)
- __asm__ __volatile__("lock; cmpxchg %3, %0" : "+m"(*dst), "=a"(result) : "a"(expected), "d"(desired) : "cc");
- #else
- #error Unsupported architecture. Please submit a feature request.
- #endif
- return result;
+
+
+ /* Buffer. */
+ pActualAttr = ((ma_pa_stream_get_buffer_attr_proc)pDevice->pContext->pulse.pa_stream_get_buffer_attr)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
+ if (pActualAttr != NULL) {
+ attr = *pActualAttr;
}
- static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_compare_and_swap_16(volatile c89atomic_uint16* dst, c89atomic_uint16 expected, c89atomic_uint16 desired)
- {
- c89atomic_uint16 result;
- #if defined(C89ATOMIC_X86) || defined(C89ATOMIC_X64)
- __asm__ __volatile__("lock; cmpxchg %3, %0" : "+m"(*dst), "=a"(result) : "a"(expected), "d"(desired) : "cc");
- #else
- #error Unsupported architecture. Please submit a feature request.
- #endif
- return result;
+
+ if (attr.tlength > 0) {
+ pDescriptorPlayback->periodCount = ma_max(attr.maxlength / attr.tlength, 1);
+ } else {
+ pDescriptorPlayback->periodCount = 1;
}
- static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_compare_and_swap_32(volatile c89atomic_uint32* dst, c89atomic_uint32 expected, c89atomic_uint32 desired)
- {
- c89atomic_uint32 result;
- #if defined(C89ATOMIC_X86) || defined(C89ATOMIC_X64)
- __asm__ __volatile__("lock; cmpxchg %3, %0" : "+m"(*dst), "=a"(result) : "a"(expected), "d"(desired) : "cc");
- #else
- #error Unsupported architecture. Please submit a feature request.
- #endif
- return result;
+
+ pDescriptorPlayback->periodSizeInFrames = attr.maxlength / ma_get_bytes_per_frame(pDescriptorPlayback->format, pDescriptorPlayback->channels) / pDescriptorPlayback->periodCount;
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Playback actual attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; internalPeriodSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsize, pDescriptorPlayback->periodSizeInFrames);
+ }
+
+
+ /*
+ We need a ring buffer for handling duplex mode. We can use the main duplex ring buffer in the main
+ part of the ma_device struct. We cannot, however, depend on ma_device_init() initializing this for
+ us later on because that will only do it if it's a fully asynchronous backend - i.e. the
+ onDeviceDataLoop callback is NULL, which is not the case for PulseAudio.
+ */
+ if (pConfig->deviceType == ma_device_type_duplex) {
+ ma_format rbFormat = (format != ma_format_unknown) ? format : pDescriptorCapture->format;
+ ma_uint32 rbChannels = (channels > 0) ? channels : pDescriptorCapture->channels;
+ ma_uint32 rbSampleRate = (sampleRate > 0) ? sampleRate : pDescriptorCapture->sampleRate;
+
+ result = ma_duplex_rb_init(rbFormat, rbChannels, rbSampleRate, pDescriptorCapture->sampleRate, pDescriptorCapture->periodSizeInFrames, &pDevice->pContext->allocationCallbacks, &pDevice->duplexRB);
+ if (result != MA_SUCCESS) {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to initialize ring buffer. %s.\n", ma_result_description(result));
+ goto on_error4;
}
- static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_compare_and_swap_64(volatile c89atomic_uint64* dst, c89atomic_uint64 expected, c89atomic_uint64 desired)
- {
- volatile c89atomic_uint64 result;
- #if defined(C89ATOMIC_X86)
- c89atomic_uint32 resultEAX;
- c89atomic_uint32 resultEDX;
- __asm__ __volatile__("push %%ebx; xchg %5, %%ebx; lock; cmpxchg8b %0; pop %%ebx" : "+m"(*dst), "=a"(resultEAX), "=d"(resultEDX) : "a"(expected & 0xFFFFFFFF), "d"(expected >> 32), "r"(desired & 0xFFFFFFFF), "c"(desired >> 32) : "cc");
- result = ((c89atomic_uint64)resultEDX << 32) | resultEAX;
- #elif defined(C89ATOMIC_X64)
- __asm__ __volatile__("lock; cmpxchg %3, %0" : "+m"(*dst), "=a"(result) : "a"(expected), "d"(desired) : "cc");
- #else
- #error Unsupported architecture. Please submit a feature request.
- #endif
+ }
+
+ return MA_SUCCESS;
+
+
+on_error4:
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ ((ma_pa_stream_disconnect_proc)pDevice->pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
+ }
+on_error3:
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ ((ma_pa_stream_unref_proc)pDevice->pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
+ }
+on_error2:
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
+ ((ma_pa_stream_disconnect_proc)pDevice->pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
+ }
+on_error1:
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
+ ((ma_pa_stream_unref_proc)pDevice->pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
+ }
+on_error0:
+ return result;
+}
+
+
+static void ma_pulse_operation_complete_callback(ma_pa_stream* pStream, int success, void* pUserData)
+{
+ ma_bool32* pIsSuccessful = (ma_bool32*)pUserData;
+ MA_ASSERT(pIsSuccessful != NULL);
+
+ *pIsSuccessful = (ma_bool32)success;
+
+ (void)pStream; /* Unused. */
+}
+
+static ma_result ma_device__cork_stream__pulse(ma_device* pDevice, ma_device_type deviceType, int cork)
+{
+ ma_context* pContext = pDevice->pContext;
+ ma_bool32 wasSuccessful;
+ ma_pa_stream* pStream;
+ ma_pa_operation* pOP;
+ ma_result result;
+
+ /* This should not be called with a duplex device type. */
+ if (deviceType == ma_device_type_duplex) {
+ return MA_INVALID_ARGS;
+ }
+
+ wasSuccessful = MA_FALSE;
+
+ pStream = (ma_pa_stream*)((deviceType == ma_device_type_capture) ? pDevice->pulse.pStreamCapture : pDevice->pulse.pStreamPlayback);
+ MA_ASSERT(pStream != NULL);
+
+ pOP = ((ma_pa_stream_cork_proc)pContext->pulse.pa_stream_cork)(pStream, cork, ma_pulse_operation_complete_callback, &wasSuccessful);
+ if (pOP == NULL) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to cork PulseAudio stream.");
+ return MA_ERROR;
+ }
+
+ result = ma_wait_for_operation_and_unref__pulse(pDevice->pContext, pDevice->pulse.pMainLoop, pOP);
+ if (result != MA_SUCCESS) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] An error occurred while waiting for the PulseAudio stream to cork.");
+ return result;
+ }
+
+ if (!wasSuccessful) {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to %s PulseAudio stream.", (cork) ? "stop" : "start");
+ return MA_ERROR;
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_start__pulse(ma_device* pDevice)
+{
+ ma_result result;
+
+ MA_ASSERT(pDevice != NULL);
+
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ result = ma_device__cork_stream__pulse(pDevice, ma_device_type_capture, 0);
+ if (result != MA_SUCCESS) {
return result;
}
- static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_exchange_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
- {
- c89atomic_uint8 result = 0;
- (void)order;
- #if defined(C89ATOMIC_X86) || defined(C89ATOMIC_X64)
- __asm__ __volatile__("lock; xchg %1, %0" : "+m"(*dst), "=a"(result) : "a"(src));
- #else
- #error Unsupported architecture. Please submit a feature request.
- #endif
+ }
+
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ /*
+ We need to fill some data before uncorking. Not doing this will result in the write callback
+ never getting fired. We're not going to abort if writing fails because I still want the device
+ to get uncorked.
+ */
+ ma_device_write_to_stream__pulse(pDevice, (ma_pa_stream*)(pDevice->pulse.pStreamPlayback), NULL); /* No need to check the result here. Always want to fall through an uncork.*/
+
+ result = ma_device__cork_stream__pulse(pDevice, ma_device_type_playback, 0);
+ if (result != MA_SUCCESS) {
return result;
}
- static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_exchange_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
- {
- c89atomic_uint16 result = 0;
- (void)order;
- #if defined(C89ATOMIC_X86) || defined(C89ATOMIC_X64)
- __asm__ __volatile__("lock; xchg %1, %0" : "+m"(*dst), "=a"(result) : "a"(src));
- #else
- #error Unsupported architecture. Please submit a feature request.
- #endif
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_stop__pulse(ma_device* pDevice)
+{
+ ma_result result;
+
+ MA_ASSERT(pDevice != NULL);
+
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ result = ma_device__cork_stream__pulse(pDevice, ma_device_type_capture, 1);
+ if (result != MA_SUCCESS) {
return result;
}
- static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_exchange_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
- {
- c89atomic_uint32 result;
- (void)order;
- #if defined(C89ATOMIC_X86) || defined(C89ATOMIC_X64)
- __asm__ __volatile__("lock; xchg %1, %0" : "+m"(*dst), "=a"(result) : "a"(src));
- #else
- #error Unsupported architecture. Please submit a feature request.
- #endif
+ }
+
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ /*
+ Ideally we would drain the device here, but there's been cases where PulseAudio seems to be
+ broken on some systems to the point where no audio processing seems to happen. When this
+ happens, draining never completes and we get stuck here. For now I'm disabling draining of
+ the device so we don't just freeze the application.
+ */
+ #if 0
+ ma_pa_operation* pOP = ((ma_pa_stream_drain_proc)pDevice->pContext->pulse.pa_stream_drain)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, ma_pulse_operation_complete_callback, &wasSuccessful);
+ ma_wait_for_operation_and_unref__pulse(pDevice->pContext, pDevice->pulse.pMainLoop, pOP);
+ #endif
+
+ result = ma_device__cork_stream__pulse(pDevice, ma_device_type_playback, 1);
+ if (result != MA_SUCCESS) {
return result;
}
- static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_exchange_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
- {
- c89atomic_uint64 result;
- (void)order;
- #if defined(C89ATOMIC_X86)
- do {
- result = *dst;
- } while (c89atomic_compare_and_swap_64(dst, result, src) != result);
- #elif defined(C89ATOMIC_X64)
- __asm__ __volatile__("lock; xchg %1, %0" : "+m"(*dst), "=a"(result) : "a"(src));
- #else
- #error Unsupported architecture. Please submit a feature request.
- #endif
- return result;
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_data_loop__pulse(ma_device* pDevice)
+{
+ int resultPA;
+
+ MA_ASSERT(pDevice != NULL);
+
+ /* NOTE: Don't start the device here. It'll be done at a higher level. */
+
+ /*
+ All data is handled through callbacks. All we need to do is iterate over the main loop and let
+ the callbacks deal with it.
+ */
+ while (ma_device_get_state(pDevice) == ma_device_state_started) {
+ resultPA = ((ma_pa_mainloop_iterate_proc)pDevice->pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pDevice->pulse.pMainLoop, 1, NULL);
+ if (resultPA < 0) {
+ break;
}
- static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_fetch_add_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
- {
- c89atomic_uint8 result;
- (void)order;
- #if defined(C89ATOMIC_X86) || defined(C89ATOMIC_X64)
- __asm__ __volatile__("lock; xadd %1, %0" : "+m"(*dst), "=a"(result) : "a"(src) : "cc");
- #else
- #error Unsupported architecture. Please submit a feature request.
- #endif
- return result;
+ }
+
+ /* NOTE: Don't stop the device here. It'll be done at a higher level. */
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_data_loop_wakeup__pulse(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
+
+ ((ma_pa_mainloop_wakeup_proc)pDevice->pContext->pulse.pa_mainloop_wakeup)((ma_pa_mainloop*)pDevice->pulse.pMainLoop);
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_uninit__pulse(ma_context* pContext)
+{
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pContext->backend == ma_backend_pulseaudio);
+
+ ((ma_pa_context_disconnect_proc)pContext->pulse.pa_context_disconnect)((ma_pa_context*)pContext->pulse.pPulseContext);
+ ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)((ma_pa_context*)pContext->pulse.pPulseContext);
+ ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)((ma_pa_mainloop*)pContext->pulse.pMainLoop);
+
+ ma_free(pContext->pulse.pServerName, &pContext->allocationCallbacks);
+ ma_free(pContext->pulse.pApplicationName, &pContext->allocationCallbacks);
+
+#ifndef MA_NO_RUNTIME_LINKING
+ ma_dlclose(pContext, pContext->pulse.pulseSO);
+#endif
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_init__pulse(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+{
+ ma_result result;
+#ifndef MA_NO_RUNTIME_LINKING
+ const char* libpulseNames[] = {
+ "libpulse.so",
+ "libpulse.so.0"
+ };
+ size_t i;
+
+ for (i = 0; i < ma_countof(libpulseNames); ++i) {
+ pContext->pulse.pulseSO = ma_dlopen(pContext, libpulseNames[i]);
+ if (pContext->pulse.pulseSO != NULL) {
+ break;
}
- static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_fetch_add_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
- {
- c89atomic_uint16 result;
- (void)order;
- #if defined(C89ATOMIC_X86) || defined(C89ATOMIC_X64)
- __asm__ __volatile__("lock; xadd %1, %0" : "+m"(*dst), "=a"(result) : "a"(src) : "cc");
- #else
- #error Unsupported architecture. Please submit a feature request.
- #endif
- return result;
+ }
+
+ if (pContext->pulse.pulseSO == NULL) {
+ return MA_NO_BACKEND;
+ }
+
+ pContext->pulse.pa_mainloop_new = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_new");
+ pContext->pulse.pa_mainloop_free = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_free");
+ pContext->pulse.pa_mainloop_quit = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_quit");
+ pContext->pulse.pa_mainloop_get_api = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_get_api");
+ pContext->pulse.pa_mainloop_iterate = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_iterate");
+ pContext->pulse.pa_mainloop_wakeup = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_wakeup");
+ pContext->pulse.pa_threaded_mainloop_new = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_new");
+ pContext->pulse.pa_threaded_mainloop_free = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_free");
+ pContext->pulse.pa_threaded_mainloop_start = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_start");
+ pContext->pulse.pa_threaded_mainloop_stop = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_stop");
+ pContext->pulse.pa_threaded_mainloop_lock = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_lock");
+ pContext->pulse.pa_threaded_mainloop_unlock = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_unlock");
+ pContext->pulse.pa_threaded_mainloop_wait = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_wait");
+ pContext->pulse.pa_threaded_mainloop_signal = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_signal");
+ pContext->pulse.pa_threaded_mainloop_accept = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_accept");
+ pContext->pulse.pa_threaded_mainloop_get_retval = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_get_retval");
+ pContext->pulse.pa_threaded_mainloop_get_api = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_get_api");
+ pContext->pulse.pa_threaded_mainloop_in_thread = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_in_thread");
+ pContext->pulse.pa_threaded_mainloop_set_name = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_set_name");
+ pContext->pulse.pa_context_new = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_new");
+ pContext->pulse.pa_context_unref = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_unref");
+ pContext->pulse.pa_context_connect = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_connect");
+ pContext->pulse.pa_context_disconnect = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_disconnect");
+ pContext->pulse.pa_context_set_state_callback = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_set_state_callback");
+ pContext->pulse.pa_context_get_state = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_get_state");
+ pContext->pulse.pa_context_get_sink_info_list = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_get_sink_info_list");
+ pContext->pulse.pa_context_get_source_info_list = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_get_source_info_list");
+ pContext->pulse.pa_context_get_sink_info_by_name = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_get_sink_info_by_name");
+ pContext->pulse.pa_context_get_source_info_by_name = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_get_source_info_by_name");
+ pContext->pulse.pa_operation_unref = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_operation_unref");
+ pContext->pulse.pa_operation_get_state = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_operation_get_state");
+ pContext->pulse.pa_channel_map_init_extend = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_channel_map_init_extend");
+ pContext->pulse.pa_channel_map_valid = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_channel_map_valid");
+ pContext->pulse.pa_channel_map_compatible = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_channel_map_compatible");
+ pContext->pulse.pa_stream_new = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_new");
+ pContext->pulse.pa_stream_unref = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_unref");
+ pContext->pulse.pa_stream_connect_playback = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_connect_playback");
+ pContext->pulse.pa_stream_connect_record = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_connect_record");
+ pContext->pulse.pa_stream_disconnect = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_disconnect");
+ pContext->pulse.pa_stream_get_state = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_get_state");
+ pContext->pulse.pa_stream_get_sample_spec = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_get_sample_spec");
+ pContext->pulse.pa_stream_get_channel_map = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_get_channel_map");
+ pContext->pulse.pa_stream_get_buffer_attr = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_get_buffer_attr");
+ pContext->pulse.pa_stream_set_buffer_attr = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_set_buffer_attr");
+ pContext->pulse.pa_stream_get_device_name = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_get_device_name");
+ pContext->pulse.pa_stream_set_write_callback = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_set_write_callback");
+ pContext->pulse.pa_stream_set_read_callback = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_set_read_callback");
+ pContext->pulse.pa_stream_set_suspended_callback = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_set_suspended_callback");
+ pContext->pulse.pa_stream_set_moved_callback = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_set_moved_callback");
+ pContext->pulse.pa_stream_is_suspended = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_is_suspended");
+ pContext->pulse.pa_stream_flush = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_flush");
+ pContext->pulse.pa_stream_drain = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_drain");
+ pContext->pulse.pa_stream_is_corked = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_is_corked");
+ pContext->pulse.pa_stream_cork = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_cork");
+ pContext->pulse.pa_stream_trigger = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_trigger");
+ pContext->pulse.pa_stream_begin_write = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_begin_write");
+ pContext->pulse.pa_stream_write = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_write");
+ pContext->pulse.pa_stream_peek = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_peek");
+ pContext->pulse.pa_stream_drop = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_drop");
+ pContext->pulse.pa_stream_writable_size = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_writable_size");
+ pContext->pulse.pa_stream_readable_size = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_readable_size");
+#else
+ /* This strange assignment system is just for type safety. */
+ ma_pa_mainloop_new_proc _pa_mainloop_new = pa_mainloop_new;
+ ma_pa_mainloop_free_proc _pa_mainloop_free = pa_mainloop_free;
+ ma_pa_mainloop_quit_proc _pa_mainloop_quit = pa_mainloop_quit;
+ ma_pa_mainloop_get_api_proc _pa_mainloop_get_api = pa_mainloop_get_api;
+ ma_pa_mainloop_iterate_proc _pa_mainloop_iterate = pa_mainloop_iterate;
+ ma_pa_mainloop_wakeup_proc _pa_mainloop_wakeup = pa_mainloop_wakeup;
+ ma_pa_threaded_mainloop_new_proc _pa_threaded_mainloop_new = pa_threaded_mainloop_new;
+ ma_pa_threaded_mainloop_free_proc _pa_threaded_mainloop_free = pa_threaded_mainloop_free;
+ ma_pa_threaded_mainloop_start_proc _pa_threaded_mainloop_start = pa_threaded_mainloop_start;
+ ma_pa_threaded_mainloop_stop_proc _pa_threaded_mainloop_stop = pa_threaded_mainloop_stop;
+ ma_pa_threaded_mainloop_lock_proc _pa_threaded_mainloop_lock = pa_threaded_mainloop_lock;
+ ma_pa_threaded_mainloop_unlock_proc _pa_threaded_mainloop_unlock = pa_threaded_mainloop_unlock;
+ ma_pa_threaded_mainloop_wait_proc _pa_threaded_mainloop_wait = pa_threaded_mainloop_wait;
+ ma_pa_threaded_mainloop_signal_proc _pa_threaded_mainloop_signal = pa_threaded_mainloop_signal;
+ ma_pa_threaded_mainloop_accept_proc _pa_threaded_mainloop_accept = pa_threaded_mainloop_accept;
+ ma_pa_threaded_mainloop_get_retval_proc _pa_threaded_mainloop_get_retval = pa_threaded_mainloop_get_retval;
+ ma_pa_threaded_mainloop_get_api_proc _pa_threaded_mainloop_get_api = pa_threaded_mainloop_get_api;
+ ma_pa_threaded_mainloop_in_thread_proc _pa_threaded_mainloop_in_thread = pa_threaded_mainloop_in_thread;
+ ma_pa_threaded_mainloop_set_name_proc _pa_threaded_mainloop_set_name = pa_threaded_mainloop_set_name;
+ ma_pa_context_new_proc _pa_context_new = pa_context_new;
+ ma_pa_context_unref_proc _pa_context_unref = pa_context_unref;
+ ma_pa_context_connect_proc _pa_context_connect = pa_context_connect;
+ ma_pa_context_disconnect_proc _pa_context_disconnect = pa_context_disconnect;
+ ma_pa_context_set_state_callback_proc _pa_context_set_state_callback = pa_context_set_state_callback;
+ ma_pa_context_get_state_proc _pa_context_get_state = pa_context_get_state;
+ ma_pa_context_get_sink_info_list_proc _pa_context_get_sink_info_list = pa_context_get_sink_info_list;
+ ma_pa_context_get_source_info_list_proc _pa_context_get_source_info_list = pa_context_get_source_info_list;
+ ma_pa_context_get_sink_info_by_name_proc _pa_context_get_sink_info_by_name = pa_context_get_sink_info_by_name;
+ ma_pa_context_get_source_info_by_name_proc _pa_context_get_source_info_by_name= pa_context_get_source_info_by_name;
+ ma_pa_operation_unref_proc _pa_operation_unref = pa_operation_unref;
+ ma_pa_operation_get_state_proc _pa_operation_get_state = pa_operation_get_state;
+ ma_pa_channel_map_init_extend_proc _pa_channel_map_init_extend = pa_channel_map_init_extend;
+ ma_pa_channel_map_valid_proc _pa_channel_map_valid = pa_channel_map_valid;
+ ma_pa_channel_map_compatible_proc _pa_channel_map_compatible = pa_channel_map_compatible;
+ ma_pa_stream_new_proc _pa_stream_new = pa_stream_new;
+ ma_pa_stream_unref_proc _pa_stream_unref = pa_stream_unref;
+ ma_pa_stream_connect_playback_proc _pa_stream_connect_playback = pa_stream_connect_playback;
+ ma_pa_stream_connect_record_proc _pa_stream_connect_record = pa_stream_connect_record;
+ ma_pa_stream_disconnect_proc _pa_stream_disconnect = pa_stream_disconnect;
+ ma_pa_stream_get_state_proc _pa_stream_get_state = pa_stream_get_state;
+ ma_pa_stream_get_sample_spec_proc _pa_stream_get_sample_spec = pa_stream_get_sample_spec;
+ ma_pa_stream_get_channel_map_proc _pa_stream_get_channel_map = pa_stream_get_channel_map;
+ ma_pa_stream_get_buffer_attr_proc _pa_stream_get_buffer_attr = pa_stream_get_buffer_attr;
+ ma_pa_stream_set_buffer_attr_proc _pa_stream_set_buffer_attr = pa_stream_set_buffer_attr;
+ ma_pa_stream_get_device_name_proc _pa_stream_get_device_name = pa_stream_get_device_name;
+ ma_pa_stream_set_write_callback_proc _pa_stream_set_write_callback = pa_stream_set_write_callback;
+ ma_pa_stream_set_read_callback_proc _pa_stream_set_read_callback = pa_stream_set_read_callback;
+ ma_pa_stream_set_suspended_callback_proc _pa_stream_set_suspended_callback = pa_stream_set_suspended_callback;
+ ma_pa_stream_set_moved_callback_proc _pa_stream_set_moved_callback = pa_stream_set_moved_callback;
+ ma_pa_stream_is_suspended_proc _pa_stream_is_suspended = pa_stream_is_suspended;
+ ma_pa_stream_flush_proc _pa_stream_flush = pa_stream_flush;
+ ma_pa_stream_drain_proc _pa_stream_drain = pa_stream_drain;
+ ma_pa_stream_is_corked_proc _pa_stream_is_corked = pa_stream_is_corked;
+ ma_pa_stream_cork_proc _pa_stream_cork = pa_stream_cork;
+ ma_pa_stream_trigger_proc _pa_stream_trigger = pa_stream_trigger;
+ ma_pa_stream_begin_write_proc _pa_stream_begin_write = pa_stream_begin_write;
+ ma_pa_stream_write_proc _pa_stream_write = pa_stream_write;
+ ma_pa_stream_peek_proc _pa_stream_peek = pa_stream_peek;
+ ma_pa_stream_drop_proc _pa_stream_drop = pa_stream_drop;
+ ma_pa_stream_writable_size_proc _pa_stream_writable_size = pa_stream_writable_size;
+ ma_pa_stream_readable_size_proc _pa_stream_readable_size = pa_stream_readable_size;
+
+ pContext->pulse.pa_mainloop_new = (ma_proc)_pa_mainloop_new;
+ pContext->pulse.pa_mainloop_free = (ma_proc)_pa_mainloop_free;
+ pContext->pulse.pa_mainloop_quit = (ma_proc)_pa_mainloop_quit;
+ pContext->pulse.pa_mainloop_get_api = (ma_proc)_pa_mainloop_get_api;
+ pContext->pulse.pa_mainloop_iterate = (ma_proc)_pa_mainloop_iterate;
+ pContext->pulse.pa_mainloop_wakeup = (ma_proc)_pa_mainloop_wakeup;
+ pContext->pulse.pa_threaded_mainloop_new = (ma_proc)_pa_threaded_mainloop_new;
+ pContext->pulse.pa_threaded_mainloop_free = (ma_proc)_pa_threaded_mainloop_free;
+ pContext->pulse.pa_threaded_mainloop_start = (ma_proc)_pa_threaded_mainloop_start;
+ pContext->pulse.pa_threaded_mainloop_stop = (ma_proc)_pa_threaded_mainloop_stop;
+ pContext->pulse.pa_threaded_mainloop_lock = (ma_proc)_pa_threaded_mainloop_lock;
+ pContext->pulse.pa_threaded_mainloop_unlock = (ma_proc)_pa_threaded_mainloop_unlock;
+ pContext->pulse.pa_threaded_mainloop_wait = (ma_proc)_pa_threaded_mainloop_wait;
+ pContext->pulse.pa_threaded_mainloop_signal = (ma_proc)_pa_threaded_mainloop_signal;
+ pContext->pulse.pa_threaded_mainloop_accept = (ma_proc)_pa_threaded_mainloop_accept;
+ pContext->pulse.pa_threaded_mainloop_get_retval = (ma_proc)_pa_threaded_mainloop_get_retval;
+ pContext->pulse.pa_threaded_mainloop_get_api = (ma_proc)_pa_threaded_mainloop_get_api;
+ pContext->pulse.pa_threaded_mainloop_in_thread = (ma_proc)_pa_threaded_mainloop_in_thread;
+ pContext->pulse.pa_threaded_mainloop_set_name = (ma_proc)_pa_threaded_mainloop_set_name;
+ pContext->pulse.pa_context_new = (ma_proc)_pa_context_new;
+ pContext->pulse.pa_context_unref = (ma_proc)_pa_context_unref;
+ pContext->pulse.pa_context_connect = (ma_proc)_pa_context_connect;
+ pContext->pulse.pa_context_disconnect = (ma_proc)_pa_context_disconnect;
+ pContext->pulse.pa_context_set_state_callback = (ma_proc)_pa_context_set_state_callback;
+ pContext->pulse.pa_context_get_state = (ma_proc)_pa_context_get_state;
+ pContext->pulse.pa_context_get_sink_info_list = (ma_proc)_pa_context_get_sink_info_list;
+ pContext->pulse.pa_context_get_source_info_list = (ma_proc)_pa_context_get_source_info_list;
+ pContext->pulse.pa_context_get_sink_info_by_name = (ma_proc)_pa_context_get_sink_info_by_name;
+ pContext->pulse.pa_context_get_source_info_by_name = (ma_proc)_pa_context_get_source_info_by_name;
+ pContext->pulse.pa_operation_unref = (ma_proc)_pa_operation_unref;
+ pContext->pulse.pa_operation_get_state = (ma_proc)_pa_operation_get_state;
+ pContext->pulse.pa_channel_map_init_extend = (ma_proc)_pa_channel_map_init_extend;
+ pContext->pulse.pa_channel_map_valid = (ma_proc)_pa_channel_map_valid;
+ pContext->pulse.pa_channel_map_compatible = (ma_proc)_pa_channel_map_compatible;
+ pContext->pulse.pa_stream_new = (ma_proc)_pa_stream_new;
+ pContext->pulse.pa_stream_unref = (ma_proc)_pa_stream_unref;
+ pContext->pulse.pa_stream_connect_playback = (ma_proc)_pa_stream_connect_playback;
+ pContext->pulse.pa_stream_connect_record = (ma_proc)_pa_stream_connect_record;
+ pContext->pulse.pa_stream_disconnect = (ma_proc)_pa_stream_disconnect;
+ pContext->pulse.pa_stream_get_state = (ma_proc)_pa_stream_get_state;
+ pContext->pulse.pa_stream_get_sample_spec = (ma_proc)_pa_stream_get_sample_spec;
+ pContext->pulse.pa_stream_get_channel_map = (ma_proc)_pa_stream_get_channel_map;
+ pContext->pulse.pa_stream_get_buffer_attr = (ma_proc)_pa_stream_get_buffer_attr;
+ pContext->pulse.pa_stream_set_buffer_attr = (ma_proc)_pa_stream_set_buffer_attr;
+ pContext->pulse.pa_stream_get_device_name = (ma_proc)_pa_stream_get_device_name;
+ pContext->pulse.pa_stream_set_write_callback = (ma_proc)_pa_stream_set_write_callback;
+ pContext->pulse.pa_stream_set_read_callback = (ma_proc)_pa_stream_set_read_callback;
+ pContext->pulse.pa_stream_set_suspended_callback = (ma_proc)_pa_stream_set_suspended_callback;
+ pContext->pulse.pa_stream_set_moved_callback = (ma_proc)_pa_stream_set_moved_callback;
+ pContext->pulse.pa_stream_is_suspended = (ma_proc)_pa_stream_is_suspended;
+ pContext->pulse.pa_stream_flush = (ma_proc)_pa_stream_flush;
+ pContext->pulse.pa_stream_drain = (ma_proc)_pa_stream_drain;
+ pContext->pulse.pa_stream_is_corked = (ma_proc)_pa_stream_is_corked;
+ pContext->pulse.pa_stream_cork = (ma_proc)_pa_stream_cork;
+ pContext->pulse.pa_stream_trigger = (ma_proc)_pa_stream_trigger;
+ pContext->pulse.pa_stream_begin_write = (ma_proc)_pa_stream_begin_write;
+ pContext->pulse.pa_stream_write = (ma_proc)_pa_stream_write;
+ pContext->pulse.pa_stream_peek = (ma_proc)_pa_stream_peek;
+ pContext->pulse.pa_stream_drop = (ma_proc)_pa_stream_drop;
+ pContext->pulse.pa_stream_writable_size = (ma_proc)_pa_stream_writable_size;
+ pContext->pulse.pa_stream_readable_size = (ma_proc)_pa_stream_readable_size;
+#endif
+
+ /* We need to make a copy of the application and server names so we can pass them to the pa_context of each device. */
+ pContext->pulse.pApplicationName = ma_copy_string(pConfig->pulse.pApplicationName, &pContext->allocationCallbacks);
+ if (pContext->pulse.pApplicationName == NULL && pConfig->pulse.pApplicationName != NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+
+ pContext->pulse.pServerName = ma_copy_string(pConfig->pulse.pServerName, &pContext->allocationCallbacks);
+ if (pContext->pulse.pServerName == NULL && pConfig->pulse.pServerName != NULL) {
+ ma_free(pContext->pulse.pApplicationName, &pContext->allocationCallbacks);
+ return MA_OUT_OF_MEMORY;
+ }
+
+ result = ma_init_pa_mainloop_and_pa_context__pulse(pContext, pConfig->pulse.pApplicationName, pConfig->pulse.pServerName, pConfig->pulse.tryAutoSpawn, &pContext->pulse.pMainLoop, &pContext->pulse.pPulseContext);
+ if (result != MA_SUCCESS) {
+ ma_free(pContext->pulse.pServerName, &pContext->allocationCallbacks);
+ ma_free(pContext->pulse.pApplicationName, &pContext->allocationCallbacks);
+ #ifndef MA_NO_RUNTIME_LINKING
+ ma_dlclose(pContext, pContext->pulse.pulseSO);
+ #endif
+ return result;
+ }
+
+ /* With pa_mainloop we run a synchronous backend, but we implement our own main loop. */
+ pCallbacks->onContextInit = ma_context_init__pulse;
+ pCallbacks->onContextUninit = ma_context_uninit__pulse;
+ pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__pulse;
+ pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__pulse;
+ pCallbacks->onDeviceInit = ma_device_init__pulse;
+ pCallbacks->onDeviceUninit = ma_device_uninit__pulse;
+ pCallbacks->onDeviceStart = ma_device_start__pulse;
+ pCallbacks->onDeviceStop = ma_device_stop__pulse;
+ pCallbacks->onDeviceRead = NULL; /* Not used because we're implementing onDeviceDataLoop. */
+ pCallbacks->onDeviceWrite = NULL; /* Not used because we're implementing onDeviceDataLoop. */
+ pCallbacks->onDeviceDataLoop = ma_device_data_loop__pulse;
+ pCallbacks->onDeviceDataLoopWakeup = ma_device_data_loop_wakeup__pulse;
+
+ return MA_SUCCESS;
+}
+#endif
+
+
+/******************************************************************************
+
+JACK Backend
+
+******************************************************************************/
+#ifdef MA_HAS_JACK
+
+/* It is assumed jack.h is available when compile-time linking is being used. */
+#ifdef MA_NO_RUNTIME_LINKING
+#include
+
+typedef jack_nframes_t ma_jack_nframes_t;
+typedef jack_options_t ma_jack_options_t;
+typedef jack_status_t ma_jack_status_t;
+typedef jack_client_t ma_jack_client_t;
+typedef jack_port_t ma_jack_port_t;
+typedef JackProcessCallback ma_JackProcessCallback;
+typedef JackBufferSizeCallback ma_JackBufferSizeCallback;
+typedef JackShutdownCallback ma_JackShutdownCallback;
+#define MA_JACK_DEFAULT_AUDIO_TYPE JACK_DEFAULT_AUDIO_TYPE
+#define ma_JackNoStartServer JackNoStartServer
+#define ma_JackPortIsInput JackPortIsInput
+#define ma_JackPortIsOutput JackPortIsOutput
+#define ma_JackPortIsPhysical JackPortIsPhysical
+#else
+typedef ma_uint32 ma_jack_nframes_t;
+typedef int ma_jack_options_t;
+typedef int ma_jack_status_t;
+typedef struct ma_jack_client_t ma_jack_client_t;
+typedef struct ma_jack_port_t ma_jack_port_t;
+typedef int (* ma_JackProcessCallback) (ma_jack_nframes_t nframes, void* arg);
+typedef int (* ma_JackBufferSizeCallback)(ma_jack_nframes_t nframes, void* arg);
+typedef void (* ma_JackShutdownCallback) (void* arg);
+#define MA_JACK_DEFAULT_AUDIO_TYPE "32 bit float mono audio"
+#define ma_JackNoStartServer 1
+#define ma_JackPortIsInput 1
+#define ma_JackPortIsOutput 2
+#define ma_JackPortIsPhysical 4
+#endif
+
+typedef ma_jack_client_t* (* ma_jack_client_open_proc) (const char* client_name, ma_jack_options_t options, ma_jack_status_t* status, ...);
+typedef int (* ma_jack_client_close_proc) (ma_jack_client_t* client);
+typedef int (* ma_jack_client_name_size_proc) (void);
+typedef int (* ma_jack_set_process_callback_proc) (ma_jack_client_t* client, ma_JackProcessCallback process_callback, void* arg);
+typedef int (* ma_jack_set_buffer_size_callback_proc)(ma_jack_client_t* client, ma_JackBufferSizeCallback bufsize_callback, void* arg);
+typedef void (* ma_jack_on_shutdown_proc) (ma_jack_client_t* client, ma_JackShutdownCallback function, void* arg);
+typedef ma_jack_nframes_t (* ma_jack_get_sample_rate_proc) (ma_jack_client_t* client);
+typedef ma_jack_nframes_t (* ma_jack_get_buffer_size_proc) (ma_jack_client_t* client);
+typedef const char** (* ma_jack_get_ports_proc) (ma_jack_client_t* client, const char* port_name_pattern, const char* type_name_pattern, unsigned long flags);
+typedef int (* ma_jack_activate_proc) (ma_jack_client_t* client);
+typedef int (* ma_jack_deactivate_proc) (ma_jack_client_t* client);
+typedef int (* ma_jack_connect_proc) (ma_jack_client_t* client, const char* source_port, const char* destination_port);
+typedef ma_jack_port_t* (* ma_jack_port_register_proc) (ma_jack_client_t* client, const char* port_name, const char* port_type, unsigned long flags, unsigned long buffer_size);
+typedef const char* (* ma_jack_port_name_proc) (const ma_jack_port_t* port);
+typedef void* (* ma_jack_port_get_buffer_proc) (ma_jack_port_t* port, ma_jack_nframes_t nframes);
+typedef void (* ma_jack_free_proc) (void* ptr);
+
+static ma_result ma_context_open_client__jack(ma_context* pContext, ma_jack_client_t** ppClient)
+{
+ size_t maxClientNameSize;
+ char clientName[256];
+ ma_jack_status_t status;
+ ma_jack_client_t* pClient;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(ppClient != NULL);
+
+ if (ppClient) {
+ *ppClient = NULL;
+ }
+
+ maxClientNameSize = ((ma_jack_client_name_size_proc)pContext->jack.jack_client_name_size)(); /* Includes null terminator. */
+ ma_strncpy_s(clientName, ma_min(sizeof(clientName), maxClientNameSize), (pContext->jack.pClientName != NULL) ? pContext->jack.pClientName : "miniaudio", (size_t)-1);
+
+ pClient = ((ma_jack_client_open_proc)pContext->jack.jack_client_open)(clientName, (pContext->jack.tryStartServer) ? 0 : ma_JackNoStartServer, &status, NULL);
+ if (pClient == NULL) {
+ return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
+ }
+
+ if (ppClient) {
+ *ppClient = pClient;
+ }
+
+ return MA_SUCCESS;
+}
+
+
+static ma_result ma_context_enumerate_devices__jack(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+{
+ ma_bool32 cbResult = MA_TRUE;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(callback != NULL);
+
+ /* Playback. */
+ if (cbResult) {
+ ma_device_info deviceInfo;
+ MA_ZERO_OBJECT(&deviceInfo);
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
+ deviceInfo.isDefault = MA_TRUE; /* JACK only uses default devices. */
+ cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
+ }
+
+ /* Capture. */
+ if (cbResult) {
+ ma_device_info deviceInfo;
+ MA_ZERO_OBJECT(&deviceInfo);
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
+ deviceInfo.isDefault = MA_TRUE; /* JACK only uses default devices. */
+ cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
+ }
+
+ (void)cbResult; /* For silencing a static analysis warning. */
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_get_device_info__jack(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+{
+ ma_jack_client_t* pClient;
+ ma_result result;
+ const char** ppPorts;
+
+ MA_ASSERT(pContext != NULL);
+
+ if (pDeviceID != NULL && pDeviceID->jack != 0) {
+ return MA_NO_DEVICE; /* Don't know the device. */
+ }
+
+ /* Name / Description */
+ if (deviceType == ma_device_type_playback) {
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
+ } else {
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
+ }
+
+ /* Jack only uses default devices. */
+ pDeviceInfo->isDefault = MA_TRUE;
+
+ /* Jack only supports f32 and has a specific channel count and sample rate. */
+ pDeviceInfo->nativeDataFormats[0].format = ma_format_f32;
+
+ /* The channel count and sample rate can only be determined by opening the device. */
+ result = ma_context_open_client__jack(pContext, &pClient);
+ if (result != MA_SUCCESS) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[JACK] Failed to open client.");
+ return result;
+ }
+
+ pDeviceInfo->nativeDataFormats[0].sampleRate = ((ma_jack_get_sample_rate_proc)pContext->jack.jack_get_sample_rate)((ma_jack_client_t*)pClient);
+ pDeviceInfo->nativeDataFormats[0].channels = 0;
+
+ ppPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ((deviceType == ma_device_type_playback) ? ma_JackPortIsInput : ma_JackPortIsOutput));
+ if (ppPorts == NULL) {
+ ((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pClient);
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[JACK] Failed to query physical ports.");
+ return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
+ }
+
+ while (ppPorts[pDeviceInfo->nativeDataFormats[0].channels] != NULL) {
+ pDeviceInfo->nativeDataFormats[0].channels += 1;
+ }
+
+ pDeviceInfo->nativeDataFormats[0].flags = 0;
+ pDeviceInfo->nativeDataFormatCount = 1;
+
+ ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppPorts);
+ ((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pClient);
+
+ (void)pContext;
+ return MA_SUCCESS;
+}
+
+
+static ma_result ma_device_uninit__jack(ma_device* pDevice)
+{
+ ma_context* pContext;
+
+ MA_ASSERT(pDevice != NULL);
+
+ pContext = pDevice->pContext;
+ MA_ASSERT(pContext != NULL);
+
+ if (pDevice->jack.pClient != NULL) {
+ ((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pDevice->jack.pClient);
+ }
+
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ ma_free(pDevice->jack.pIntermediaryBufferCapture, &pDevice->pContext->allocationCallbacks);
+ ma_free(pDevice->jack.ppPortsCapture, &pDevice->pContext->allocationCallbacks);
+ }
+
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ ma_free(pDevice->jack.pIntermediaryBufferPlayback, &pDevice->pContext->allocationCallbacks);
+ ma_free(pDevice->jack.ppPortsPlayback, &pDevice->pContext->allocationCallbacks);
+ }
+
+ return MA_SUCCESS;
+}
+
+static void ma_device__jack_shutdown_callback(void* pUserData)
+{
+ /* JACK died. Stop the device. */
+ ma_device* pDevice = (ma_device*)pUserData;
+ MA_ASSERT(pDevice != NULL);
+
+ ma_device_stop(pDevice);
+}
+
+static int ma_device__jack_buffer_size_callback(ma_jack_nframes_t frameCount, void* pUserData)
+{
+ ma_device* pDevice = (ma_device*)pUserData;
+ MA_ASSERT(pDevice != NULL);
+
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ size_t newBufferSize = frameCount * (pDevice->capture.internalChannels * ma_get_bytes_per_sample(pDevice->capture.internalFormat));
+ float* pNewBuffer = (float*)ma_calloc(newBufferSize, &pDevice->pContext->allocationCallbacks);
+ if (pNewBuffer == NULL) {
+ return MA_OUT_OF_MEMORY;
}
- static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_fetch_add_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
- {
- c89atomic_uint32 result;
- (void)order;
- #if defined(C89ATOMIC_X86) || defined(C89ATOMIC_X64)
- __asm__ __volatile__("lock; xadd %1, %0" : "+m"(*dst), "=a"(result) : "a"(src) : "cc");
- #else
- #error Unsupported architecture. Please submit a feature request.
- #endif
- return result;
+
+ ma_free(pDevice->jack.pIntermediaryBufferCapture, &pDevice->pContext->allocationCallbacks);
+
+ pDevice->jack.pIntermediaryBufferCapture = pNewBuffer;
+ pDevice->playback.internalPeriodSizeInFrames = frameCount;
+ }
+
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ size_t newBufferSize = frameCount * (pDevice->playback.internalChannels * ma_get_bytes_per_sample(pDevice->playback.internalFormat));
+ float* pNewBuffer = (float*)ma_calloc(newBufferSize, &pDevice->pContext->allocationCallbacks);
+ if (pNewBuffer == NULL) {
+ return MA_OUT_OF_MEMORY;
}
- static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_fetch_add_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
- {
- #if defined(C89ATOMIC_X86)
- c89atomic_uint64 oldValue;
- c89atomic_uint64 newValue;
- (void)order;
- do {
- oldValue = *dst;
- newValue = oldValue + src;
- } while (c89atomic_compare_and_swap_64(dst, oldValue, newValue) != oldValue);
- return oldValue;
- #elif defined(C89ATOMIC_X64)
- c89atomic_uint64 result;
- (void)order;
- __asm__ __volatile__("lock; xadd %1, %0" : "+m"(*dst), "=a"(result) : "a"(src) : "cc");
- return result;
- #endif
+
+ ma_free(pDevice->jack.pIntermediaryBufferPlayback, &pDevice->pContext->allocationCallbacks);
+
+ pDevice->jack.pIntermediaryBufferPlayback = pNewBuffer;
+ pDevice->playback.internalPeriodSizeInFrames = frameCount;
+ }
+
+ return 0;
+}
+
+static int ma_device__jack_process_callback(ma_jack_nframes_t frameCount, void* pUserData)
+{
+ ma_device* pDevice;
+ ma_context* pContext;
+ ma_uint32 iChannel;
+
+ pDevice = (ma_device*)pUserData;
+ MA_ASSERT(pDevice != NULL);
+
+ pContext = pDevice->pContext;
+ MA_ASSERT(pContext != NULL);
+
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ /* Channels need to be interleaved. */
+ for (iChannel = 0; iChannel < pDevice->capture.internalChannels; ++iChannel) {
+ const float* pSrc = (const float*)((ma_jack_port_get_buffer_proc)pContext->jack.jack_port_get_buffer)((ma_jack_port_t*)pDevice->jack.ppPortsCapture[iChannel], frameCount);
+ if (pSrc != NULL) {
+ float* pDst = pDevice->jack.pIntermediaryBufferCapture + iChannel;
+ ma_jack_nframes_t iFrame;
+ for (iFrame = 0; iFrame < frameCount; ++iFrame) {
+ *pDst = *pSrc;
+
+ pDst += pDevice->capture.internalChannels;
+ pSrc += 1;
+ }
+ }
}
- static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_fetch_sub_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
- {
- c89atomic_uint8 oldValue;
- c89atomic_uint8 newValue;
- do {
- oldValue = *dst;
- newValue = (c89atomic_uint8)(oldValue - src);
- } while (c89atomic_compare_and_swap_8(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ ma_device_handle_backend_data_callback(pDevice, NULL, pDevice->jack.pIntermediaryBufferCapture, frameCount);
+ }
+
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ ma_device_handle_backend_data_callback(pDevice, pDevice->jack.pIntermediaryBufferPlayback, NULL, frameCount);
+
+ /* Channels need to be deinterleaved. */
+ for (iChannel = 0; iChannel < pDevice->playback.internalChannels; ++iChannel) {
+ float* pDst = (float*)((ma_jack_port_get_buffer_proc)pContext->jack.jack_port_get_buffer)((ma_jack_port_t*)pDevice->jack.ppPortsPlayback[iChannel], frameCount);
+ if (pDst != NULL) {
+ const float* pSrc = pDevice->jack.pIntermediaryBufferPlayback + iChannel;
+ ma_jack_nframes_t iFrame;
+ for (iFrame = 0; iFrame < frameCount; ++iFrame) {
+ *pDst = *pSrc;
+
+ pDst += 1;
+ pSrc += pDevice->playback.internalChannels;
+ }
+ }
}
- static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_fetch_sub_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
- {
- c89atomic_uint16 oldValue;
- c89atomic_uint16 newValue;
- do {
- oldValue = *dst;
- newValue = (c89atomic_uint16)(oldValue - src);
- } while (c89atomic_compare_and_swap_16(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+ }
+
+ return 0;
+}
+
+static ma_result ma_device_init__jack(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+{
+ ma_result result;
+ ma_uint32 periodSizeInFrames;
+
+ MA_ASSERT(pConfig != NULL);
+ MA_ASSERT(pDevice != NULL);
+
+ if (pConfig->deviceType == ma_device_type_loopback) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Loopback mode not supported.");
+ return MA_DEVICE_TYPE_NOT_SUPPORTED;
+ }
+
+ /* Only supporting default devices with JACK. */
+ if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->pDeviceID != NULL && pDescriptorPlayback->pDeviceID->jack != 0) ||
+ ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->pDeviceID != NULL && pDescriptorCapture->pDeviceID->jack != 0)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Only default devices are supported.");
+ return MA_NO_DEVICE;
+ }
+
+ /* No exclusive mode with the JACK backend. */
+ if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) ||
+ ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Exclusive mode not supported.");
+ return MA_SHARE_MODE_NOT_SUPPORTED;
+ }
+
+ /* Open the client. */
+ result = ma_context_open_client__jack(pDevice->pContext, (ma_jack_client_t**)&pDevice->jack.pClient);
+ if (result != MA_SUCCESS) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to open client.");
+ return result;
+ }
+
+ /* Callbacks. */
+ if (((ma_jack_set_process_callback_proc)pDevice->pContext->jack.jack_set_process_callback)((ma_jack_client_t*)pDevice->jack.pClient, ma_device__jack_process_callback, pDevice) != 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to set process callback.");
+ return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
+ }
+ if (((ma_jack_set_buffer_size_callback_proc)pDevice->pContext->jack.jack_set_buffer_size_callback)((ma_jack_client_t*)pDevice->jack.pClient, ma_device__jack_buffer_size_callback, pDevice) != 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to set buffer size callback.");
+ return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
+ }
+
+ ((ma_jack_on_shutdown_proc)pDevice->pContext->jack.jack_on_shutdown)((ma_jack_client_t*)pDevice->jack.pClient, ma_device__jack_shutdown_callback, pDevice);
+
+
+ /* The buffer size in frames can change. */
+ periodSizeInFrames = ((ma_jack_get_buffer_size_proc)pDevice->pContext->jack.jack_get_buffer_size)((ma_jack_client_t*)pDevice->jack.pClient);
+
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
+ ma_uint32 iPort;
+ const char** ppPorts;
+
+ pDescriptorCapture->format = ma_format_f32;
+ pDescriptorCapture->channels = 0;
+ pDescriptorCapture->sampleRate = ((ma_jack_get_sample_rate_proc)pDevice->pContext->jack.jack_get_sample_rate)((ma_jack_client_t*)pDevice->jack.pClient);
+ ma_channel_map_init_standard(ma_standard_channel_map_alsa, pDescriptorCapture->channelMap, ma_countof(pDescriptorCapture->channelMap), pDescriptorCapture->channels);
+
+ ppPorts = ((ma_jack_get_ports_proc)pDevice->pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ma_JackPortIsOutput);
+ if (ppPorts == NULL) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to query physical ports.");
+ return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
}
- static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_fetch_sub_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
- {
- c89atomic_uint32 oldValue;
- c89atomic_uint32 newValue;
- do {
- oldValue = *dst;
- newValue = oldValue - src;
- } while (c89atomic_compare_and_swap_32(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ /* Need to count the number of ports first so we can allocate some memory. */
+ while (ppPorts[pDescriptorCapture->channels] != NULL) {
+ pDescriptorCapture->channels += 1;
}
- static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_fetch_sub_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
- {
- c89atomic_uint64 oldValue;
- c89atomic_uint64 newValue;
- do {
- oldValue = *dst;
- newValue = oldValue - src;
- } while (c89atomic_compare_and_swap_64(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ pDevice->jack.ppPortsCapture = (ma_ptr*)ma_malloc(sizeof(*pDevice->jack.ppPortsCapture) * pDescriptorCapture->channels, &pDevice->pContext->allocationCallbacks);
+ if (pDevice->jack.ppPortsCapture == NULL) {
+ return MA_OUT_OF_MEMORY;
}
- static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_fetch_and_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
- {
- c89atomic_uint8 oldValue;
- c89atomic_uint8 newValue;
- do {
- oldValue = *dst;
- newValue = (c89atomic_uint8)(oldValue & src);
- } while (c89atomic_compare_and_swap_8(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ for (iPort = 0; iPort < pDescriptorCapture->channels; iPort += 1) {
+ char name[64];
+ ma_strcpy_s(name, sizeof(name), "capture");
+ ma_itoa_s((int)iPort, name+7, sizeof(name)-7, 10); /* 7 = length of "capture" */
+
+ pDevice->jack.ppPortsCapture[iPort] = ((ma_jack_port_register_proc)pDevice->pContext->jack.jack_port_register)((ma_jack_client_t*)pDevice->jack.pClient, name, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsInput, 0);
+ if (pDevice->jack.ppPortsCapture[iPort] == NULL) {
+ ((ma_jack_free_proc)pDevice->pContext->jack.jack_free)((void*)ppPorts);
+ ma_device_uninit__jack(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to register ports.");
+ return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
+ }
}
- static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_fetch_and_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
- {
- c89atomic_uint16 oldValue;
- c89atomic_uint16 newValue;
- do {
- oldValue = *dst;
- newValue = (c89atomic_uint16)(oldValue & src);
- } while (c89atomic_compare_and_swap_16(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ ((ma_jack_free_proc)pDevice->pContext->jack.jack_free)((void*)ppPorts);
+
+ pDescriptorCapture->periodSizeInFrames = periodSizeInFrames;
+ pDescriptorCapture->periodCount = 1; /* There's no notion of a period in JACK. Just set to 1. */
+
+ pDevice->jack.pIntermediaryBufferCapture = (float*)ma_calloc(pDescriptorCapture->periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels), &pDevice->pContext->allocationCallbacks);
+ if (pDevice->jack.pIntermediaryBufferCapture == NULL) {
+ ma_device_uninit__jack(pDevice);
+ return MA_OUT_OF_MEMORY;
}
- static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_fetch_and_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
- {
- c89atomic_uint32 oldValue;
- c89atomic_uint32 newValue;
- do {
- oldValue = *dst;
- newValue = oldValue & src;
- } while (c89atomic_compare_and_swap_32(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+ }
+
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ ma_uint32 iPort;
+ const char** ppPorts;
+
+ pDescriptorPlayback->format = ma_format_f32;
+ pDescriptorPlayback->channels = 0;
+ pDescriptorPlayback->sampleRate = ((ma_jack_get_sample_rate_proc)pDevice->pContext->jack.jack_get_sample_rate)((ma_jack_client_t*)pDevice->jack.pClient);
+ ma_channel_map_init_standard(ma_standard_channel_map_alsa, pDescriptorPlayback->channelMap, ma_countof(pDescriptorPlayback->channelMap), pDescriptorPlayback->channels);
+
+ ppPorts = ((ma_jack_get_ports_proc)pDevice->pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ma_JackPortIsInput);
+ if (ppPorts == NULL) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to query physical ports.");
+ return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
}
- static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_fetch_and_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
- {
- c89atomic_uint64 oldValue;
- c89atomic_uint64 newValue;
- do {
- oldValue = *dst;
- newValue = oldValue & src;
- } while (c89atomic_compare_and_swap_64(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ /* Need to count the number of ports first so we can allocate some memory. */
+ while (ppPorts[pDescriptorPlayback->channels] != NULL) {
+ pDescriptorPlayback->channels += 1;
}
- static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_fetch_xor_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
- {
- c89atomic_uint8 oldValue;
- c89atomic_uint8 newValue;
- do {
- oldValue = *dst;
- newValue = (c89atomic_uint8)(oldValue ^ src);
- } while (c89atomic_compare_and_swap_8(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ pDevice->jack.ppPortsPlayback = (ma_ptr*)ma_malloc(sizeof(*pDevice->jack.ppPortsPlayback) * pDescriptorPlayback->channels, &pDevice->pContext->allocationCallbacks);
+ if (pDevice->jack.ppPortsPlayback == NULL) {
+ ma_free(pDevice->jack.ppPortsCapture, &pDevice->pContext->allocationCallbacks);
+ return MA_OUT_OF_MEMORY;
}
- static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_fetch_xor_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
- {
- c89atomic_uint16 oldValue;
- c89atomic_uint16 newValue;
- do {
- oldValue = *dst;
- newValue = (c89atomic_uint16)(oldValue ^ src);
- } while (c89atomic_compare_and_swap_16(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ for (iPort = 0; iPort < pDescriptorPlayback->channels; iPort += 1) {
+ char name[64];
+ ma_strcpy_s(name, sizeof(name), "playback");
+ ma_itoa_s((int)iPort, name+8, sizeof(name)-8, 10); /* 8 = length of "playback" */
+
+ pDevice->jack.ppPortsPlayback[iPort] = ((ma_jack_port_register_proc)pDevice->pContext->jack.jack_port_register)((ma_jack_client_t*)pDevice->jack.pClient, name, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsOutput, 0);
+ if (pDevice->jack.ppPortsPlayback[iPort] == NULL) {
+ ((ma_jack_free_proc)pDevice->pContext->jack.jack_free)((void*)ppPorts);
+ ma_device_uninit__jack(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to register ports.");
+ return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
+ }
}
- static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_fetch_xor_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
- {
- c89atomic_uint32 oldValue;
- c89atomic_uint32 newValue;
- do {
- oldValue = *dst;
- newValue = oldValue ^ src;
- } while (c89atomic_compare_and_swap_32(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ ((ma_jack_free_proc)pDevice->pContext->jack.jack_free)((void*)ppPorts);
+
+ pDescriptorPlayback->periodSizeInFrames = periodSizeInFrames;
+ pDescriptorPlayback->periodCount = 1; /* There's no notion of a period in JACK. Just set to 1. */
+
+ pDevice->jack.pIntermediaryBufferPlayback = (float*)ma_calloc(pDescriptorPlayback->periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorPlayback->format, pDescriptorPlayback->channels), &pDevice->pContext->allocationCallbacks);
+ if (pDevice->jack.pIntermediaryBufferPlayback == NULL) {
+ ma_device_uninit__jack(pDevice);
+ return MA_OUT_OF_MEMORY;
}
- static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_fetch_xor_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
- {
- c89atomic_uint64 oldValue;
- c89atomic_uint64 newValue;
- do {
- oldValue = *dst;
- newValue = oldValue ^ src;
- } while (c89atomic_compare_and_swap_64(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+ }
+
+ return MA_SUCCESS;
+}
+
+
+static ma_result ma_device_start__jack(ma_device* pDevice)
+{
+ ma_context* pContext = pDevice->pContext;
+ int resultJACK;
+ size_t i;
+
+ resultJACK = ((ma_jack_activate_proc)pContext->jack.jack_activate)((ma_jack_client_t*)pDevice->jack.pClient);
+ if (resultJACK != 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to activate the JACK client.");
+ return MA_FAILED_TO_START_BACKEND_DEVICE;
+ }
+
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ const char** ppServerPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ma_JackPortIsOutput);
+ if (ppServerPorts == NULL) {
+ ((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to retrieve physical ports.");
+ return MA_ERROR;
}
- static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_fetch_or_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8 src, c89atomic_memory_order order)
- {
- c89atomic_uint8 oldValue;
- c89atomic_uint8 newValue;
- do {
- oldValue = *dst;
- newValue = (c89atomic_uint8)(oldValue | src);
- } while (c89atomic_compare_and_swap_8(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ for (i = 0; ppServerPorts[i] != NULL; ++i) {
+ const char* pServerPort = ppServerPorts[i];
+ const char* pClientPort = ((ma_jack_port_name_proc)pContext->jack.jack_port_name)((ma_jack_port_t*)pDevice->jack.ppPortsCapture[i]);
+
+ resultJACK = ((ma_jack_connect_proc)pContext->jack.jack_connect)((ma_jack_client_t*)pDevice->jack.pClient, pServerPort, pClientPort);
+ if (resultJACK != 0) {
+ ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts);
+ ((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to connect ports.");
+ return MA_ERROR;
+ }
}
- static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_fetch_or_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16 src, c89atomic_memory_order order)
- {
- c89atomic_uint16 oldValue;
- c89atomic_uint16 newValue;
- do {
- oldValue = *dst;
- newValue = (c89atomic_uint16)(oldValue | src);
- } while (c89atomic_compare_and_swap_16(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts);
+ }
+
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ const char** ppServerPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ma_JackPortIsInput);
+ if (ppServerPorts == NULL) {
+ ((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to retrieve physical ports.");
+ return MA_ERROR;
}
- static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_fetch_or_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32 src, c89atomic_memory_order order)
- {
- c89atomic_uint32 oldValue;
- c89atomic_uint32 newValue;
- do {
- oldValue = *dst;
- newValue = oldValue | src;
- } while (c89atomic_compare_and_swap_32(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ for (i = 0; ppServerPorts[i] != NULL; ++i) {
+ const char* pServerPort = ppServerPorts[i];
+ const char* pClientPort = ((ma_jack_port_name_proc)pContext->jack.jack_port_name)((ma_jack_port_t*)pDevice->jack.ppPortsPlayback[i]);
+
+ resultJACK = ((ma_jack_connect_proc)pContext->jack.jack_connect)((ma_jack_client_t*)pDevice->jack.pClient, pClientPort, pServerPort);
+ if (resultJACK != 0) {
+ ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts);
+ ((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to connect ports.");
+ return MA_ERROR;
+ }
}
- static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_fetch_or_explicit_64(volatile c89atomic_uint64* dst, c89atomic_uint64 src, c89atomic_memory_order order)
- {
- c89atomic_uint64 oldValue;
- c89atomic_uint64 newValue;
- do {
- oldValue = *dst;
- newValue = oldValue | src;
- } while (c89atomic_compare_and_swap_64(dst, oldValue, newValue) != oldValue);
- (void)order;
- return oldValue;
+
+ ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts);
+ }
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_stop__jack(ma_device* pDevice)
+{
+ ma_context* pContext = pDevice->pContext;
+
+ if (((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient) != 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] An error occurred when deactivating the JACK client.");
+ return MA_ERROR;
+ }
+
+ ma_device__on_notification_stopped(pDevice);
+
+ return MA_SUCCESS;
+}
+
+
+static ma_result ma_context_uninit__jack(ma_context* pContext)
+{
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pContext->backend == ma_backend_jack);
+
+ ma_free(pContext->jack.pClientName, &pContext->allocationCallbacks);
+ pContext->jack.pClientName = NULL;
+
+#ifndef MA_NO_RUNTIME_LINKING
+ ma_dlclose(pContext, pContext->jack.jackSO);
+#endif
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_init__jack(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+{
+#ifndef MA_NO_RUNTIME_LINKING
+ const char* libjackNames[] = {
+#ifdef MA_WIN32
+ "libjack.dll",
+ "libjack64.dll"
+#else
+ "libjack.so",
+ "libjack.so.0"
+#endif
+ };
+ size_t i;
+
+ for (i = 0; i < ma_countof(libjackNames); ++i) {
+ pContext->jack.jackSO = ma_dlopen(pContext, libjackNames[i]);
+ if (pContext->jack.jackSO != NULL) {
+ break;
+ }
+ }
+
+ if (pContext->jack.jackSO == NULL) {
+ return MA_NO_BACKEND;
+ }
+
+ pContext->jack.jack_client_open = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_client_open");
+ pContext->jack.jack_client_close = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_client_close");
+ pContext->jack.jack_client_name_size = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_client_name_size");
+ pContext->jack.jack_set_process_callback = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_set_process_callback");
+ pContext->jack.jack_set_buffer_size_callback = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_set_buffer_size_callback");
+ pContext->jack.jack_on_shutdown = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_on_shutdown");
+ pContext->jack.jack_get_sample_rate = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_get_sample_rate");
+ pContext->jack.jack_get_buffer_size = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_get_buffer_size");
+ pContext->jack.jack_get_ports = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_get_ports");
+ pContext->jack.jack_activate = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_activate");
+ pContext->jack.jack_deactivate = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_deactivate");
+ pContext->jack.jack_connect = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_connect");
+ pContext->jack.jack_port_register = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_port_register");
+ pContext->jack.jack_port_name = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_port_name");
+ pContext->jack.jack_port_get_buffer = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_port_get_buffer");
+ pContext->jack.jack_free = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_free");
+#else
+ /*
+ This strange assignment system is here just to ensure type safety of miniaudio's function pointer
+ types. If anything differs slightly the compiler should throw a warning.
+ */
+ ma_jack_client_open_proc _jack_client_open = jack_client_open;
+ ma_jack_client_close_proc _jack_client_close = jack_client_close;
+ ma_jack_client_name_size_proc _jack_client_name_size = jack_client_name_size;
+ ma_jack_set_process_callback_proc _jack_set_process_callback = jack_set_process_callback;
+ ma_jack_set_buffer_size_callback_proc _jack_set_buffer_size_callback = jack_set_buffer_size_callback;
+ ma_jack_on_shutdown_proc _jack_on_shutdown = jack_on_shutdown;
+ ma_jack_get_sample_rate_proc _jack_get_sample_rate = jack_get_sample_rate;
+ ma_jack_get_buffer_size_proc _jack_get_buffer_size = jack_get_buffer_size;
+ ma_jack_get_ports_proc _jack_get_ports = jack_get_ports;
+ ma_jack_activate_proc _jack_activate = jack_activate;
+ ma_jack_deactivate_proc _jack_deactivate = jack_deactivate;
+ ma_jack_connect_proc _jack_connect = jack_connect;
+ ma_jack_port_register_proc _jack_port_register = jack_port_register;
+ ma_jack_port_name_proc _jack_port_name = jack_port_name;
+ ma_jack_port_get_buffer_proc _jack_port_get_buffer = jack_port_get_buffer;
+ ma_jack_free_proc _jack_free = jack_free;
+
+ pContext->jack.jack_client_open = (ma_proc)_jack_client_open;
+ pContext->jack.jack_client_close = (ma_proc)_jack_client_close;
+ pContext->jack.jack_client_name_size = (ma_proc)_jack_client_name_size;
+ pContext->jack.jack_set_process_callback = (ma_proc)_jack_set_process_callback;
+ pContext->jack.jack_set_buffer_size_callback = (ma_proc)_jack_set_buffer_size_callback;
+ pContext->jack.jack_on_shutdown = (ma_proc)_jack_on_shutdown;
+ pContext->jack.jack_get_sample_rate = (ma_proc)_jack_get_sample_rate;
+ pContext->jack.jack_get_buffer_size = (ma_proc)_jack_get_buffer_size;
+ pContext->jack.jack_get_ports = (ma_proc)_jack_get_ports;
+ pContext->jack.jack_activate = (ma_proc)_jack_activate;
+ pContext->jack.jack_deactivate = (ma_proc)_jack_deactivate;
+ pContext->jack.jack_connect = (ma_proc)_jack_connect;
+ pContext->jack.jack_port_register = (ma_proc)_jack_port_register;
+ pContext->jack.jack_port_name = (ma_proc)_jack_port_name;
+ pContext->jack.jack_port_get_buffer = (ma_proc)_jack_port_get_buffer;
+ pContext->jack.jack_free = (ma_proc)_jack_free;
+#endif
+
+ if (pConfig->jack.pClientName != NULL) {
+ pContext->jack.pClientName = ma_copy_string(pConfig->jack.pClientName, &pContext->allocationCallbacks);
+ }
+ pContext->jack.tryStartServer = pConfig->jack.tryStartServer;
+
+ /*
+ Getting here means the JACK library is installed, but it doesn't necessarily mean it's usable. We need to quickly test this by connecting
+ a temporary client.
+ */
+ {
+ ma_jack_client_t* pDummyClient;
+ ma_result result = ma_context_open_client__jack(pContext, &pDummyClient);
+ if (result != MA_SUCCESS) {
+ ma_free(pContext->jack.pClientName, &pContext->allocationCallbacks);
+ #ifndef MA_NO_RUNTIME_LINKING
+ ma_dlclose(pContext, pContext->jack.jackSO);
+ #endif
+ return MA_NO_BACKEND;
}
+
+ ((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pDummyClient);
+ }
+
+
+ pCallbacks->onContextInit = ma_context_init__jack;
+ pCallbacks->onContextUninit = ma_context_uninit__jack;
+ pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__jack;
+ pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__jack;
+ pCallbacks->onDeviceInit = ma_device_init__jack;
+ pCallbacks->onDeviceUninit = ma_device_uninit__jack;
+ pCallbacks->onDeviceStart = ma_device_start__jack;
+ pCallbacks->onDeviceStop = ma_device_stop__jack;
+ pCallbacks->onDeviceRead = NULL; /* Not used because JACK is asynchronous. */
+ pCallbacks->onDeviceWrite = NULL; /* Not used because JACK is asynchronous. */
+ pCallbacks->onDeviceDataLoop = NULL; /* Not used because JACK is asynchronous. */
+
+ return MA_SUCCESS;
+}
+#endif /* JACK */
+
+
+
+/******************************************************************************
+
+Core Audio Backend
+
+References
+==========
+- Technical Note TN2091: Device input using the HAL Output Audio Unit
+ https://developer.apple.com/library/archive/technotes/tn2091/_index.html
+
+******************************************************************************/
+#ifdef MA_HAS_COREAUDIO
+#include
+
+#if defined(TARGET_OS_IPHONE) && TARGET_OS_IPHONE == 1
+ #define MA_APPLE_MOBILE
+ #if defined(TARGET_OS_TV) && TARGET_OS_TV == 1
+ #define MA_APPLE_TV
#endif
- #define c89atomic_signal_fence(order) c89atomic_thread_fence(order)
- static C89ATOMIC_INLINE c89atomic_uint8 c89atomic_load_explicit_8(volatile const c89atomic_uint8* ptr, c89atomic_memory_order order)
+ #if defined(TARGET_OS_WATCH) && TARGET_OS_WATCH == 1
+ #define MA_APPLE_WATCH
+ #endif
+ #if __has_feature(objc_arc)
+ #define MA_BRIDGE_TRANSFER __bridge_transfer
+ #define MA_BRIDGE_RETAINED __bridge_retained
+ #else
+ #define MA_BRIDGE_TRANSFER
+ #define MA_BRIDGE_RETAINED
+ #endif
+#else
+ #define MA_APPLE_DESKTOP
+#endif
+
+#if defined(MA_APPLE_DESKTOP)
+#include
+#else
+#include
+#endif
+
+#include
+
+/* CoreFoundation */
+typedef Boolean (* ma_CFStringGetCString_proc)(CFStringRef theString, char* buffer, CFIndex bufferSize, CFStringEncoding encoding);
+typedef void (* ma_CFRelease_proc)(CFTypeRef cf);
+
+/* CoreAudio */
+#if defined(MA_APPLE_DESKTOP)
+typedef OSStatus (* ma_AudioObjectGetPropertyData_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, UInt32 inQualifierDataSize, const void* inQualifierData, UInt32* ioDataSize, void* outData);
+typedef OSStatus (* ma_AudioObjectGetPropertyDataSize_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, UInt32 inQualifierDataSize, const void* inQualifierData, UInt32* outDataSize);
+typedef OSStatus (* ma_AudioObjectSetPropertyData_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, UInt32 inQualifierDataSize, const void* inQualifierData, UInt32 inDataSize, const void* inData);
+typedef OSStatus (* ma_AudioObjectAddPropertyListener_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, AudioObjectPropertyListenerProc inListener, void* inClientData);
+typedef OSStatus (* ma_AudioObjectRemovePropertyListener_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, AudioObjectPropertyListenerProc inListener, void* inClientData);
+#endif
+
+/* AudioToolbox */
+typedef AudioComponent (* ma_AudioComponentFindNext_proc)(AudioComponent inComponent, const AudioComponentDescription* inDesc);
+typedef OSStatus (* ma_AudioComponentInstanceDispose_proc)(AudioComponentInstance inInstance);
+typedef OSStatus (* ma_AudioComponentInstanceNew_proc)(AudioComponent inComponent, AudioComponentInstance* outInstance);
+typedef OSStatus (* ma_AudioOutputUnitStart_proc)(AudioUnit inUnit);
+typedef OSStatus (* ma_AudioOutputUnitStop_proc)(AudioUnit inUnit);
+typedef OSStatus (* ma_AudioUnitAddPropertyListener_proc)(AudioUnit inUnit, AudioUnitPropertyID inID, AudioUnitPropertyListenerProc inProc, void* inProcUserData);
+typedef OSStatus (* ma_AudioUnitGetPropertyInfo_proc)(AudioUnit inUnit, AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, UInt32* outDataSize, Boolean* outWriteable);
+typedef OSStatus (* ma_AudioUnitGetProperty_proc)(AudioUnit inUnit, AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, void* outData, UInt32* ioDataSize);
+typedef OSStatus (* ma_AudioUnitSetProperty_proc)(AudioUnit inUnit, AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, const void* inData, UInt32 inDataSize);
+typedef OSStatus (* ma_AudioUnitInitialize_proc)(AudioUnit inUnit);
+typedef OSStatus (* ma_AudioUnitRender_proc)(AudioUnit inUnit, AudioUnitRenderActionFlags* ioActionFlags, const AudioTimeStamp* inTimeStamp, UInt32 inOutputBusNumber, UInt32 inNumberFrames, AudioBufferList* ioData);
+
+
+#define MA_COREAUDIO_OUTPUT_BUS 0
+#define MA_COREAUDIO_INPUT_BUS 1
+
+#if defined(MA_APPLE_DESKTOP)
+static ma_result ma_device_reinit_internal__coreaudio(ma_device* pDevice, ma_device_type deviceType, ma_bool32 disposePreviousAudioUnit);
+#endif
+
+/*
+Core Audio
+
+So far, Core Audio has been the worst backend to work with due to being both unintuitive and having almost no documentation
+apart from comments in the headers (which admittedly are quite good). For my own purposes, and for anybody out there whose
+needing to figure out how this darn thing works, I'm going to outline a few things here.
+
+Since miniaudio is a fairly low-level API, one of the things it needs is control over specific devices, and it needs to be
+able to identify whether or not it can be used as playback and/or capture. The AudioObject API is the only one I've seen
+that supports this level of detail. There was some public domain sample code I stumbled across that used the AudioComponent
+and AudioUnit APIs, but I couldn't see anything that gave low-level control over device selection and capabilities (the
+distinction between playback and capture in particular). Therefore, miniaudio is using the AudioObject API.
+
+Most (all?) functions in the AudioObject API take a AudioObjectID as it's input. This is the device identifier. When
+retrieving global information, such as the device list, you use kAudioObjectSystemObject. When retrieving device-specific
+data, you pass in the ID for that device. In order to retrieve device-specific IDs you need to enumerate over each of the
+devices. This is done using the AudioObjectGetPropertyDataSize() and AudioObjectGetPropertyData() APIs which seem to be
+the central APIs for retrieving information about the system and specific devices.
+
+To use the AudioObjectGetPropertyData() API you need to use the notion of a property address. A property address is a
+structure with three variables and is used to identify which property you are getting or setting. The first is the "selector"
+which is basically the specific property that you're wanting to retrieve or set. The second is the "scope", which is
+typically set to kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyScopeInput for input-specific properties and
+kAudioObjectPropertyScopeOutput for output-specific properties. The last is the "element" which is always set to
+kAudioObjectPropertyElementMaster in miniaudio's case. I don't know of any cases where this would be set to anything different.
+
+Back to the earlier issue of device retrieval, you first use the AudioObjectGetPropertyDataSize() API to retrieve the size
+of the raw data which is just a list of AudioDeviceID's. You use the kAudioObjectSystemObject AudioObjectID, and a property
+address with the kAudioHardwarePropertyDevices selector and the kAudioObjectPropertyScopeGlobal scope. Once you have the
+size, allocate a block of memory of that size and then call AudioObjectGetPropertyData(). The data is just a list of
+AudioDeviceID's so just do "dataSize/sizeof(AudioDeviceID)" to know the device count.
+*/
+
+static ma_result ma_result_from_OSStatus(OSStatus status)
+{
+ switch (status)
{
- (void)order;
- return c89atomic_compare_and_swap_8((c89atomic_uint8*)ptr, 0, 0);
+ case noErr: return MA_SUCCESS;
+ #if defined(MA_APPLE_DESKTOP)
+ case kAudioHardwareNotRunningError: return MA_DEVICE_NOT_STARTED;
+ case kAudioHardwareUnspecifiedError: return MA_ERROR;
+ case kAudioHardwareUnknownPropertyError: return MA_INVALID_ARGS;
+ case kAudioHardwareBadPropertySizeError: return MA_INVALID_OPERATION;
+ case kAudioHardwareIllegalOperationError: return MA_INVALID_OPERATION;
+ case kAudioHardwareBadObjectError: return MA_INVALID_ARGS;
+ case kAudioHardwareBadDeviceError: return MA_INVALID_ARGS;
+ case kAudioHardwareBadStreamError: return MA_INVALID_ARGS;
+ case kAudioHardwareUnsupportedOperationError: return MA_INVALID_OPERATION;
+ case kAudioDeviceUnsupportedFormatError: return MA_FORMAT_NOT_SUPPORTED;
+ case kAudioDevicePermissionsError: return MA_ACCESS_DENIED;
+ #endif
+ default: return MA_ERROR;
}
- static C89ATOMIC_INLINE c89atomic_uint16 c89atomic_load_explicit_16(volatile const c89atomic_uint16* ptr, c89atomic_memory_order order)
+}
+
+#if 0
+static ma_channel ma_channel_from_AudioChannelBitmap(AudioChannelBitmap bit)
+{
+ switch (bit)
{
- (void)order;
- return c89atomic_compare_and_swap_16((c89atomic_uint16*)ptr, 0, 0);
+ case kAudioChannelBit_Left: return MA_CHANNEL_LEFT;
+ case kAudioChannelBit_Right: return MA_CHANNEL_RIGHT;
+ case kAudioChannelBit_Center: return MA_CHANNEL_FRONT_CENTER;
+ case kAudioChannelBit_LFEScreen: return MA_CHANNEL_LFE;
+ case kAudioChannelBit_LeftSurround: return MA_CHANNEL_BACK_LEFT;
+ case kAudioChannelBit_RightSurround: return MA_CHANNEL_BACK_RIGHT;
+ case kAudioChannelBit_LeftCenter: return MA_CHANNEL_FRONT_LEFT_CENTER;
+ case kAudioChannelBit_RightCenter: return MA_CHANNEL_FRONT_RIGHT_CENTER;
+ case kAudioChannelBit_CenterSurround: return MA_CHANNEL_BACK_CENTER;
+ case kAudioChannelBit_LeftSurroundDirect: return MA_CHANNEL_SIDE_LEFT;
+ case kAudioChannelBit_RightSurroundDirect: return MA_CHANNEL_SIDE_RIGHT;
+ case kAudioChannelBit_TopCenterSurround: return MA_CHANNEL_TOP_CENTER;
+ case kAudioChannelBit_VerticalHeightLeft: return MA_CHANNEL_TOP_FRONT_LEFT;
+ case kAudioChannelBit_VerticalHeightCenter: return MA_CHANNEL_TOP_FRONT_CENTER;
+ case kAudioChannelBit_VerticalHeightRight: return MA_CHANNEL_TOP_FRONT_RIGHT;
+ case kAudioChannelBit_TopBackLeft: return MA_CHANNEL_TOP_BACK_LEFT;
+ case kAudioChannelBit_TopBackCenter: return MA_CHANNEL_TOP_BACK_CENTER;
+ case kAudioChannelBit_TopBackRight: return MA_CHANNEL_TOP_BACK_RIGHT;
+ default: return MA_CHANNEL_NONE;
}
- static C89ATOMIC_INLINE c89atomic_uint32 c89atomic_load_explicit_32(volatile const c89atomic_uint32* ptr, c89atomic_memory_order order)
- {
- (void)order;
- return c89atomic_compare_and_swap_32((c89atomic_uint32*)ptr, 0, 0);
+}
+#endif
+
+static ma_result ma_format_from_AudioStreamBasicDescription(const AudioStreamBasicDescription* pDescription, ma_format* pFormatOut)
+{
+ MA_ASSERT(pDescription != NULL);
+ MA_ASSERT(pFormatOut != NULL);
+
+ *pFormatOut = ma_format_unknown; /* Safety. */
+
+ /* There's a few things miniaudio doesn't support. */
+ if (pDescription->mFormatID != kAudioFormatLinearPCM) {
+ return MA_FORMAT_NOT_SUPPORTED;
}
- static C89ATOMIC_INLINE c89atomic_uint64 c89atomic_load_explicit_64(volatile const c89atomic_uint64* ptr, c89atomic_memory_order order)
- {
- (void)order;
- return c89atomic_compare_and_swap_64((c89atomic_uint64*)ptr, 0, 0);
+
+ /* We don't support any non-packed formats that are aligned high. */
+ if ((pDescription->mFormatFlags & kLinearPCMFormatFlagIsAlignedHigh) != 0) {
+ return MA_FORMAT_NOT_SUPPORTED;
}
- #define c89atomic_store_explicit_8( dst, src, order) (void)c89atomic_exchange_explicit_8 (dst, src, order)
- #define c89atomic_store_explicit_16(dst, src, order) (void)c89atomic_exchange_explicit_16(dst, src, order)
- #define c89atomic_store_explicit_32(dst, src, order) (void)c89atomic_exchange_explicit_32(dst, src, order)
- #define c89atomic_store_explicit_64(dst, src, order) (void)c89atomic_exchange_explicit_64(dst, src, order)
- #define c89atomic_test_and_set_explicit_8( dst, order) c89atomic_exchange_explicit_8 (dst, 1, order)
- #define c89atomic_test_and_set_explicit_16(dst, order) c89atomic_exchange_explicit_16(dst, 1, order)
- #define c89atomic_test_and_set_explicit_32(dst, order) c89atomic_exchange_explicit_32(dst, 1, order)
- #define c89atomic_test_and_set_explicit_64(dst, order) c89atomic_exchange_explicit_64(dst, 1, order)
- #define c89atomic_clear_explicit_8( dst, order) c89atomic_store_explicit_8 (dst, 0, order)
- #define c89atomic_clear_explicit_16(dst, order) c89atomic_store_explicit_16(dst, 0, order)
- #define c89atomic_clear_explicit_32(dst, order) c89atomic_store_explicit_32(dst, 0, order)
- #define c89atomic_clear_explicit_64(dst, order) c89atomic_store_explicit_64(dst, 0, order)
- typedef c89atomic_uint8 c89atomic_flag;
- #define c89atomic_flag_test_and_set_explicit(ptr, order) (c89atomic_bool)c89atomic_test_and_set_explicit_8(ptr, order)
- #define c89atomic_flag_clear_explicit(ptr, order) c89atomic_clear_explicit_8(ptr, order)
- #define c89atoimc_flag_load_explicit(ptr, order) c89atomic_load_explicit_8(ptr, order)
-#endif
-#if !defined(C89ATOMIC_HAS_NATIVE_COMPARE_EXCHANGE)
- #if defined(C89ATOMIC_HAS_8)
- c89atomic_bool c89atomic_compare_exchange_strong_explicit_8(volatile c89atomic_uint8* dst, c89atomic_uint8* expected, c89atomic_uint8 desired, c89atomic_memory_order successOrder, c89atomic_memory_order failureOrder)
- {
- c89atomic_uint8 expectedValue;
- c89atomic_uint8 result;
- (void)successOrder;
- (void)failureOrder;
- expectedValue = c89atomic_load_explicit_8(expected, c89atomic_memory_order_seq_cst);
- result = c89atomic_compare_and_swap_8(dst, expectedValue, desired);
- if (result == expectedValue) {
- return 1;
- } else {
- c89atomic_store_explicit_8(expected, result, failureOrder);
- return 0;
- }
- }
- #endif
- #if defined(C89ATOMIC_HAS_16)
- c89atomic_bool c89atomic_compare_exchange_strong_explicit_16(volatile c89atomic_uint16* dst, c89atomic_uint16* expected, c89atomic_uint16 desired, c89atomic_memory_order successOrder, c89atomic_memory_order failureOrder)
- {
- c89atomic_uint16 expectedValue;
- c89atomic_uint16 result;
- (void)successOrder;
- (void)failureOrder;
- expectedValue = c89atomic_load_explicit_16(expected, c89atomic_memory_order_seq_cst);
- result = c89atomic_compare_and_swap_16(dst, expectedValue, desired);
- if (result == expectedValue) {
- return 1;
- } else {
- c89atomic_store_explicit_16(expected, result, failureOrder);
- return 0;
- }
+
+ /* Only supporting native-endian. */
+ if ((ma_is_little_endian() && (pDescription->mFormatFlags & kAudioFormatFlagIsBigEndian) != 0) || (ma_is_big_endian() && (pDescription->mFormatFlags & kAudioFormatFlagIsBigEndian) == 0)) {
+ return MA_FORMAT_NOT_SUPPORTED;
+ }
+
+ /* We are not currently supporting non-interleaved formats (this will be added in a future version of miniaudio). */
+ /*if ((pDescription->mFormatFlags & kAudioFormatFlagIsNonInterleaved) != 0) {
+ return MA_FORMAT_NOT_SUPPORTED;
+ }*/
+
+ if ((pDescription->mFormatFlags & kLinearPCMFormatFlagIsFloat) != 0) {
+ if (pDescription->mBitsPerChannel == 32) {
+ *pFormatOut = ma_format_f32;
+ return MA_SUCCESS;
}
- #endif
- #if defined(C89ATOMIC_HAS_32)
- c89atomic_bool c89atomic_compare_exchange_strong_explicit_32(volatile c89atomic_uint32* dst, c89atomic_uint32* expected, c89atomic_uint32 desired, c89atomic_memory_order successOrder, c89atomic_memory_order failureOrder)
- {
- c89atomic_uint32 expectedValue;
- c89atomic_uint32 result;
- (void)successOrder;
- (void)failureOrder;
- expectedValue = c89atomic_load_explicit_32(expected, c89atomic_memory_order_seq_cst);
- result = c89atomic_compare_and_swap_32(dst, expectedValue, desired);
- if (result == expectedValue) {
- return 1;
- } else {
- c89atomic_store_explicit_32(expected, result, failureOrder);
- return 0;
+ } else {
+ if ((pDescription->mFormatFlags & kLinearPCMFormatFlagIsSignedInteger) != 0) {
+ if (pDescription->mBitsPerChannel == 16) {
+ *pFormatOut = ma_format_s16;
+ return MA_SUCCESS;
+ } else if (pDescription->mBitsPerChannel == 24) {
+ if (pDescription->mBytesPerFrame == (pDescription->mBitsPerChannel/8 * pDescription->mChannelsPerFrame)) {
+ *pFormatOut = ma_format_s24;
+ return MA_SUCCESS;
+ } else {
+ if (pDescription->mBytesPerFrame/pDescription->mChannelsPerFrame == sizeof(ma_int32)) {
+ /* TODO: Implement ma_format_s24_32. */
+ /**pFormatOut = ma_format_s24_32;*/
+ /*return MA_SUCCESS;*/
+ return MA_FORMAT_NOT_SUPPORTED;
+ }
+ }
+ } else if (pDescription->mBitsPerChannel == 32) {
+ *pFormatOut = ma_format_s32;
+ return MA_SUCCESS;
}
- }
- #endif
- #if defined(C89ATOMIC_HAS_64)
- c89atomic_bool c89atomic_compare_exchange_strong_explicit_64(volatile c89atomic_uint64* dst, volatile c89atomic_uint64* expected, c89atomic_uint64 desired, c89atomic_memory_order successOrder, c89atomic_memory_order failureOrder)
- {
- c89atomic_uint64 expectedValue;
- c89atomic_uint64 result;
- (void)successOrder;
- (void)failureOrder;
- expectedValue = c89atomic_load_explicit_64(expected, c89atomic_memory_order_seq_cst);
- result = c89atomic_compare_and_swap_64(dst, expectedValue, desired);
- if (result == expectedValue) {
- return 1;
- } else {
- c89atomic_store_explicit_64(expected, result, failureOrder);
- return 0;
+ } else {
+ if (pDescription->mBitsPerChannel == 8) {
+ *pFormatOut = ma_format_u8;
+ return MA_SUCCESS;
}
}
- #endif
- #define c89atomic_compare_exchange_weak_explicit_8( dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_strong_explicit_8 (dst, expected, desired, successOrder, failureOrder)
- #define c89atomic_compare_exchange_weak_explicit_16(dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_strong_explicit_16(dst, expected, desired, successOrder, failureOrder)
- #define c89atomic_compare_exchange_weak_explicit_32(dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_strong_explicit_32(dst, expected, desired, successOrder, failureOrder)
- #define c89atomic_compare_exchange_weak_explicit_64(dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_strong_explicit_64(dst, expected, desired, successOrder, failureOrder)
-#endif
-#if !defined(C89ATOMIC_HAS_NATIVE_IS_LOCK_FREE)
- static C89ATOMIC_INLINE c89atomic_bool c89atomic_is_lock_free_8(volatile void* ptr)
- {
- (void)ptr;
- return 1;
- }
- static C89ATOMIC_INLINE c89atomic_bool c89atomic_is_lock_free_16(volatile void* ptr)
- {
- (void)ptr;
- return 1;
- }
- static C89ATOMIC_INLINE c89atomic_bool c89atomic_is_lock_free_32(volatile void* ptr)
- {
- (void)ptr;
- return 1;
}
- static C89ATOMIC_INLINE c89atomic_bool c89atomic_is_lock_free_64(volatile void* ptr)
+
+ /* Getting here means the format is not supported. */
+ return MA_FORMAT_NOT_SUPPORTED;
+}
+
+#if defined(MA_APPLE_DESKTOP)
+static ma_channel ma_channel_from_AudioChannelLabel(AudioChannelLabel label)
+{
+ switch (label)
{
- (void)ptr;
- #if defined(C89ATOMIC_64BIT)
- return 1;
- #else
- #if defined(C89ATOMIC_X86) || defined(C89ATOMIC_X64)
- return 1;
- #else
- return 0;
- #endif
+ case kAudioChannelLabel_Unknown: return MA_CHANNEL_NONE;
+ case kAudioChannelLabel_Unused: return MA_CHANNEL_NONE;
+ case kAudioChannelLabel_UseCoordinates: return MA_CHANNEL_NONE;
+ case kAudioChannelLabel_Left: return MA_CHANNEL_LEFT;
+ case kAudioChannelLabel_Right: return MA_CHANNEL_RIGHT;
+ case kAudioChannelLabel_Center: return MA_CHANNEL_FRONT_CENTER;
+ case kAudioChannelLabel_LFEScreen: return MA_CHANNEL_LFE;
+ case kAudioChannelLabel_LeftSurround: return MA_CHANNEL_BACK_LEFT;
+ case kAudioChannelLabel_RightSurround: return MA_CHANNEL_BACK_RIGHT;
+ case kAudioChannelLabel_LeftCenter: return MA_CHANNEL_FRONT_LEFT_CENTER;
+ case kAudioChannelLabel_RightCenter: return MA_CHANNEL_FRONT_RIGHT_CENTER;
+ case kAudioChannelLabel_CenterSurround: return MA_CHANNEL_BACK_CENTER;
+ case kAudioChannelLabel_LeftSurroundDirect: return MA_CHANNEL_SIDE_LEFT;
+ case kAudioChannelLabel_RightSurroundDirect: return MA_CHANNEL_SIDE_RIGHT;
+ case kAudioChannelLabel_TopCenterSurround: return MA_CHANNEL_TOP_CENTER;
+ case kAudioChannelLabel_VerticalHeightLeft: return MA_CHANNEL_TOP_FRONT_LEFT;
+ case kAudioChannelLabel_VerticalHeightCenter: return MA_CHANNEL_TOP_FRONT_CENTER;
+ case kAudioChannelLabel_VerticalHeightRight: return MA_CHANNEL_TOP_FRONT_RIGHT;
+ case kAudioChannelLabel_TopBackLeft: return MA_CHANNEL_TOP_BACK_LEFT;
+ case kAudioChannelLabel_TopBackCenter: return MA_CHANNEL_TOP_BACK_CENTER;
+ case kAudioChannelLabel_TopBackRight: return MA_CHANNEL_TOP_BACK_RIGHT;
+ case kAudioChannelLabel_RearSurroundLeft: return MA_CHANNEL_BACK_LEFT;
+ case kAudioChannelLabel_RearSurroundRight: return MA_CHANNEL_BACK_RIGHT;
+ case kAudioChannelLabel_LeftWide: return MA_CHANNEL_SIDE_LEFT;
+ case kAudioChannelLabel_RightWide: return MA_CHANNEL_SIDE_RIGHT;
+ case kAudioChannelLabel_LFE2: return MA_CHANNEL_LFE;
+ case kAudioChannelLabel_LeftTotal: return MA_CHANNEL_LEFT;
+ case kAudioChannelLabel_RightTotal: return MA_CHANNEL_RIGHT;
+ case kAudioChannelLabel_HearingImpaired: return MA_CHANNEL_NONE;
+ case kAudioChannelLabel_Narration: return MA_CHANNEL_MONO;
+ case kAudioChannelLabel_Mono: return MA_CHANNEL_MONO;
+ case kAudioChannelLabel_DialogCentricMix: return MA_CHANNEL_MONO;
+ case kAudioChannelLabel_CenterSurroundDirect: return MA_CHANNEL_BACK_CENTER;
+ case kAudioChannelLabel_Haptic: return MA_CHANNEL_NONE;
+ case kAudioChannelLabel_Ambisonic_W: return MA_CHANNEL_NONE;
+ case kAudioChannelLabel_Ambisonic_X: return MA_CHANNEL_NONE;
+ case kAudioChannelLabel_Ambisonic_Y: return MA_CHANNEL_NONE;
+ case kAudioChannelLabel_Ambisonic_Z: return MA_CHANNEL_NONE;
+ case kAudioChannelLabel_MS_Mid: return MA_CHANNEL_LEFT;
+ case kAudioChannelLabel_MS_Side: return MA_CHANNEL_RIGHT;
+ case kAudioChannelLabel_XY_X: return MA_CHANNEL_LEFT;
+ case kAudioChannelLabel_XY_Y: return MA_CHANNEL_RIGHT;
+ case kAudioChannelLabel_HeadphonesLeft: return MA_CHANNEL_LEFT;
+ case kAudioChannelLabel_HeadphonesRight: return MA_CHANNEL_RIGHT;
+ case kAudioChannelLabel_ClickTrack: return MA_CHANNEL_NONE;
+ case kAudioChannelLabel_ForeignLanguage: return MA_CHANNEL_NONE;
+ case kAudioChannelLabel_Discrete: return MA_CHANNEL_NONE;
+ case kAudioChannelLabel_Discrete_0: return MA_CHANNEL_AUX_0;
+ case kAudioChannelLabel_Discrete_1: return MA_CHANNEL_AUX_1;
+ case kAudioChannelLabel_Discrete_2: return MA_CHANNEL_AUX_2;
+ case kAudioChannelLabel_Discrete_3: return MA_CHANNEL_AUX_3;
+ case kAudioChannelLabel_Discrete_4: return MA_CHANNEL_AUX_4;
+ case kAudioChannelLabel_Discrete_5: return MA_CHANNEL_AUX_5;
+ case kAudioChannelLabel_Discrete_6: return MA_CHANNEL_AUX_6;
+ case kAudioChannelLabel_Discrete_7: return MA_CHANNEL_AUX_7;
+ case kAudioChannelLabel_Discrete_8: return MA_CHANNEL_AUX_8;
+ case kAudioChannelLabel_Discrete_9: return MA_CHANNEL_AUX_9;
+ case kAudioChannelLabel_Discrete_10: return MA_CHANNEL_AUX_10;
+ case kAudioChannelLabel_Discrete_11: return MA_CHANNEL_AUX_11;
+ case kAudioChannelLabel_Discrete_12: return MA_CHANNEL_AUX_12;
+ case kAudioChannelLabel_Discrete_13: return MA_CHANNEL_AUX_13;
+ case kAudioChannelLabel_Discrete_14: return MA_CHANNEL_AUX_14;
+ case kAudioChannelLabel_Discrete_15: return MA_CHANNEL_AUX_15;
+ case kAudioChannelLabel_Discrete_65535: return MA_CHANNEL_NONE;
+
+ #if 0 /* Introduced in a later version of macOS. */
+ case kAudioChannelLabel_HOA_ACN: return MA_CHANNEL_NONE;
+ case kAudioChannelLabel_HOA_ACN_0: return MA_CHANNEL_AUX_0;
+ case kAudioChannelLabel_HOA_ACN_1: return MA_CHANNEL_AUX_1;
+ case kAudioChannelLabel_HOA_ACN_2: return MA_CHANNEL_AUX_2;
+ case kAudioChannelLabel_HOA_ACN_3: return MA_CHANNEL_AUX_3;
+ case kAudioChannelLabel_HOA_ACN_4: return MA_CHANNEL_AUX_4;
+ case kAudioChannelLabel_HOA_ACN_5: return MA_CHANNEL_AUX_5;
+ case kAudioChannelLabel_HOA_ACN_6: return MA_CHANNEL_AUX_6;
+ case kAudioChannelLabel_HOA_ACN_7: return MA_CHANNEL_AUX_7;
+ case kAudioChannelLabel_HOA_ACN_8: return MA_CHANNEL_AUX_8;
+ case kAudioChannelLabel_HOA_ACN_9: return MA_CHANNEL_AUX_9;
+ case kAudioChannelLabel_HOA_ACN_10: return MA_CHANNEL_AUX_10;
+ case kAudioChannelLabel_HOA_ACN_11: return MA_CHANNEL_AUX_11;
+ case kAudioChannelLabel_HOA_ACN_12: return MA_CHANNEL_AUX_12;
+ case kAudioChannelLabel_HOA_ACN_13: return MA_CHANNEL_AUX_13;
+ case kAudioChannelLabel_HOA_ACN_14: return MA_CHANNEL_AUX_14;
+ case kAudioChannelLabel_HOA_ACN_15: return MA_CHANNEL_AUX_15;
+ case kAudioChannelLabel_HOA_ACN_65024: return MA_CHANNEL_NONE;
#endif
+
+ default: return MA_CHANNEL_NONE;
}
+}
+
+static ma_result ma_get_channel_map_from_AudioChannelLayout(AudioChannelLayout* pChannelLayout, ma_channel* pChannelMap, size_t channelMapCap)
+{
+ MA_ASSERT(pChannelLayout != NULL);
+
+ if (pChannelLayout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelDescriptions) {
+ UInt32 iChannel;
+ for (iChannel = 0; iChannel < pChannelLayout->mNumberChannelDescriptions && iChannel < channelMapCap; ++iChannel) {
+ pChannelMap[iChannel] = ma_channel_from_AudioChannelLabel(pChannelLayout->mChannelDescriptions[iChannel].mChannelLabel);
+ }
+ } else
+#if 0
+ if (pChannelLayout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelBitmap) {
+ /* This is the same kind of system that's used by Windows audio APIs. */
+ UInt32 iChannel = 0;
+ UInt32 iBit;
+ AudioChannelBitmap bitmap = pChannelLayout->mChannelBitmap;
+ for (iBit = 0; iBit < 32 && iChannel < channelMapCap; ++iBit) {
+ AudioChannelBitmap bit = bitmap & (1 << iBit);
+ if (bit != 0) {
+ pChannelMap[iChannel++] = ma_channel_from_AudioChannelBit(bit);
+ }
+ }
+ } else
#endif
-#if defined(C89ATOMIC_64BIT)
- static C89ATOMIC_INLINE c89atomic_bool c89atomic_is_lock_free_ptr(volatile void** ptr)
{
- return c89atomic_is_lock_free_64((volatile c89atomic_uint64*)ptr);
- }
- static C89ATOMIC_INLINE void* c89atomic_load_explicit_ptr(volatile void** ptr, c89atomic_memory_order order)
- {
- return (void*)c89atomic_load_explicit_64((volatile c89atomic_uint64*)ptr, order);
+ /*
+ Need to use the tag to determine the channel map. For now I'm just assuming a default channel map, but later on this should
+ be updated to determine the mapping based on the tag.
+ */
+ UInt32 channelCount;
+
+ /* Our channel map retrieval APIs below take 32-bit integers, so we'll want to clamp the channel map capacity. */
+ if (channelMapCap > 0xFFFFFFFF) {
+ channelMapCap = 0xFFFFFFFF;
+ }
+
+ channelCount = ma_min(AudioChannelLayoutTag_GetNumberOfChannels(pChannelLayout->mChannelLayoutTag), (UInt32)channelMapCap);
+
+ switch (pChannelLayout->mChannelLayoutTag)
+ {
+ case kAudioChannelLayoutTag_Mono:
+ case kAudioChannelLayoutTag_Stereo:
+ case kAudioChannelLayoutTag_StereoHeadphones:
+ case kAudioChannelLayoutTag_MatrixStereo:
+ case kAudioChannelLayoutTag_MidSide:
+ case kAudioChannelLayoutTag_XY:
+ case kAudioChannelLayoutTag_Binaural:
+ case kAudioChannelLayoutTag_Ambisonic_B_Format:
+ {
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, channelCount);
+ } break;
+
+ case kAudioChannelLayoutTag_Octagonal:
+ {
+ pChannelMap[7] = MA_CHANNEL_SIDE_RIGHT;
+ pChannelMap[6] = MA_CHANNEL_SIDE_LEFT;
+ } /* Intentional fallthrough. */
+ case kAudioChannelLayoutTag_Hexagonal:
+ {
+ pChannelMap[5] = MA_CHANNEL_BACK_CENTER;
+ } /* Intentional fallthrough. */
+ case kAudioChannelLayoutTag_Pentagonal:
+ {
+ pChannelMap[4] = MA_CHANNEL_FRONT_CENTER;
+ } /* Intentional fallghrough. */
+ case kAudioChannelLayoutTag_Quadraphonic:
+ {
+ pChannelMap[3] = MA_CHANNEL_BACK_RIGHT;
+ pChannelMap[2] = MA_CHANNEL_BACK_LEFT;
+ pChannelMap[1] = MA_CHANNEL_RIGHT;
+ pChannelMap[0] = MA_CHANNEL_LEFT;
+ } break;
+
+ /* TODO: Add support for more tags here. */
+
+ default:
+ {
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, channelCount);
+ } break;
+ }
}
- static C89ATOMIC_INLINE void c89atomic_store_explicit_ptr(volatile void** dst, void* src, c89atomic_memory_order order)
- {
- c89atomic_store_explicit_64((volatile c89atomic_uint64*)dst, (c89atomic_uint64)src, order);
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_get_device_object_ids__coreaudio(ma_context* pContext, UInt32* pDeviceCount, AudioObjectID** ppDeviceObjectIDs) /* NOTE: Free the returned buffer with ma_free(). */
+{
+ AudioObjectPropertyAddress propAddressDevices;
+ UInt32 deviceObjectsDataSize;
+ OSStatus status;
+ AudioObjectID* pDeviceObjectIDs;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pDeviceCount != NULL);
+ MA_ASSERT(ppDeviceObjectIDs != NULL);
+
+ /* Safety. */
+ *pDeviceCount = 0;
+ *ppDeviceObjectIDs = NULL;
+
+ propAddressDevices.mSelector = kAudioHardwarePropertyDevices;
+ propAddressDevices.mScope = kAudioObjectPropertyScopeGlobal;
+ propAddressDevices.mElement = kAudioObjectPropertyElementMaster;
+
+ status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(kAudioObjectSystemObject, &propAddressDevices, 0, NULL, &deviceObjectsDataSize);
+ if (status != noErr) {
+ return ma_result_from_OSStatus(status);
}
- static C89ATOMIC_INLINE void* c89atomic_exchange_explicit_ptr(volatile void** dst, void* src, c89atomic_memory_order order)
- {
- return (void*)c89atomic_exchange_explicit_64((volatile c89atomic_uint64*)dst, (c89atomic_uint64)src, order);
+
+ pDeviceObjectIDs = (AudioObjectID*)ma_malloc(deviceObjectsDataSize, &pContext->allocationCallbacks);
+ if (pDeviceObjectIDs == NULL) {
+ return MA_OUT_OF_MEMORY;
}
- static C89ATOMIC_INLINE c89atomic_bool c89atomic_compare_exchange_strong_explicit_ptr(volatile void** dst, volatile void** expected, void* desired, c89atomic_memory_order successOrder, c89atomic_memory_order failureOrder)
- {
- return c89atomic_compare_exchange_strong_explicit_64((volatile c89atomic_uint64*)dst, (c89atomic_uint64*)expected, (c89atomic_uint64)desired, successOrder, failureOrder);
+
+ status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(kAudioObjectSystemObject, &propAddressDevices, 0, NULL, &deviceObjectsDataSize, pDeviceObjectIDs);
+ if (status != noErr) {
+ ma_free(pDeviceObjectIDs, &pContext->allocationCallbacks);
+ return ma_result_from_OSStatus(status);
}
- static C89ATOMIC_INLINE c89atomic_bool c89atomic_compare_exchange_weak_explicit_ptr(volatile void** dst, volatile void** expected, void* desired, c89atomic_memory_order successOrder, c89atomic_memory_order failureOrder)
- {
- return c89atomic_compare_exchange_weak_explicit_64((volatile c89atomic_uint64*)dst, (c89atomic_uint64*)expected, (c89atomic_uint64)desired, successOrder, failureOrder);
+
+ *pDeviceCount = deviceObjectsDataSize / sizeof(AudioObjectID);
+ *ppDeviceObjectIDs = pDeviceObjectIDs;
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_get_AudioObject_uid_as_CFStringRef(ma_context* pContext, AudioObjectID objectID, CFStringRef* pUID)
+{
+ AudioObjectPropertyAddress propAddress;
+ UInt32 dataSize;
+ OSStatus status;
+
+ MA_ASSERT(pContext != NULL);
+
+ propAddress.mSelector = kAudioDevicePropertyDeviceUID;
+ propAddress.mScope = kAudioObjectPropertyScopeGlobal;
+ propAddress.mElement = kAudioObjectPropertyElementMaster;
+
+ dataSize = sizeof(*pUID);
+ status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(objectID, &propAddress, 0, NULL, &dataSize, pUID);
+ if (status != noErr) {
+ return ma_result_from_OSStatus(status);
}
- static C89ATOMIC_INLINE void* c89atomic_compare_and_swap_ptr(volatile void** dst, void* expected, void* desired)
- {
- return (void*)c89atomic_compare_and_swap_64((volatile c89atomic_uint64*)dst, (c89atomic_uint64)expected, (c89atomic_uint64)desired);
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_get_AudioObject_uid(ma_context* pContext, AudioObjectID objectID, size_t bufferSize, char* bufferOut)
+{
+ CFStringRef uid;
+ ma_result result;
+
+ MA_ASSERT(pContext != NULL);
+
+ result = ma_get_AudioObject_uid_as_CFStringRef(pContext, objectID, &uid);
+ if (result != MA_SUCCESS) {
+ return result;
}
-#elif defined(C89ATOMIC_32BIT)
- static C89ATOMIC_INLINE c89atomic_bool c89atomic_is_lock_free_ptr(volatile void** ptr)
- {
- return c89atomic_is_lock_free_32((volatile c89atomic_uint32*)ptr);
+
+ if (!((ma_CFStringGetCString_proc)pContext->coreaudio.CFStringGetCString)(uid, bufferOut, bufferSize, kCFStringEncodingUTF8)) {
+ return MA_ERROR;
}
- static C89ATOMIC_INLINE void* c89atomic_load_explicit_ptr(volatile void** ptr, c89atomic_memory_order order)
- {
- return (void*)c89atomic_load_explicit_32((volatile c89atomic_uint32*)ptr, order);
+
+ ((ma_CFRelease_proc)pContext->coreaudio.CFRelease)(uid);
+ return MA_SUCCESS;
+}
+
+static ma_result ma_get_AudioObject_name(ma_context* pContext, AudioObjectID objectID, size_t bufferSize, char* bufferOut)
+{
+ AudioObjectPropertyAddress propAddress;
+ CFStringRef deviceName = NULL;
+ UInt32 dataSize;
+ OSStatus status;
+
+ MA_ASSERT(pContext != NULL);
+
+ propAddress.mSelector = kAudioDevicePropertyDeviceNameCFString;
+ propAddress.mScope = kAudioObjectPropertyScopeGlobal;
+ propAddress.mElement = kAudioObjectPropertyElementMaster;
+
+ dataSize = sizeof(deviceName);
+ status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(objectID, &propAddress, 0, NULL, &dataSize, &deviceName);
+ if (status != noErr) {
+ return ma_result_from_OSStatus(status);
}
- static C89ATOMIC_INLINE void c89atomic_store_explicit_ptr(volatile void** dst, void* src, c89atomic_memory_order order)
- {
- c89atomic_store_explicit_32((volatile c89atomic_uint32*)dst, (c89atomic_uint32)src, order);
+
+ if (!((ma_CFStringGetCString_proc)pContext->coreaudio.CFStringGetCString)(deviceName, bufferOut, bufferSize, kCFStringEncodingUTF8)) {
+ return MA_ERROR;
}
- static C89ATOMIC_INLINE void* c89atomic_exchange_explicit_ptr(volatile void** dst, void* src, c89atomic_memory_order order)
- {
- return (void*)c89atomic_exchange_explicit_32((volatile c89atomic_uint32*)dst, (c89atomic_uint32)src, order);
+
+ ((ma_CFRelease_proc)pContext->coreaudio.CFRelease)(deviceName);
+ return MA_SUCCESS;
+}
+
+static ma_bool32 ma_does_AudioObject_support_scope(ma_context* pContext, AudioObjectID deviceObjectID, AudioObjectPropertyScope scope)
+{
+ AudioObjectPropertyAddress propAddress;
+ UInt32 dataSize;
+ OSStatus status;
+ AudioBufferList* pBufferList;
+ ma_bool32 isSupported;
+
+ MA_ASSERT(pContext != NULL);
+
+ /* To know whether or not a device is an input device we need ot look at the stream configuration. If it has an output channel it's a playback device. */
+ propAddress.mSelector = kAudioDevicePropertyStreamConfiguration;
+ propAddress.mScope = scope;
+ propAddress.mElement = kAudioObjectPropertyElementMaster;
+
+ status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(deviceObjectID, &propAddress, 0, NULL, &dataSize);
+ if (status != noErr) {
+ return MA_FALSE;
}
- static C89ATOMIC_INLINE c89atomic_bool c89atomic_compare_exchange_strong_explicit_ptr(volatile void** dst, void** expected, void* desired, c89atomic_memory_order successOrder, c89atomic_memory_order failureOrder)
- {
- return c89atomic_compare_exchange_strong_explicit_32((volatile c89atomic_uint32*)dst, (c89atomic_uint32*)expected, (c89atomic_uint32)desired, successOrder, failureOrder);
+
+ pBufferList = (AudioBufferList*)ma_malloc(dataSize, &pContext->allocationCallbacks);
+ if (pBufferList == NULL) {
+ return MA_FALSE; /* Out of memory. */
}
- static C89ATOMIC_INLINE c89atomic_bool c89atomic_compare_exchange_weak_explicit_ptr(volatile void** dst, volatile void** expected, void* desired, c89atomic_memory_order successOrder, c89atomic_memory_order failureOrder)
- {
- return c89atomic_compare_exchange_weak_explicit_32((volatile c89atomic_uint32*)dst, (c89atomic_uint32*)expected, (c89atomic_uint32)desired, successOrder, failureOrder);
+
+ status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pBufferList);
+ if (status != noErr) {
+ ma_free(pBufferList, &pContext->allocationCallbacks);
+ return MA_FALSE;
}
- static C89ATOMIC_INLINE void* c89atomic_compare_and_swap_ptr(volatile void** dst, void* expected, void* desired)
- {
- return (void*)c89atomic_compare_and_swap_32((volatile c89atomic_uint32*)dst, (c89atomic_uint32)expected, (c89atomic_uint32)desired);
+
+ isSupported = MA_FALSE;
+ if (pBufferList->mNumberBuffers > 0) {
+ isSupported = MA_TRUE;
}
-#else
- #error Unsupported architecture.
-#endif
-#define c89atomic_flag_test_and_set(ptr) c89atomic_flag_test_and_set_explicit(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_flag_clear(ptr) c89atomic_flag_clear_explicit(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_store_ptr(dst, src) c89atomic_store_explicit_ptr((volatile void**)dst, (void*)src, c89atomic_memory_order_seq_cst)
-#define c89atomic_load_ptr(ptr) c89atomic_load_explicit_ptr((volatile void**)ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_exchange_ptr(dst, src) c89atomic_exchange_explicit_ptr((volatile void**)dst, (void*)src, c89atomic_memory_order_seq_cst)
-#define c89atomic_compare_exchange_strong_ptr(dst, expected, desired) c89atomic_compare_exchange_strong_explicit_ptr((volatile void**)dst, (volatile void**)expected, (void*)desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
-#define c89atomic_compare_exchange_weak_ptr(dst, expected, desired) c89atomic_compare_exchange_weak_explicit_ptr((volatile void**)dst, (volatile void**)expected, (void*)desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
-#define c89atomic_test_and_set_8( ptr) c89atomic_test_and_set_explicit_8( ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_test_and_set_16(ptr) c89atomic_test_and_set_explicit_16(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_test_and_set_32(ptr) c89atomic_test_and_set_explicit_32(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_test_and_set_64(ptr) c89atomic_test_and_set_explicit_64(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_clear_8( ptr) c89atomic_clear_explicit_8( ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_clear_16(ptr) c89atomic_clear_explicit_16(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_clear_32(ptr) c89atomic_clear_explicit_32(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_clear_64(ptr) c89atomic_clear_explicit_64(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_store_8( dst, src) c89atomic_store_explicit_8( dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_store_16(dst, src) c89atomic_store_explicit_16(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_store_32(dst, src) c89atomic_store_explicit_32(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_store_64(dst, src) c89atomic_store_explicit_64(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_load_8( ptr) c89atomic_load_explicit_8( ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_load_16(ptr) c89atomic_load_explicit_16(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_load_32(ptr) c89atomic_load_explicit_32(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_load_64(ptr) c89atomic_load_explicit_64(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_exchange_8( dst, src) c89atomic_exchange_explicit_8( dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_exchange_16(dst, src) c89atomic_exchange_explicit_16(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_exchange_32(dst, src) c89atomic_exchange_explicit_32(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_exchange_64(dst, src) c89atomic_exchange_explicit_64(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_compare_exchange_strong_8( dst, expected, desired) c89atomic_compare_exchange_strong_explicit_8( dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
-#define c89atomic_compare_exchange_strong_16(dst, expected, desired) c89atomic_compare_exchange_strong_explicit_16(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
-#define c89atomic_compare_exchange_strong_32(dst, expected, desired) c89atomic_compare_exchange_strong_explicit_32(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
-#define c89atomic_compare_exchange_strong_64(dst, expected, desired) c89atomic_compare_exchange_strong_explicit_64(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
-#define c89atomic_compare_exchange_weak_8( dst, expected, desired) c89atomic_compare_exchange_weak_explicit_8( dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
-#define c89atomic_compare_exchange_weak_16( dst, expected, desired) c89atomic_compare_exchange_weak_explicit_16(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
-#define c89atomic_compare_exchange_weak_32( dst, expected, desired) c89atomic_compare_exchange_weak_explicit_32(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
-#define c89atomic_compare_exchange_weak_64( dst, expected, desired) c89atomic_compare_exchange_weak_explicit_64(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_add_8( dst, src) c89atomic_fetch_add_explicit_8( dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_add_16(dst, src) c89atomic_fetch_add_explicit_16(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_add_32(dst, src) c89atomic_fetch_add_explicit_32(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_add_64(dst, src) c89atomic_fetch_add_explicit_64(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_sub_8( dst, src) c89atomic_fetch_sub_explicit_8( dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_sub_16(dst, src) c89atomic_fetch_sub_explicit_16(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_sub_32(dst, src) c89atomic_fetch_sub_explicit_32(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_sub_64(dst, src) c89atomic_fetch_sub_explicit_64(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_or_8( dst, src) c89atomic_fetch_or_explicit_8( dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_or_16(dst, src) c89atomic_fetch_or_explicit_16(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_or_32(dst, src) c89atomic_fetch_or_explicit_32(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_or_64(dst, src) c89atomic_fetch_or_explicit_64(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_xor_8( dst, src) c89atomic_fetch_xor_explicit_8( dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_xor_16(dst, src) c89atomic_fetch_xor_explicit_16(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_xor_32(dst, src) c89atomic_fetch_xor_explicit_32(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_xor_64(dst, src) c89atomic_fetch_xor_explicit_64(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_and_8( dst, src) c89atomic_fetch_and_explicit_8 (dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_and_16(dst, src) c89atomic_fetch_and_explicit_16(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_and_32(dst, src) c89atomic_fetch_and_explicit_32(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_and_64(dst, src) c89atomic_fetch_and_explicit_64(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_test_and_set_explicit_i8( ptr, order) (c89atomic_int8 )c89atomic_test_and_set_explicit_8( (c89atomic_uint8* )ptr, order)
-#define c89atomic_test_and_set_explicit_i16(ptr, order) (c89atomic_int16)c89atomic_test_and_set_explicit_16((c89atomic_uint16*)ptr, order)
-#define c89atomic_test_and_set_explicit_i32(ptr, order) (c89atomic_int32)c89atomic_test_and_set_explicit_32((c89atomic_uint32*)ptr, order)
-#define c89atomic_test_and_set_explicit_i64(ptr, order) (c89atomic_int64)c89atomic_test_and_set_explicit_64((c89atomic_uint64*)ptr, order)
-#define c89atomic_clear_explicit_i8( ptr, order) c89atomic_clear_explicit_8( (c89atomic_uint8* )ptr, order)
-#define c89atomic_clear_explicit_i16(ptr, order) c89atomic_clear_explicit_16((c89atomic_uint16*)ptr, order)
-#define c89atomic_clear_explicit_i32(ptr, order) c89atomic_clear_explicit_32((c89atomic_uint32*)ptr, order)
-#define c89atomic_clear_explicit_i64(ptr, order) c89atomic_clear_explicit_64((c89atomic_uint64*)ptr, order)
-#define c89atomic_store_explicit_i8( dst, src, order) (c89atomic_int8 )c89atomic_store_explicit_8( (c89atomic_uint8* )dst, (c89atomic_uint8 )src, order)
-#define c89atomic_store_explicit_i16(dst, src, order) (c89atomic_int16)c89atomic_store_explicit_16((c89atomic_uint16*)dst, (c89atomic_uint16)src, order)
-#define c89atomic_store_explicit_i32(dst, src, order) (c89atomic_int32)c89atomic_store_explicit_32((c89atomic_uint32*)dst, (c89atomic_uint32)src, order)
-#define c89atomic_store_explicit_i64(dst, src, order) (c89atomic_int64)c89atomic_store_explicit_64((c89atomic_uint64*)dst, (c89atomic_uint64)src, order)
-#define c89atomic_load_explicit_i8( ptr, order) (c89atomic_int8 )c89atomic_load_explicit_8( (c89atomic_uint8* )ptr, order)
-#define c89atomic_load_explicit_i16(ptr, order) (c89atomic_int16)c89atomic_load_explicit_16((c89atomic_uint16*)ptr, order)
-#define c89atomic_load_explicit_i32(ptr, order) (c89atomic_int32)c89atomic_load_explicit_32((c89atomic_uint32*)ptr, order)
-#define c89atomic_load_explicit_i64(ptr, order) (c89atomic_int64)c89atomic_load_explicit_64((c89atomic_uint64*)ptr, order)
-#define c89atomic_exchange_explicit_i8( dst, src, order) (c89atomic_int8 )c89atomic_exchange_explicit_8 ((c89atomic_uint8* )dst, (c89atomic_uint8 )src, order)
-#define c89atomic_exchange_explicit_i16(dst, src, order) (c89atomic_int16)c89atomic_exchange_explicit_16((c89atomic_uint16*)dst, (c89atomic_uint16)src, order)
-#define c89atomic_exchange_explicit_i32(dst, src, order) (c89atomic_int32)c89atomic_exchange_explicit_32((c89atomic_uint32*)dst, (c89atomic_uint32)src, order)
-#define c89atomic_exchange_explicit_i64(dst, src, order) (c89atomic_int64)c89atomic_exchange_explicit_64((c89atomic_uint64*)dst, (c89atomic_uint64)src, order)
-#define c89atomic_compare_exchange_strong_explicit_i8( dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_strong_explicit_8( (c89atomic_uint8* )dst, (c89atomic_uint8* )expected, (c89atomic_uint8 )desired, successOrder, failureOrder)
-#define c89atomic_compare_exchange_strong_explicit_i16(dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_strong_explicit_16((c89atomic_uint16*)dst, (c89atomic_uint16*)expected, (c89atomic_uint16)desired, successOrder, failureOrder)
-#define c89atomic_compare_exchange_strong_explicit_i32(dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_strong_explicit_32((c89atomic_uint32*)dst, (c89atomic_uint32*)expected, (c89atomic_uint32)desired, successOrder, failureOrder)
-#define c89atomic_compare_exchange_strong_explicit_i64(dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_strong_explicit_64((c89atomic_uint64*)dst, (c89atomic_uint64*)expected, (c89atomic_uint64)desired, successOrder, failureOrder)
-#define c89atomic_compare_exchange_weak_explicit_i8( dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_weak_explicit_8( (c89atomic_uint8* )dst, (c89atomic_uint8* )expected, (c89atomic_uint8 )desired, successOrder, failureOrder)
-#define c89atomic_compare_exchange_weak_explicit_i16(dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_weak_explicit_16((c89atomic_uint16*)dst, (c89atomic_uint16*)expected, (c89atomic_uint16)desired, successOrder, failureOrder)
-#define c89atomic_compare_exchange_weak_explicit_i32(dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_weak_explicit_32((c89atomic_uint32*)dst, (c89atomic_uint32*)expected, (c89atomic_uint32)desired, successOrder, failureOrder)
-#define c89atomic_compare_exchange_weak_explicit_i64(dst, expected, desired, successOrder, failureOrder) c89atomic_compare_exchange_weak_explicit_64((c89atomic_uint64*)dst, (c89atomic_uint64*)expected, (c89atomic_uint64)desired, successOrder, failureOrder)
-#define c89atomic_fetch_add_explicit_i8( dst, src, order) (c89atomic_int8 )c89atomic_fetch_add_explicit_8( (c89atomic_uint8* )dst, (c89atomic_uint8 )src, order)
-#define c89atomic_fetch_add_explicit_i16(dst, src, order) (c89atomic_int16)c89atomic_fetch_add_explicit_16((c89atomic_uint16*)dst, (c89atomic_uint16)src, order)
-#define c89atomic_fetch_add_explicit_i32(dst, src, order) (c89atomic_int32)c89atomic_fetch_add_explicit_32((c89atomic_uint32*)dst, (c89atomic_uint32)src, order)
-#define c89atomic_fetch_add_explicit_i64(dst, src, order) (c89atomic_int64)c89atomic_fetch_add_explicit_64((c89atomic_uint64*)dst, (c89atomic_uint64)src, order)
-#define c89atomic_fetch_sub_explicit_i8( dst, src, order) (c89atomic_int8 )c89atomic_fetch_sub_explicit_8( (c89atomic_uint8* )dst, (c89atomic_uint8 )src, order)
-#define c89atomic_fetch_sub_explicit_i16(dst, src, order) (c89atomic_int16)c89atomic_fetch_sub_explicit_16((c89atomic_uint16*)dst, (c89atomic_uint16)src, order)
-#define c89atomic_fetch_sub_explicit_i32(dst, src, order) (c89atomic_int32)c89atomic_fetch_sub_explicit_32((c89atomic_uint32*)dst, (c89atomic_uint32)src, order)
-#define c89atomic_fetch_sub_explicit_i64(dst, src, order) (c89atomic_int64)c89atomic_fetch_sub_explicit_64((c89atomic_uint64*)dst, (c89atomic_uint64)src, order)
-#define c89atomic_fetch_or_explicit_i8( dst, src, order) (c89atomic_int8 )c89atomic_fetch_or_explicit_8( (c89atomic_uint8* )dst, (c89atomic_uint8 )src, order)
-#define c89atomic_fetch_or_explicit_i16(dst, src, order) (c89atomic_int16)c89atomic_fetch_or_explicit_16((c89atomic_uint16*)dst, (c89atomic_uint16)src, order)
-#define c89atomic_fetch_or_explicit_i32(dst, src, order) (c89atomic_int32)c89atomic_fetch_or_explicit_32((c89atomic_uint32*)dst, (c89atomic_uint32)src, order)
-#define c89atomic_fetch_or_explicit_i64(dst, src, order) (c89atomic_int64)c89atomic_fetch_or_explicit_64((c89atomic_uint64*)dst, (c89atomic_uint64)src, order)
-#define c89atomic_fetch_xor_explicit_i8( dst, src, order) (c89atomic_int8 )c89atomic_fetch_xor_explicit_8( (c89atomic_uint8* )dst, (c89atomic_uint8 )src, order)
-#define c89atomic_fetch_xor_explicit_i16(dst, src, order) (c89atomic_int16)c89atomic_fetch_xor_explicit_16((c89atomic_uint16*)dst, (c89atomic_uint16)src, order)
-#define c89atomic_fetch_xor_explicit_i32(dst, src, order) (c89atomic_int32)c89atomic_fetch_xor_explicit_32((c89atomic_uint32*)dst, (c89atomic_uint32)src, order)
-#define c89atomic_fetch_xor_explicit_i64(dst, src, order) (c89atomic_int64)c89atomic_fetch_xor_explicit_64((c89atomic_uint64*)dst, (c89atomic_uint64)src, order)
-#define c89atomic_fetch_and_explicit_i8( dst, src, order) (c89atomic_int8 )c89atomic_fetch_and_explicit_8( (c89atomic_uint8* )dst, (c89atomic_uint8 )src, order)
-#define c89atomic_fetch_and_explicit_i16(dst, src, order) (c89atomic_int16)c89atomic_fetch_and_explicit_16((c89atomic_uint16*)dst, (c89atomic_uint16)src, order)
-#define c89atomic_fetch_and_explicit_i32(dst, src, order) (c89atomic_int32)c89atomic_fetch_and_explicit_32((c89atomic_uint32*)dst, (c89atomic_uint32)src, order)
-#define c89atomic_fetch_and_explicit_i64(dst, src, order) (c89atomic_int64)c89atomic_fetch_and_explicit_64((c89atomic_uint64*)dst, (c89atomic_uint64)src, order)
-#define c89atomic_test_and_set_i8( ptr) c89atomic_test_and_set_explicit_i8( ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_test_and_set_i16(ptr) c89atomic_test_and_set_explicit_i16(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_test_and_set_i32(ptr) c89atomic_test_and_set_explicit_i32(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_test_and_set_i64(ptr) c89atomic_test_and_set_explicit_i64(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_clear_i8( ptr) c89atomic_clear_explicit_i8( ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_clear_i16(ptr) c89atomic_clear_explicit_i16(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_clear_i32(ptr) c89atomic_clear_explicit_i32(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_clear_i64(ptr) c89atomic_clear_explicit_i64(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_store_i8( dst, src) c89atomic_store_explicit_i8( dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_store_i16(dst, src) c89atomic_store_explicit_i16(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_store_i32(dst, src) c89atomic_store_explicit_i32(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_store_i64(dst, src) c89atomic_store_explicit_i64(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_load_i8( ptr) c89atomic_load_explicit_i8( ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_load_i16(ptr) c89atomic_load_explicit_i16(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_load_i32(ptr) c89atomic_load_explicit_i32(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_load_i64(ptr) c89atomic_load_explicit_i64(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_exchange_i8( dst, src) c89atomic_exchange_explicit_i8( dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_exchange_i16(dst, src) c89atomic_exchange_explicit_i16(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_exchange_i32(dst, src) c89atomic_exchange_explicit_i32(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_exchange_i64(dst, src) c89atomic_exchange_explicit_i64(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_compare_exchange_strong_i8( dst, expected, desired) c89atomic_compare_exchange_strong_explicit_i8( dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
-#define c89atomic_compare_exchange_strong_i16(dst, expected, desired) c89atomic_compare_exchange_strong_explicit_i16(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
-#define c89atomic_compare_exchange_strong_i32(dst, expected, desired) c89atomic_compare_exchange_strong_explicit_i32(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
-#define c89atomic_compare_exchange_strong_i64(dst, expected, desired) c89atomic_compare_exchange_strong_explicit_i64(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
-#define c89atomic_compare_exchange_weak_i8( dst, expected, desired) c89atomic_compare_exchange_weak_explicit_i8( dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
-#define c89atomic_compare_exchange_weak_i16(dst, expected, desired) c89atomic_compare_exchange_weak_explicit_i16(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
-#define c89atomic_compare_exchange_weak_i32(dst, expected, desired) c89atomic_compare_exchange_weak_explicit_i32(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
-#define c89atomic_compare_exchange_weak_i64(dst, expected, desired) c89atomic_compare_exchange_weak_explicit_i64(dst, expected, desired, c89atomic_memory_order_seq_cst, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_add_i8( dst, src) c89atomic_fetch_add_explicit_i8( dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_add_i16(dst, src) c89atomic_fetch_add_explicit_i16(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_add_i32(dst, src) c89atomic_fetch_add_explicit_i32(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_add_i64(dst, src) c89atomic_fetch_add_explicit_i64(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_sub_i8( dst, src) c89atomic_fetch_sub_explicit_i8( dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_sub_i16(dst, src) c89atomic_fetch_sub_explicit_i16(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_sub_i32(dst, src) c89atomic_fetch_sub_explicit_i32(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_sub_i64(dst, src) c89atomic_fetch_sub_explicit_i64(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_or_i8( dst, src) c89atomic_fetch_or_explicit_i8( dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_or_i16(dst, src) c89atomic_fetch_or_explicit_i16(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_or_i32(dst, src) c89atomic_fetch_or_explicit_i32(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_or_i64(dst, src) c89atomic_fetch_or_explicit_i64(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_xor_i8( dst, src) c89atomic_fetch_xor_explicit_i8( dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_xor_i16(dst, src) c89atomic_fetch_xor_explicit_i16(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_xor_i32(dst, src) c89atomic_fetch_xor_explicit_i32(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_xor_i64(dst, src) c89atomic_fetch_xor_explicit_i64(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_and_i8( dst, src) c89atomic_fetch_and_explicit_i8( dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_and_i16(dst, src) c89atomic_fetch_and_explicit_i16(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_and_i32(dst, src) c89atomic_fetch_and_explicit_i32(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_fetch_and_i64(dst, src) c89atomic_fetch_and_explicit_i64(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_compare_and_swap_i8( dst, expected, dedsired) (c89atomic_int8 )c89atomic_compare_and_swap_8( (c89atomic_uint8* )dst, (c89atomic_uint8 )expected, (c89atomic_uint8 )dedsired)
-#define c89atomic_compare_and_swap_i16(dst, expected, dedsired) (c89atomic_int16)c89atomic_compare_and_swap_16((c89atomic_uint16*)dst, (c89atomic_uint16)expected, (c89atomic_uint16)dedsired)
-#define c89atomic_compare_and_swap_i32(dst, expected, dedsired) (c89atomic_int32)c89atomic_compare_and_swap_32((c89atomic_uint32*)dst, (c89atomic_uint32)expected, (c89atomic_uint32)dedsired)
-#define c89atomic_compare_and_swap_i64(dst, expected, dedsired) (c89atomic_int64)c89atomic_compare_and_swap_64((c89atomic_uint64*)dst, (c89atomic_uint64)expected, (c89atomic_uint64)dedsired)
-typedef union
-{
- c89atomic_uint32 i;
- float f;
-} c89atomic_if32;
-typedef union
-{
- c89atomic_uint64 i;
- double f;
-} c89atomic_if64;
-#define c89atomic_clear_explicit_f32(ptr, order) c89atomic_clear_explicit_32((c89atomic_uint32*)ptr, order)
-#define c89atomic_clear_explicit_f64(ptr, order) c89atomic_clear_explicit_64((c89atomic_uint64*)ptr, order)
-static C89ATOMIC_INLINE void c89atomic_store_explicit_f32(volatile float* dst, float src, c89atomic_memory_order order)
-{
- c89atomic_if32 x;
- x.f = src;
- c89atomic_store_explicit_32((volatile c89atomic_uint32*)dst, x.i, order);
-}
-static C89ATOMIC_INLINE void c89atomic_store_explicit_f64(volatile double* dst, double src, c89atomic_memory_order order)
-{
- c89atomic_if64 x;
- x.f = src;
- c89atomic_store_explicit_64((volatile c89atomic_uint64*)dst, x.i, order);
-}
-static C89ATOMIC_INLINE float c89atomic_load_explicit_f32(volatile float* ptr, c89atomic_memory_order order)
-{
- c89atomic_if32 r;
- r.i = c89atomic_load_explicit_32((volatile c89atomic_uint32*)ptr, order);
- return r.f;
+
+ ma_free(pBufferList, &pContext->allocationCallbacks);
+ return isSupported;
}
-static C89ATOMIC_INLINE double c89atomic_load_explicit_f64(volatile double* ptr, c89atomic_memory_order order)
+
+static ma_bool32 ma_does_AudioObject_support_playback(ma_context* pContext, AudioObjectID deviceObjectID)
{
- c89atomic_if64 r;
- r.i = c89atomic_load_explicit_64((volatile c89atomic_uint64*)ptr, order);
- return r.f;
+ return ma_does_AudioObject_support_scope(pContext, deviceObjectID, kAudioObjectPropertyScopeOutput);
}
-static C89ATOMIC_INLINE float c89atomic_exchange_explicit_f32(volatile float* dst, float src, c89atomic_memory_order order)
+
+static ma_bool32 ma_does_AudioObject_support_capture(ma_context* pContext, AudioObjectID deviceObjectID)
{
- c89atomic_if32 r;
- c89atomic_if32 x;
- x.f = src;
- r.i = c89atomic_exchange_explicit_32((volatile c89atomic_uint32*)dst, x.i, order);
- return r.f;
+ return ma_does_AudioObject_support_scope(pContext, deviceObjectID, kAudioObjectPropertyScopeInput);
}
-static C89ATOMIC_INLINE double c89atomic_exchange_explicit_f64(volatile double* dst, double src, c89atomic_memory_order order)
+
+
+static ma_result ma_get_AudioObject_stream_descriptions(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, UInt32* pDescriptionCount, AudioStreamRangedDescription** ppDescriptions) /* NOTE: Free the returned pointer with ma_free(). */
{
- c89atomic_if64 r;
- c89atomic_if64 x;
- x.f = src;
- r.i = c89atomic_exchange_explicit_64((volatile c89atomic_uint64*)dst, x.i, order);
- return r.f;
+ AudioObjectPropertyAddress propAddress;
+ UInt32 dataSize;
+ OSStatus status;
+ AudioStreamRangedDescription* pDescriptions;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pDescriptionCount != NULL);
+ MA_ASSERT(ppDescriptions != NULL);
+
+ /*
+ TODO: Experiment with kAudioStreamPropertyAvailablePhysicalFormats instead of (or in addition to) kAudioStreamPropertyAvailableVirtualFormats. My
+ MacBook Pro uses s24/32 format, however, which miniaudio does not currently support.
+ */
+ propAddress.mSelector = kAudioStreamPropertyAvailableVirtualFormats; /*kAudioStreamPropertyAvailablePhysicalFormats;*/
+ propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput;
+ propAddress.mElement = kAudioObjectPropertyElementMaster;
+
+ status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(deviceObjectID, &propAddress, 0, NULL, &dataSize);
+ if (status != noErr) {
+ return ma_result_from_OSStatus(status);
+ }
+
+ pDescriptions = (AudioStreamRangedDescription*)ma_malloc(dataSize, &pContext->allocationCallbacks);
+ if (pDescriptions == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+
+ status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pDescriptions);
+ if (status != noErr) {
+ ma_free(pDescriptions, &pContext->allocationCallbacks);
+ return ma_result_from_OSStatus(status);
+ }
+
+ *pDescriptionCount = dataSize / sizeof(*pDescriptions);
+ *ppDescriptions = pDescriptions;
+ return MA_SUCCESS;
}
-#define c89atomic_clear_f32(ptr) (float )c89atomic_clear_explicit_f32(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_clear_f64(ptr) (double)c89atomic_clear_explicit_f64(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_store_f32(dst, src) c89atomic_store_explicit_f32(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_store_f64(dst, src) c89atomic_store_explicit_f64(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_load_f32(ptr) (float )c89atomic_load_explicit_f32(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_load_f64(ptr) (double)c89atomic_load_explicit_f64(ptr, c89atomic_memory_order_seq_cst)
-#define c89atomic_exchange_f32(dst, src) (float )c89atomic_exchange_explicit_f32(dst, src, c89atomic_memory_order_seq_cst)
-#define c89atomic_exchange_f64(dst, src) (double)c89atomic_exchange_explicit_f64(dst, src, c89atomic_memory_order_seq_cst)
-typedef c89atomic_flag c89atomic_spinlock;
-static C89ATOMIC_INLINE void c89atomic_spinlock_lock(volatile c89atomic_spinlock* pSpinlock)
+
+
+static ma_result ma_get_AudioObject_channel_layout(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, AudioChannelLayout** ppChannelLayout) /* NOTE: Free the returned pointer with ma_free(). */
{
- for (;;) {
- if (c89atomic_flag_test_and_set_explicit(pSpinlock, c89atomic_memory_order_acquire) == 0) {
- break;
- }
- while (c89atoimc_flag_load_explicit(pSpinlock, c89atomic_memory_order_relaxed) == 1) {
- }
+ AudioObjectPropertyAddress propAddress;
+ UInt32 dataSize;
+ OSStatus status;
+ AudioChannelLayout* pChannelLayout;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(ppChannelLayout != NULL);
+
+ *ppChannelLayout = NULL; /* Safety. */
+
+ propAddress.mSelector = kAudioDevicePropertyPreferredChannelLayout;
+ propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput;
+ propAddress.mElement = kAudioObjectPropertyElementMaster;
+
+ status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(deviceObjectID, &propAddress, 0, NULL, &dataSize);
+ if (status != noErr) {
+ return ma_result_from_OSStatus(status);
+ }
+
+ pChannelLayout = (AudioChannelLayout*)ma_malloc(dataSize, &pContext->allocationCallbacks);
+ if (pChannelLayout == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+
+ status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pChannelLayout);
+ if (status != noErr) {
+ ma_free(pChannelLayout, &pContext->allocationCallbacks);
+ return ma_result_from_OSStatus(status);
}
+
+ *ppChannelLayout = pChannelLayout;
+ return MA_SUCCESS;
}
-static C89ATOMIC_INLINE void c89atomic_spinlock_unlock(volatile c89atomic_spinlock* pSpinlock)
+
+static ma_result ma_get_AudioObject_channel_count(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32* pChannelCount)
{
- c89atomic_flag_clear_explicit(pSpinlock, c89atomic_memory_order_release);
-}
-#if defined(__cplusplus)
-}
-#endif
-#endif
-/* c89atomic.h end */
+ AudioChannelLayout* pChannelLayout;
+ ma_result result;
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pChannelCount != NULL);
+ *pChannelCount = 0; /* Safety. */
-MA_API ma_uint64 ma_calculate_frame_count_after_resampling(ma_uint32 sampleRateOut, ma_uint32 sampleRateIn, ma_uint64 frameCountIn)
+ result = ma_get_AudioObject_channel_layout(pContext, deviceObjectID, deviceType, &pChannelLayout);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ if (pChannelLayout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelDescriptions) {
+ *pChannelCount = pChannelLayout->mNumberChannelDescriptions;
+ } else if (pChannelLayout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelBitmap) {
+ *pChannelCount = ma_count_set_bits(pChannelLayout->mChannelBitmap);
+ } else {
+ *pChannelCount = AudioChannelLayoutTag_GetNumberOfChannels(pChannelLayout->mChannelLayoutTag);
+ }
+
+ ma_free(pChannelLayout, &pContext->allocationCallbacks);
+ return MA_SUCCESS;
+}
+
+#if 0
+static ma_result ma_get_AudioObject_channel_map(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_channel* pChannelMap, size_t channelMapCap)
{
- /* For robustness we're going to use a resampler object to calculate this since that already has a way of calculating this. */
+ AudioChannelLayout* pChannelLayout;
ma_result result;
- ma_uint64 frameCountOut;
- ma_resampler_config config;
- ma_resampler resampler;
- if (sampleRateOut == sampleRateIn) {
- return frameCountIn;
+ MA_ASSERT(pContext != NULL);
+
+ result = ma_get_AudioObject_channel_layout(pContext, deviceObjectID, deviceType, &pChannelLayout);
+ if (result != MA_SUCCESS) {
+ return result; /* Rather than always failing here, would it be more robust to simply assume a default? */
}
- config = ma_resampler_config_init(ma_format_s16, 1, sampleRateIn, sampleRateOut, ma_resample_algorithm_linear);
- result = ma_resampler_init(&config, &resampler);
+ result = ma_get_channel_map_from_AudioChannelLayout(pChannelLayout, pChannelMap, channelMapCap);
if (result != MA_SUCCESS) {
- return 0;
+ ma_free(pChannelLayout, &pContext->allocationCallbacks);
+ return result;
}
- frameCountOut = ma_resampler_get_expected_output_frame_count(&resampler, frameCountIn);
+ ma_free(pChannelLayout, &pContext->allocationCallbacks);
+ return result;
+}
+#endif
+
+static ma_result ma_get_AudioObject_sample_rates(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, UInt32* pSampleRateRangesCount, AudioValueRange** ppSampleRateRanges) /* NOTE: Free the returned pointer with ma_free(). */
+{
+ AudioObjectPropertyAddress propAddress;
+ UInt32 dataSize;
+ OSStatus status;
+ AudioValueRange* pSampleRateRanges;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pSampleRateRangesCount != NULL);
+ MA_ASSERT(ppSampleRateRanges != NULL);
- ma_resampler_uninit(&resampler);
- return frameCountOut;
-}
+ /* Safety. */
+ *pSampleRateRangesCount = 0;
+ *ppSampleRateRanges = NULL;
-#ifndef MA_DATA_CONVERTER_STACK_BUFFER_SIZE
-#define MA_DATA_CONVERTER_STACK_BUFFER_SIZE 4096
-#endif
+ propAddress.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
+ propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput;
+ propAddress.mElement = kAudioObjectPropertyElementMaster;
+ status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(deviceObjectID, &propAddress, 0, NULL, &dataSize);
+ if (status != noErr) {
+ return ma_result_from_OSStatus(status);
+ }
+ pSampleRateRanges = (AudioValueRange*)ma_malloc(dataSize, &pContext->allocationCallbacks);
+ if (pSampleRateRanges == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
-#if defined(MA_WIN32)
-static ma_result ma_result_from_GetLastError(DWORD error)
-{
- switch (error)
- {
- case ERROR_SUCCESS: return MA_SUCCESS;
- case ERROR_PATH_NOT_FOUND: return MA_DOES_NOT_EXIST;
- case ERROR_TOO_MANY_OPEN_FILES: return MA_TOO_MANY_OPEN_FILES;
- case ERROR_NOT_ENOUGH_MEMORY: return MA_OUT_OF_MEMORY;
- case ERROR_DISK_FULL: return MA_NO_SPACE;
- case ERROR_HANDLE_EOF: return MA_AT_END;
- case ERROR_NEGATIVE_SEEK: return MA_BAD_SEEK;
- case ERROR_INVALID_PARAMETER: return MA_INVALID_ARGS;
- case ERROR_ACCESS_DENIED: return MA_ACCESS_DENIED;
- case ERROR_SEM_TIMEOUT: return MA_TIMEOUT;
- case ERROR_FILE_NOT_FOUND: return MA_DOES_NOT_EXIST;
- default: break;
+ status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pSampleRateRanges);
+ if (status != noErr) {
+ ma_free(pSampleRateRanges, &pContext->allocationCallbacks);
+ return ma_result_from_OSStatus(status);
}
- return MA_ERROR;
+ *pSampleRateRangesCount = dataSize / sizeof(*pSampleRateRanges);
+ *ppSampleRateRanges = pSampleRateRanges;
+ return MA_SUCCESS;
}
-#endif /* MA_WIN32 */
+#if 0
+static ma_result ma_get_AudioObject_get_closest_sample_rate(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32 sampleRateIn, ma_uint32* pSampleRateOut)
+{
+ UInt32 sampleRateRangeCount;
+ AudioValueRange* pSampleRateRanges;
+ ma_result result;
-/*******************************************************************************
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pSampleRateOut != NULL);
-Threading
+ *pSampleRateOut = 0; /* Safety. */
-*******************************************************************************/
-#ifndef MA_NO_THREADING
-#ifdef MA_WIN32
- #define MA_THREADCALL WINAPI
- typedef unsigned long ma_thread_result;
-#else
- #define MA_THREADCALL
- typedef void* ma_thread_result;
-#endif
-typedef ma_thread_result (MA_THREADCALL * ma_thread_entry_proc)(void* pData);
+ result = ma_get_AudioObject_sample_rates(pContext, deviceObjectID, deviceType, &sampleRateRangeCount, &pSampleRateRanges);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
-static MA_INLINE ma_result ma_spinlock_lock_ex(volatile ma_spinlock* pSpinlock, ma_bool32 yield)
-{
- if (pSpinlock == NULL) {
- return MA_INVALID_ARGS;
+ if (sampleRateRangeCount == 0) {
+ ma_free(pSampleRateRanges, &pContext->allocationCallbacks);
+ return MA_ERROR; /* Should never hit this case should we? */
}
- for (;;) {
- if (c89atomic_exchange_explicit_32(pSpinlock, 1, c89atomic_memory_order_acquire) == 0) {
- break;
+ if (sampleRateIn == 0) {
+ /* Search in order of miniaudio's preferred priority. */
+ UInt32 iMALSampleRate;
+ for (iMALSampleRate = 0; iMALSampleRate < ma_countof(g_maStandardSampleRatePriorities); ++iMALSampleRate) {
+ ma_uint32 malSampleRate = g_maStandardSampleRatePriorities[iMALSampleRate];
+ UInt32 iCASampleRate;
+ for (iCASampleRate = 0; iCASampleRate < sampleRateRangeCount; ++iCASampleRate) {
+ AudioValueRange caSampleRate = pSampleRateRanges[iCASampleRate];
+ if (caSampleRate.mMinimum <= malSampleRate && caSampleRate.mMaximum >= malSampleRate) {
+ *pSampleRateOut = malSampleRate;
+ ma_free(pSampleRateRanges, &pContext->allocationCallbacks);
+ return MA_SUCCESS;
+ }
+ }
}
- while (c89atomic_load_explicit_32(pSpinlock, c89atomic_memory_order_relaxed) == 1) {
- if (yield) {
- ma_yield();
+ /*
+ If we get here it means none of miniaudio's standard sample rates matched any of the supported sample rates from the device. In this
+ case we just fall back to the first one reported by Core Audio.
+ */
+ MA_ASSERT(sampleRateRangeCount > 0);
+
+ *pSampleRateOut = pSampleRateRanges[0].mMinimum;
+ ma_free(pSampleRateRanges, &pContext->allocationCallbacks);
+ return MA_SUCCESS;
+ } else {
+ /* Find the closest match to this sample rate. */
+ UInt32 currentAbsoluteDifference = INT32_MAX;
+ UInt32 iCurrentClosestRange = (UInt32)-1;
+ UInt32 iRange;
+ for (iRange = 0; iRange < sampleRateRangeCount; ++iRange) {
+ if (pSampleRateRanges[iRange].mMinimum <= sampleRateIn && pSampleRateRanges[iRange].mMaximum >= sampleRateIn) {
+ *pSampleRateOut = sampleRateIn;
+ ma_free(pSampleRateRanges, &pContext->allocationCallbacks);
+ return MA_SUCCESS;
+ } else {
+ UInt32 absoluteDifference;
+ if (pSampleRateRanges[iRange].mMinimum > sampleRateIn) {
+ absoluteDifference = pSampleRateRanges[iRange].mMinimum - sampleRateIn;
+ } else {
+ absoluteDifference = sampleRateIn - pSampleRateRanges[iRange].mMaximum;
+ }
+
+ if (currentAbsoluteDifference > absoluteDifference) {
+ currentAbsoluteDifference = absoluteDifference;
+ iCurrentClosestRange = iRange;
+ }
}
}
+
+ MA_ASSERT(iCurrentClosestRange != (UInt32)-1);
+
+ *pSampleRateOut = pSampleRateRanges[iCurrentClosestRange].mMinimum;
+ ma_free(pSampleRateRanges, &pContext->allocationCallbacks);
+ return MA_SUCCESS;
}
- return MA_SUCCESS;
+ /* Should never get here, but it would mean we weren't able to find any suitable sample rates. */
+ /*ma_free(pSampleRateRanges, &pContext->allocationCallbacks);*/
+ /*return MA_ERROR;*/
}
+#endif
-MA_API ma_result ma_spinlock_lock(volatile ma_spinlock* pSpinlock)
+static ma_result ma_get_AudioObject_closest_buffer_size_in_frames(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32 bufferSizeInFramesIn, ma_uint32* pBufferSizeInFramesOut)
{
- return ma_spinlock_lock_ex(pSpinlock, MA_TRUE);
-}
+ AudioObjectPropertyAddress propAddress;
+ AudioValueRange bufferSizeRange;
+ UInt32 dataSize;
+ OSStatus status;
-MA_API ma_result ma_spinlock_lock_noyield(volatile ma_spinlock* pSpinlock)
-{
- return ma_spinlock_lock_ex(pSpinlock, MA_FALSE);
-}
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pBufferSizeInFramesOut != NULL);
-MA_API ma_result ma_spinlock_unlock(volatile ma_spinlock* pSpinlock)
-{
- if (pSpinlock == NULL) {
- return MA_INVALID_ARGS;
- }
+ *pBufferSizeInFramesOut = 0; /* Safety. */
- c89atomic_store_explicit_32(pSpinlock, 0, c89atomic_memory_order_release);
- return MA_SUCCESS;
-}
+ propAddress.mSelector = kAudioDevicePropertyBufferFrameSizeRange;
+ propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput;
+ propAddress.mElement = kAudioObjectPropertyElementMaster;
-#ifdef MA_WIN32
-static int ma_thread_priority_to_win32(ma_thread_priority priority)
-{
- switch (priority) {
- case ma_thread_priority_idle: return THREAD_PRIORITY_IDLE;
- case ma_thread_priority_lowest: return THREAD_PRIORITY_LOWEST;
- case ma_thread_priority_low: return THREAD_PRIORITY_BELOW_NORMAL;
- case ma_thread_priority_normal: return THREAD_PRIORITY_NORMAL;
- case ma_thread_priority_high: return THREAD_PRIORITY_ABOVE_NORMAL;
- case ma_thread_priority_highest: return THREAD_PRIORITY_HIGHEST;
- case ma_thread_priority_realtime: return THREAD_PRIORITY_TIME_CRITICAL;
- default: return THREAD_PRIORITY_NORMAL;
+ dataSize = sizeof(bufferSizeRange);
+ status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, &bufferSizeRange);
+ if (status != noErr) {
+ return ma_result_from_OSStatus(status);
}
-}
-static ma_result ma_thread_create__win32(ma_thread* pThread, ma_thread_priority priority, size_t stackSize, ma_thread_entry_proc entryProc, void* pData)
-{
- *pThread = CreateThread(NULL, stackSize, entryProc, pData, 0, NULL);
- if (*pThread == NULL) {
- return ma_result_from_GetLastError(GetLastError());
+ /* This is just a clamp. */
+ if (bufferSizeInFramesIn < bufferSizeRange.mMinimum) {
+ *pBufferSizeInFramesOut = (ma_uint32)bufferSizeRange.mMinimum;
+ } else if (bufferSizeInFramesIn > bufferSizeRange.mMaximum) {
+ *pBufferSizeInFramesOut = (ma_uint32)bufferSizeRange.mMaximum;
+ } else {
+ *pBufferSizeInFramesOut = bufferSizeInFramesIn;
}
- SetThreadPriority((HANDLE)*pThread, ma_thread_priority_to_win32(priority));
-
return MA_SUCCESS;
}
-static void ma_thread_wait__win32(ma_thread* pThread)
+static ma_result ma_set_AudioObject_buffer_size_in_frames(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32* pPeriodSizeInOut)
{
- WaitForSingleObject((HANDLE)*pThread, INFINITE);
- CloseHandle((HANDLE)*pThread);
-}
+ ma_result result;
+ ma_uint32 chosenBufferSizeInFrames;
+ AudioObjectPropertyAddress propAddress;
+ UInt32 dataSize;
+ OSStatus status;
+ MA_ASSERT(pContext != NULL);
-static ma_result ma_mutex_init__win32(ma_mutex* pMutex)
-{
- *pMutex = CreateEventW(NULL, FALSE, TRUE, NULL);
- if (*pMutex == NULL) {
- return ma_result_from_GetLastError(GetLastError());
+ result = ma_get_AudioObject_closest_buffer_size_in_frames(pContext, deviceObjectID, deviceType, *pPeriodSizeInOut, &chosenBufferSizeInFrames);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ /* Try setting the size of the buffer... If this fails we just use whatever is currently set. */
+ propAddress.mSelector = kAudioDevicePropertyBufferFrameSize;
+ propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput;
+ propAddress.mElement = kAudioObjectPropertyElementMaster;
+
+ ((ma_AudioObjectSetPropertyData_proc)pContext->coreaudio.AudioObjectSetPropertyData)(deviceObjectID, &propAddress, 0, NULL, sizeof(chosenBufferSizeInFrames), &chosenBufferSizeInFrames);
+
+ /* Get the actual size of the buffer. */
+ dataSize = sizeof(*pPeriodSizeInOut);
+ status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, &chosenBufferSizeInFrames);
+ if (status != noErr) {
+ return ma_result_from_OSStatus(status);
}
+ *pPeriodSizeInOut = chosenBufferSizeInFrames;
return MA_SUCCESS;
}
-static void ma_mutex_uninit__win32(ma_mutex* pMutex)
+static ma_result ma_find_default_AudioObjectID(ma_context* pContext, ma_device_type deviceType, AudioObjectID* pDeviceObjectID)
{
- CloseHandle((HANDLE)*pMutex);
-}
+ AudioObjectPropertyAddress propAddressDefaultDevice;
+ UInt32 defaultDeviceObjectIDSize = sizeof(AudioObjectID);
+ AudioObjectID defaultDeviceObjectID;
+ OSStatus status;
-static void ma_mutex_lock__win32(ma_mutex* pMutex)
-{
- WaitForSingleObject((HANDLE)*pMutex, INFINITE);
-}
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pDeviceObjectID != NULL);
-static void ma_mutex_unlock__win32(ma_mutex* pMutex)
-{
- SetEvent((HANDLE)*pMutex);
-}
+ /* Safety. */
+ *pDeviceObjectID = 0;
+ propAddressDefaultDevice.mScope = kAudioObjectPropertyScopeGlobal;
+ propAddressDefaultDevice.mElement = kAudioObjectPropertyElementMaster;
+ if (deviceType == ma_device_type_playback) {
+ propAddressDefaultDevice.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
+ } else {
+ propAddressDefaultDevice.mSelector = kAudioHardwarePropertyDefaultInputDevice;
+ }
-static ma_result ma_event_init__win32(ma_event* pEvent)
-{
- *pEvent = CreateEventW(NULL, FALSE, FALSE, NULL);
- if (*pEvent == NULL) {
- return ma_result_from_GetLastError(GetLastError());
+ defaultDeviceObjectIDSize = sizeof(AudioObjectID);
+ status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(kAudioObjectSystemObject, &propAddressDefaultDevice, 0, NULL, &defaultDeviceObjectIDSize, &defaultDeviceObjectID);
+ if (status == noErr) {
+ *pDeviceObjectID = defaultDeviceObjectID;
+ return MA_SUCCESS;
}
- return MA_SUCCESS;
+ /* If we get here it means we couldn't find the device. */
+ return MA_NO_DEVICE;
}
-static void ma_event_uninit__win32(ma_event* pEvent)
+static ma_result ma_find_AudioObjectID(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, AudioObjectID* pDeviceObjectID)
{
- CloseHandle((HANDLE)*pEvent);
-}
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pDeviceObjectID != NULL);
-static ma_result ma_event_wait__win32(ma_event* pEvent)
-{
- DWORD result = WaitForSingleObject((HANDLE)*pEvent, INFINITE);
- if (result == WAIT_OBJECT_0) {
- return MA_SUCCESS;
- }
+ /* Safety. */
+ *pDeviceObjectID = 0;
- if (result == WAIT_TIMEOUT) {
- return MA_TIMEOUT;
- }
+ if (pDeviceID == NULL) {
+ /* Default device. */
+ return ma_find_default_AudioObjectID(pContext, deviceType, pDeviceObjectID);
+ } else {
+ /* Explicit device. */
+ UInt32 deviceCount;
+ AudioObjectID* pDeviceObjectIDs;
+ ma_result result;
+ UInt32 iDevice;
- return ma_result_from_GetLastError(GetLastError());
-}
+ result = ma_get_device_object_ids__coreaudio(pContext, &deviceCount, &pDeviceObjectIDs);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
-static ma_result ma_event_signal__win32(ma_event* pEvent)
-{
- BOOL result = SetEvent((HANDLE)*pEvent);
- if (result == 0) {
- return ma_result_from_GetLastError(GetLastError());
- }
+ for (iDevice = 0; iDevice < deviceCount; ++iDevice) {
+ AudioObjectID deviceObjectID = pDeviceObjectIDs[iDevice];
- return MA_SUCCESS;
-}
+ char uid[256];
+ if (ma_get_AudioObject_uid(pContext, deviceObjectID, sizeof(uid), uid) != MA_SUCCESS) {
+ continue;
+ }
+ if (deviceType == ma_device_type_playback) {
+ if (ma_does_AudioObject_support_playback(pContext, deviceObjectID)) {
+ if (strcmp(uid, pDeviceID->coreaudio) == 0) {
+ *pDeviceObjectID = deviceObjectID;
+ ma_free(pDeviceObjectIDs, &pContext->allocationCallbacks);
+ return MA_SUCCESS;
+ }
+ }
+ } else {
+ if (ma_does_AudioObject_support_capture(pContext, deviceObjectID)) {
+ if (strcmp(uid, pDeviceID->coreaudio) == 0) {
+ *pDeviceObjectID = deviceObjectID;
+ ma_free(pDeviceObjectIDs, &pContext->allocationCallbacks);
+ return MA_SUCCESS;
+ }
+ }
+ }
+ }
-static ma_result ma_semaphore_init__win32(int initialValue, ma_semaphore* pSemaphore)
-{
- *pSemaphore = CreateSemaphoreW(NULL, (LONG)initialValue, LONG_MAX, NULL);
- if (*pSemaphore == NULL) {
- return ma_result_from_GetLastError(GetLastError());
+ ma_free(pDeviceObjectIDs, &pContext->allocationCallbacks);
}
- return MA_SUCCESS;
+ /* If we get here it means we couldn't find the device. */
+ return MA_NO_DEVICE;
}
-static void ma_semaphore_uninit__win32(ma_semaphore* pSemaphore)
-{
- CloseHandle((HANDLE)*pSemaphore);
-}
-static ma_result ma_semaphore_wait__win32(ma_semaphore* pSemaphore)
+static ma_result ma_find_best_format__coreaudio(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_format format, ma_uint32 channels, ma_uint32 sampleRate, const AudioStreamBasicDescription* pOrigFormat, AudioStreamBasicDescription* pFormat)
{
- DWORD result = WaitForSingleObject((HANDLE)*pSemaphore, INFINITE);
- if (result == WAIT_OBJECT_0) {
- return MA_SUCCESS;
+ UInt32 deviceFormatDescriptionCount;
+ AudioStreamRangedDescription* pDeviceFormatDescriptions;
+ ma_result result;
+ ma_uint32 desiredSampleRate;
+ ma_uint32 desiredChannelCount;
+ ma_format desiredFormat;
+ AudioStreamBasicDescription bestDeviceFormatSoFar;
+ ma_bool32 hasSupportedFormat;
+ UInt32 iFormat;
+
+ result = ma_get_AudioObject_stream_descriptions(pContext, deviceObjectID, deviceType, &deviceFormatDescriptionCount, &pDeviceFormatDescriptions);
+ if (result != MA_SUCCESS) {
+ return result;
}
- if (result == WAIT_TIMEOUT) {
- return MA_TIMEOUT;
+ desiredSampleRate = sampleRate;
+ if (desiredSampleRate == 0) {
+ desiredSampleRate = pOrigFormat->mSampleRate;
}
- return ma_result_from_GetLastError(GetLastError());
-}
+ desiredChannelCount = channels;
+ if (desiredChannelCount == 0) {
+ desiredChannelCount = pOrigFormat->mChannelsPerFrame;
+ }
-static ma_result ma_semaphore_release__win32(ma_semaphore* pSemaphore)
-{
- BOOL result = ReleaseSemaphore((HANDLE)*pSemaphore, 1, NULL);
- if (result == 0) {
- return ma_result_from_GetLastError(GetLastError());
+ desiredFormat = format;
+ if (desiredFormat == ma_format_unknown) {
+ result = ma_format_from_AudioStreamBasicDescription(pOrigFormat, &desiredFormat);
+ if (result != MA_SUCCESS || desiredFormat == ma_format_unknown) {
+ desiredFormat = g_maFormatPriorities[0];
+ }
}
- return MA_SUCCESS;
-}
-#endif
+ /*
+ If we get here it means we don't have an exact match to what the client is asking for. We'll need to find the closest one. The next
+ loop will check for formats that have the same sample rate to what we're asking for. If there is, we prefer that one in all cases.
+ */
+ MA_ZERO_OBJECT(&bestDeviceFormatSoFar);
+ hasSupportedFormat = MA_FALSE;
+ for (iFormat = 0; iFormat < deviceFormatDescriptionCount; ++iFormat) {
+ ma_format format;
+ ma_result formatResult = ma_format_from_AudioStreamBasicDescription(&pDeviceFormatDescriptions[iFormat].mFormat, &format);
+ if (formatResult == MA_SUCCESS && format != ma_format_unknown) {
+ hasSupportedFormat = MA_TRUE;
+ bestDeviceFormatSoFar = pDeviceFormatDescriptions[iFormat].mFormat;
+ break;
+ }
+ }
-#ifdef MA_POSIX
-static ma_result ma_thread_create__posix(ma_thread* pThread, ma_thread_priority priority, size_t stackSize, ma_thread_entry_proc entryProc, void* pData)
-{
- int result;
- pthread_attr_t* pAttr = NULL;
+ if (!hasSupportedFormat) {
+ ma_free(pDeviceFormatDescriptions, &pContext->allocationCallbacks);
+ return MA_FORMAT_NOT_SUPPORTED;
+ }
-#if !defined(__EMSCRIPTEN__)
- /* Try setting the thread priority. It's not critical if anything fails here. */
- pthread_attr_t attr;
- if (pthread_attr_init(&attr) == 0) {
- int scheduler = -1;
- if (priority == ma_thread_priority_idle) {
-#ifdef SCHED_IDLE
- if (pthread_attr_setschedpolicy(&attr, SCHED_IDLE) == 0) {
- scheduler = SCHED_IDLE;
- }
-#endif
- } else if (priority == ma_thread_priority_realtime) {
-#ifdef SCHED_FIFO
- if (pthread_attr_setschedpolicy(&attr, SCHED_FIFO) == 0) {
- scheduler = SCHED_FIFO;
- }
-#endif
-#ifdef MA_LINUX
- } else {
- scheduler = sched_getscheduler(0);
-#endif
- }
- if (stackSize > 0) {
- pthread_attr_setstacksize(&attr, stackSize);
+ for (iFormat = 0; iFormat < deviceFormatDescriptionCount; ++iFormat) {
+ AudioStreamBasicDescription thisDeviceFormat = pDeviceFormatDescriptions[iFormat].mFormat;
+ ma_format thisSampleFormat;
+ ma_result formatResult;
+ ma_format bestSampleFormatSoFar;
+
+ /* If the format is not supported by miniaudio we need to skip this one entirely. */
+ formatResult = ma_format_from_AudioStreamBasicDescription(&pDeviceFormatDescriptions[iFormat].mFormat, &thisSampleFormat);
+ if (formatResult != MA_SUCCESS || thisSampleFormat == ma_format_unknown) {
+ continue; /* The format is not supported by miniaudio. Skip. */
}
- if (scheduler != -1) {
- int priorityMin = sched_get_priority_min(scheduler);
- int priorityMax = sched_get_priority_max(scheduler);
- int priorityStep = (priorityMax - priorityMin) / 7; /* 7 = number of priorities supported by miniaudio. */
+ ma_format_from_AudioStreamBasicDescription(&bestDeviceFormatSoFar, &bestSampleFormatSoFar);
- struct sched_param sched;
- if (pthread_attr_getschedparam(&attr, &sched) == 0) {
- if (priority == ma_thread_priority_idle) {
- sched.sched_priority = priorityMin;
- } else if (priority == ma_thread_priority_realtime) {
- sched.sched_priority = priorityMax;
+ /* Getting here means the format is supported by miniaudio which makes this format a candidate. */
+ if (thisDeviceFormat.mSampleRate != desiredSampleRate) {
+ /*
+ The sample rate does not match, but this format could still be usable, although it's a very low priority. If the best format
+ so far has an equal sample rate we can just ignore this one.
+ */
+ if (bestDeviceFormatSoFar.mSampleRate == desiredSampleRate) {
+ continue; /* The best sample rate so far has the same sample rate as what we requested which means it's still the best so far. Skip this format. */
+ } else {
+ /* In this case, neither the best format so far nor this one have the same sample rate. Check the channel count next. */
+ if (thisDeviceFormat.mChannelsPerFrame != desiredChannelCount) {
+ /* This format has a different sample rate _and_ a different channel count. */
+ if (bestDeviceFormatSoFar.mChannelsPerFrame == desiredChannelCount) {
+ continue; /* No change to the best format. */
+ } else {
+ /*
+ Both this format and the best so far have different sample rates and different channel counts. Whichever has the
+ best format is the new best.
+ */
+ if (ma_get_format_priority_index(thisSampleFormat) < ma_get_format_priority_index(bestSampleFormatSoFar)) {
+ bestDeviceFormatSoFar = thisDeviceFormat;
+ continue;
+ } else {
+ continue; /* No change to the best format. */
+ }
+ }
} else {
- sched.sched_priority += ((int)priority + 5) * priorityStep; /* +5 because the lowest priority is -5. */
- if (sched.sched_priority < priorityMin) {
- sched.sched_priority = priorityMin;
+ /* This format has a different sample rate but the desired channel count. */
+ if (bestDeviceFormatSoFar.mChannelsPerFrame == desiredChannelCount) {
+ /* Both this format and the best so far have the desired channel count. Whichever has the best format is the new best. */
+ if (ma_get_format_priority_index(thisSampleFormat) < ma_get_format_priority_index(bestSampleFormatSoFar)) {
+ bestDeviceFormatSoFar = thisDeviceFormat;
+ continue;
+ } else {
+ continue; /* No change to the best format for now. */
+ }
+ } else {
+ /* This format has the desired channel count, but the best so far does not. We have a new best. */
+ bestDeviceFormatSoFar = thisDeviceFormat;
+ continue;
+ }
+ }
+ }
+ } else {
+ /*
+ The sample rates match which makes this format a very high priority contender. If the best format so far has a different
+ sample rate it needs to be replaced with this one.
+ */
+ if (bestDeviceFormatSoFar.mSampleRate != desiredSampleRate) {
+ bestDeviceFormatSoFar = thisDeviceFormat;
+ continue;
+ } else {
+ /* In this case both this format and the best format so far have the same sample rate. Check the channel count next. */
+ if (thisDeviceFormat.mChannelsPerFrame == desiredChannelCount) {
+ /*
+ In this case this format has the same channel count as what the client is requesting. If the best format so far has
+ a different count, this one becomes the new best.
+ */
+ if (bestDeviceFormatSoFar.mChannelsPerFrame != desiredChannelCount) {
+ bestDeviceFormatSoFar = thisDeviceFormat;
+ continue;
+ } else {
+ /* In this case both this format and the best so far have the ideal sample rate and channel count. Check the format. */
+ if (thisSampleFormat == desiredFormat) {
+ bestDeviceFormatSoFar = thisDeviceFormat;
+ break; /* Found the exact match. */
+ } else {
+ /* The formats are different. The new best format is the one with the highest priority format according to miniaudio. */
+ if (ma_get_format_priority_index(thisSampleFormat) < ma_get_format_priority_index(bestSampleFormatSoFar)) {
+ bestDeviceFormatSoFar = thisDeviceFormat;
+ continue;
+ } else {
+ continue; /* No change to the best format for now. */
+ }
+ }
}
- if (sched.sched_priority > priorityMax) {
- sched.sched_priority = priorityMax;
+ } else {
+ /*
+ In this case the channel count is different to what the client has requested. If the best so far has the same channel
+ count as the requested count then it remains the best.
+ */
+ if (bestDeviceFormatSoFar.mChannelsPerFrame == desiredChannelCount) {
+ continue;
+ } else {
+ /*
+ This is the case where both have the same sample rate (good) but different channel counts. Right now both have about
+ the same priority, but we need to compare the format now.
+ */
+ if (thisSampleFormat == bestSampleFormatSoFar) {
+ if (ma_get_format_priority_index(thisSampleFormat) < ma_get_format_priority_index(bestSampleFormatSoFar)) {
+ bestDeviceFormatSoFar = thisDeviceFormat;
+ continue;
+ } else {
+ continue; /* No change to the best format for now. */
+ }
+ }
}
}
-
- if (pthread_attr_setschedparam(&attr, &sched) == 0) {
- pAttr = &attr;
- }
}
}
}
-#else
- /* It's the emscripten build. We'll have a few unused parameters. */
- (void)priority;
- (void)stackSize;
-#endif
-
- result = pthread_create(pThread, pAttr, entryProc, pData);
-
- /* The thread attributes object is no longer required. */
- if (pAttr != NULL) {
- pthread_attr_destroy(pAttr);
- }
- if (result != 0) {
- return ma_result_from_errno(result);
- }
+ *pFormat = bestDeviceFormatSoFar;
+ ma_free(pDeviceFormatDescriptions, &pContext->allocationCallbacks);
return MA_SUCCESS;
}
-static void ma_thread_wait__posix(ma_thread* pThread)
+static ma_result ma_get_AudioUnit_channel_map(ma_context* pContext, AudioUnit audioUnit, ma_device_type deviceType, ma_channel* pChannelMap, size_t channelMapCap)
{
- pthread_join(*pThread, NULL);
- pthread_detach(*pThread);
-}
+ AudioUnitScope deviceScope;
+ AudioUnitElement deviceBus;
+ UInt32 channelLayoutSize;
+ OSStatus status;
+ AudioChannelLayout* pChannelLayout;
+ ma_result result;
+ MA_ASSERT(pContext != NULL);
-static ma_result ma_mutex_init__posix(ma_mutex* pMutex)
-{
- int result = pthread_mutex_init((pthread_mutex_t*)pMutex, NULL);
- if (result != 0) {
- return ma_result_from_errno(result);
+ if (deviceType == ma_device_type_playback) {
+ deviceScope = kAudioUnitScope_Input;
+ deviceBus = MA_COREAUDIO_OUTPUT_BUS;
+ } else {
+ deviceScope = kAudioUnitScope_Output;
+ deviceBus = MA_COREAUDIO_INPUT_BUS;
}
- return MA_SUCCESS;
-}
-
-static void ma_mutex_uninit__posix(ma_mutex* pMutex)
-{
- pthread_mutex_destroy((pthread_mutex_t*)pMutex);
-}
-
-static void ma_mutex_lock__posix(ma_mutex* pMutex)
-{
- pthread_mutex_lock((pthread_mutex_t*)pMutex);
-}
-
-static void ma_mutex_unlock__posix(ma_mutex* pMutex)
-{
- pthread_mutex_unlock((pthread_mutex_t*)pMutex);
-}
-
+ status = ((ma_AudioUnitGetPropertyInfo_proc)pContext->coreaudio.AudioUnitGetPropertyInfo)(audioUnit, kAudioUnitProperty_AudioChannelLayout, deviceScope, deviceBus, &channelLayoutSize, NULL);
+ if (status != noErr) {
+ return ma_result_from_OSStatus(status);
+ }
-static ma_result ma_event_init__posix(ma_event* pEvent)
-{
- int result;
+ pChannelLayout = (AudioChannelLayout*)ma_malloc(channelLayoutSize, &pContext->allocationCallbacks);
+ if (pChannelLayout == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
- result = pthread_mutex_init(&pEvent->lock, NULL);
- if (result != 0) {
- return ma_result_from_errno(result);
+ status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(audioUnit, kAudioUnitProperty_AudioChannelLayout, deviceScope, deviceBus, pChannelLayout, &channelLayoutSize);
+ if (status != noErr) {
+ ma_free(pChannelLayout, &pContext->allocationCallbacks);
+ return ma_result_from_OSStatus(status);
}
- result = pthread_cond_init(&pEvent->cond, NULL);
- if (result != 0) {
- pthread_mutex_destroy(&pEvent->lock);
- return ma_result_from_errno(result);
+ result = ma_get_channel_map_from_AudioChannelLayout(pChannelLayout, pChannelMap, channelMapCap);
+ if (result != MA_SUCCESS) {
+ ma_free(pChannelLayout, &pContext->allocationCallbacks);
+ return result;
}
- pEvent->value = 0;
+ ma_free(pChannelLayout, &pContext->allocationCallbacks);
return MA_SUCCESS;
}
+#endif /* MA_APPLE_DESKTOP */
-static void ma_event_uninit__posix(ma_event* pEvent)
-{
- pthread_cond_destroy(&pEvent->cond);
- pthread_mutex_destroy(&pEvent->lock);
-}
-static ma_result ma_event_wait__posix(ma_event* pEvent)
+#if !defined(MA_APPLE_DESKTOP)
+static void ma_AVAudioSessionPortDescription_to_device_info(AVAudioSessionPortDescription* pPortDesc, ma_device_info* pInfo)
{
- pthread_mutex_lock(&pEvent->lock);
- {
- while (pEvent->value == 0) {
- pthread_cond_wait(&pEvent->cond, &pEvent->lock);
- }
- pEvent->value = 0; /* Auto-reset. */
- }
- pthread_mutex_unlock(&pEvent->lock);
-
- return MA_SUCCESS;
+ MA_ZERO_OBJECT(pInfo);
+ ma_strncpy_s(pInfo->name, sizeof(pInfo->name), [pPortDesc.portName UTF8String], (size_t)-1);
+ ma_strncpy_s(pInfo->id.coreaudio, sizeof(pInfo->id.coreaudio), [pPortDesc.UID UTF8String], (size_t)-1);
}
+#endif
-static ma_result ma_event_signal__posix(ma_event* pEvent)
+static ma_result ma_context_enumerate_devices__coreaudio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
{
- pthread_mutex_lock(&pEvent->lock);
- {
- pEvent->value = 1;
- pthread_cond_signal(&pEvent->cond);
- }
- pthread_mutex_unlock(&pEvent->lock);
-
- return MA_SUCCESS;
-}
-
+#if defined(MA_APPLE_DESKTOP)
+ UInt32 deviceCount;
+ AudioObjectID* pDeviceObjectIDs;
+ AudioObjectID defaultDeviceObjectIDPlayback;
+ AudioObjectID defaultDeviceObjectIDCapture;
+ ma_result result;
+ UInt32 iDevice;
-static ma_result ma_semaphore_init__posix(int initialValue, ma_semaphore* pSemaphore)
-{
- int result;
+ ma_find_default_AudioObjectID(pContext, ma_device_type_playback, &defaultDeviceObjectIDPlayback); /* OK if this fails. */
+ ma_find_default_AudioObjectID(pContext, ma_device_type_capture, &defaultDeviceObjectIDCapture); /* OK if this fails. */
- if (pSemaphore == NULL) {
- return MA_INVALID_ARGS;
+ result = ma_get_device_object_ids__coreaudio(pContext, &deviceCount, &pDeviceObjectIDs);
+ if (result != MA_SUCCESS) {
+ return result;
}
- pSemaphore->value = initialValue;
+ for (iDevice = 0; iDevice < deviceCount; ++iDevice) {
+ AudioObjectID deviceObjectID = pDeviceObjectIDs[iDevice];
+ ma_device_info info;
- result = pthread_mutex_init(&pSemaphore->lock, NULL);
- if (result != 0) {
- return ma_result_from_errno(result); /* Failed to create mutex. */
- }
+ MA_ZERO_OBJECT(&info);
+ if (ma_get_AudioObject_uid(pContext, deviceObjectID, sizeof(info.id.coreaudio), info.id.coreaudio) != MA_SUCCESS) {
+ continue;
+ }
+ if (ma_get_AudioObject_name(pContext, deviceObjectID, sizeof(info.name), info.name) != MA_SUCCESS) {
+ continue;
+ }
- result = pthread_cond_init(&pSemaphore->cond, NULL);
- if (result != 0) {
- pthread_mutex_destroy(&pSemaphore->lock);
- return ma_result_from_errno(result); /* Failed to create condition variable. */
- }
+ if (ma_does_AudioObject_support_playback(pContext, deviceObjectID)) {
+ if (deviceObjectID == defaultDeviceObjectIDPlayback) {
+ info.isDefault = MA_TRUE;
+ }
- return MA_SUCCESS;
-}
+ if (!callback(pContext, ma_device_type_playback, &info, pUserData)) {
+ break;
+ }
+ }
+ if (ma_does_AudioObject_support_capture(pContext, deviceObjectID)) {
+ if (deviceObjectID == defaultDeviceObjectIDCapture) {
+ info.isDefault = MA_TRUE;
+ }
-static void ma_semaphore_uninit__posix(ma_semaphore* pSemaphore)
-{
- if (pSemaphore == NULL) {
- return;
+ if (!callback(pContext, ma_device_type_capture, &info, pUserData)) {
+ break;
+ }
+ }
}
- pthread_cond_destroy(&pSemaphore->cond);
- pthread_mutex_destroy(&pSemaphore->lock);
-}
+ ma_free(pDeviceObjectIDs, &pContext->allocationCallbacks);
+#else
+ ma_device_info info;
+ NSArray *pInputs = [[[AVAudioSession sharedInstance] currentRoute] inputs];
+ NSArray *pOutputs = [[[AVAudioSession sharedInstance] currentRoute] outputs];
-static ma_result ma_semaphore_wait__posix(ma_semaphore* pSemaphore)
-{
- if (pSemaphore == NULL) {
- return MA_INVALID_ARGS;
+ for (AVAudioSessionPortDescription* pPortDesc in pOutputs) {
+ ma_AVAudioSessionPortDescription_to_device_info(pPortDesc, &info);
+ if (!callback(pContext, ma_device_type_playback, &info, pUserData)) {
+ return MA_SUCCESS;
+ }
}
- pthread_mutex_lock(&pSemaphore->lock);
- {
- /* We need to wait on a condition variable before escaping. We can't return from this function until the semaphore has been signaled. */
- while (pSemaphore->value == 0) {
- pthread_cond_wait(&pSemaphore->cond, &pSemaphore->lock);
+ for (AVAudioSessionPortDescription* pPortDesc in pInputs) {
+ ma_AVAudioSessionPortDescription_to_device_info(pPortDesc, &info);
+ if (!callback(pContext, ma_device_type_capture, &info, pUserData)) {
+ return MA_SUCCESS;
}
-
- pSemaphore->value -= 1;
}
- pthread_mutex_unlock(&pSemaphore->lock);
+#endif
return MA_SUCCESS;
}
-static ma_result ma_semaphore_release__posix(ma_semaphore* pSemaphore)
+static ma_result ma_context_get_device_info__coreaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
{
- if (pSemaphore == NULL) {
- return MA_INVALID_ARGS;
- }
+ ma_result result;
- pthread_mutex_lock(&pSemaphore->lock);
- {
- pSemaphore->value += 1;
- pthread_cond_signal(&pSemaphore->cond);
- }
- pthread_mutex_unlock(&pSemaphore->lock);
+ MA_ASSERT(pContext != NULL);
- return MA_SUCCESS;
-}
-#endif
+#if defined(MA_APPLE_DESKTOP)
+ /* Desktop */
+ {
+ AudioObjectID deviceObjectID;
+ AudioObjectID defaultDeviceObjectID;
+ UInt32 streamDescriptionCount;
+ AudioStreamRangedDescription* pStreamDescriptions;
+ UInt32 iStreamDescription;
+ UInt32 sampleRateRangeCount;
+ AudioValueRange* pSampleRateRanges;
-typedef struct
-{
- ma_thread_entry_proc entryProc;
- void* pData;
- ma_allocation_callbacks allocationCallbacks;
-} ma_thread_proxy_data;
+ ma_find_default_AudioObjectID(pContext, deviceType, &defaultDeviceObjectID); /* OK if this fails. */
-static ma_thread_result MA_THREADCALL ma_thread_entry_proxy(void* pData)
-{
- ma_thread_proxy_data* pProxyData = (ma_thread_proxy_data*)pData;
- ma_thread_entry_proc entryProc;
- void* pEntryProcData;
- ma_thread_result result;
+ result = ma_find_AudioObjectID(pContext, deviceType, pDeviceID, &deviceObjectID);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- #if defined(MA_ON_THREAD_ENTRY)
- MA_ON_THREAD_ENTRY
- #endif
+ result = ma_get_AudioObject_uid(pContext, deviceObjectID, sizeof(pDeviceInfo->id.coreaudio), pDeviceInfo->id.coreaudio);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- entryProc = pProxyData->entryProc;
- pEntryProcData = pProxyData->pData;
+ result = ma_get_AudioObject_name(pContext, deviceObjectID, sizeof(pDeviceInfo->name), pDeviceInfo->name);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- /* Free the proxy data before getting into the real thread entry proc. */
- ma_free(pProxyData, &pProxyData->allocationCallbacks);
+ if (deviceObjectID == defaultDeviceObjectID) {
+ pDeviceInfo->isDefault = MA_TRUE;
+ }
- result = entryProc(pEntryProcData);
+ /*
+ There could be a large number of permutations here. Fortunately there is only a single channel count
+ being reported which reduces this quite a bit. For sample rates we're only reporting those that are
+ one of miniaudio's recognized "standard" rates. If there are still more formats than can fit into
+ our fixed sized array we'll just need to truncate them. This is unlikely and will probably only happen
+ if some driver performs software data conversion and therefore reports every possible format and
+ sample rate.
+ */
+ pDeviceInfo->nativeDataFormatCount = 0;
- #if defined(MA_ON_THREAD_EXIT)
- MA_ON_THREAD_EXIT
- #endif
+ /* Formats. */
+ {
+ ma_format uniqueFormats[ma_format_count];
+ ma_uint32 uniqueFormatCount = 0;
+ ma_uint32 channels;
- return result;
-}
+ /* Channels. */
+ result = ma_get_AudioObject_channel_count(pContext, deviceObjectID, deviceType, &channels);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
-static ma_result ma_thread_create(ma_thread* pThread, ma_thread_priority priority, size_t stackSize, ma_thread_entry_proc entryProc, void* pData, const ma_allocation_callbacks* pAllocationCallbacks)
-{
- ma_result result;
- ma_thread_proxy_data* pProxyData;
+ /* Formats. */
+ result = ma_get_AudioObject_stream_descriptions(pContext, deviceObjectID, deviceType, &streamDescriptionCount, &pStreamDescriptions);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- if (pThread == NULL || entryProc == NULL) {
- return MA_FALSE;
- }
+ for (iStreamDescription = 0; iStreamDescription < streamDescriptionCount; ++iStreamDescription) {
+ ma_format format;
+ ma_bool32 hasFormatBeenHandled = MA_FALSE;
+ ma_uint32 iOutputFormat;
+ ma_uint32 iSampleRate;
- pProxyData = (ma_thread_proxy_data*)ma_malloc(sizeof(*pProxyData), pAllocationCallbacks); /* Will be freed by the proxy entry proc. */
- if (pProxyData == NULL) {
- return MA_OUT_OF_MEMORY;
- }
+ result = ma_format_from_AudioStreamBasicDescription(&pStreamDescriptions[iStreamDescription].mFormat, &format);
+ if (result != MA_SUCCESS) {
+ continue;
+ }
- pProxyData->entryProc = entryProc;
- pProxyData->pData = pData;
- ma_allocation_callbacks_init_copy(&pProxyData->allocationCallbacks, pAllocationCallbacks);
+ MA_ASSERT(format != ma_format_unknown);
-#ifdef MA_WIN32
- result = ma_thread_create__win32(pThread, priority, stackSize, ma_thread_entry_proxy, pProxyData);
-#endif
-#ifdef MA_POSIX
- result = ma_thread_create__posix(pThread, priority, stackSize, ma_thread_entry_proxy, pProxyData);
-#endif
+ /* Make sure the format isn't already in the output list. */
+ for (iOutputFormat = 0; iOutputFormat < uniqueFormatCount; ++iOutputFormat) {
+ if (uniqueFormats[iOutputFormat] == format) {
+ hasFormatBeenHandled = MA_TRUE;
+ break;
+ }
+ }
- if (result != MA_SUCCESS) {
- ma_free(pProxyData, pAllocationCallbacks);
- return result;
- }
+ /* If we've already handled this format just skip it. */
+ if (hasFormatBeenHandled) {
+ continue;
+ }
- return MA_SUCCESS;
-}
+ uniqueFormats[uniqueFormatCount] = format;
+ uniqueFormatCount += 1;
-static void ma_thread_wait(ma_thread* pThread)
-{
- if (pThread == NULL) {
- return;
- }
+ /* Sample Rates */
+ result = ma_get_AudioObject_sample_rates(pContext, deviceObjectID, deviceType, &sampleRateRangeCount, &pSampleRateRanges);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
-#ifdef MA_WIN32
- ma_thread_wait__win32(pThread);
-#endif
-#ifdef MA_POSIX
- ma_thread_wait__posix(pThread);
-#endif
-}
+ /*
+ Annoyingly Core Audio reports a sample rate range. We just get all the standard rates that are
+ between this range.
+ */
+ for (iSampleRate = 0; iSampleRate < sampleRateRangeCount; ++iSampleRate) {
+ ma_uint32 iStandardSampleRate;
+ for (iStandardSampleRate = 0; iStandardSampleRate < ma_countof(g_maStandardSampleRatePriorities); iStandardSampleRate += 1) {
+ ma_uint32 standardSampleRate = g_maStandardSampleRatePriorities[iStandardSampleRate];
+ if (standardSampleRate >= pSampleRateRanges[iSampleRate].mMinimum && standardSampleRate <= pSampleRateRanges[iSampleRate].mMaximum) {
+ /* We have a new data format. Add it to the list. */
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = format;
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = channels;
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = standardSampleRate;
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = 0;
+ pDeviceInfo->nativeDataFormatCount += 1;
+ if (pDeviceInfo->nativeDataFormatCount >= ma_countof(pDeviceInfo->nativeDataFormats)) {
+ break; /* No more room for any more formats. */
+ }
+ }
+ }
+ }
-MA_API ma_result ma_mutex_init(ma_mutex* pMutex)
-{
- if (pMutex == NULL) {
- MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */
- return MA_INVALID_ARGS;
- }
+ ma_free(pSampleRateRanges, &pContext->allocationCallbacks);
-#ifdef MA_WIN32
- return ma_mutex_init__win32(pMutex);
-#endif
-#ifdef MA_POSIX
- return ma_mutex_init__posix(pMutex);
-#endif
-}
+ if (pDeviceInfo->nativeDataFormatCount >= ma_countof(pDeviceInfo->nativeDataFormats)) {
+ break; /* No more room for any more formats. */
+ }
+ }
-MA_API void ma_mutex_uninit(ma_mutex* pMutex)
-{
- if (pMutex == NULL) {
- return;
+ ma_free(pStreamDescriptions, &pContext->allocationCallbacks);
+ }
}
+#else
+ /* Mobile */
+ {
+ AudioComponentDescription desc;
+ AudioComponent component;
+ AudioUnit audioUnit;
+ OSStatus status;
+ AudioUnitScope formatScope;
+ AudioUnitElement formatElement;
+ AudioStreamBasicDescription bestFormat;
+ UInt32 propSize;
-#ifdef MA_WIN32
- ma_mutex_uninit__win32(pMutex);
-#endif
-#ifdef MA_POSIX
- ma_mutex_uninit__posix(pMutex);
-#endif
-}
+ /* We want to ensure we use a consistent device name to device enumeration. */
+ if (pDeviceID != NULL && pDeviceID->coreaudio[0] != '\0') {
+ ma_bool32 found = MA_FALSE;
+ if (deviceType == ma_device_type_playback) {
+ NSArray *pOutputs = [[[AVAudioSession sharedInstance] currentRoute] outputs];
+ for (AVAudioSessionPortDescription* pPortDesc in pOutputs) {
+ if (strcmp(pDeviceID->coreaudio, [pPortDesc.UID UTF8String]) == 0) {
+ ma_AVAudioSessionPortDescription_to_device_info(pPortDesc, pDeviceInfo);
+ found = MA_TRUE;
+ break;
+ }
+ }
+ } else {
+ NSArray *pInputs = [[[AVAudioSession sharedInstance] currentRoute] inputs];
+ for (AVAudioSessionPortDescription* pPortDesc in pInputs) {
+ if (strcmp(pDeviceID->coreaudio, [pPortDesc.UID UTF8String]) == 0) {
+ ma_AVAudioSessionPortDescription_to_device_info(pPortDesc, pDeviceInfo);
+ found = MA_TRUE;
+ break;
+ }
+ }
+ }
-MA_API void ma_mutex_lock(ma_mutex* pMutex)
-{
- if (pMutex == NULL) {
- MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */
- return;
- }
+ if (!found) {
+ return MA_DOES_NOT_EXIST;
+ }
+ } else {
+ if (deviceType == ma_device_type_playback) {
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
+ } else {
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
+ }
+ }
-#ifdef MA_WIN32
- ma_mutex_lock__win32(pMutex);
-#endif
-#ifdef MA_POSIX
- ma_mutex_lock__posix(pMutex);
-#endif
-}
-MA_API void ma_mutex_unlock(ma_mutex* pMutex)
-{
- if (pMutex == NULL) {
- MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */
- return;
-}
+ /*
+ Retrieving device information is more annoying on mobile than desktop. For simplicity I'm locking this down to whatever format is
+ reported on a temporary I/O unit. The problem, however, is that this doesn't return a value for the sample rate which we need to
+ retrieve from the AVAudioSession shared instance.
+ */
+ desc.componentType = kAudioUnitType_Output;
+ desc.componentSubType = kAudioUnitSubType_RemoteIO;
+ desc.componentManufacturer = kAudioUnitManufacturer_Apple;
+ desc.componentFlags = 0;
+ desc.componentFlagsMask = 0;
+
+ component = ((ma_AudioComponentFindNext_proc)pContext->coreaudio.AudioComponentFindNext)(NULL, &desc);
+ if (component == NULL) {
+ return MA_FAILED_TO_INIT_BACKEND;
+ }
-#ifdef MA_WIN32
- ma_mutex_unlock__win32(pMutex);
-#endif
-#ifdef MA_POSIX
- ma_mutex_unlock__posix(pMutex);
-#endif
-}
+ status = ((ma_AudioComponentInstanceNew_proc)pContext->coreaudio.AudioComponentInstanceNew)(component, &audioUnit);
+ if (status != noErr) {
+ return ma_result_from_OSStatus(status);
+ }
+ formatScope = (deviceType == ma_device_type_playback) ? kAudioUnitScope_Input : kAudioUnitScope_Output;
+ formatElement = (deviceType == ma_device_type_playback) ? MA_COREAUDIO_OUTPUT_BUS : MA_COREAUDIO_INPUT_BUS;
-MA_API ma_result ma_event_init(ma_event* pEvent)
-{
- if (pEvent == NULL) {
- MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */
- return MA_INVALID_ARGS;
- }
+ propSize = sizeof(bestFormat);
+ status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, &propSize);
+ if (status != noErr) {
+ ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(audioUnit);
+ return ma_result_from_OSStatus(status);
+ }
-#ifdef MA_WIN32
- return ma_event_init__win32(pEvent);
-#endif
-#ifdef MA_POSIX
- return ma_event_init__posix(pEvent);
-#endif
-}
+ ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(audioUnit);
+ audioUnit = NULL;
-#if 0
-static ma_result ma_event_alloc_and_init(ma_event** ppEvent, ma_allocation_callbacks* pAllocationCallbacks)
-{
- ma_result result;
- ma_event* pEvent;
+ /* Only a single format is being reported for iOS. */
+ pDeviceInfo->nativeDataFormatCount = 1;
- if (ppEvent == NULL) {
- return MA_INVALID_ARGS;
- }
+ result = ma_format_from_AudioStreamBasicDescription(&bestFormat, &pDeviceInfo->nativeDataFormats[0].format);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- *ppEvent = NULL;
+ pDeviceInfo->nativeDataFormats[0].channels = bestFormat.mChannelsPerFrame;
- pEvent = ma_malloc(sizeof(*pEvent), pAllocationCallbacks/*, MA_ALLOCATION_TYPE_EVENT*/);
- if (pEvent == NULL) {
- return MA_OUT_OF_MEMORY;
- }
+ /*
+ It looks like Apple are wanting to push the whole AVAudioSession thing. Thus, we need to use that to determine device settings. To do
+ this we just get the shared instance and inspect.
+ */
+ @autoreleasepool {
+ AVAudioSession* pAudioSession = [AVAudioSession sharedInstance];
+ MA_ASSERT(pAudioSession != NULL);
- result = ma_event_init(pEvent);
- if (result != MA_SUCCESS) {
- ma_free(pEvent, pAllocationCallbacks/*, MA_ALLOCATION_TYPE_EVENT*/);
- return result;
+ pDeviceInfo->nativeDataFormats[0].sampleRate = (ma_uint32)pAudioSession.sampleRate;
+ }
}
+#endif
- *ppEvent = pEvent;
- return result;
+ (void)pDeviceInfo; /* Unused. */
+ return MA_SUCCESS;
}
-#endif
-MA_API void ma_event_uninit(ma_event* pEvent)
+static AudioBufferList* ma_allocate_AudioBufferList__coreaudio(ma_uint32 sizeInFrames, ma_format format, ma_uint32 channels, ma_stream_layout layout, const ma_allocation_callbacks* pAllocationCallbacks)
{
- if (pEvent == NULL) {
- return;
- }
+ AudioBufferList* pBufferList;
+ UInt32 audioBufferSizeInBytes;
+ size_t allocationSize;
-#ifdef MA_WIN32
- ma_event_uninit__win32(pEvent);
-#endif
-#ifdef MA_POSIX
- ma_event_uninit__posix(pEvent);
-#endif
-}
+ MA_ASSERT(sizeInFrames > 0);
+ MA_ASSERT(format != ma_format_unknown);
+ MA_ASSERT(channels > 0);
-#if 0
-static void ma_event_uninit_and_free(ma_event* pEvent, ma_allocation_callbacks* pAllocationCallbacks)
-{
- if (pEvent == NULL) {
- return;
+ allocationSize = sizeof(AudioBufferList) - sizeof(AudioBuffer); /* Subtract sizeof(AudioBuffer) because that part is dynamically sized. */
+ if (layout == ma_stream_layout_interleaved) {
+ /* Interleaved case. This is the simple case because we just have one buffer. */
+ allocationSize += sizeof(AudioBuffer) * 1;
+ } else {
+ /* Non-interleaved case. This is the more complex case because there's more than one buffer. */
+ allocationSize += sizeof(AudioBuffer) * channels;
}
- ma_event_uninit(pEvent);
- ma_free(pEvent, pAllocationCallbacks/*, MA_ALLOCATION_TYPE_EVENT*/);
-}
-#endif
+ allocationSize += sizeInFrames * ma_get_bytes_per_frame(format, channels);
-MA_API ma_result ma_event_wait(ma_event* pEvent)
-{
- if (pEvent == NULL) {
- MA_ASSERT(MA_FALSE); /* Fire an assert to the caller is aware of this bug. */
- return MA_INVALID_ARGS;
+ pBufferList = (AudioBufferList*)ma_malloc(allocationSize, pAllocationCallbacks);
+ if (pBufferList == NULL) {
+ return NULL;
}
-#ifdef MA_WIN32
- return ma_event_wait__win32(pEvent);
-#endif
-#ifdef MA_POSIX
- return ma_event_wait__posix(pEvent);
-#endif
-}
+ audioBufferSizeInBytes = (UInt32)(sizeInFrames * ma_get_bytes_per_sample(format));
-MA_API ma_result ma_event_signal(ma_event* pEvent)
-{
- if (pEvent == NULL) {
- MA_ASSERT(MA_FALSE); /* Fire an assert to the caller is aware of this bug. */
- return MA_INVALID_ARGS;
+ if (layout == ma_stream_layout_interleaved) {
+ pBufferList->mNumberBuffers = 1;
+ pBufferList->mBuffers[0].mNumberChannels = channels;
+ pBufferList->mBuffers[0].mDataByteSize = audioBufferSizeInBytes * channels;
+ pBufferList->mBuffers[0].mData = (ma_uint8*)pBufferList + sizeof(AudioBufferList);
+ } else {
+ ma_uint32 iBuffer;
+ pBufferList->mNumberBuffers = channels;
+ for (iBuffer = 0; iBuffer < pBufferList->mNumberBuffers; ++iBuffer) {
+ pBufferList->mBuffers[iBuffer].mNumberChannels = 1;
+ pBufferList->mBuffers[iBuffer].mDataByteSize = audioBufferSizeInBytes;
+ pBufferList->mBuffers[iBuffer].mData = (ma_uint8*)pBufferList + ((sizeof(AudioBufferList) - sizeof(AudioBuffer)) + (sizeof(AudioBuffer) * channels)) + (audioBufferSizeInBytes * iBuffer);
+ }
}
-#ifdef MA_WIN32
- return ma_event_signal__win32(pEvent);
-#endif
-#ifdef MA_POSIX
- return ma_event_signal__posix(pEvent);
-#endif
+ return pBufferList;
}
-
-MA_API ma_result ma_semaphore_init(int initialValue, ma_semaphore* pSemaphore)
+static ma_result ma_device_realloc_AudioBufferList__coreaudio(ma_device* pDevice, ma_uint32 sizeInFrames, ma_format format, ma_uint32 channels, ma_stream_layout layout)
{
- if (pSemaphore == NULL) {
- MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */
- return MA_INVALID_ARGS;
- }
+ MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(format != ma_format_unknown);
+ MA_ASSERT(channels > 0);
-#ifdef MA_WIN32
- return ma_semaphore_init__win32(initialValue, pSemaphore);
-#endif
-#ifdef MA_POSIX
- return ma_semaphore_init__posix(initialValue, pSemaphore);
-#endif
-}
+ /* Only resize the buffer if necessary. */
+ if (pDevice->coreaudio.audioBufferCapInFrames < sizeInFrames) {
+ AudioBufferList* pNewAudioBufferList;
-MA_API void ma_semaphore_uninit(ma_semaphore* pSemaphore)
-{
- if (pSemaphore == NULL) {
- MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */
- return;
+ pNewAudioBufferList = ma_allocate_AudioBufferList__coreaudio(sizeInFrames, format, channels, layout, &pDevice->pContext->allocationCallbacks);
+ if (pNewAudioBufferList == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+
+ /* At this point we'll have a new AudioBufferList and we can free the old one. */
+ ma_free(pDevice->coreaudio.pAudioBufferList, &pDevice->pContext->allocationCallbacks);
+ pDevice->coreaudio.pAudioBufferList = pNewAudioBufferList;
+ pDevice->coreaudio.audioBufferCapInFrames = sizeInFrames;
}
-#ifdef MA_WIN32
- ma_semaphore_uninit__win32(pSemaphore);
-#endif
-#ifdef MA_POSIX
- ma_semaphore_uninit__posix(pSemaphore);
-#endif
+ /* Getting here means the capacity of the audio is fine. */
+ return MA_SUCCESS;
}
-MA_API ma_result ma_semaphore_wait(ma_semaphore* pSemaphore)
+
+static OSStatus ma_on_output__coreaudio(void* pUserData, AudioUnitRenderActionFlags* pActionFlags, const AudioTimeStamp* pTimeStamp, UInt32 busNumber, UInt32 frameCount, AudioBufferList* pBufferList)
{
- if (pSemaphore == NULL) {
- MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */
- return MA_INVALID_ARGS;
- }
+ ma_device* pDevice = (ma_device*)pUserData;
+ ma_stream_layout layout;
-#ifdef MA_WIN32
- return ma_semaphore_wait__win32(pSemaphore);
-#endif
-#ifdef MA_POSIX
- return ma_semaphore_wait__posix(pSemaphore);
-#endif
-}
+ MA_ASSERT(pDevice != NULL);
-MA_API ma_result ma_semaphore_release(ma_semaphore* pSemaphore)
-{
- if (pSemaphore == NULL) {
- MA_ASSERT(MA_FALSE); /* Fire an assert so the caller is aware of this bug. */
- return MA_INVALID_ARGS;
+ /*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "INFO: Output Callback: busNumber=%d, frameCount=%d, mNumberBuffers=%d\n", (int)busNumber, (int)frameCount, (int)pBufferList->mNumberBuffers);*/
+
+ /* We need to check whether or not we are outputting interleaved or non-interleaved samples. The way we do this is slightly different for each type. */
+ layout = ma_stream_layout_interleaved;
+ if (pBufferList->mBuffers[0].mNumberChannels != pDevice->playback.internalChannels) {
+ layout = ma_stream_layout_deinterleaved;
}
-#ifdef MA_WIN32
- return ma_semaphore_release__win32(pSemaphore);
-#endif
-#ifdef MA_POSIX
- return ma_semaphore_release__posix(pSemaphore);
-#endif
-}
-#else
-/* MA_NO_THREADING is set which means threading is disabled. Threading is required by some API families. If any of these are enabled we need to throw an error. */
-#ifndef MA_NO_DEVICE_IO
-#error "MA_NO_THREADING cannot be used without MA_NO_DEVICE_IO";
-#endif
-#endif /* MA_NO_THREADING */
+ if (layout == ma_stream_layout_interleaved) {
+ /* For now we can assume everything is interleaved. */
+ UInt32 iBuffer;
+ for (iBuffer = 0; iBuffer < pBufferList->mNumberBuffers; ++iBuffer) {
+ if (pBufferList->mBuffers[iBuffer].mNumberChannels == pDevice->playback.internalChannels) {
+ ma_uint32 frameCountForThisBuffer = pBufferList->mBuffers[iBuffer].mDataByteSize / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
+ if (frameCountForThisBuffer > 0) {
+ ma_device_handle_backend_data_callback(pDevice, pBufferList->mBuffers[iBuffer].mData, NULL, frameCountForThisBuffer);
+ }
+ /*a_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", (int)frameCount, (int)pBufferList->mBuffers[iBuffer].mNumberChannels, (int)pBufferList->mBuffers[iBuffer].mDataByteSize);*/
+ } else {
+ /*
+ This case is where the number of channels in the output buffer do not match our internal channels. It could mean that it's
+ not interleaved, in which case we can't handle right now since miniaudio does not yet support non-interleaved streams. We just
+ output silence here.
+ */
+ MA_ZERO_MEMORY(pBufferList->mBuffers[iBuffer].mData, pBufferList->mBuffers[iBuffer].mDataByteSize);
+ /*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " WARNING: Outputting silence. frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", (int)frameCount, (int)pBufferList->mBuffers[iBuffer].mNumberChannels, (int)pBufferList->mBuffers[iBuffer].mDataByteSize);*/
+ }
+ }
+ } else {
+ /* This is the deinterleaved case. We need to update each buffer in groups of internalChannels. This assumes each buffer is the same size. */
+ MA_ASSERT(pDevice->playback.internalChannels <= MA_MAX_CHANNELS); /* This should heve been validated at initialization time. */
+ /*
+ For safety we'll check that the internal channels is a multiple of the buffer count. If it's not it means something
+ very strange has happened and we're not going to support it.
+ */
+ if ((pBufferList->mNumberBuffers % pDevice->playback.internalChannels) == 0) {
+ ma_uint8 tempBuffer[4096];
+ UInt32 iBuffer;
-/************************************************************************************************************************************************************
-*************************************************************************************************************************************************************
+ for (iBuffer = 0; iBuffer < pBufferList->mNumberBuffers; iBuffer += pDevice->playback.internalChannels) {
+ ma_uint32 frameCountPerBuffer = pBufferList->mBuffers[iBuffer].mDataByteSize / ma_get_bytes_per_sample(pDevice->playback.internalFormat);
+ ma_uint32 framesRemaining = frameCountPerBuffer;
-DEVICE I/O
-==========
+ while (framesRemaining > 0) {
+ void* ppDeinterleavedBuffers[MA_MAX_CHANNELS];
+ ma_uint32 iChannel;
+ ma_uint32 framesToRead = sizeof(tempBuffer) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
+ if (framesToRead > framesRemaining) {
+ framesToRead = framesRemaining;
+ }
-*************************************************************************************************************************************************************
-************************************************************************************************************************************************************/
-#ifndef MA_NO_DEVICE_IO
-#ifdef MA_WIN32
- #include
- #include
- #include
-#endif
+ ma_device_handle_backend_data_callback(pDevice, tempBuffer, NULL, framesToRead);
-#if defined(MA_APPLE) && (__MAC_OS_X_VERSION_MIN_REQUIRED < 101200)
- #include /* For mach_absolute_time() */
-#endif
+ for (iChannel = 0; iChannel < pDevice->playback.internalChannels; ++iChannel) {
+ ppDeinterleavedBuffers[iChannel] = (void*)ma_offset_ptr(pBufferList->mBuffers[iBuffer+iChannel].mData, (frameCountPerBuffer - framesRemaining) * ma_get_bytes_per_sample(pDevice->playback.internalFormat));
+ }
-#ifdef MA_POSIX
- #include
- #include
- #include
-#endif
+ ma_deinterleave_pcm_frames(pDevice->playback.internalFormat, pDevice->playback.internalChannels, framesToRead, tempBuffer, ppDeinterleavedBuffers);
-/*
-Unfortunately using runtime linking for pthreads causes problems. This has occurred for me when testing on FreeBSD. When
-using runtime linking, deadlocks can occur (for me it happens when loading data from fread()). It turns out that doing
-compile-time linking fixes this. I'm not sure why this happens, but the safest way I can think of to fix this is to simply
-disable runtime linking by default. To enable runtime linking, #define this before the implementation of this file. I am
-not officially supporting this, but I'm leaving it here in case it's useful for somebody, somewhere.
-*/
-/*#define MA_USE_RUNTIME_LINKING_FOR_PTHREAD*/
+ framesRemaining -= framesToRead;
+ }
+ }
+ }
+ }
-/* Disable run-time linking on certain backends. */
-#ifndef MA_NO_RUNTIME_LINKING
- #if defined(MA_EMSCRIPTEN)
- #define MA_NO_RUNTIME_LINKING
- #endif
-#endif
+ (void)pActionFlags;
+ (void)pTimeStamp;
+ (void)busNumber;
+ (void)frameCount;
+ return noErr;
+}
-MA_API void ma_device_info_add_native_data_format(ma_device_info* pDeviceInfo, ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 flags)
+static OSStatus ma_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFlags* pActionFlags, const AudioTimeStamp* pTimeStamp, UInt32 busNumber, UInt32 frameCount, AudioBufferList* pUnusedBufferList)
{
- if (pDeviceInfo == NULL) {
- return;
- }
+ ma_device* pDevice = (ma_device*)pUserData;
+ AudioBufferList* pRenderedBufferList;
+ ma_result result;
+ ma_stream_layout layout;
+ ma_uint32 iBuffer;
+ OSStatus status;
- if (pDeviceInfo->nativeDataFormatCount < ma_countof(pDeviceInfo->nativeDataFormats)) {
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = format;
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = channels;
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = sampleRate;
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = flags;
- pDeviceInfo->nativeDataFormatCount += 1;
+ MA_ASSERT(pDevice != NULL);
+
+ pRenderedBufferList = (AudioBufferList*)pDevice->coreaudio.pAudioBufferList;
+ MA_ASSERT(pRenderedBufferList);
+
+ /* We need to check whether or not we are outputting interleaved or non-interleaved samples. The way we do this is slightly different for each type. */
+ layout = ma_stream_layout_interleaved;
+ if (pRenderedBufferList->mBuffers[0].mNumberChannels != pDevice->capture.internalChannels) {
+ layout = ma_stream_layout_deinterleaved;
}
-}
+ /*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "INFO: Input Callback: busNumber=%d, frameCount=%d, mNumberBuffers=%d\n", (int)busNumber, (int)frameCount, (int)pRenderedBufferList->mNumberBuffers);*/
-MA_API const char* ma_get_backend_name(ma_backend backend)
-{
- switch (backend)
- {
- case ma_backend_wasapi: return "WASAPI";
- case ma_backend_dsound: return "DirectSound";
- case ma_backend_winmm: return "WinMM";
- case ma_backend_coreaudio: return "Core Audio";
- case ma_backend_sndio: return "sndio";
- case ma_backend_audio4: return "audio(4)";
- case ma_backend_oss: return "OSS";
- case ma_backend_pulseaudio: return "PulseAudio";
- case ma_backend_alsa: return "ALSA";
- case ma_backend_jack: return "JACK";
- case ma_backend_aaudio: return "AAudio";
- case ma_backend_opensl: return "OpenSL|ES";
- case ma_backend_webaudio: return "Web Audio";
- case ma_backend_custom: return "Custom";
- case ma_backend_null: return "Null";
- default: return "Unknown";
+ /*
+ There has been a situation reported where frame count passed into this function is greater than the capacity of
+ our capture buffer. There doesn't seem to be a reliable way to determine what the maximum frame count will be,
+ so we need to instead resort to dynamically reallocating our buffer to ensure it's large enough to capture the
+ number of frames requested by this callback.
+ */
+ result = ma_device_realloc_AudioBufferList__coreaudio(pDevice, frameCount, pDevice->capture.internalFormat, pDevice->capture.internalChannels, layout);
+ if (result != MA_SUCCESS) {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "Failed to allocate AudioBufferList for capture.\n");
+ return noErr;
}
-}
+
+ pRenderedBufferList = (AudioBufferList*)pDevice->coreaudio.pAudioBufferList;
+ MA_ASSERT(pRenderedBufferList);
-MA_API ma_bool32 ma_is_backend_enabled(ma_backend backend)
-{
/*
- This looks a little bit gross, but we want all backends to be included in the switch to avoid warnings on some compilers
- about some enums not being handled by the switch statement.
+ When you call AudioUnitRender(), Core Audio tries to be helpful by setting the mDataByteSize to the number of bytes
+ that were actually rendered. The problem with this is that the next call can fail with -50 due to the size no longer
+ being set to the capacity of the buffer, but instead the size in bytes of the previous render. This will cause a
+ problem when a future call to this callback specifies a larger number of frames.
+
+ To work around this we need to explicitly set the size of each buffer to their respective size in bytes.
*/
- switch (backend)
- {
- case ma_backend_wasapi:
- #if defined(MA_HAS_WASAPI)
- return MA_TRUE;
- #else
- return MA_FALSE;
- #endif
- case ma_backend_dsound:
- #if defined(MA_HAS_DSOUND)
- return MA_TRUE;
- #else
- return MA_FALSE;
- #endif
- case ma_backend_winmm:
- #if defined(MA_HAS_WINMM)
- return MA_TRUE;
- #else
- return MA_FALSE;
- #endif
- case ma_backend_coreaudio:
- #if defined(MA_HAS_COREAUDIO)
- return MA_TRUE;
- #else
- return MA_FALSE;
- #endif
- case ma_backend_sndio:
- #if defined(MA_HAS_SNDIO)
- return MA_TRUE;
- #else
- return MA_FALSE;
- #endif
- case ma_backend_audio4:
- #if defined(MA_HAS_AUDIO4)
- return MA_TRUE;
- #else
- return MA_FALSE;
- #endif
- case ma_backend_oss:
- #if defined(MA_HAS_OSS)
- return MA_TRUE;
- #else
- return MA_FALSE;
- #endif
- case ma_backend_pulseaudio:
- #if defined(MA_HAS_PULSEAUDIO)
- return MA_TRUE;
- #else
- return MA_FALSE;
- #endif
- case ma_backend_alsa:
- #if defined(MA_HAS_ALSA)
- return MA_TRUE;
- #else
- return MA_FALSE;
- #endif
- case ma_backend_jack:
- #if defined(MA_HAS_JACK)
- return MA_TRUE;
- #else
- return MA_FALSE;
- #endif
- case ma_backend_aaudio:
- #if defined(MA_HAS_AAUDIO)
- return MA_TRUE;
- #else
- return MA_FALSE;
- #endif
- case ma_backend_opensl:
- #if defined(MA_HAS_OPENSL)
- return MA_TRUE;
- #else
- return MA_FALSE;
- #endif
- case ma_backend_webaudio:
- #if defined(MA_HAS_WEBAUDIO)
- return MA_TRUE;
- #else
- return MA_FALSE;
- #endif
- case ma_backend_custom:
- #if defined(MA_HAS_CUSTOM)
- return MA_TRUE;
- #else
- return MA_FALSE;
- #endif
- case ma_backend_null:
- #if defined(MA_HAS_NULL)
- return MA_TRUE;
- #else
- return MA_FALSE;
- #endif
+ for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; ++iBuffer) {
+ pRenderedBufferList->mBuffers[iBuffer].mDataByteSize = pDevice->coreaudio.audioBufferCapInFrames * ma_get_bytes_per_sample(pDevice->capture.internalFormat) * pRenderedBufferList->mBuffers[iBuffer].mNumberChannels;
+ }
+
+ status = ((ma_AudioUnitRender_proc)pDevice->pContext->coreaudio.AudioUnitRender)((AudioUnit)pDevice->coreaudio.audioUnitCapture, pActionFlags, pTimeStamp, busNumber, frameCount, pRenderedBufferList);
+ if (status != noErr) {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " ERROR: AudioUnitRender() failed with %d.\n", (int)status);
+ return status;
+ }
+
+ if (layout == ma_stream_layout_interleaved) {
+ for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; ++iBuffer) {
+ if (pRenderedBufferList->mBuffers[iBuffer].mNumberChannels == pDevice->capture.internalChannels) {
+ ma_device_handle_backend_data_callback(pDevice, NULL, pRenderedBufferList->mBuffers[iBuffer].mData, frameCount);
+ /*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " mDataByteSize=%d.\n", (int)pRenderedBufferList->mBuffers[iBuffer].mDataByteSize);*/
+ } else {
+ /*
+ This case is where the number of channels in the output buffer do not match our internal channels. It could mean that it's
+ not interleaved, in which case we can't handle right now since miniaudio does not yet support non-interleaved streams.
+ */
+ ma_uint8 silentBuffer[4096];
+ ma_uint32 framesRemaining;
+
+ MA_ZERO_MEMORY(silentBuffer, sizeof(silentBuffer));
+
+ framesRemaining = frameCount;
+ while (framesRemaining > 0) {
+ ma_uint32 framesToSend = sizeof(silentBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
+ if (framesToSend > framesRemaining) {
+ framesToSend = framesRemaining;
+ }
- default: return MA_FALSE;
+ ma_device_handle_backend_data_callback(pDevice, NULL, silentBuffer, framesToSend);
+
+ framesRemaining -= framesToSend;
+ }
+
+ /*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " WARNING: Outputting silence. frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", (int)frameCount, (int)pRenderedBufferList->mBuffers[iBuffer].mNumberChannels, (int)pRenderedBufferList->mBuffers[iBuffer].mDataByteSize);*/
+ }
+ }
+ } else {
+ /* This is the deinterleaved case. We need to interleave the audio data before sending it to the client. This assumes each buffer is the same size. */
+ MA_ASSERT(pDevice->capture.internalChannels <= MA_MAX_CHANNELS); /* This should have been validated at initialization time. */
+
+ /*
+ For safety we'll check that the internal channels is a multiple of the buffer count. If it's not it means something
+ very strange has happened and we're not going to support it.
+ */
+ if ((pRenderedBufferList->mNumberBuffers % pDevice->capture.internalChannels) == 0) {
+ ma_uint8 tempBuffer[4096];
+ for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; iBuffer += pDevice->capture.internalChannels) {
+ ma_uint32 framesRemaining = frameCount;
+ while (framesRemaining > 0) {
+ void* ppDeinterleavedBuffers[MA_MAX_CHANNELS];
+ ma_uint32 iChannel;
+ ma_uint32 framesToSend = sizeof(tempBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
+ if (framesToSend > framesRemaining) {
+ framesToSend = framesRemaining;
+ }
+
+ for (iChannel = 0; iChannel < pDevice->capture.internalChannels; ++iChannel) {
+ ppDeinterleavedBuffers[iChannel] = (void*)ma_offset_ptr(pRenderedBufferList->mBuffers[iBuffer+iChannel].mData, (frameCount - framesRemaining) * ma_get_bytes_per_sample(pDevice->capture.internalFormat));
+ }
+
+ ma_interleave_pcm_frames(pDevice->capture.internalFormat, pDevice->capture.internalChannels, framesToSend, (const void**)ppDeinterleavedBuffers, tempBuffer);
+ ma_device_handle_backend_data_callback(pDevice, NULL, tempBuffer, framesToSend);
+
+ framesRemaining -= framesToSend;
+ }
+ }
+ }
}
+
+ (void)pActionFlags;
+ (void)pTimeStamp;
+ (void)busNumber;
+ (void)frameCount;
+ (void)pUnusedBufferList;
+
+ return noErr;
}
-MA_API ma_result ma_get_enabled_backends(ma_backend* pBackends, size_t backendCap, size_t* pBackendCount)
+static void on_start_stop__coreaudio(void* pUserData, AudioUnit audioUnit, AudioUnitPropertyID propertyID, AudioUnitScope scope, AudioUnitElement element)
{
- size_t backendCount;
- size_t iBackend;
- ma_result result = MA_SUCCESS;
+ ma_device* pDevice = (ma_device*)pUserData;
+ MA_ASSERT(pDevice != NULL);
- if (pBackendCount == NULL) {
- return MA_INVALID_ARGS;
+ /* Don't do anything if it looks like we're just reinitializing due to a device switch. */
+ if (((audioUnit == pDevice->coreaudio.audioUnitPlayback) && pDevice->coreaudio.isSwitchingPlaybackDevice) ||
+ ((audioUnit == pDevice->coreaudio.audioUnitCapture) && pDevice->coreaudio.isSwitchingCaptureDevice)) {
+ return;
}
- backendCount = 0;
+ /*
+ There's been a report of a deadlock here when triggered by ma_device_uninit(). It looks like
+ AudioUnitGetProprty (called below) and AudioComponentInstanceDispose (called in ma_device_uninit)
+ can try waiting on the same lock. I'm going to try working around this by not calling any Core
+ Audio APIs in the callback when the device has been stopped or uninitialized.
+ */
+ if (ma_device_get_state(pDevice) == ma_device_state_uninitialized || ma_device_get_state(pDevice) == ma_device_state_stopping || ma_device_get_state(pDevice) == ma_device_state_stopped) {
+ ma_device__on_notification_stopped(pDevice);
+ } else {
+ UInt32 isRunning;
+ UInt32 isRunningSize = sizeof(isRunning);
+ OSStatus status = ((ma_AudioUnitGetProperty_proc)pDevice->pContext->coreaudio.AudioUnitGetProperty)(audioUnit, kAudioOutputUnitProperty_IsRunning, scope, element, &isRunning, &isRunningSize);
+ if (status != noErr) {
+ goto done; /* Don't really know what to do in this case... just ignore it, I suppose... */
+ }
- for (iBackend = 0; iBackend <= ma_backend_null; iBackend += 1) {
- ma_backend backend = (ma_backend)iBackend;
+ if (!isRunning) {
+ /*
+ The stop event is a bit annoying in Core Audio because it will be called when we automatically switch the default device. Some scenarios to consider:
- if (ma_is_backend_enabled(backend)) {
- /* The backend is enabled. Try adding it to the list. If there's no room, MA_NO_SPACE needs to be returned. */
- if (backendCount == backendCap) {
- result = MA_NO_SPACE;
- break;
- } else {
- pBackends[backendCount] = backend;
- backendCount += 1;
+ 1) When the device is unplugged, this will be called _before_ the default device change notification.
+ 2) When the device is changed via the default device change notification, this will be called _after_ the switch.
+
+ For case #1, we just check if there's a new default device available. If so, we just ignore the stop event. For case #2 we check a flag.
+ */
+ if (((audioUnit == pDevice->coreaudio.audioUnitPlayback) && pDevice->coreaudio.isDefaultPlaybackDevice) ||
+ ((audioUnit == pDevice->coreaudio.audioUnitCapture) && pDevice->coreaudio.isDefaultCaptureDevice)) {
+ /*
+ It looks like the device is switching through an external event, such as the user unplugging the device or changing the default device
+ via the operating system's sound settings. If we're re-initializing the device, we just terminate because we want the stopping of the
+ device to be seamless to the client (we don't want them receiving the stopped event and thinking that the device has stopped when it
+ hasn't!).
+ */
+ if (((audioUnit == pDevice->coreaudio.audioUnitPlayback) && pDevice->coreaudio.isSwitchingPlaybackDevice) ||
+ ((audioUnit == pDevice->coreaudio.audioUnitCapture) && pDevice->coreaudio.isSwitchingCaptureDevice)) {
+ goto done;
+ }
+
+ /*
+ Getting here means the device is not reinitializing which means it may have been unplugged. From what I can see, it looks like Core Audio
+ will try switching to the new default device seamlessly. We need to somehow find a way to determine whether or not Core Audio will most
+ likely be successful in switching to the new device.
+
+ TODO: Try to predict if Core Audio will switch devices. If not, the stopped callback needs to be posted.
+ */
+ goto done;
}
+
+ /* Getting here means we need to stop the device. */
+ ma_device__on_notification_stopped(pDevice);
}
}
- if (pBackendCount != NULL) {
- *pBackendCount = backendCount;
- }
+ (void)propertyID; /* Unused. */
- return result;
+done:
+ /* Always signal the stop event. It's possible for the "else" case to get hit which can happen during an interruption. */
+ ma_event_signal(&pDevice->coreaudio.stopEvent);
}
-MA_API ma_bool32 ma_is_loopback_supported(ma_backend backend)
+#if defined(MA_APPLE_DESKTOP)
+static ma_spinlock g_DeviceTrackingInitLock_CoreAudio = 0; /* A spinlock for mutal exclusion of the init/uninit of the global tracking data. Initialization to 0 is what we need. */
+static ma_uint32 g_DeviceTrackingInitCounter_CoreAudio = 0;
+static ma_mutex g_DeviceTrackingMutex_CoreAudio;
+static ma_device** g_ppTrackedDevices_CoreAudio = NULL;
+static ma_uint32 g_TrackedDeviceCap_CoreAudio = 0;
+static ma_uint32 g_TrackedDeviceCount_CoreAudio = 0;
+
+static OSStatus ma_default_device_changed__coreaudio(AudioObjectID objectID, UInt32 addressCount, const AudioObjectPropertyAddress* pAddresses, void* pUserData)
{
- switch (backend)
- {
- case ma_backend_wasapi: return MA_TRUE;
- case ma_backend_dsound: return MA_FALSE;
- case ma_backend_winmm: return MA_FALSE;
- case ma_backend_coreaudio: return MA_FALSE;
- case ma_backend_sndio: return MA_FALSE;
- case ma_backend_audio4: return MA_FALSE;
- case ma_backend_oss: return MA_FALSE;
- case ma_backend_pulseaudio: return MA_FALSE;
- case ma_backend_alsa: return MA_FALSE;
- case ma_backend_jack: return MA_FALSE;
- case ma_backend_aaudio: return MA_FALSE;
- case ma_backend_opensl: return MA_FALSE;
- case ma_backend_webaudio: return MA_FALSE;
- case ma_backend_custom: return MA_FALSE; /* <-- Will depend on the implementation of the backend. */
- case ma_backend_null: return MA_FALSE;
- default: return MA_FALSE;
+ ma_device_type deviceType;
+
+ /* Not sure if I really need to check this, but it makes me feel better. */
+ if (addressCount == 0) {
+ return noErr;
}
-}
+ if (pAddresses[0].mSelector == kAudioHardwarePropertyDefaultOutputDevice) {
+ deviceType = ma_device_type_playback;
+ } else if (pAddresses[0].mSelector == kAudioHardwarePropertyDefaultInputDevice) {
+ deviceType = ma_device_type_capture;
+ } else {
+ return noErr; /* Should never hit this. */
+ }
+ ma_mutex_lock(&g_DeviceTrackingMutex_CoreAudio);
+ {
+ ma_uint32 iDevice;
+ for (iDevice = 0; iDevice < g_TrackedDeviceCount_CoreAudio; iDevice += 1) {
+ ma_result reinitResult;
+ ma_device* pDevice;
-#ifdef MA_WIN32
-/* WASAPI error codes. */
-#define MA_AUDCLNT_E_NOT_INITIALIZED ((HRESULT)0x88890001)
-#define MA_AUDCLNT_E_ALREADY_INITIALIZED ((HRESULT)0x88890002)
-#define MA_AUDCLNT_E_WRONG_ENDPOINT_TYPE ((HRESULT)0x88890003)
-#define MA_AUDCLNT_E_DEVICE_INVALIDATED ((HRESULT)0x88890004)
-#define MA_AUDCLNT_E_NOT_STOPPED ((HRESULT)0x88890005)
-#define MA_AUDCLNT_E_BUFFER_TOO_LARGE ((HRESULT)0x88890006)
-#define MA_AUDCLNT_E_OUT_OF_ORDER ((HRESULT)0x88890007)
-#define MA_AUDCLNT_E_UNSUPPORTED_FORMAT ((HRESULT)0x88890008)
-#define MA_AUDCLNT_E_INVALID_SIZE ((HRESULT)0x88890009)
-#define MA_AUDCLNT_E_DEVICE_IN_USE ((HRESULT)0x8889000A)
-#define MA_AUDCLNT_E_BUFFER_OPERATION_PENDING ((HRESULT)0x8889000B)
-#define MA_AUDCLNT_E_THREAD_NOT_REGISTERED ((HRESULT)0x8889000C)
-#define MA_AUDCLNT_E_NO_SINGLE_PROCESS ((HRESULT)0x8889000D)
-#define MA_AUDCLNT_E_EXCLUSIVE_MODE_NOT_ALLOWED ((HRESULT)0x8889000E)
-#define MA_AUDCLNT_E_ENDPOINT_CREATE_FAILED ((HRESULT)0x8889000F)
-#define MA_AUDCLNT_E_SERVICE_NOT_RUNNING ((HRESULT)0x88890010)
-#define MA_AUDCLNT_E_EVENTHANDLE_NOT_EXPECTED ((HRESULT)0x88890011)
-#define MA_AUDCLNT_E_EXCLUSIVE_MODE_ONLY ((HRESULT)0x88890012)
-#define MA_AUDCLNT_E_BUFDURATION_PERIOD_NOT_EQUAL ((HRESULT)0x88890013)
-#define MA_AUDCLNT_E_EVENTHANDLE_NOT_SET ((HRESULT)0x88890014)
-#define MA_AUDCLNT_E_INCORRECT_BUFFER_SIZE ((HRESULT)0x88890015)
-#define MA_AUDCLNT_E_BUFFER_SIZE_ERROR ((HRESULT)0x88890016)
-#define MA_AUDCLNT_E_CPUUSAGE_EXCEEDED ((HRESULT)0x88890017)
-#define MA_AUDCLNT_E_BUFFER_ERROR ((HRESULT)0x88890018)
-#define MA_AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED ((HRESULT)0x88890019)
-#define MA_AUDCLNT_E_INVALID_DEVICE_PERIOD ((HRESULT)0x88890020)
-#define MA_AUDCLNT_E_INVALID_STREAM_FLAG ((HRESULT)0x88890021)
-#define MA_AUDCLNT_E_ENDPOINT_OFFLOAD_NOT_CAPABLE ((HRESULT)0x88890022)
-#define MA_AUDCLNT_E_OUT_OF_OFFLOAD_RESOURCES ((HRESULT)0x88890023)
-#define MA_AUDCLNT_E_OFFLOAD_MODE_ONLY ((HRESULT)0x88890024)
-#define MA_AUDCLNT_E_NONOFFLOAD_MODE_ONLY ((HRESULT)0x88890025)
-#define MA_AUDCLNT_E_RESOURCES_INVALIDATED ((HRESULT)0x88890026)
-#define MA_AUDCLNT_E_RAW_MODE_UNSUPPORTED ((HRESULT)0x88890027)
-#define MA_AUDCLNT_E_ENGINE_PERIODICITY_LOCKED ((HRESULT)0x88890028)
-#define MA_AUDCLNT_E_ENGINE_FORMAT_LOCKED ((HRESULT)0x88890029)
-#define MA_AUDCLNT_E_HEADTRACKING_ENABLED ((HRESULT)0x88890030)
-#define MA_AUDCLNT_E_HEADTRACKING_UNSUPPORTED ((HRESULT)0x88890040)
-#define MA_AUDCLNT_S_BUFFER_EMPTY ((HRESULT)0x08890001)
-#define MA_AUDCLNT_S_THREAD_ALREADY_REGISTERED ((HRESULT)0x08890002)
-#define MA_AUDCLNT_S_POSITION_STALLED ((HRESULT)0x08890003)
+ pDevice = g_ppTrackedDevices_CoreAudio[iDevice];
+ if (pDevice->type == deviceType || pDevice->type == ma_device_type_duplex) {
+ if (deviceType == ma_device_type_playback) {
+ pDevice->coreaudio.isSwitchingPlaybackDevice = MA_TRUE;
+ reinitResult = ma_device_reinit_internal__coreaudio(pDevice, deviceType, MA_TRUE);
+ pDevice->coreaudio.isSwitchingPlaybackDevice = MA_FALSE;
+ } else {
+ pDevice->coreaudio.isSwitchingCaptureDevice = MA_TRUE;
+ reinitResult = ma_device_reinit_internal__coreaudio(pDevice, deviceType, MA_TRUE);
+ pDevice->coreaudio.isSwitchingCaptureDevice = MA_FALSE;
+ }
-#define MA_DS_OK ((HRESULT)0)
-#define MA_DS_NO_VIRTUALIZATION ((HRESULT)0x0878000A)
-#define MA_DSERR_ALLOCATED ((HRESULT)0x8878000A)
-#define MA_DSERR_CONTROLUNAVAIL ((HRESULT)0x8878001E)
-#define MA_DSERR_INVALIDPARAM ((HRESULT)0x80070057) /*E_INVALIDARG*/
-#define MA_DSERR_INVALIDCALL ((HRESULT)0x88780032)
-#define MA_DSERR_GENERIC ((HRESULT)0x80004005) /*E_FAIL*/
-#define MA_DSERR_PRIOLEVELNEEDED ((HRESULT)0x88780046)
-#define MA_DSERR_OUTOFMEMORY ((HRESULT)0x8007000E) /*E_OUTOFMEMORY*/
-#define MA_DSERR_BADFORMAT ((HRESULT)0x88780064)
-#define MA_DSERR_UNSUPPORTED ((HRESULT)0x80004001) /*E_NOTIMPL*/
-#define MA_DSERR_NODRIVER ((HRESULT)0x88780078)
-#define MA_DSERR_ALREADYINITIALIZED ((HRESULT)0x88780082)
-#define MA_DSERR_NOAGGREGATION ((HRESULT)0x80040110) /*CLASS_E_NOAGGREGATION*/
-#define MA_DSERR_BUFFERLOST ((HRESULT)0x88780096)
-#define MA_DSERR_OTHERAPPHASPRIO ((HRESULT)0x887800A0)
-#define MA_DSERR_UNINITIALIZED ((HRESULT)0x887800AA)
-#define MA_DSERR_NOINTERFACE ((HRESULT)0x80004002) /*E_NOINTERFACE*/
-#define MA_DSERR_ACCESSDENIED ((HRESULT)0x80070005) /*E_ACCESSDENIED*/
-#define MA_DSERR_BUFFERTOOSMALL ((HRESULT)0x887800B4)
-#define MA_DSERR_DS8_REQUIRED ((HRESULT)0x887800BE)
-#define MA_DSERR_SENDLOOP ((HRESULT)0x887800C8)
-#define MA_DSERR_BADSENDBUFFERGUID ((HRESULT)0x887800D2)
-#define MA_DSERR_OBJECTNOTFOUND ((HRESULT)0x88781161)
-#define MA_DSERR_FXUNAVAILABLE ((HRESULT)0x887800DC)
+ if (reinitResult == MA_SUCCESS) {
+ ma_device__post_init_setup(pDevice, deviceType);
-static ma_result ma_result_from_HRESULT(HRESULT hr)
+ /* Restart the device if required. If this fails we need to stop the device entirely. */
+ if (ma_device_get_state(pDevice) == ma_device_state_started) {
+ OSStatus status;
+ if (deviceType == ma_device_type_playback) {
+ status = ((ma_AudioOutputUnitStart_proc)pDevice->pContext->coreaudio.AudioOutputUnitStart)((AudioUnit)pDevice->coreaudio.audioUnitPlayback);
+ if (status != noErr) {
+ if (pDevice->type == ma_device_type_duplex) {
+ ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitCapture);
+ }
+ ma_device__set_state(pDevice, ma_device_state_stopped);
+ }
+ } else if (deviceType == ma_device_type_capture) {
+ status = ((ma_AudioOutputUnitStart_proc)pDevice->pContext->coreaudio.AudioOutputUnitStart)((AudioUnit)pDevice->coreaudio.audioUnitCapture);
+ if (status != noErr) {
+ if (pDevice->type == ma_device_type_duplex) {
+ ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitPlayback);
+ }
+ ma_device__set_state(pDevice, ma_device_state_stopped);
+ }
+ }
+ }
+
+ ma_device__on_notification_rerouted(pDevice);
+ }
+ }
+ }
+ }
+ ma_mutex_unlock(&g_DeviceTrackingMutex_CoreAudio);
+
+ /* Unused parameters. */
+ (void)objectID;
+ (void)pUserData;
+
+ return noErr;
+}
+
+static ma_result ma_context__init_device_tracking__coreaudio(ma_context* pContext)
{
- switch (hr)
+ MA_ASSERT(pContext != NULL);
+
+ ma_spinlock_lock(&g_DeviceTrackingInitLock_CoreAudio);
{
- case NOERROR: return MA_SUCCESS;
- /*case S_OK: return MA_SUCCESS;*/
+ /* Don't do anything if we've already initializd device tracking. */
+ if (g_DeviceTrackingInitCounter_CoreAudio == 0) {
+ AudioObjectPropertyAddress propAddress;
+ propAddress.mScope = kAudioObjectPropertyScopeGlobal;
+ propAddress.mElement = kAudioObjectPropertyElementMaster;
- case E_POINTER: return MA_INVALID_ARGS;
- case E_UNEXPECTED: return MA_ERROR;
- case E_NOTIMPL: return MA_NOT_IMPLEMENTED;
- case E_OUTOFMEMORY: return MA_OUT_OF_MEMORY;
- case E_INVALIDARG: return MA_INVALID_ARGS;
- case E_NOINTERFACE: return MA_API_NOT_FOUND;
- case E_HANDLE: return MA_INVALID_ARGS;
- case E_ABORT: return MA_ERROR;
- case E_FAIL: return MA_ERROR;
- case E_ACCESSDENIED: return MA_ACCESS_DENIED;
+ ma_mutex_init(&g_DeviceTrackingMutex_CoreAudio);
- /* WASAPI */
- case MA_AUDCLNT_E_NOT_INITIALIZED: return MA_DEVICE_NOT_INITIALIZED;
- case MA_AUDCLNT_E_ALREADY_INITIALIZED: return MA_DEVICE_ALREADY_INITIALIZED;
- case MA_AUDCLNT_E_WRONG_ENDPOINT_TYPE: return MA_INVALID_ARGS;
- case MA_AUDCLNT_E_DEVICE_INVALIDATED: return MA_UNAVAILABLE;
- case MA_AUDCLNT_E_NOT_STOPPED: return MA_DEVICE_NOT_STOPPED;
- case MA_AUDCLNT_E_BUFFER_TOO_LARGE: return MA_TOO_BIG;
- case MA_AUDCLNT_E_OUT_OF_ORDER: return MA_INVALID_OPERATION;
- case MA_AUDCLNT_E_UNSUPPORTED_FORMAT: return MA_FORMAT_NOT_SUPPORTED;
- case MA_AUDCLNT_E_INVALID_SIZE: return MA_INVALID_ARGS;
- case MA_AUDCLNT_E_DEVICE_IN_USE: return MA_BUSY;
- case MA_AUDCLNT_E_BUFFER_OPERATION_PENDING: return MA_INVALID_OPERATION;
- case MA_AUDCLNT_E_THREAD_NOT_REGISTERED: return MA_DOES_NOT_EXIST;
- case MA_AUDCLNT_E_NO_SINGLE_PROCESS: return MA_INVALID_OPERATION;
- case MA_AUDCLNT_E_EXCLUSIVE_MODE_NOT_ALLOWED: return MA_SHARE_MODE_NOT_SUPPORTED;
- case MA_AUDCLNT_E_ENDPOINT_CREATE_FAILED: return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
- case MA_AUDCLNT_E_SERVICE_NOT_RUNNING: return MA_NOT_CONNECTED;
- case MA_AUDCLNT_E_EVENTHANDLE_NOT_EXPECTED: return MA_INVALID_ARGS;
- case MA_AUDCLNT_E_EXCLUSIVE_MODE_ONLY: return MA_SHARE_MODE_NOT_SUPPORTED;
- case MA_AUDCLNT_E_BUFDURATION_PERIOD_NOT_EQUAL: return MA_INVALID_ARGS;
- case MA_AUDCLNT_E_EVENTHANDLE_NOT_SET: return MA_INVALID_ARGS;
- case MA_AUDCLNT_E_INCORRECT_BUFFER_SIZE: return MA_INVALID_ARGS;
- case MA_AUDCLNT_E_BUFFER_SIZE_ERROR: return MA_INVALID_ARGS;
- case MA_AUDCLNT_E_CPUUSAGE_EXCEEDED: return MA_ERROR;
- case MA_AUDCLNT_E_BUFFER_ERROR: return MA_ERROR;
- case MA_AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED: return MA_INVALID_ARGS;
- case MA_AUDCLNT_E_INVALID_DEVICE_PERIOD: return MA_INVALID_ARGS;
- case MA_AUDCLNT_E_INVALID_STREAM_FLAG: return MA_INVALID_ARGS;
- case MA_AUDCLNT_E_ENDPOINT_OFFLOAD_NOT_CAPABLE: return MA_INVALID_OPERATION;
- case MA_AUDCLNT_E_OUT_OF_OFFLOAD_RESOURCES: return MA_OUT_OF_MEMORY;
- case MA_AUDCLNT_E_OFFLOAD_MODE_ONLY: return MA_INVALID_OPERATION;
- case MA_AUDCLNT_E_NONOFFLOAD_MODE_ONLY: return MA_INVALID_OPERATION;
- case MA_AUDCLNT_E_RESOURCES_INVALIDATED: return MA_INVALID_DATA;
- case MA_AUDCLNT_E_RAW_MODE_UNSUPPORTED: return MA_INVALID_OPERATION;
- case MA_AUDCLNT_E_ENGINE_PERIODICITY_LOCKED: return MA_INVALID_OPERATION;
- case MA_AUDCLNT_E_ENGINE_FORMAT_LOCKED: return MA_INVALID_OPERATION;
- case MA_AUDCLNT_E_HEADTRACKING_ENABLED: return MA_INVALID_OPERATION;
- case MA_AUDCLNT_E_HEADTRACKING_UNSUPPORTED: return MA_INVALID_OPERATION;
- case MA_AUDCLNT_S_BUFFER_EMPTY: return MA_NO_SPACE;
- case MA_AUDCLNT_S_THREAD_ALREADY_REGISTERED: return MA_ALREADY_EXISTS;
- case MA_AUDCLNT_S_POSITION_STALLED: return MA_ERROR;
+ propAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice;
+ ((ma_AudioObjectAddPropertyListener_proc)pContext->coreaudio.AudioObjectAddPropertyListener)(kAudioObjectSystemObject, &propAddress, &ma_default_device_changed__coreaudio, NULL);
- /* DirectSound */
- /*case MA_DS_OK: return MA_SUCCESS;*/ /* S_OK */
- case MA_DS_NO_VIRTUALIZATION: return MA_SUCCESS;
- case MA_DSERR_ALLOCATED: return MA_ALREADY_IN_USE;
- case MA_DSERR_CONTROLUNAVAIL: return MA_INVALID_OPERATION;
- /*case MA_DSERR_INVALIDPARAM: return MA_INVALID_ARGS;*/ /* E_INVALIDARG */
- case MA_DSERR_INVALIDCALL: return MA_INVALID_OPERATION;
- /*case MA_DSERR_GENERIC: return MA_ERROR;*/ /* E_FAIL */
- case MA_DSERR_PRIOLEVELNEEDED: return MA_INVALID_OPERATION;
- /*case MA_DSERR_OUTOFMEMORY: return MA_OUT_OF_MEMORY;*/ /* E_OUTOFMEMORY */
- case MA_DSERR_BADFORMAT: return MA_FORMAT_NOT_SUPPORTED;
- /*case MA_DSERR_UNSUPPORTED: return MA_NOT_IMPLEMENTED;*/ /* E_NOTIMPL */
- case MA_DSERR_NODRIVER: return MA_FAILED_TO_INIT_BACKEND;
- case MA_DSERR_ALREADYINITIALIZED: return MA_DEVICE_ALREADY_INITIALIZED;
- case MA_DSERR_NOAGGREGATION: return MA_ERROR;
- case MA_DSERR_BUFFERLOST: return MA_UNAVAILABLE;
- case MA_DSERR_OTHERAPPHASPRIO: return MA_ACCESS_DENIED;
- case MA_DSERR_UNINITIALIZED: return MA_DEVICE_NOT_INITIALIZED;
- /*case MA_DSERR_NOINTERFACE: return MA_API_NOT_FOUND;*/ /* E_NOINTERFACE */
- /*case MA_DSERR_ACCESSDENIED: return MA_ACCESS_DENIED;*/ /* E_ACCESSDENIED */
- case MA_DSERR_BUFFERTOOSMALL: return MA_NO_SPACE;
- case MA_DSERR_DS8_REQUIRED: return MA_INVALID_OPERATION;
- case MA_DSERR_SENDLOOP: return MA_DEADLOCK;
- case MA_DSERR_BADSENDBUFFERGUID: return MA_INVALID_ARGS;
- case MA_DSERR_OBJECTNOTFOUND: return MA_NO_DEVICE;
- case MA_DSERR_FXUNAVAILABLE: return MA_UNAVAILABLE;
+ propAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
+ ((ma_AudioObjectAddPropertyListener_proc)pContext->coreaudio.AudioObjectAddPropertyListener)(kAudioObjectSystemObject, &propAddress, &ma_default_device_changed__coreaudio, NULL);
- default: return MA_ERROR;
+ }
+ g_DeviceTrackingInitCounter_CoreAudio += 1;
}
+ ma_spinlock_unlock(&g_DeviceTrackingInitLock_CoreAudio);
+
+ return MA_SUCCESS;
}
-typedef HRESULT (WINAPI * MA_PFN_CoInitializeEx)(LPVOID pvReserved, DWORD dwCoInit);
-typedef void (WINAPI * MA_PFN_CoUninitialize)(void);
-typedef HRESULT (WINAPI * MA_PFN_CoCreateInstance)(REFCLSID rclsid, LPUNKNOWN pUnkOuter, DWORD dwClsContext, REFIID riid, LPVOID *ppv);
-typedef void (WINAPI * MA_PFN_CoTaskMemFree)(LPVOID pv);
-typedef HRESULT (WINAPI * MA_PFN_PropVariantClear)(PROPVARIANT *pvar);
-typedef int (WINAPI * MA_PFN_StringFromGUID2)(const GUID* const rguid, LPOLESTR lpsz, int cchMax);
+static ma_result ma_context__uninit_device_tracking__coreaudio(ma_context* pContext)
+{
+ MA_ASSERT(pContext != NULL);
-typedef HWND (WINAPI * MA_PFN_GetForegroundWindow)(void);
-typedef HWND (WINAPI * MA_PFN_GetDesktopWindow)(void);
+ ma_spinlock_lock(&g_DeviceTrackingInitLock_CoreAudio);
+ {
+ if (g_DeviceTrackingInitCounter_CoreAudio > 0)
+ g_DeviceTrackingInitCounter_CoreAudio -= 1;
-/* Microsoft documents these APIs as returning LSTATUS, but the Win32 API shipping with some compilers do not define it. It's just a LONG. */
-typedef LONG (WINAPI * MA_PFN_RegOpenKeyExA)(HKEY hKey, LPCSTR lpSubKey, DWORD ulOptions, REGSAM samDesired, PHKEY phkResult);
-typedef LONG (WINAPI * MA_PFN_RegCloseKey)(HKEY hKey);
-typedef LONG (WINAPI * MA_PFN_RegQueryValueExA)(HKEY hKey, LPCSTR lpValueName, LPDWORD lpReserved, LPDWORD lpType, LPBYTE lpData, LPDWORD lpcbData);
-#endif
+ if (g_DeviceTrackingInitCounter_CoreAudio == 0) {
+ AudioObjectPropertyAddress propAddress;
+ propAddress.mScope = kAudioObjectPropertyScopeGlobal;
+ propAddress.mElement = kAudioObjectPropertyElementMaster;
+ propAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice;
+ ((ma_AudioObjectRemovePropertyListener_proc)pContext->coreaudio.AudioObjectRemovePropertyListener)(kAudioObjectSystemObject, &propAddress, &ma_default_device_changed__coreaudio, NULL);
-#define MA_DEFAULT_PLAYBACK_DEVICE_NAME "Default Playback Device"
-#define MA_DEFAULT_CAPTURE_DEVICE_NAME "Default Capture Device"
+ propAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
+ ((ma_AudioObjectRemovePropertyListener_proc)pContext->coreaudio.AudioObjectRemovePropertyListener)(kAudioObjectSystemObject, &propAddress, &ma_default_device_changed__coreaudio, NULL);
+ /* At this point there should be no tracked devices. If not there's an error somewhere. */
+ if (g_ppTrackedDevices_CoreAudio != NULL) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_WARNING, "You have uninitialized all contexts while an associated device is still active.");
+ ma_spinlock_unlock(&g_DeviceTrackingInitLock_CoreAudio);
+ return MA_INVALID_OPERATION;
+ }
-/* Posts a log message. */
-static void ma_post_log_message(ma_context* pContext, ma_device* pDevice, ma_uint32 logLevel, const char* message)
-{
- if (pContext == NULL) {
- if (pDevice != NULL) {
- pContext = pDevice->pContext;
+ ma_mutex_uninit(&g_DeviceTrackingMutex_CoreAudio);
}
}
+ ma_spinlock_unlock(&g_DeviceTrackingInitLock_CoreAudio);
- if (pContext == NULL) {
- return;
- }
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device__track__coreaudio(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
+
+ ma_mutex_lock(&g_DeviceTrackingMutex_CoreAudio);
+ {
+ /* Allocate memory if required. */
+ if (g_TrackedDeviceCap_CoreAudio <= g_TrackedDeviceCount_CoreAudio) {
+ ma_uint32 newCap;
+ ma_device** ppNewDevices;
- ma_log_post(ma_context_get_log(pContext), logLevel, message); /* <-- This will deal with MA_DEBUG_OUTPUT. */
+ newCap = g_TrackedDeviceCap_CoreAudio * 2;
+ if (newCap == 0) {
+ newCap = 1;
+ }
- /* Legacy. */
-#if defined(MA_LOG_LEVEL)
- if (logLevel <= MA_LOG_LEVEL) {
- ma_log_proc onLog;
+ ppNewDevices = (ma_device**)ma_realloc(g_ppTrackedDevices_CoreAudio, sizeof(*g_ppTrackedDevices_CoreAudio)*newCap, &pDevice->pContext->allocationCallbacks);
+ if (ppNewDevices == NULL) {
+ ma_mutex_unlock(&g_DeviceTrackingMutex_CoreAudio);
+ return MA_OUT_OF_MEMORY;
+ }
- onLog = pContext->logCallback;
- if (onLog) {
- onLog(pContext, pDevice, logLevel, message);
+ g_ppTrackedDevices_CoreAudio = ppNewDevices;
+ g_TrackedDeviceCap_CoreAudio = newCap;
}
+
+ g_ppTrackedDevices_CoreAudio[g_TrackedDeviceCount_CoreAudio] = pDevice;
+ g_TrackedDeviceCount_CoreAudio += 1;
}
-#endif
-}
+ ma_mutex_unlock(&g_DeviceTrackingMutex_CoreAudio);
-/* Posts an log message. Throw a breakpoint in here if you're needing to debug. The return value is always "resultCode". */
-static ma_result ma_context_post_error(ma_context* pContext, ma_device* pDevice, ma_uint32 logLevel, const char* message, ma_result resultCode)
-{
- ma_post_log_message(pContext, pDevice, logLevel, message);
- return resultCode;
+ return MA_SUCCESS;
}
-static ma_result ma_post_error(ma_device* pDevice, ma_uint32 logLevel, const char* message, ma_result resultCode)
+static ma_result ma_device__untrack__coreaudio(ma_device* pDevice)
{
- return ma_context_post_error(ma_device_get_context(pDevice), pDevice, logLevel, message, resultCode);
-}
+ MA_ASSERT(pDevice != NULL);
+ ma_mutex_lock(&g_DeviceTrackingMutex_CoreAudio);
+ {
+ ma_uint32 iDevice;
+ for (iDevice = 0; iDevice < g_TrackedDeviceCount_CoreAudio; iDevice += 1) {
+ if (g_ppTrackedDevices_CoreAudio[iDevice] == pDevice) {
+ /* We've found the device. We now need to remove it from the list. */
+ ma_uint32 jDevice;
+ for (jDevice = iDevice; jDevice < g_TrackedDeviceCount_CoreAudio-1; jDevice += 1) {
+ g_ppTrackedDevices_CoreAudio[jDevice] = g_ppTrackedDevices_CoreAudio[jDevice+1];
+ }
+ g_TrackedDeviceCount_CoreAudio -= 1;
+ /* If there's nothing else in the list we need to free memory. */
+ if (g_TrackedDeviceCount_CoreAudio == 0) {
+ ma_free(g_ppTrackedDevices_CoreAudio, &pDevice->pContext->allocationCallbacks);
+ g_ppTrackedDevices_CoreAudio = NULL;
+ g_TrackedDeviceCap_CoreAudio = 0;
+ }
-/*******************************************************************************
+ break;
+ }
+ }
+ }
+ ma_mutex_unlock(&g_DeviceTrackingMutex_CoreAudio);
-Timing
+ return MA_SUCCESS;
+}
+#endif
-*******************************************************************************/
-#ifdef MA_WIN32
- static LARGE_INTEGER g_ma_TimerFrequency; /* <-- Initialized to zero since it's static. */
- static void ma_timer_init(ma_timer* pTimer)
- {
- LARGE_INTEGER counter;
+#if defined(MA_APPLE_MOBILE)
+@interface ma_ios_notification_handler:NSObject {
+ ma_device* m_pDevice;
+}
+@end
- if (g_ma_TimerFrequency.QuadPart == 0) {
- QueryPerformanceFrequency(&g_ma_TimerFrequency);
- }
+@implementation ma_ios_notification_handler
+-(id)init:(ma_device*)pDevice
+{
+ self = [super init];
+ m_pDevice = pDevice;
- QueryPerformanceCounter(&counter);
- pTimer->counter = counter.QuadPart;
- }
+ /* For route changes. */
+ [[NSNotificationCenter defaultCenter] addObserver:self selector:@selector(handle_route_change:) name:AVAudioSessionRouteChangeNotification object:[AVAudioSession sharedInstance]];
- static double ma_timer_get_time_in_seconds(ma_timer* pTimer)
- {
- LARGE_INTEGER counter;
- if (!QueryPerformanceCounter(&counter)) {
- return 0;
- }
+ /* For interruptions. */
+ [[NSNotificationCenter defaultCenter] addObserver:self selector:@selector(handle_interruption:) name:AVAudioSessionInterruptionNotification object:[AVAudioSession sharedInstance]];
- return (double)(counter.QuadPart - pTimer->counter) / g_ma_TimerFrequency.QuadPart;
- }
-#elif defined(MA_APPLE) && (__MAC_OS_X_VERSION_MIN_REQUIRED < 101200)
- static ma_uint64 g_ma_TimerFrequency = 0;
- static void ma_timer_init(ma_timer* pTimer)
- {
- mach_timebase_info_data_t baseTime;
- mach_timebase_info(&baseTime);
- g_ma_TimerFrequency = (baseTime.denom * 1e9) / baseTime.numer;
+ return self;
+}
- pTimer->counter = mach_absolute_time();
- }
+-(void)dealloc
+{
+ [self remove_handler];
+}
- static double ma_timer_get_time_in_seconds(ma_timer* pTimer)
- {
- ma_uint64 newTimeCounter = mach_absolute_time();
- ma_uint64 oldTimeCounter = pTimer->counter;
+-(void)remove_handler
+{
+ [[NSNotificationCenter defaultCenter] removeObserver:self name:AVAudioSessionRouteChangeNotification object:nil];
+ [[NSNotificationCenter defaultCenter] removeObserver:self name:AVAudioSessionInterruptionNotification object:nil];
+}
- return (newTimeCounter - oldTimeCounter) / g_ma_TimerFrequency;
- }
-#elif defined(MA_EMSCRIPTEN)
- static MA_INLINE void ma_timer_init(ma_timer* pTimer)
+-(void)handle_interruption:(NSNotification*)pNotification
+{
+ NSInteger type = [[[pNotification userInfo] objectForKey:AVAudioSessionInterruptionTypeKey] integerValue];
+ switch (type)
{
- pTimer->counterD = emscripten_get_now();
- }
+ case AVAudioSessionInterruptionTypeBegan:
+ {
+ ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_INFO, "[Core Audio] Interruption: AVAudioSessionInterruptionTypeBegan\n");
- static MA_INLINE double ma_timer_get_time_in_seconds(ma_timer* pTimer)
- {
- return (emscripten_get_now() - pTimer->counterD) / 1000; /* Emscripten is in milliseconds. */
- }
-#else
- #if _POSIX_C_SOURCE >= 199309L
- #if defined(CLOCK_MONOTONIC)
- #define MA_CLOCK_ID CLOCK_MONOTONIC
- #else
- #define MA_CLOCK_ID CLOCK_REALTIME
- #endif
+ /*
+ Core Audio will have stopped the internal device automatically, but we need explicitly
+ stop it at a higher level to ensure miniaudio-specific state is updated for consistency.
+ */
+ ma_device_stop(m_pDevice);
- static void ma_timer_init(ma_timer* pTimer)
+ /*
+ Fire the notification after the device has been stopped to ensure it's in the correct
+ state when the notification handler is invoked.
+ */
+ ma_device__on_notification_interruption_began(m_pDevice);
+ } break;
+
+ case AVAudioSessionInterruptionTypeEnded:
{
- struct timespec newTime;
- clock_gettime(MA_CLOCK_ID, &newTime);
+ ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_INFO, "[Core Audio] Interruption: AVAudioSessionInterruptionTypeEnded\n");
+ ma_device__on_notification_interruption_ended(m_pDevice);
+ } break;
+ }
+}
- pTimer->counter = (newTime.tv_sec * 1000000000) + newTime.tv_nsec;
- }
+-(void)handle_route_change:(NSNotification*)pNotification
+{
+ AVAudioSession* pSession = [AVAudioSession sharedInstance];
- static double ma_timer_get_time_in_seconds(ma_timer* pTimer)
+ NSInteger reason = [[[pNotification userInfo] objectForKey:AVAudioSessionRouteChangeReasonKey] integerValue];
+ switch (reason)
+ {
+ case AVAudioSessionRouteChangeReasonOldDeviceUnavailable:
{
- ma_uint64 newTimeCounter;
- ma_uint64 oldTimeCounter;
-
- struct timespec newTime;
- clock_gettime(MA_CLOCK_ID, &newTime);
+ ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_INFO, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonOldDeviceUnavailable\n");
+ } break;
- newTimeCounter = (newTime.tv_sec * 1000000000) + newTime.tv_nsec;
- oldTimeCounter = pTimer->counter;
+ case AVAudioSessionRouteChangeReasonNewDeviceAvailable:
+ {
+ ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_INFO, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonNewDeviceAvailable\n");
+ } break;
- return (newTimeCounter - oldTimeCounter) / 1000000000.0;
- }
- #else
- static void ma_timer_init(ma_timer* pTimer)
+ case AVAudioSessionRouteChangeReasonNoSuitableRouteForCategory:
{
- struct timeval newTime;
- gettimeofday(&newTime, NULL);
+ ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_INFO, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonNoSuitableRouteForCategory\n");
+ } break;
- pTimer->counter = (newTime.tv_sec * 1000000) + newTime.tv_usec;
- }
+ case AVAudioSessionRouteChangeReasonWakeFromSleep:
+ {
+ ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_INFO, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonWakeFromSleep\n");
+ } break;
- static double ma_timer_get_time_in_seconds(ma_timer* pTimer)
+ case AVAudioSessionRouteChangeReasonOverride:
{
- ma_uint64 newTimeCounter;
- ma_uint64 oldTimeCounter;
+ ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_INFO, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonOverride\n");
+ } break;
- struct timeval newTime;
- gettimeofday(&newTime, NULL);
+ case AVAudioSessionRouteChangeReasonCategoryChange:
+ {
+ ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_INFO, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonCategoryChange\n");
+ } break;
- newTimeCounter = (newTime.tv_sec * 1000000) + newTime.tv_usec;
- oldTimeCounter = pTimer->counter;
+ case AVAudioSessionRouteChangeReasonUnknown:
+ default:
+ {
+ ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_INFO, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonUnknown\n");
+ } break;
+ }
- return (newTimeCounter - oldTimeCounter) / 1000000.0;
- }
- #endif
+ ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_DEBUG, "[Core Audio] Changing Route. inputNumberChannels=%d; outputNumberOfChannels=%d\n", (int)pSession.inputNumberOfChannels, (int)pSession.outputNumberOfChannels);
+
+ /* Let the application know about the route change. */
+ ma_device__on_notification_rerouted(m_pDevice);
+}
+@end
#endif
-
-/*******************************************************************************
-
-Dynamic Linking
-
-*******************************************************************************/
-MA_API ma_handle ma_dlopen(ma_context* pContext, const char* filename)
+static ma_result ma_device_uninit__coreaudio(ma_device* pDevice)
{
- ma_handle handle;
-
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "Loading library: %s\n", filename);
-
-#ifdef _WIN32
-#ifdef MA_WIN32_DESKTOP
- handle = (ma_handle)LoadLibraryA(filename);
-#else
- /* *sigh* It appears there is no ANSI version of LoadPackagedLibrary()... */
- WCHAR filenameW[4096];
- if (MultiByteToWideChar(CP_UTF8, 0, filename, -1, filenameW, sizeof(filenameW)) == 0) {
- handle = NULL;
- } else {
- handle = (ma_handle)LoadPackagedLibrary(filenameW, 0);
- }
-#endif
-#else
- handle = (ma_handle)dlopen(filename, RTLD_NOW);
-#endif
+ MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(ma_device_get_state(pDevice) == ma_device_state_uninitialized);
+#if defined(MA_APPLE_DESKTOP)
/*
- I'm not considering failure to load a library an error nor a warning because seamlessly falling through to a lower-priority
- backend is a deliberate design choice. Instead I'm logging it as an informational message.
+ Make sure we're no longer tracking the device. It doesn't matter if we call this for a non-default device because it'll
+ just gracefully ignore it.
*/
- if (handle == NULL) {
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_INFO, "Failed to load library: %s\n", filename);
+ ma_device__untrack__coreaudio(pDevice);
+#endif
+#if defined(MA_APPLE_MOBILE)
+ if (pDevice->coreaudio.pNotificationHandler != NULL) {
+ ma_ios_notification_handler* pNotificationHandler = (MA_BRIDGE_TRANSFER ma_ios_notification_handler*)pDevice->coreaudio.pNotificationHandler;
+ [pNotificationHandler remove_handler];
}
+#endif
- (void)pContext; /* It's possible for pContext to be unused. */
- return handle;
-}
+ if (pDevice->coreaudio.audioUnitCapture != NULL) {
+ ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitCapture);
+ }
+ if (pDevice->coreaudio.audioUnitPlayback != NULL) {
+ ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitPlayback);
+ }
-MA_API void ma_dlclose(ma_context* pContext, ma_handle handle)
-{
-#ifdef _WIN32
- FreeLibrary((HMODULE)handle);
-#else
- dlclose((void*)handle);
-#endif
+ if (pDevice->coreaudio.pAudioBufferList) {
+ ma_free(pDevice->coreaudio.pAudioBufferList, &pDevice->pContext->allocationCallbacks);
+ }
- (void)pContext;
+ return MA_SUCCESS;
}
-MA_API ma_proc ma_dlsym(ma_context* pContext, ma_handle handle, const char* symbol)
+typedef struct
{
- ma_proc proc;
+ ma_bool32 allowNominalSampleRateChange;
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "Loading symbol: %s\n", symbol);
+ /* Input. */
+ ma_format formatIn;
+ ma_uint32 channelsIn;
+ ma_uint32 sampleRateIn;
+ ma_channel channelMapIn[MA_MAX_CHANNELS];
+ ma_uint32 periodSizeInFramesIn;
+ ma_uint32 periodSizeInMillisecondsIn;
+ ma_uint32 periodsIn;
+ ma_share_mode shareMode;
+ ma_performance_profile performanceProfile;
+ ma_bool32 registerStopEvent;
-#ifdef _WIN32
- proc = (ma_proc)GetProcAddress((HMODULE)handle, symbol);
-#else
-#if defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8))
- #pragma GCC diagnostic push
- #pragma GCC diagnostic ignored "-Wpedantic"
-#endif
- proc = (ma_proc)dlsym((void*)handle, symbol);
-#if defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8))
- #pragma GCC diagnostic pop
-#endif
+ /* Output. */
+#if defined(MA_APPLE_DESKTOP)
+ AudioObjectID deviceObjectID;
#endif
+ AudioComponent component;
+ AudioUnit audioUnit;
+ AudioBufferList* pAudioBufferList; /* Only used for input devices. */
+ ma_format formatOut;
+ ma_uint32 channelsOut;
+ ma_uint32 sampleRateOut;
+ ma_channel channelMapOut[MA_MAX_CHANNELS];
+ ma_uint32 periodSizeInFramesOut;
+ ma_uint32 periodsOut;
+ char deviceName[256];
+} ma_device_init_internal_data__coreaudio;
- if (proc == NULL) {
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_WARNING, "Failed to load symbol: %s\n", symbol);
- }
-
- (void)pContext; /* It's possible for pContext to be unused. */
- return proc;
-}
-
-
-#if 0
-static ma_uint32 ma_get_closest_standard_sample_rate(ma_uint32 sampleRateIn)
+static ma_result ma_device_init_internal__coreaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_init_internal_data__coreaudio* pData, void* pDevice_DoNotReference) /* <-- pDevice is typed as void* intentionally so as to avoid accidentally referencing it. */
{
- ma_uint32 closestRate = 0;
- ma_uint32 closestDiff = 0xFFFFFFFF;
- size_t iStandardRate;
+ ma_result result;
+ OSStatus status;
+ UInt32 enableIOFlag;
+ AudioStreamBasicDescription bestFormat;
+ UInt32 actualPeriodSizeInFrames;
+ AURenderCallbackStruct callbackInfo;
+#if defined(MA_APPLE_DESKTOP)
+ AudioObjectID deviceObjectID;
+#endif
- for (iStandardRate = 0; iStandardRate < ma_countof(g_maStandardSampleRatePriorities); ++iStandardRate) {
- ma_uint32 standardRate = g_maStandardSampleRatePriorities[iStandardRate];
- ma_uint32 diff;
+ /* This API should only be used for a single device type: playback or capture. No full-duplex mode. */
+ if (deviceType == ma_device_type_duplex) {
+ return MA_INVALID_ARGS;
+ }
- if (sampleRateIn > standardRate) {
- diff = sampleRateIn - standardRate;
- } else {
- diff = standardRate - sampleRateIn;
- }
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(deviceType == ma_device_type_playback || deviceType == ma_device_type_capture);
- if (diff == 0) {
- return standardRate; /* The input sample rate is a standard rate. */
- }
+#if defined(MA_APPLE_DESKTOP)
+ pData->deviceObjectID = 0;
+#endif
+ pData->component = NULL;
+ pData->audioUnit = NULL;
+ pData->pAudioBufferList = NULL;
- if (closestDiff > diff) {
- closestDiff = diff;
- closestRate = standardRate;
- }
+#if defined(MA_APPLE_DESKTOP)
+ result = ma_find_AudioObjectID(pContext, deviceType, pDeviceID, &deviceObjectID);
+ if (result != MA_SUCCESS) {
+ return result;
}
- return closestRate;
-}
+ pData->deviceObjectID = deviceObjectID;
#endif
+ /* Core audio doesn't really use the notion of a period so we can leave this unmodified, but not too over the top. */
+ pData->periodsOut = pData->periodsIn;
+ if (pData->periodsOut == 0) {
+ pData->periodsOut = MA_DEFAULT_PERIODS;
+ }
+ if (pData->periodsOut > 16) {
+ pData->periodsOut = 16;
+ }
-static void ma_device__on_data(ma_device* pDevice, void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount)
-{
- float masterVolumeFactor;
- ma_device_get_master_volume(pDevice, &masterVolumeFactor); /* Use ma_device_get_master_volume() to ensure the volume is loaded atomically. */
+ /* Audio unit. */
+ status = ((ma_AudioComponentInstanceNew_proc)pContext->coreaudio.AudioComponentInstanceNew)((AudioComponent)pContext->coreaudio.component, (AudioUnit*)&pData->audioUnit);
+ if (status != noErr) {
+ return ma_result_from_OSStatus(status);
+ }
- if (pDevice->onData) {
- if (!pDevice->noPreZeroedOutputBuffer && pFramesOut != NULL) {
- ma_silence_pcm_frames(pFramesOut, frameCount, pDevice->playback.format, pDevice->playback.channels);
- }
-
- /* Volume control of input makes things a bit awkward because the input buffer is read-only. We'll need to use a temp buffer and loop in this case. */
- if (pFramesIn != NULL && masterVolumeFactor < 1) {
- ma_uint8 tempFramesIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
- ma_uint32 bpfCapture = ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels);
- ma_uint32 bpfPlayback = ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
- ma_uint32 totalFramesProcessed = 0;
- while (totalFramesProcessed < frameCount) {
- ma_uint32 framesToProcessThisIteration = frameCount - totalFramesProcessed;
- if (framesToProcessThisIteration > sizeof(tempFramesIn)/bpfCapture) {
- framesToProcessThisIteration = sizeof(tempFramesIn)/bpfCapture;
- }
- ma_copy_and_apply_volume_factor_pcm_frames(tempFramesIn, ma_offset_ptr(pFramesIn, totalFramesProcessed*bpfCapture), framesToProcessThisIteration, pDevice->capture.format, pDevice->capture.channels, masterVolumeFactor);
+ /* The input/output buses need to be explicitly enabled and disabled. We set the flag based on the output unit first, then we just swap it for input. */
+ enableIOFlag = 1;
+ if (deviceType == ma_device_type_capture) {
+ enableIOFlag = 0;
+ }
- pDevice->onData(pDevice, ma_offset_ptr(pFramesOut, totalFramesProcessed*bpfPlayback), tempFramesIn, framesToProcessThisIteration);
+ status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, MA_COREAUDIO_OUTPUT_BUS, &enableIOFlag, sizeof(enableIOFlag));
+ if (status != noErr) {
+ ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
+ return ma_result_from_OSStatus(status);
+ }
- totalFramesProcessed += framesToProcessThisIteration;
- }
- } else {
- pDevice->onData(pDevice, pFramesOut, pFramesIn, frameCount);
- }
+ enableIOFlag = (enableIOFlag == 0) ? 1 : 0;
+ status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, MA_COREAUDIO_INPUT_BUS, &enableIOFlag, sizeof(enableIOFlag));
+ if (status != noErr) {
+ ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
+ return ma_result_from_OSStatus(status);
+ }
- /* Volume control and clipping for playback devices. */
- if (pFramesOut != NULL) {
- if (masterVolumeFactor < 1) {
- if (pFramesIn == NULL) { /* <-- In full-duplex situations, the volume will have been applied to the input samples before the data callback. Applying it again post-callback will incorrectly compound it. */
- ma_apply_volume_factor_pcm_frames(pFramesOut, frameCount, pDevice->playback.format, pDevice->playback.channels, masterVolumeFactor);
+
+ /* Set the device to use with this audio unit. This is only used on desktop since we are using defaults on mobile. */
+#if defined(MA_APPLE_DESKTOP)
+ status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &deviceObjectID, sizeof(deviceObjectID));
+ if (status != noErr) {
+ ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
+ return ma_result_from_OSStatus(result);
+ }
+#else
+ /*
+ For some reason it looks like Apple is only allowing selection of the input device. There does not appear to be any way to change
+ the default output route. I have no idea why this is like this, but for now we'll only be able to configure capture devices.
+ */
+ if (pDeviceID != NULL) {
+ if (deviceType == ma_device_type_capture) {
+ ma_bool32 found = MA_FALSE;
+ NSArray *pInputs = [[[AVAudioSession sharedInstance] currentRoute] inputs];
+ for (AVAudioSessionPortDescription* pPortDesc in pInputs) {
+ if (strcmp(pDeviceID->coreaudio, [pPortDesc.UID UTF8String]) == 0) {
+ [[AVAudioSession sharedInstance] setPreferredInput:pPortDesc error:nil];
+ found = MA_TRUE;
+ break;
}
}
- if (!pDevice->noClip && pDevice->playback.format == ma_format_f32) {
- ma_clip_pcm_frames_f32((float*)pFramesOut, frameCount, pDevice->playback.channels);
+ if (found == MA_FALSE) {
+ return MA_DOES_NOT_EXIST;
}
}
}
-}
-
-
-
-/* A helper function for reading sample data from the client. */
-static void ma_device__read_frames_from_client(ma_device* pDevice, ma_uint32 frameCount, void* pFramesOut)
-{
- MA_ASSERT(pDevice != NULL);
- MA_ASSERT(frameCount > 0);
- MA_ASSERT(pFramesOut != NULL);
-
- if (pDevice->playback.converter.isPassthrough) {
- ma_device__on_data(pDevice, pFramesOut, NULL, frameCount);
- } else {
- ma_result result;
- ma_uint64 totalFramesReadOut;
- ma_uint64 totalFramesReadIn;
- void* pRunningFramesOut;
-
- totalFramesReadOut = 0;
- totalFramesReadIn = 0;
- pRunningFramesOut = pFramesOut;
-
- while (totalFramesReadOut < frameCount) {
- ma_uint8 pIntermediaryBuffer[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In client format. */
- ma_uint64 intermediaryBufferCap = sizeof(pIntermediaryBuffer) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
- ma_uint64 framesToReadThisIterationIn;
- ma_uint64 framesReadThisIterationIn;
- ma_uint64 framesToReadThisIterationOut;
- ma_uint64 framesReadThisIterationOut;
- ma_uint64 requiredInputFrameCount;
-
- framesToReadThisIterationOut = (frameCount - totalFramesReadOut);
- framesToReadThisIterationIn = framesToReadThisIterationOut;
- if (framesToReadThisIterationIn > intermediaryBufferCap) {
- framesToReadThisIterationIn = intermediaryBufferCap;
- }
+#endif
- requiredInputFrameCount = ma_data_converter_get_required_input_frame_count(&pDevice->playback.converter, framesToReadThisIterationOut);
- if (framesToReadThisIterationIn > requiredInputFrameCount) {
- framesToReadThisIterationIn = requiredInputFrameCount;
- }
+ /*
+ Format. This is the hardest part of initialization because there's a few variables to take into account.
+ 1) The format must be supported by the device.
+ 2) The format must be supported miniaudio.
+ 3) There's a priority that miniaudio prefers.
- if (framesToReadThisIterationIn > 0) {
- ma_device__on_data(pDevice, pIntermediaryBuffer, NULL, (ma_uint32)framesToReadThisIterationIn);
- totalFramesReadIn += framesToReadThisIterationIn;
- }
+ Ideally we would like to use a format that's as close to the hardware as possible so we can get as close to a passthrough as possible. The
+ most important property is the sample rate. miniaudio can do format conversion for any sample rate and channel count, but cannot do the same
+ for the sample data format. If the sample data format is not supported by miniaudio it must be ignored completely.
- /*
- At this point we have our decoded data in input format and now we need to convert to output format. Note that even if we didn't read any
- input frames, we still want to try processing frames because there may some output frames generated from cached input data.
- */
- framesReadThisIterationIn = framesToReadThisIterationIn;
- framesReadThisIterationOut = framesToReadThisIterationOut;
- result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, pIntermediaryBuffer, &framesReadThisIterationIn, pRunningFramesOut, &framesReadThisIterationOut);
- if (result != MA_SUCCESS) {
- break;
- }
+ On mobile platforms this is a bit different. We just force the use of whatever the audio unit's current format is set to.
+ */
+ {
+ AudioStreamBasicDescription origFormat;
+ UInt32 origFormatSize = sizeof(origFormat);
+ AudioUnitScope formatScope = (deviceType == ma_device_type_playback) ? kAudioUnitScope_Input : kAudioUnitScope_Output;
+ AudioUnitElement formatElement = (deviceType == ma_device_type_playback) ? MA_COREAUDIO_OUTPUT_BUS : MA_COREAUDIO_INPUT_BUS;
- totalFramesReadOut += framesReadThisIterationOut;
- pRunningFramesOut = ma_offset_ptr(pRunningFramesOut, framesReadThisIterationOut * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels));
+ if (deviceType == ma_device_type_playback) {
+ status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, MA_COREAUDIO_OUTPUT_BUS, &origFormat, &origFormatSize);
+ } else {
+ status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, MA_COREAUDIO_INPUT_BUS, &origFormat, &origFormatSize);
+ }
+ if (status != noErr) {
+ ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
+ return ma_result_from_OSStatus(status);
+ }
- if (framesReadThisIterationIn == 0 && framesReadThisIterationOut == 0) {
- break; /* We're done. */
- }
+ #if defined(MA_APPLE_DESKTOP)
+ result = ma_find_best_format__coreaudio(pContext, deviceObjectID, deviceType, pData->formatIn, pData->channelsIn, pData->sampleRateIn, &origFormat, &bestFormat);
+ if (result != MA_SUCCESS) {
+ ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
+ return result;
}
- }
-}
-/* A helper for sending sample data to the client. */
-static void ma_device__send_frames_to_client(ma_device* pDevice, ma_uint32 frameCountInDeviceFormat, const void* pFramesInDeviceFormat)
-{
- MA_ASSERT(pDevice != NULL);
- MA_ASSERT(frameCountInDeviceFormat > 0);
- MA_ASSERT(pFramesInDeviceFormat != NULL);
+ /*
+ Technical Note TN2091: Device input using the HAL Output Audio Unit
+ https://developer.apple.com/library/archive/technotes/tn2091/_index.html
- if (pDevice->capture.converter.isPassthrough) {
- ma_device__on_data(pDevice, NULL, pFramesInDeviceFormat, frameCountInDeviceFormat);
- } else {
- ma_result result;
- ma_uint8 pFramesInClientFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
- ma_uint64 framesInClientFormatCap = sizeof(pFramesInClientFormat) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels);
- ma_uint64 totalDeviceFramesProcessed = 0;
- ma_uint64 totalClientFramesProcessed = 0;
- const void* pRunningFramesInDeviceFormat = pFramesInDeviceFormat;
+ This documentation says the following:
- /* We just keep going until we've exhaused all of our input frames and cannot generate any more output frames. */
- for (;;) {
- ma_uint64 deviceFramesProcessedThisIteration;
- ma_uint64 clientFramesProcessedThisIteration;
+ The internal AudioConverter can handle any *simple* conversion. Typically, this means that a client can specify ANY
+ variant of the PCM formats. Consequently, the device's sample rate should match the desired sample rate. If sample rate
+ conversion is needed, it can be accomplished by buffering the input and converting the data on a separate thread with
+ another AudioConverter.
- deviceFramesProcessedThisIteration = (frameCountInDeviceFormat - totalDeviceFramesProcessed);
- clientFramesProcessedThisIteration = framesInClientFormatCap;
+ The important part here is the mention that it can handle *simple* conversions, which does *not* include sample rate. We
+ therefore want to ensure the sample rate stays consistent. This document is specifically for input, but I'm going to play it
+ safe and apply the same rule to output as well.
- result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningFramesInDeviceFormat, &deviceFramesProcessedThisIteration, pFramesInClientFormat, &clientFramesProcessedThisIteration);
- if (result != MA_SUCCESS) {
- break;
- }
+ I have tried going against the documentation by setting the sample rate anyway, but this just results in AudioUnitRender()
+ returning a result code of -10863. I have also tried changing the format directly on the input scope on the input bus, but
+ this just results in `ca_require: IsStreamFormatWritable(inScope, inElement) NotWritable` when trying to set the format.
- if (clientFramesProcessedThisIteration > 0) {
- ma_device__on_data(pDevice, NULL, pFramesInClientFormat, (ma_uint32)clientFramesProcessedThisIteration); /* Safe cast. */
- }
+ Something that does seem to work, however, has been setting the nominal sample rate on the deivce object. The problem with
+ this, however, is that it actually changes the sample rate at the operating system level and not just the application. This
+ could be intrusive to the user, however, so I don't think it's wise to make this the default. Instead I'm making this a
+ configuration option. When the `coreaudio.allowNominalSampleRateChange` config option is set to true, changing the sample
+ rate will be allowed. Otherwise it'll be fixed to the current sample rate. To check the system-defined sample rate, run
+ the Audio MIDI Setup program that comes installed on macOS and observe how the sample rate changes as the sample rate is
+ changed by miniaudio.
+ */
+ if (pData->allowNominalSampleRateChange) {
+ AudioValueRange sampleRateRange;
+ AudioObjectPropertyAddress propAddress;
- pRunningFramesInDeviceFormat = ma_offset_ptr(pRunningFramesInDeviceFormat, deviceFramesProcessedThisIteration * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels));
- totalDeviceFramesProcessed += deviceFramesProcessedThisIteration;
- totalClientFramesProcessed += clientFramesProcessedThisIteration;
+ sampleRateRange.mMinimum = bestFormat.mSampleRate;
+ sampleRateRange.mMaximum = bestFormat.mSampleRate;
- if (deviceFramesProcessedThisIteration == 0 && clientFramesProcessedThisIteration == 0) {
- break; /* We're done. */
+ propAddress.mSelector = kAudioDevicePropertyNominalSampleRate;
+ propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput;
+ propAddress.mElement = kAudioObjectPropertyElementMaster;
+
+ status = ((ma_AudioObjectSetPropertyData_proc)pContext->coreaudio.AudioObjectSetPropertyData)(deviceObjectID, &propAddress, 0, NULL, sizeof(sampleRateRange), &sampleRateRange);
+ if (status != noErr) {
+ bestFormat.mSampleRate = origFormat.mSampleRate;
}
+ } else {
+ bestFormat.mSampleRate = origFormat.mSampleRate;
}
- }
-}
-
-static ma_result ma_device__handle_duplex_callback_capture(ma_device* pDevice, ma_uint32 frameCountInDeviceFormat, const void* pFramesInDeviceFormat, ma_pcm_rb* pRB)
-{
- ma_result result;
- ma_uint32 totalDeviceFramesProcessed = 0;
- const void* pRunningFramesInDeviceFormat = pFramesInDeviceFormat;
- MA_ASSERT(pDevice != NULL);
- MA_ASSERT(frameCountInDeviceFormat > 0);
- MA_ASSERT(pFramesInDeviceFormat != NULL);
- MA_ASSERT(pRB != NULL);
+ status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, sizeof(bestFormat));
+ if (status != noErr) {
+ /* We failed to set the format, so fall back to the current format of the audio unit. */
+ bestFormat = origFormat;
+ }
+ #else
+ bestFormat = origFormat;
- /* Write to the ring buffer. The ring buffer is in the client format which means we need to convert. */
- for (;;) {
- ma_uint32 framesToProcessInDeviceFormat = (frameCountInDeviceFormat - totalDeviceFramesProcessed);
- ma_uint32 framesToProcessInClientFormat = MA_DATA_CONVERTER_STACK_BUFFER_SIZE / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels);
- ma_uint64 framesProcessedInDeviceFormat;
- ma_uint64 framesProcessedInClientFormat;
- void* pFramesInClientFormat;
+ /*
+ Sample rate is a little different here because for some reason kAudioUnitProperty_StreamFormat returns 0... Oh well. We need to instead try
+ setting the sample rate to what the user has requested and then just see the results of it. Need to use some Objective-C here for this since
+ it depends on Apple's AVAudioSession API. To do this we just get the shared AVAudioSession instance and then set it. Note that from what I
+ can tell, it looks like the sample rate is shared between playback and capture for everything.
+ */
+ @autoreleasepool {
+ AVAudioSession* pAudioSession = [AVAudioSession sharedInstance];
+ MA_ASSERT(pAudioSession != NULL);
- result = ma_pcm_rb_acquire_write(pRB, &framesToProcessInClientFormat, &pFramesInClientFormat);
- if (result != MA_SUCCESS) {
- ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "Failed to acquire capture PCM frames from ring buffer.", result);
- break;
- }
+ [pAudioSession setPreferredSampleRate:(double)pData->sampleRateIn error:nil];
+ bestFormat.mSampleRate = pAudioSession.sampleRate;
- if (framesToProcessInClientFormat == 0) {
- if (ma_pcm_rb_pointer_distance(pRB) == (ma_int32)ma_pcm_rb_get_subbuffer_size(pRB)) {
- break; /* Overrun. Not enough room in the ring buffer for input frame. Excess frames are dropped. */
+ /*
+ I've had a report that the channel count returned by AudioUnitGetProperty above is inconsistent with
+ AVAudioSession outputNumberOfChannels. I'm going to try using the AVAudioSession values instead.
+ */
+ if (deviceType == ma_device_type_playback) {
+ bestFormat.mChannelsPerFrame = (UInt32)pAudioSession.outputNumberOfChannels;
+ }
+ if (deviceType == ma_device_type_capture) {
+ bestFormat.mChannelsPerFrame = (UInt32)pAudioSession.inputNumberOfChannels;
}
}
- /* Convert. */
- framesProcessedInDeviceFormat = framesToProcessInDeviceFormat;
- framesProcessedInClientFormat = framesToProcessInClientFormat;
- result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningFramesInDeviceFormat, &framesProcessedInDeviceFormat, pFramesInClientFormat, &framesProcessedInClientFormat);
- if (result != MA_SUCCESS) {
- break;
+ status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, sizeof(bestFormat));
+ if (status != noErr) {
+ ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
+ return ma_result_from_OSStatus(status);
}
+ #endif
- result = ma_pcm_rb_commit_write(pRB, (ma_uint32)framesProcessedInClientFormat, pFramesInClientFormat); /* Safe cast. */
+ result = ma_format_from_AudioStreamBasicDescription(&bestFormat, &pData->formatOut);
if (result != MA_SUCCESS) {
- ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "Failed to commit capture PCM frames to ring buffer.", result);
- break;
+ ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
+ return result;
}
- pRunningFramesInDeviceFormat = ma_offset_ptr(pRunningFramesInDeviceFormat, framesProcessedInDeviceFormat * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels));
- totalDeviceFramesProcessed += (ma_uint32)framesProcessedInDeviceFormat; /* Safe cast. */
-
- /* We're done when we're unable to process any client nor device frames. */
- if (framesProcessedInClientFormat == 0 && framesProcessedInDeviceFormat == 0) {
- break; /* Done. */
+ if (pData->formatOut == ma_format_unknown) {
+ ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
+ return MA_FORMAT_NOT_SUPPORTED;
}
- }
-
- return MA_SUCCESS;
-}
-static ma_result ma_device__handle_duplex_callback_playback(ma_device* pDevice, ma_uint32 frameCount, void* pFramesInInternalFormat, ma_pcm_rb* pRB)
-{
- ma_result result;
- ma_uint8 playbackFramesInExternalFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
- ma_uint8 silentInputFrames[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
- ma_uint32 totalFramesToReadFromClient;
- ma_uint32 totalFramesReadFromClient;
- ma_uint32 totalFramesReadOut = 0;
+ pData->channelsOut = bestFormat.mChannelsPerFrame;
+ pData->sampleRateOut = bestFormat.mSampleRate;
+ }
- MA_ASSERT(pDevice != NULL);
- MA_ASSERT(frameCount > 0);
- MA_ASSERT(pFramesInInternalFormat != NULL);
- MA_ASSERT(pRB != NULL);
+ /* Clamp the channel count for safety. */
+ if (pData->channelsOut > MA_MAX_CHANNELS) {
+ pData->channelsOut = MA_MAX_CHANNELS;
+ }
/*
- Sitting in the ring buffer should be captured data from the capture callback in external format. If there's not enough data in there for
- the whole frameCount frames we just use silence instead for the input data.
+ Internal channel map. This is weird in my testing. If I use the AudioObject to get the
+ channel map, the channel descriptions are set to "Unknown" for some reason. To work around
+ this it looks like retrieving it from the AudioUnit will work. However, and this is where
+ it gets weird, it doesn't seem to work with capture devices, nor at all on iOS... Therefore
+ I'm going to fall back to a default assumption in these cases.
*/
- MA_ZERO_MEMORY(silentInputFrames, sizeof(silentInputFrames));
-
- /* We need to calculate how many output frames are required to be read from the client to completely fill frameCount internal frames. */
- totalFramesToReadFromClient = (ma_uint32)ma_data_converter_get_required_input_frame_count(&pDevice->playback.converter, frameCount);
- totalFramesReadFromClient = 0;
- while (totalFramesReadFromClient < totalFramesToReadFromClient && ma_device_is_started(pDevice)) {
- ma_uint32 framesRemainingFromClient;
- ma_uint32 framesToProcessFromClient;
- ma_uint32 inputFrameCount;
- void* pInputFrames;
-
- framesRemainingFromClient = (totalFramesToReadFromClient - totalFramesReadFromClient);
- framesToProcessFromClient = sizeof(playbackFramesInExternalFormat) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
- if (framesToProcessFromClient > framesRemainingFromClient) {
- framesToProcessFromClient = framesRemainingFromClient;
+#if defined(MA_APPLE_DESKTOP)
+ result = ma_get_AudioUnit_channel_map(pContext, pData->audioUnit, deviceType, pData->channelMapOut, pData->channelsOut);
+ if (result != MA_SUCCESS) {
+ #if 0
+ /* Try falling back to the channel map from the AudioObject. */
+ result = ma_get_AudioObject_channel_map(pContext, deviceObjectID, deviceType, pData->channelMapOut, pData->channelsOut);
+ if (result != MA_SUCCESS) {
+ return result;
}
+ #else
+ /* Fall back to default assumptions. */
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pData->channelMapOut, ma_countof(pData->channelMapOut), pData->channelsOut);
+ #endif
+ }
+#else
+ /* TODO: Figure out how to get the channel map using AVAudioSession. */
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pData->channelMapOut, ma_countof(pData->channelMapOut), pData->channelsOut);
+#endif
- /* We need to grab captured samples before firing the callback. If there's not enough input samples we just pass silence. */
- inputFrameCount = framesToProcessFromClient;
- result = ma_pcm_rb_acquire_read(pRB, &inputFrameCount, &pInputFrames);
- if (result == MA_SUCCESS) {
- if (inputFrameCount > 0) {
- /* Use actual input frames. */
- ma_device__on_data(pDevice, playbackFramesInExternalFormat, pInputFrames, inputFrameCount);
- } else {
- if (ma_pcm_rb_pointer_distance(pRB) == 0) {
- break; /* Underrun. */
- }
- }
- /* We're done with the captured samples. */
- result = ma_pcm_rb_commit_read(pRB, inputFrameCount, pInputFrames);
- if (result != MA_SUCCESS) {
- break; /* Don't know what to do here... Just abandon ship. */
+ /* Buffer size. Not allowing this to be configurable on iOS. */
+ if (pData->periodSizeInFramesIn == 0) {
+ if (pData->periodSizeInMillisecondsIn == 0) {
+ if (pData->performanceProfile == ma_performance_profile_low_latency) {
+ actualPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY, pData->sampleRateOut);
+ } else {
+ actualPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE, pData->sampleRateOut);
}
} else {
- /* Use silent input frames. */
- inputFrameCount = ma_min(
- sizeof(playbackFramesInExternalFormat) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels),
- sizeof(silentInputFrames) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels)
- );
-
- ma_device__on_data(pDevice, playbackFramesInExternalFormat, silentInputFrames, inputFrameCount);
+ actualPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pData->periodSizeInMillisecondsIn, pData->sampleRateOut);
}
+ } else {
+ actualPeriodSizeInFrames = pData->periodSizeInFramesIn;
+ }
- /* We have samples in external format so now we need to convert to internal format and output to the device. */
- {
- ma_uint64 framesConvertedIn = inputFrameCount;
- ma_uint64 framesConvertedOut = (frameCount - totalFramesReadOut);
- ma_data_converter_process_pcm_frames(&pDevice->playback.converter, playbackFramesInExternalFormat, &framesConvertedIn, pFramesInInternalFormat, &framesConvertedOut);
-
- totalFramesReadFromClient += (ma_uint32)framesConvertedIn; /* Safe cast. */
- totalFramesReadOut += (ma_uint32)framesConvertedOut; /* Safe cast. */
- pFramesInInternalFormat = ma_offset_ptr(pFramesInInternalFormat, framesConvertedOut * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels));
- }
+#if defined(MA_APPLE_DESKTOP)
+ result = ma_set_AudioObject_buffer_size_in_frames(pContext, deviceObjectID, deviceType, &actualPeriodSizeInFrames);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+#else
+ /*
+ On iOS, the size of the IO buffer needs to be specified in seconds and is a floating point
+ number. I don't trust any potential truncation errors due to converting from float to integer
+ so I'm going to explicitly set the actual period size to the next power of 2.
+ */
+ @autoreleasepool {
+ AVAudioSession* pAudioSession = [AVAudioSession sharedInstance];
+ MA_ASSERT(pAudioSession != NULL);
+
+ [pAudioSession setPreferredIOBufferDuration:((float)actualPeriodSizeInFrames / pAudioSession.sampleRate) error:nil];
+ actualPeriodSizeInFrames = ma_next_power_of_2((ma_uint32)(pAudioSession.IOBufferDuration * pAudioSession.sampleRate));
}
+#endif
- return MA_SUCCESS;
-}
-/* A helper for changing the state of the device. */
-static MA_INLINE void ma_device__set_state(ma_device* pDevice, ma_uint32 newState)
-{
- c89atomic_exchange_32(&pDevice->state, newState);
-}
+ /*
+ During testing I discovered that the buffer size can be too big. You'll get an error like this:
+ kAudioUnitErr_TooManyFramesToProcess : inFramesToProcess=4096, mMaxFramesPerSlice=512
-#ifdef MA_WIN32
- GUID MA_GUID_KSDATAFORMAT_SUBTYPE_PCM = {0x00000001, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}};
- GUID MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT = {0x00000003, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}};
- /*GUID MA_GUID_KSDATAFORMAT_SUBTYPE_ALAW = {0x00000006, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}};*/
- /*GUID MA_GUID_KSDATAFORMAT_SUBTYPE_MULAW = {0x00000007, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}};*/
-#endif
+ Note how inFramesToProcess is smaller than mMaxFramesPerSlice. To fix, we need to set kAudioUnitProperty_MaximumFramesPerSlice to that
+ of the size of our buffer, or do it the other way around and set our buffer size to the kAudioUnitProperty_MaximumFramesPerSlice.
+ */
+ status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Global, 0, &actualPeriodSizeInFrames, sizeof(actualPeriodSizeInFrames));
+ if (status != noErr) {
+ ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
+ return ma_result_from_OSStatus(status);
+ }
+ pData->periodSizeInFramesOut = (ma_uint32)actualPeriodSizeInFrames;
+ /* We need a buffer list if this is an input device. We render into this in the input callback. */
+ if (deviceType == ma_device_type_capture) {
+ ma_bool32 isInterleaved = (bestFormat.mFormatFlags & kAudioFormatFlagIsNonInterleaved) == 0;
+ AudioBufferList* pBufferList;
-MA_API ma_uint32 ma_get_format_priority_index(ma_format format) /* Lower = better. */
-{
- ma_uint32 i;
- for (i = 0; i < ma_countof(g_maFormatPriorities); ++i) {
- if (g_maFormatPriorities[i] == format) {
- return i;
+ pBufferList = ma_allocate_AudioBufferList__coreaudio(pData->periodSizeInFramesOut, pData->formatOut, pData->channelsOut, (isInterleaved) ? ma_stream_layout_interleaved : ma_stream_layout_deinterleaved, &pContext->allocationCallbacks);
+ if (pBufferList == NULL) {
+ ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
+ return MA_OUT_OF_MEMORY;
}
- }
-
- /* Getting here means the format could not be found or is equal to ma_format_unknown. */
- return (ma_uint32)-1;
-}
-
-static ma_result ma_device__post_init_setup(ma_device* pDevice, ma_device_type deviceType);
+ pData->pAudioBufferList = pBufferList;
+ }
-static ma_bool32 ma_device_descriptor_is_valid(const ma_device_descriptor* pDeviceDescriptor)
-{
- if (pDeviceDescriptor == NULL) {
- return MA_FALSE;
+ /* Callbacks. */
+ callbackInfo.inputProcRefCon = pDevice_DoNotReference;
+ if (deviceType == ma_device_type_playback) {
+ callbackInfo.inputProc = ma_on_output__coreaudio;
+ status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Global, 0, &callbackInfo, sizeof(callbackInfo));
+ if (status != noErr) {
+ ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
+ return ma_result_from_OSStatus(status);
+ }
+ } else {
+ callbackInfo.inputProc = ma_on_input__coreaudio;
+ status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &callbackInfo, sizeof(callbackInfo));
+ if (status != noErr) {
+ ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
+ return ma_result_from_OSStatus(status);
+ }
}
- if (pDeviceDescriptor->format == ma_format_unknown) {
- return MA_FALSE;
+ /* We need to listen for stop events. */
+ if (pData->registerStopEvent) {
+ status = ((ma_AudioUnitAddPropertyListener_proc)pContext->coreaudio.AudioUnitAddPropertyListener)(pData->audioUnit, kAudioOutputUnitProperty_IsRunning, on_start_stop__coreaudio, pDevice_DoNotReference);
+ if (status != noErr) {
+ ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
+ return ma_result_from_OSStatus(status);
+ }
}
- if (pDeviceDescriptor->channels < MA_MIN_CHANNELS || pDeviceDescriptor->channels > MA_MAX_CHANNELS) {
- return MA_FALSE;
+ /* Initialize the audio unit. */
+ status = ((ma_AudioUnitInitialize_proc)pContext->coreaudio.AudioUnitInitialize)(pData->audioUnit);
+ if (status != noErr) {
+ ma_free(pData->pAudioBufferList, &pContext->allocationCallbacks);
+ pData->pAudioBufferList = NULL;
+ ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
+ return ma_result_from_OSStatus(status);
}
- if (pDeviceDescriptor->sampleRate == 0) {
- return MA_FALSE;
+ /* Grab the name. */
+#if defined(MA_APPLE_DESKTOP)
+ ma_get_AudioObject_name(pContext, deviceObjectID, sizeof(pData->deviceName), pData->deviceName);
+#else
+ if (deviceType == ma_device_type_playback) {
+ ma_strcpy_s(pData->deviceName, sizeof(pData->deviceName), MA_DEFAULT_PLAYBACK_DEVICE_NAME);
+ } else {
+ ma_strcpy_s(pData->deviceName, sizeof(pData->deviceName), MA_DEFAULT_CAPTURE_DEVICE_NAME);
}
+#endif
- return MA_TRUE;
+ return result;
}
-
-static ma_result ma_device_audio_thread__default_read_write(ma_device* pDevice)
+#if defined(MA_APPLE_DESKTOP)
+static ma_result ma_device_reinit_internal__coreaudio(ma_device* pDevice, ma_device_type deviceType, ma_bool32 disposePreviousAudioUnit)
{
- ma_result result = MA_SUCCESS;
- ma_bool32 exitLoop = MA_FALSE;
- ma_uint8 capturedDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
- ma_uint8 playbackDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
- ma_uint32 capturedDeviceDataCapInFrames = 0;
- ma_uint32 playbackDeviceDataCapInFrames = 0;
+ ma_device_init_internal_data__coreaudio data;
+ ma_result result;
- MA_ASSERT(pDevice != NULL);
+ /* This should only be called for playback or capture, not duplex. */
+ if (deviceType == ma_device_type_duplex) {
+ return MA_INVALID_ARGS;
+ }
- /* Just some quick validation on the device type and the available callbacks. */
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) {
- if (pDevice->pContext->callbacks.onDeviceRead == NULL) {
- return MA_NOT_IMPLEMENTED;
+ data.allowNominalSampleRateChange = MA_FALSE; /* Don't change the nominal sample rate when switching devices. */
+
+ if (deviceType == ma_device_type_capture) {
+ data.formatIn = pDevice->capture.format;
+ data.channelsIn = pDevice->capture.channels;
+ data.sampleRateIn = pDevice->sampleRate;
+ MA_COPY_MEMORY(data.channelMapIn, pDevice->capture.channelMap, sizeof(pDevice->capture.channelMap));
+ data.shareMode = pDevice->capture.shareMode;
+ data.performanceProfile = pDevice->coreaudio.originalPerformanceProfile;
+ data.registerStopEvent = MA_TRUE;
+
+ if (disposePreviousAudioUnit) {
+ ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitCapture);
+ ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitCapture);
+ }
+ if (pDevice->coreaudio.pAudioBufferList) {
+ ma_free(pDevice->coreaudio.pAudioBufferList, &pDevice->pContext->allocationCallbacks);
}
+ } else if (deviceType == ma_device_type_playback) {
+ data.formatIn = pDevice->playback.format;
+ data.channelsIn = pDevice->playback.channels;
+ data.sampleRateIn = pDevice->sampleRate;
+ MA_COPY_MEMORY(data.channelMapIn, pDevice->playback.channelMap, sizeof(pDevice->playback.channelMap));
+ data.shareMode = pDevice->playback.shareMode;
+ data.performanceProfile = pDevice->coreaudio.originalPerformanceProfile;
+ data.registerStopEvent = (pDevice->type != ma_device_type_duplex);
- capturedDeviceDataCapInFrames = sizeof(capturedDeviceData) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
+ if (disposePreviousAudioUnit) {
+ ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitPlayback);
+ ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitPlayback);
+ }
}
+ data.periodSizeInFramesIn = pDevice->coreaudio.originalPeriodSizeInFrames;
+ data.periodSizeInMillisecondsIn = pDevice->coreaudio.originalPeriodSizeInMilliseconds;
+ data.periodsIn = pDevice->coreaudio.originalPeriods;
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- if (pDevice->pContext->callbacks.onDeviceWrite == NULL) {
- return MA_NOT_IMPLEMENTED;
- }
+ /* Need at least 3 periods for duplex. */
+ if (data.periodsIn < 3 && pDevice->type == ma_device_type_duplex) {
+ data.periodsIn = 3;
+ }
- playbackDeviceDataCapInFrames = sizeof(playbackDeviceData) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
+ result = ma_device_init_internal__coreaudio(pDevice->pContext, deviceType, NULL, &data, (void*)pDevice);
+ if (result != MA_SUCCESS) {
+ return result;
}
- /* NOTE: The device was started outside of this function, in the worker thread. */
+ if (deviceType == ma_device_type_capture) {
+ #if defined(MA_APPLE_DESKTOP)
+ pDevice->coreaudio.deviceObjectIDCapture = (ma_uint32)data.deviceObjectID;
+ ma_get_AudioObject_uid(pDevice->pContext, pDevice->coreaudio.deviceObjectIDCapture, sizeof(pDevice->capture.id.coreaudio), pDevice->capture.id.coreaudio);
+ #endif
+ pDevice->coreaudio.audioUnitCapture = (ma_ptr)data.audioUnit;
+ pDevice->coreaudio.pAudioBufferList = (ma_ptr)data.pAudioBufferList;
+ pDevice->coreaudio.audioBufferCapInFrames = data.periodSizeInFramesOut;
- while (ma_device_get_state(pDevice) == MA_STATE_STARTED && !exitLoop) {
- switch (pDevice->type) {
- case ma_device_type_duplex:
- {
- /* The process is: onDeviceRead() -> convert -> callback -> convert -> onDeviceWrite() */
- ma_uint32 totalCapturedDeviceFramesProcessed = 0;
- ma_uint32 capturedDevicePeriodSizeInFrames = ma_min(pDevice->capture.internalPeriodSizeInFrames, pDevice->playback.internalPeriodSizeInFrames);
+ pDevice->capture.internalFormat = data.formatOut;
+ pDevice->capture.internalChannels = data.channelsOut;
+ pDevice->capture.internalSampleRate = data.sampleRateOut;
+ MA_COPY_MEMORY(pDevice->capture.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut));
+ pDevice->capture.internalPeriodSizeInFrames = data.periodSizeInFramesOut;
+ pDevice->capture.internalPeriods = data.periodsOut;
+ } else if (deviceType == ma_device_type_playback) {
+ #if defined(MA_APPLE_DESKTOP)
+ pDevice->coreaudio.deviceObjectIDPlayback = (ma_uint32)data.deviceObjectID;
+ ma_get_AudioObject_uid(pDevice->pContext, pDevice->coreaudio.deviceObjectIDPlayback, sizeof(pDevice->playback.id.coreaudio), pDevice->playback.id.coreaudio);
+ #endif
+ pDevice->coreaudio.audioUnitPlayback = (ma_ptr)data.audioUnit;
- while (totalCapturedDeviceFramesProcessed < capturedDevicePeriodSizeInFrames) {
- ma_uint32 capturedDeviceFramesRemaining;
- ma_uint32 capturedDeviceFramesProcessed;
- ma_uint32 capturedDeviceFramesToProcess;
- ma_uint32 capturedDeviceFramesToTryProcessing = capturedDevicePeriodSizeInFrames - totalCapturedDeviceFramesProcessed;
- if (capturedDeviceFramesToTryProcessing > capturedDeviceDataCapInFrames) {
- capturedDeviceFramesToTryProcessing = capturedDeviceDataCapInFrames;
- }
+ pDevice->playback.internalFormat = data.formatOut;
+ pDevice->playback.internalChannels = data.channelsOut;
+ pDevice->playback.internalSampleRate = data.sampleRateOut;
+ MA_COPY_MEMORY(pDevice->playback.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut));
+ pDevice->playback.internalPeriodSizeInFrames = data.periodSizeInFramesOut;
+ pDevice->playback.internalPeriods = data.periodsOut;
+ }
- result = pDevice->pContext->callbacks.onDeviceRead(pDevice, capturedDeviceData, capturedDeviceFramesToTryProcessing, &capturedDeviceFramesToProcess);
- if (result != MA_SUCCESS) {
- exitLoop = MA_TRUE;
- break;
- }
+ return MA_SUCCESS;
+}
+#endif /* MA_APPLE_DESKTOP */
- capturedDeviceFramesRemaining = capturedDeviceFramesToProcess;
- capturedDeviceFramesProcessed = 0;
+static ma_result ma_device_init__coreaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+{
+ ma_result result;
- /* At this point we have our captured data in device format and we now need to convert it to client format. */
- for (;;) {
- ma_uint8 capturedClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
- ma_uint8 playbackClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
- ma_uint32 capturedClientDataCapInFrames = sizeof(capturedClientData) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels);
- ma_uint32 playbackClientDataCapInFrames = sizeof(playbackClientData) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
- ma_uint64 capturedClientFramesToProcessThisIteration = ma_min(capturedClientDataCapInFrames, playbackClientDataCapInFrames);
- ma_uint64 capturedDeviceFramesToProcessThisIteration = capturedDeviceFramesRemaining;
- ma_uint8* pRunningCapturedDeviceFrames = ma_offset_ptr(capturedDeviceData, capturedDeviceFramesProcessed * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels));
+ MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(pConfig != NULL);
- /* Convert capture data from device format to client format. */
- result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningCapturedDeviceFrames, &capturedDeviceFramesToProcessThisIteration, capturedClientData, &capturedClientFramesToProcessThisIteration);
- if (result != MA_SUCCESS) {
- break;
- }
+ if (pConfig->deviceType == ma_device_type_loopback) {
+ return MA_DEVICE_TYPE_NOT_SUPPORTED;
+ }
- /*
- If we weren't able to generate any output frames it must mean we've exhaused all of our input. The only time this would not be the case is if capturedClientData was too small
- which should never be the case when it's of the size MA_DATA_CONVERTER_STACK_BUFFER_SIZE.
- */
- if (capturedClientFramesToProcessThisIteration == 0) {
- break;
- }
+ /* No exclusive mode with the Core Audio backend for now. */
+ if (((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive) ||
+ ((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive)) {
+ return MA_SHARE_MODE_NOT_SUPPORTED;
+ }
- ma_device__on_data(pDevice, playbackClientData, capturedClientData, (ma_uint32)capturedClientFramesToProcessThisIteration); /* Safe cast .*/
+ /* Capture needs to be initialized first. */
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
+ ma_device_init_internal_data__coreaudio data;
+ data.allowNominalSampleRateChange = pConfig->coreaudio.allowNominalSampleRateChange;
+ data.formatIn = pDescriptorCapture->format;
+ data.channelsIn = pDescriptorCapture->channels;
+ data.sampleRateIn = pDescriptorCapture->sampleRate;
+ MA_COPY_MEMORY(data.channelMapIn, pDescriptorCapture->channelMap, sizeof(pDescriptorCapture->channelMap));
+ data.periodSizeInFramesIn = pDescriptorCapture->periodSizeInFrames;
+ data.periodSizeInMillisecondsIn = pDescriptorCapture->periodSizeInMilliseconds;
+ data.periodsIn = pDescriptorCapture->periodCount;
+ data.shareMode = pDescriptorCapture->shareMode;
+ data.performanceProfile = pConfig->performanceProfile;
+ data.registerStopEvent = MA_TRUE;
- capturedDeviceFramesProcessed += (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */
- capturedDeviceFramesRemaining -= (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */
+ /* Need at least 3 periods for duplex. */
+ if (data.periodsIn < 3 && pConfig->deviceType == ma_device_type_duplex) {
+ data.periodsIn = 3;
+ }
- /* At this point the playbackClientData buffer should be holding data that needs to be written to the device. */
- for (;;) {
- ma_uint64 convertedClientFrameCount = capturedClientFramesToProcessThisIteration;
- ma_uint64 convertedDeviceFrameCount = playbackDeviceDataCapInFrames;
- result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, playbackClientData, &convertedClientFrameCount, playbackDeviceData, &convertedDeviceFrameCount);
- if (result != MA_SUCCESS) {
- break;
- }
+ result = ma_device_init_internal__coreaudio(pDevice->pContext, ma_device_type_capture, pDescriptorCapture->pDeviceID, &data, (void*)pDevice);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- result = pDevice->pContext->callbacks.onDeviceWrite(pDevice, playbackDeviceData, (ma_uint32)convertedDeviceFrameCount, NULL); /* Safe cast. */
- if (result != MA_SUCCESS) {
- exitLoop = MA_TRUE;
- break;
- }
+ pDevice->coreaudio.isDefaultCaptureDevice = (pConfig->capture.pDeviceID == NULL);
+ #if defined(MA_APPLE_DESKTOP)
+ pDevice->coreaudio.deviceObjectIDCapture = (ma_uint32)data.deviceObjectID;
+ #endif
+ pDevice->coreaudio.audioUnitCapture = (ma_ptr)data.audioUnit;
+ pDevice->coreaudio.pAudioBufferList = (ma_ptr)data.pAudioBufferList;
+ pDevice->coreaudio.audioBufferCapInFrames = data.periodSizeInFramesOut;
+ pDevice->coreaudio.originalPeriodSizeInFrames = pDescriptorCapture->periodSizeInFrames;
+ pDevice->coreaudio.originalPeriodSizeInMilliseconds = pDescriptorCapture->periodSizeInMilliseconds;
+ pDevice->coreaudio.originalPeriods = pDescriptorCapture->periodCount;
+ pDevice->coreaudio.originalPerformanceProfile = pConfig->performanceProfile;
- capturedClientFramesToProcessThisIteration -= (ma_uint32)convertedClientFrameCount; /* Safe cast. */
- if (capturedClientFramesToProcessThisIteration == 0) {
- break;
- }
- }
+ pDescriptorCapture->format = data.formatOut;
+ pDescriptorCapture->channels = data.channelsOut;
+ pDescriptorCapture->sampleRate = data.sampleRateOut;
+ MA_COPY_MEMORY(pDescriptorCapture->channelMap, data.channelMapOut, sizeof(data.channelMapOut));
+ pDescriptorCapture->periodSizeInFrames = data.periodSizeInFramesOut;
+ pDescriptorCapture->periodCount = data.periodsOut;
- /* In case an error happened from ma_device_write__null()... */
- if (result != MA_SUCCESS) {
- exitLoop = MA_TRUE;
- break;
- }
- }
+ #if defined(MA_APPLE_DESKTOP)
+ ma_get_AudioObject_uid(pDevice->pContext, pDevice->coreaudio.deviceObjectIDCapture, sizeof(pDevice->capture.id.coreaudio), pDevice->capture.id.coreaudio);
+
+ /*
+ If we are using the default device we'll need to listen for changes to the system's default device so we can seemlessly
+ switch the device in the background.
+ */
+ if (pConfig->capture.pDeviceID == NULL) {
+ ma_device__track__coreaudio(pDevice);
+ }
+ #endif
+ }
- /* Make sure we don't get stuck in the inner loop. */
- if (capturedDeviceFramesProcessed == 0) {
- break;
- }
+ /* Playback. */
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ ma_device_init_internal_data__coreaudio data;
+ data.allowNominalSampleRateChange = pConfig->coreaudio.allowNominalSampleRateChange;
+ data.formatIn = pDescriptorPlayback->format;
+ data.channelsIn = pDescriptorPlayback->channels;
+ data.sampleRateIn = pDescriptorPlayback->sampleRate;
+ MA_COPY_MEMORY(data.channelMapIn, pDescriptorPlayback->channelMap, sizeof(pDescriptorPlayback->channelMap));
+ data.shareMode = pDescriptorPlayback->shareMode;
+ data.performanceProfile = pConfig->performanceProfile;
- totalCapturedDeviceFramesProcessed += capturedDeviceFramesProcessed;
+ /* In full-duplex mode we want the playback buffer to be the same size as the capture buffer. */
+ if (pConfig->deviceType == ma_device_type_duplex) {
+ data.periodSizeInFramesIn = pDescriptorCapture->periodSizeInFrames;
+ data.periodsIn = pDescriptorCapture->periodCount;
+ data.registerStopEvent = MA_FALSE;
+ } else {
+ data.periodSizeInFramesIn = pDescriptorPlayback->periodSizeInFrames;
+ data.periodSizeInMillisecondsIn = pDescriptorPlayback->periodSizeInMilliseconds;
+ data.periodsIn = pDescriptorPlayback->periodCount;
+ data.registerStopEvent = MA_TRUE;
+ }
+
+ result = ma_device_init_internal__coreaudio(pDevice->pContext, ma_device_type_playback, pDescriptorPlayback->pDeviceID, &data, (void*)pDevice);
+ if (result != MA_SUCCESS) {
+ if (pConfig->deviceType == ma_device_type_duplex) {
+ ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitCapture);
+ if (pDevice->coreaudio.pAudioBufferList) {
+ ma_free(pDevice->coreaudio.pAudioBufferList, &pDevice->pContext->allocationCallbacks);
}
- } break;
+ }
+ return result;
+ }
- case ma_device_type_capture:
- case ma_device_type_loopback:
- {
- ma_uint32 periodSizeInFrames = pDevice->capture.internalPeriodSizeInFrames;
- ma_uint32 framesReadThisPeriod = 0;
- while (framesReadThisPeriod < periodSizeInFrames) {
- ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesReadThisPeriod;
- ma_uint32 framesProcessed;
- ma_uint32 framesToReadThisIteration = framesRemainingInPeriod;
- if (framesToReadThisIteration > capturedDeviceDataCapInFrames) {
- framesToReadThisIteration = capturedDeviceDataCapInFrames;
- }
+ pDevice->coreaudio.isDefaultPlaybackDevice = (pConfig->playback.pDeviceID == NULL);
+ #if defined(MA_APPLE_DESKTOP)
+ pDevice->coreaudio.deviceObjectIDPlayback = (ma_uint32)data.deviceObjectID;
+ #endif
+ pDevice->coreaudio.audioUnitPlayback = (ma_ptr)data.audioUnit;
+ pDevice->coreaudio.originalPeriodSizeInFrames = pDescriptorPlayback->periodSizeInFrames;
+ pDevice->coreaudio.originalPeriodSizeInMilliseconds = pDescriptorPlayback->periodSizeInMilliseconds;
+ pDevice->coreaudio.originalPeriods = pDescriptorPlayback->periodCount;
+ pDevice->coreaudio.originalPerformanceProfile = pConfig->performanceProfile;
- result = pDevice->pContext->callbacks.onDeviceRead(pDevice, capturedDeviceData, framesToReadThisIteration, &framesProcessed);
- if (result != MA_SUCCESS) {
- exitLoop = MA_TRUE;
- break;
- }
+ pDescriptorPlayback->format = data.formatOut;
+ pDescriptorPlayback->channels = data.channelsOut;
+ pDescriptorPlayback->sampleRate = data.sampleRateOut;
+ MA_COPY_MEMORY(pDescriptorPlayback->channelMap, data.channelMapOut, sizeof(data.channelMapOut));
+ pDescriptorPlayback->periodSizeInFrames = data.periodSizeInFramesOut;
+ pDescriptorPlayback->periodCount = data.periodsOut;
+
+ #if defined(MA_APPLE_DESKTOP)
+ ma_get_AudioObject_uid(pDevice->pContext, pDevice->coreaudio.deviceObjectIDPlayback, sizeof(pDevice->playback.id.coreaudio), pDevice->playback.id.coreaudio);
+
+ /*
+ If we are using the default device we'll need to listen for changes to the system's default device so we can seemlessly
+ switch the device in the background.
+ */
+ if (pDescriptorPlayback->pDeviceID == NULL && (pConfig->deviceType != ma_device_type_duplex || pDescriptorCapture->pDeviceID != NULL)) {
+ ma_device__track__coreaudio(pDevice);
+ }
+ #endif
+ }
- /* Make sure we don't get stuck in the inner loop. */
- if (framesProcessed == 0) {
- break;
- }
- ma_device__send_frames_to_client(pDevice, framesProcessed, capturedDeviceData);
- framesReadThisPeriod += framesProcessed;
- }
- } break;
+ /*
+ When stopping the device, a callback is called on another thread. We need to wait for this callback
+ before returning from ma_device_stop(). This event is used for this.
+ */
+ ma_event_init(&pDevice->coreaudio.stopEvent);
- case ma_device_type_playback:
- {
- /* We write in chunks of the period size, but use a stack allocated buffer for the intermediary. */
- ma_uint32 periodSizeInFrames = pDevice->playback.internalPeriodSizeInFrames;
- ma_uint32 framesWrittenThisPeriod = 0;
- while (framesWrittenThisPeriod < periodSizeInFrames) {
- ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesWrittenThisPeriod;
- ma_uint32 framesProcessed;
- ma_uint32 framesToWriteThisIteration = framesRemainingInPeriod;
- if (framesToWriteThisIteration > playbackDeviceDataCapInFrames) {
- framesToWriteThisIteration = playbackDeviceDataCapInFrames;
- }
+ /*
+ We need to detect when a route has changed so we can update the data conversion pipeline accordingly. This is done
+ differently on non-Desktop Apple platforms.
+ */
+#if defined(MA_APPLE_MOBILE)
+ pDevice->coreaudio.pNotificationHandler = (MA_BRIDGE_RETAINED void*)[[ma_ios_notification_handler alloc] init:pDevice];
+#endif
- ma_device__read_frames_from_client(pDevice, framesToWriteThisIteration, playbackDeviceData);
+ return MA_SUCCESS;
+}
- result = pDevice->pContext->callbacks.onDeviceWrite(pDevice, playbackDeviceData, framesToWriteThisIteration, &framesProcessed);
- if (result != MA_SUCCESS) {
- exitLoop = MA_TRUE;
- break;
- }
- /* Make sure we don't get stuck in the inner loop. */
- if (framesProcessed == 0) {
- break;
- }
+static ma_result ma_device_start__coreaudio(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
- framesWrittenThisPeriod += framesProcessed;
- }
- } break;
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ OSStatus status = ((ma_AudioOutputUnitStart_proc)pDevice->pContext->coreaudio.AudioOutputUnitStart)((AudioUnit)pDevice->coreaudio.audioUnitCapture);
+ if (status != noErr) {
+ return ma_result_from_OSStatus(status);
+ }
+ }
- /* Should never get here. */
- default: break;
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ OSStatus status = ((ma_AudioOutputUnitStart_proc)pDevice->pContext->coreaudio.AudioOutputUnitStart)((AudioUnit)pDevice->coreaudio.audioUnitPlayback);
+ if (status != noErr) {
+ if (pDevice->type == ma_device_type_duplex) {
+ ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitCapture);
+ }
+ return ma_result_from_OSStatus(status);
}
}
- return result;
+ return MA_SUCCESS;
}
+static ma_result ma_device_stop__coreaudio(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
+ /* It's not clear from the documentation whether or not AudioOutputUnitStop() actually drains the device or not. */
-/*******************************************************************************
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ OSStatus status = ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitCapture);
+ if (status != noErr) {
+ return ma_result_from_OSStatus(status);
+ }
+ }
-Null Backend
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ OSStatus status = ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitPlayback);
+ if (status != noErr) {
+ return ma_result_from_OSStatus(status);
+ }
+ }
-*******************************************************************************/
-#ifdef MA_HAS_NULL
+ /* We need to wait for the callback to finish before returning. */
+ ma_event_wait(&pDevice->coreaudio.stopEvent);
+ return MA_SUCCESS;
+}
-#define MA_DEVICE_OP_NONE__NULL 0
-#define MA_DEVICE_OP_START__NULL 1
-#define MA_DEVICE_OP_SUSPEND__NULL 2
-#define MA_DEVICE_OP_KILL__NULL 3
-static ma_thread_result MA_THREADCALL ma_device_thread__null(void* pData)
+static ma_result ma_context_uninit__coreaudio(ma_context* pContext)
{
- ma_device* pDevice = (ma_device*)pData;
- MA_ASSERT(pDevice != NULL);
-
- for (;;) { /* Keep the thread alive until the device is uninitialized. */
- ma_uint32 operation;
-
- /* Wait for an operation to be requested. */
- ma_event_wait(&pDevice->null_device.operationEvent);
-
- /* At this point an event should have been triggered. */
- operation = pDevice->null_device.operation;
-
- /* Starting the device needs to put the thread into a loop. */
- if (operation == MA_DEVICE_OP_START__NULL) {
- /* Reset the timer just in case. */
- ma_timer_init(&pDevice->null_device.timer);
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pContext->backend == ma_backend_coreaudio);
- /* Getting here means a suspend or kill operation has been requested. */
- pDevice->null_device.operationResult = MA_SUCCESS;
- ma_event_signal(&pDevice->null_device.operationCompletionEvent);
- ma_semaphore_release(&pDevice->null_device.operationSemaphore);
- continue;
+#if defined(MA_APPLE_MOBILE)
+ if (!pContext->coreaudio.noAudioSessionDeactivate) {
+ if (![[AVAudioSession sharedInstance] setActive:false error:nil]) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "Failed to deactivate audio session.");
+ return MA_FAILED_TO_INIT_BACKEND;
}
+ }
+#endif
- /* Suspending the device means we need to stop the timer and just continue the loop. */
- if (operation == MA_DEVICE_OP_SUSPEND__NULL) {
- /* We need to add the current run time to the prior run time, then reset the timer. */
- pDevice->null_device.priorRunTime += ma_timer_get_time_in_seconds(&pDevice->null_device.timer);
- ma_timer_init(&pDevice->null_device.timer);
+#if !defined(MA_NO_RUNTIME_LINKING) && !defined(MA_APPLE_MOBILE)
+ ma_dlclose(pContext, pContext->coreaudio.hAudioUnit);
+ ma_dlclose(pContext, pContext->coreaudio.hCoreAudio);
+ ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation);
+#endif
- /* We're done. */
- pDevice->null_device.operationResult = MA_SUCCESS;
- ma_event_signal(&pDevice->null_device.operationCompletionEvent);
- ma_semaphore_release(&pDevice->null_device.operationSemaphore);
- continue;
- }
+#if !defined(MA_APPLE_MOBILE)
+ ma_context__uninit_device_tracking__coreaudio(pContext);
+#endif
- /* Killing the device means we need to get out of this loop so that this thread can terminate. */
- if (operation == MA_DEVICE_OP_KILL__NULL) {
- pDevice->null_device.operationResult = MA_SUCCESS;
- ma_event_signal(&pDevice->null_device.operationCompletionEvent);
- ma_semaphore_release(&pDevice->null_device.operationSemaphore);
- break;
- }
+ (void)pContext;
+ return MA_SUCCESS;
+}
- /* Getting a signal on a "none" operation probably means an error. Return invalid operation. */
- if (operation == MA_DEVICE_OP_NONE__NULL) {
- MA_ASSERT(MA_FALSE); /* <-- Trigger this in debug mode to ensure developers are aware they're doing something wrong (or there's a bug in a miniaudio). */
- pDevice->null_device.operationResult = MA_INVALID_OPERATION;
- ma_event_signal(&pDevice->null_device.operationCompletionEvent);
- ma_semaphore_release(&pDevice->null_device.operationSemaphore);
- continue; /* Continue the loop. Don't terminate. */
- }
- }
+#if defined(MA_APPLE_MOBILE) && defined(__IPHONE_12_0)
+static AVAudioSessionCategory ma_to_AVAudioSessionCategory(ma_ios_session_category category)
+{
+ /* The "default" and "none" categories are treated different and should not be used as an input into this function. */
+ MA_ASSERT(category != ma_ios_session_category_default);
+ MA_ASSERT(category != ma_ios_session_category_none);
- return (ma_thread_result)0;
+ switch (category) {
+ case ma_ios_session_category_ambient: return AVAudioSessionCategoryAmbient;
+ case ma_ios_session_category_solo_ambient: return AVAudioSessionCategorySoloAmbient;
+ case ma_ios_session_category_playback: return AVAudioSessionCategoryPlayback;
+ case ma_ios_session_category_record: return AVAudioSessionCategoryRecord;
+ case ma_ios_session_category_play_and_record: return AVAudioSessionCategoryPlayAndRecord;
+ case ma_ios_session_category_multi_route: return AVAudioSessionCategoryMultiRoute;
+ case ma_ios_session_category_none: return AVAudioSessionCategoryAmbient;
+ case ma_ios_session_category_default: return AVAudioSessionCategoryAmbient;
+ default: return AVAudioSessionCategoryAmbient;
+ }
}
+#endif
-static ma_result ma_device_do_operation__null(ma_device* pDevice, ma_uint32 operation)
+static ma_result ma_context_init__coreaudio(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
{
+#if !defined(MA_APPLE_MOBILE)
ma_result result;
+#endif
- /*
- TODO: Need to review this and consider just using mutual exclusion. I think the original motivation
- for this was to just post the event to a queue and return immediately, but that has since changed
- and now this function is synchronous. I think this can be simplified to just use a mutex.
- */
+ MA_ASSERT(pConfig != NULL);
+ MA_ASSERT(pContext != NULL);
- /*
- The first thing to do is wait for an operation slot to become available. We only have a single slot for this, but we could extend this later
- to support queing of operations.
- */
- result = ma_semaphore_wait(&pDevice->null_device.operationSemaphore);
- if (result != MA_SUCCESS) {
- return result; /* Failed to wait for the event. */
- }
+#if defined(MA_APPLE_MOBILE)
+ @autoreleasepool {
+ AVAudioSession* pAudioSession = [AVAudioSession sharedInstance];
+ AVAudioSessionCategoryOptions options = pConfig->coreaudio.sessionCategoryOptions;
- /*
- When we get here it means the background thread is not referencing the operation code and it can be changed. After changing this we need to
- signal an event to the worker thread to let it know that it can start work.
- */
- pDevice->null_device.operation = operation;
+ MA_ASSERT(pAudioSession != NULL);
- /* Once the operation code has been set, the worker thread can start work. */
- if (ma_event_signal(&pDevice->null_device.operationEvent) != MA_SUCCESS) {
- return MA_ERROR;
- }
+ if (pConfig->coreaudio.sessionCategory == ma_ios_session_category_default) {
+ /*
+ I'm going to use trial and error to determine our default session category. First we'll try PlayAndRecord. If that fails
+ we'll try Playback and if that fails we'll try record. If all of these fail we'll just not set the category.
+ */
+ #if !defined(MA_APPLE_TV) && !defined(MA_APPLE_WATCH)
+ options |= AVAudioSessionCategoryOptionDefaultToSpeaker;
+ #endif
- /* We want everything to be synchronous so we're going to wait for the worker thread to complete it's operation. */
- if (ma_event_wait(&pDevice->null_device.operationCompletionEvent) != MA_SUCCESS) {
- return MA_ERROR;
- }
+ if ([pAudioSession setCategory: AVAudioSessionCategoryPlayAndRecord withOptions:options error:nil]) {
+ /* Using PlayAndRecord */
+ } else if ([pAudioSession setCategory: AVAudioSessionCategoryPlayback withOptions:options error:nil]) {
+ /* Using Playback */
+ } else if ([pAudioSession setCategory: AVAudioSessionCategoryRecord withOptions:options error:nil]) {
+ /* Using Record */
+ } else {
+ /* Leave as default? */
+ }
+ } else {
+ if (pConfig->coreaudio.sessionCategory != ma_ios_session_category_none) {
+ #if defined(__IPHONE_12_0)
+ if (![pAudioSession setCategory: ma_to_AVAudioSessionCategory(pConfig->coreaudio.sessionCategory) withOptions:options error:nil]) {
+ return MA_INVALID_OPERATION; /* Failed to set session category. */
+ }
+ #else
+ /* Ignore the session category on version 11 and older, but post a warning. */
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_WARNING, "Session category only supported in iOS 12 and newer.");
+ #endif
+ }
+ }
- return pDevice->null_device.operationResult;
-}
+ if (!pConfig->coreaudio.noAudioSessionActivate) {
+ if (![pAudioSession setActive:true error:nil]) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "Failed to activate audio session.");
+ return MA_FAILED_TO_INIT_BACKEND;
+ }
+ }
+ }
+#endif
-static ma_uint64 ma_device_get_total_run_time_in_frames__null(ma_device* pDevice)
-{
- ma_uint32 internalSampleRate;
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- internalSampleRate = pDevice->capture.internalSampleRate;
- } else {
- internalSampleRate = pDevice->playback.internalSampleRate;
+#if !defined(MA_NO_RUNTIME_LINKING) && !defined(MA_APPLE_MOBILE)
+ pContext->coreaudio.hCoreFoundation = ma_dlopen(pContext, "CoreFoundation.framework/CoreFoundation");
+ if (pContext->coreaudio.hCoreFoundation == NULL) {
+ return MA_API_NOT_FOUND;
}
- return (ma_uint64)((pDevice->null_device.priorRunTime + ma_timer_get_time_in_seconds(&pDevice->null_device.timer)) * internalSampleRate);
-}
+ pContext->coreaudio.CFStringGetCString = ma_dlsym(pContext, pContext->coreaudio.hCoreFoundation, "CFStringGetCString");
+ pContext->coreaudio.CFRelease = ma_dlsym(pContext, pContext->coreaudio.hCoreFoundation, "CFRelease");
-static ma_result ma_context_enumerate_devices__null(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
-{
- ma_bool32 cbResult = MA_TRUE;
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(callback != NULL);
+ pContext->coreaudio.hCoreAudio = ma_dlopen(pContext, "CoreAudio.framework/CoreAudio");
+ if (pContext->coreaudio.hCoreAudio == NULL) {
+ ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation);
+ return MA_API_NOT_FOUND;
+ }
- /* Playback. */
- if (cbResult) {
- ma_device_info deviceInfo;
- MA_ZERO_OBJECT(&deviceInfo);
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), "NULL Playback Device", (size_t)-1);
- deviceInfo.isDefault = MA_TRUE; /* Only one playback and capture device for the null backend, so might as well mark as default. */
- cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
+ pContext->coreaudio.AudioObjectGetPropertyData = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectGetPropertyData");
+ pContext->coreaudio.AudioObjectGetPropertyDataSize = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectGetPropertyDataSize");
+ pContext->coreaudio.AudioObjectSetPropertyData = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectSetPropertyData");
+ pContext->coreaudio.AudioObjectAddPropertyListener = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectAddPropertyListener");
+ pContext->coreaudio.AudioObjectRemovePropertyListener = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectRemovePropertyListener");
+
+ /*
+ It looks like Apple has moved some APIs from AudioUnit into AudioToolbox on more recent versions of macOS. They are still
+ defined in AudioUnit, but just in case they decide to remove them from there entirely I'm going to implement a fallback.
+ The way it'll work is that it'll first try AudioUnit, and if the required symbols are not present there we'll fall back to
+ AudioToolbox.
+ */
+ pContext->coreaudio.hAudioUnit = ma_dlopen(pContext, "AudioUnit.framework/AudioUnit");
+ if (pContext->coreaudio.hAudioUnit == NULL) {
+ ma_dlclose(pContext, pContext->coreaudio.hCoreAudio);
+ ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation);
+ return MA_API_NOT_FOUND;
}
- /* Capture. */
- if (cbResult) {
- ma_device_info deviceInfo;
- MA_ZERO_OBJECT(&deviceInfo);
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), "NULL Capture Device", (size_t)-1);
- deviceInfo.isDefault = MA_TRUE; /* Only one playback and capture device for the null backend, so might as well mark as default. */
- cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
+ if (ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioComponentFindNext") == NULL) {
+ /* Couldn't find the required symbols in AudioUnit, so fall back to AudioToolbox. */
+ ma_dlclose(pContext, pContext->coreaudio.hAudioUnit);
+ pContext->coreaudio.hAudioUnit = ma_dlopen(pContext, "AudioToolbox.framework/AudioToolbox");
+ if (pContext->coreaudio.hAudioUnit == NULL) {
+ ma_dlclose(pContext, pContext->coreaudio.hCoreAudio);
+ ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation);
+ return MA_API_NOT_FOUND;
+ }
}
- (void)cbResult; /* Silence a static analysis warning. */
+ pContext->coreaudio.AudioComponentFindNext = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioComponentFindNext");
+ pContext->coreaudio.AudioComponentInstanceDispose = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioComponentInstanceDispose");
+ pContext->coreaudio.AudioComponentInstanceNew = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioComponentInstanceNew");
+ pContext->coreaudio.AudioOutputUnitStart = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioOutputUnitStart");
+ pContext->coreaudio.AudioOutputUnitStop = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioOutputUnitStop");
+ pContext->coreaudio.AudioUnitAddPropertyListener = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitAddPropertyListener");
+ pContext->coreaudio.AudioUnitGetPropertyInfo = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitGetPropertyInfo");
+ pContext->coreaudio.AudioUnitGetProperty = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitGetProperty");
+ pContext->coreaudio.AudioUnitSetProperty = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitSetProperty");
+ pContext->coreaudio.AudioUnitInitialize = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitInitialize");
+ pContext->coreaudio.AudioUnitRender = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitRender");
+#else
+ pContext->coreaudio.CFStringGetCString = (ma_proc)CFStringGetCString;
+ pContext->coreaudio.CFRelease = (ma_proc)CFRelease;
- return MA_SUCCESS;
-}
+ #if defined(MA_APPLE_DESKTOP)
+ pContext->coreaudio.AudioObjectGetPropertyData = (ma_proc)AudioObjectGetPropertyData;
+ pContext->coreaudio.AudioObjectGetPropertyDataSize = (ma_proc)AudioObjectGetPropertyDataSize;
+ pContext->coreaudio.AudioObjectSetPropertyData = (ma_proc)AudioObjectSetPropertyData;
+ pContext->coreaudio.AudioObjectAddPropertyListener = (ma_proc)AudioObjectAddPropertyListener;
+ pContext->coreaudio.AudioObjectRemovePropertyListener = (ma_proc)AudioObjectRemovePropertyListener;
+ #endif
-static ma_result ma_context_get_device_info__null(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
-{
- MA_ASSERT(pContext != NULL);
+ pContext->coreaudio.AudioComponentFindNext = (ma_proc)AudioComponentFindNext;
+ pContext->coreaudio.AudioComponentInstanceDispose = (ma_proc)AudioComponentInstanceDispose;
+ pContext->coreaudio.AudioComponentInstanceNew = (ma_proc)AudioComponentInstanceNew;
+ pContext->coreaudio.AudioOutputUnitStart = (ma_proc)AudioOutputUnitStart;
+ pContext->coreaudio.AudioOutputUnitStop = (ma_proc)AudioOutputUnitStop;
+ pContext->coreaudio.AudioUnitAddPropertyListener = (ma_proc)AudioUnitAddPropertyListener;
+ pContext->coreaudio.AudioUnitGetPropertyInfo = (ma_proc)AudioUnitGetPropertyInfo;
+ pContext->coreaudio.AudioUnitGetProperty = (ma_proc)AudioUnitGetProperty;
+ pContext->coreaudio.AudioUnitSetProperty = (ma_proc)AudioUnitSetProperty;
+ pContext->coreaudio.AudioUnitInitialize = (ma_proc)AudioUnitInitialize;
+ pContext->coreaudio.AudioUnitRender = (ma_proc)AudioUnitRender;
+#endif
- if (pDeviceID != NULL && pDeviceID->nullbackend != 0) {
- return MA_NO_DEVICE; /* Don't know the device. */
+ /* Audio component. */
+ {
+ AudioComponentDescription desc;
+ desc.componentType = kAudioUnitType_Output;
+ #if defined(MA_APPLE_DESKTOP)
+ desc.componentSubType = kAudioUnitSubType_HALOutput;
+ #else
+ desc.componentSubType = kAudioUnitSubType_RemoteIO;
+ #endif
+ desc.componentManufacturer = kAudioUnitManufacturer_Apple;
+ desc.componentFlags = 0;
+ desc.componentFlagsMask = 0;
+
+ pContext->coreaudio.component = ((ma_AudioComponentFindNext_proc)pContext->coreaudio.AudioComponentFindNext)(NULL, &desc);
+ if (pContext->coreaudio.component == NULL) {
+ #if !defined(MA_NO_RUNTIME_LINKING) && !defined(MA_APPLE_MOBILE)
+ ma_dlclose(pContext, pContext->coreaudio.hAudioUnit);
+ ma_dlclose(pContext, pContext->coreaudio.hCoreAudio);
+ ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation);
+ #endif
+ return MA_FAILED_TO_INIT_BACKEND;
+ }
}
- /* Name / Description */
- if (deviceType == ma_device_type_playback) {
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), "NULL Playback Device", (size_t)-1);
- } else {
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), "NULL Capture Device", (size_t)-1);
+#if !defined(MA_APPLE_MOBILE)
+ result = ma_context__init_device_tracking__coreaudio(pContext);
+ if (result != MA_SUCCESS) {
+ #if !defined(MA_NO_RUNTIME_LINKING) && !defined(MA_APPLE_MOBILE)
+ ma_dlclose(pContext, pContext->coreaudio.hAudioUnit);
+ ma_dlclose(pContext, pContext->coreaudio.hCoreAudio);
+ ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation);
+ #endif
+ return result;
}
+#endif
- pDeviceInfo->isDefault = MA_TRUE; /* Only one playback and capture device for the null backend, so might as well mark as default. */
+ pContext->coreaudio.noAudioSessionDeactivate = pConfig->coreaudio.noAudioSessionDeactivate;
- /* Support everything on the null backend. */
- pDeviceInfo->nativeDataFormats[0].format = ma_format_unknown;
- pDeviceInfo->nativeDataFormats[0].channels = 0;
- pDeviceInfo->nativeDataFormats[0].sampleRate = 0;
- pDeviceInfo->nativeDataFormats[0].flags = 0;
- pDeviceInfo->nativeDataFormatCount = 1;
+ pCallbacks->onContextInit = ma_context_init__coreaudio;
+ pCallbacks->onContextUninit = ma_context_uninit__coreaudio;
+ pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__coreaudio;
+ pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__coreaudio;
+ pCallbacks->onDeviceInit = ma_device_init__coreaudio;
+ pCallbacks->onDeviceUninit = ma_device_uninit__coreaudio;
+ pCallbacks->onDeviceStart = ma_device_start__coreaudio;
+ pCallbacks->onDeviceStop = ma_device_stop__coreaudio;
+ pCallbacks->onDeviceRead = NULL;
+ pCallbacks->onDeviceWrite = NULL;
+ pCallbacks->onDeviceDataLoop = NULL;
- (void)pContext;
return MA_SUCCESS;
}
+#endif /* Core Audio */
-static ma_result ma_device_uninit__null(ma_device* pDevice)
-{
- MA_ASSERT(pDevice != NULL);
- /* Keep it clean and wait for the device thread to finish before returning. */
- ma_device_do_operation__null(pDevice, MA_DEVICE_OP_KILL__NULL);
+/******************************************************************************
- /* Wait for the thread to finish before continuing. */
- ma_thread_wait(&pDevice->null_device.deviceThread);
+sndio Backend
- /* At this point the loop in the device thread is as good as terminated so we can uninitialize our events. */
- ma_semaphore_uninit(&pDevice->null_device.operationSemaphore);
- ma_event_uninit(&pDevice->null_device.operationCompletionEvent);
- ma_event_uninit(&pDevice->null_device.operationEvent);
+******************************************************************************/
+#ifdef MA_HAS_SNDIO
+#include
- return MA_SUCCESS;
-}
+/*
+Only supporting OpenBSD. This did not work very well at all on FreeBSD when I tried it. Not sure if this is due
+to miniaudio's implementation or if it's some kind of system configuration issue, but basically the default device
+just doesn't emit any sound, or at times you'll hear tiny pieces. I will consider enabling this when there's
+demand for it or if I can get it tested and debugged more thoroughly.
+*/
+#if 0
+#if defined(__NetBSD__) || defined(__OpenBSD__)
+#include
+#endif
+#if defined(__FreeBSD__) || defined(__DragonFly__)
+#include
+#endif
+#endif
-static ma_result ma_device_init__null(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+#define MA_SIO_DEVANY "default"
+#define MA_SIO_PLAY 1
+#define MA_SIO_REC 2
+#define MA_SIO_NENC 8
+#define MA_SIO_NCHAN 8
+#define MA_SIO_NRATE 16
+#define MA_SIO_NCONF 4
+
+struct ma_sio_hdl; /* <-- Opaque */
+
+struct ma_sio_par
{
- ma_result result;
+ unsigned int bits;
+ unsigned int bps;
+ unsigned int sig;
+ unsigned int le;
+ unsigned int msb;
+ unsigned int rchan;
+ unsigned int pchan;
+ unsigned int rate;
+ unsigned int bufsz;
+ unsigned int xrun;
+ unsigned int round;
+ unsigned int appbufsz;
+ int __pad[3];
+ unsigned int __magic;
+};
- MA_ASSERT(pDevice != NULL);
+struct ma_sio_enc
+{
+ unsigned int bits;
+ unsigned int bps;
+ unsigned int sig;
+ unsigned int le;
+ unsigned int msb;
+};
- MA_ZERO_OBJECT(&pDevice->null_device);
+struct ma_sio_conf
+{
+ unsigned int enc;
+ unsigned int rchan;
+ unsigned int pchan;
+ unsigned int rate;
+};
- if (pConfig->deviceType == ma_device_type_loopback) {
- return MA_DEVICE_TYPE_NOT_SUPPORTED;
- }
+struct ma_sio_cap
+{
+ struct ma_sio_enc enc[MA_SIO_NENC];
+ unsigned int rchan[MA_SIO_NCHAN];
+ unsigned int pchan[MA_SIO_NCHAN];
+ unsigned int rate[MA_SIO_NRATE];
+ int __pad[7];
+ unsigned int nconf;
+ struct ma_sio_conf confs[MA_SIO_NCONF];
+};
- /* The null backend supports everything exactly as we specify it. */
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
- pDescriptorCapture->format = (pDescriptorCapture->format != ma_format_unknown) ? pDescriptorCapture->format : MA_DEFAULT_FORMAT;
- pDescriptorCapture->channels = (pDescriptorCapture->channels != 0) ? pDescriptorCapture->channels : MA_DEFAULT_CHANNELS;
- pDescriptorCapture->sampleRate = (pDescriptorCapture->sampleRate != 0) ? pDescriptorCapture->sampleRate : MA_DEFAULT_SAMPLE_RATE;
+typedef struct ma_sio_hdl* (* ma_sio_open_proc) (const char*, unsigned int, int);
+typedef void (* ma_sio_close_proc) (struct ma_sio_hdl*);
+typedef int (* ma_sio_setpar_proc) (struct ma_sio_hdl*, struct ma_sio_par*);
+typedef int (* ma_sio_getpar_proc) (struct ma_sio_hdl*, struct ma_sio_par*);
+typedef int (* ma_sio_getcap_proc) (struct ma_sio_hdl*, struct ma_sio_cap*);
+typedef size_t (* ma_sio_write_proc) (struct ma_sio_hdl*, const void*, size_t);
+typedef size_t (* ma_sio_read_proc) (struct ma_sio_hdl*, void*, size_t);
+typedef int (* ma_sio_start_proc) (struct ma_sio_hdl*);
+typedef int (* ma_sio_stop_proc) (struct ma_sio_hdl*);
+typedef int (* ma_sio_initpar_proc)(struct ma_sio_par*);
- if (pDescriptorCapture->channelMap[0] == MA_CHANNEL_NONE) {
- ma_get_standard_channel_map(ma_standard_channel_map_default, pDescriptorCapture->channels, pDescriptorCapture->channelMap);
+static ma_uint32 ma_get_standard_sample_rate_priority_index__sndio(ma_uint32 sampleRate) /* Lower = higher priority */
+{
+ ma_uint32 i;
+ for (i = 0; i < ma_countof(g_maStandardSampleRatePriorities); ++i) {
+ if (g_maStandardSampleRatePriorities[i] == sampleRate) {
+ return i;
}
-
- pDescriptorCapture->periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptorCapture, pDescriptorCapture->sampleRate, pConfig->performanceProfile);
}
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- pDescriptorPlayback->format = (pDescriptorPlayback->format != ma_format_unknown) ? pDescriptorPlayback->format : MA_DEFAULT_FORMAT;
- pDescriptorPlayback->channels = (pDescriptorPlayback->channels != 0) ? pDescriptorPlayback->channels : MA_DEFAULT_CHANNELS;
- pDescriptorPlayback->sampleRate = (pDescriptorPlayback->sampleRate != 0) ? pDescriptorPlayback->sampleRate : MA_DEFAULT_SAMPLE_RATE;
-
- if (pDescriptorPlayback->channelMap[0] == MA_CHANNEL_NONE) {
- ma_get_standard_channel_map(ma_standard_channel_map_default, pDescriptorPlayback->channels, pDescriptorPlayback->channelMap);
- }
+ return (ma_uint32)-1;
+}
- pDescriptorPlayback->periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptorPlayback, pDescriptorPlayback->sampleRate, pConfig->performanceProfile);
+static ma_format ma_format_from_sio_enc__sndio(unsigned int bits, unsigned int bps, unsigned int sig, unsigned int le, unsigned int msb)
+{
+ /* We only support native-endian right now. */
+ if ((ma_is_little_endian() && le == 0) || (ma_is_big_endian() && le == 1)) {
+ return ma_format_unknown;
}
- /*
- In order to get timing right, we need to create a thread that does nothing but keeps track of the timer. This timer is started when the
- first period is "written" to it, and then stopped in ma_device_stop__null().
- */
- result = ma_event_init(&pDevice->null_device.operationEvent);
- if (result != MA_SUCCESS) {
- return result;
+ if (bits == 8 && bps == 1 && sig == 0) {
+ return ma_format_u8;
}
-
- result = ma_event_init(&pDevice->null_device.operationCompletionEvent);
- if (result != MA_SUCCESS) {
- return result;
+ if (bits == 16 && bps == 2 && sig == 1) {
+ return ma_format_s16;
}
-
- result = ma_semaphore_init(1, &pDevice->null_device.operationSemaphore); /* <-- It's important that the initial value is set to 1. */
- if (result != MA_SUCCESS) {
- return result;
+ if (bits == 24 && bps == 3 && sig == 1) {
+ return ma_format_s24;
}
-
- result = ma_thread_create(&pDevice->null_device.deviceThread, pDevice->pContext->threadPriority, 0, ma_device_thread__null, pDevice, &pDevice->pContext->allocationCallbacks);
- if (result != MA_SUCCESS) {
- return result;
+ if (bits == 24 && bps == 4 && sig == 1 && msb == 0) {
+ /*return ma_format_s24_32;*/
+ }
+ if (bits == 32 && bps == 4 && sig == 1) {
+ return ma_format_s32;
}
- return MA_SUCCESS;
+ return ma_format_unknown;
}
-static ma_result ma_device_start__null(ma_device* pDevice)
+static ma_format ma_find_best_format_from_sio_cap__sndio(struct ma_sio_cap* caps)
{
- MA_ASSERT(pDevice != NULL);
+ ma_format bestFormat;
+ unsigned int iConfig;
- ma_device_do_operation__null(pDevice, MA_DEVICE_OP_START__NULL);
+ MA_ASSERT(caps != NULL);
- c89atomic_exchange_32(&pDevice->null_device.isStarted, MA_TRUE);
- return MA_SUCCESS;
-}
+ bestFormat = ma_format_unknown;
+ for (iConfig = 0; iConfig < caps->nconf; iConfig += 1) {
+ unsigned int iEncoding;
+ for (iEncoding = 0; iEncoding < MA_SIO_NENC; iEncoding += 1) {
+ unsigned int bits;
+ unsigned int bps;
+ unsigned int sig;
+ unsigned int le;
+ unsigned int msb;
+ ma_format format;
-static ma_result ma_device_stop__null(ma_device* pDevice)
-{
- MA_ASSERT(pDevice != NULL);
+ if ((caps->confs[iConfig].enc & (1UL << iEncoding)) == 0) {
+ continue;
+ }
- ma_device_do_operation__null(pDevice, MA_DEVICE_OP_SUSPEND__NULL);
+ bits = caps->enc[iEncoding].bits;
+ bps = caps->enc[iEncoding].bps;
+ sig = caps->enc[iEncoding].sig;
+ le = caps->enc[iEncoding].le;
+ msb = caps->enc[iEncoding].msb;
+ format = ma_format_from_sio_enc__sndio(bits, bps, sig, le, msb);
+ if (format == ma_format_unknown) {
+ continue; /* Format not supported. */
+ }
- c89atomic_exchange_32(&pDevice->null_device.isStarted, MA_FALSE);
- return MA_SUCCESS;
+ if (bestFormat == ma_format_unknown) {
+ bestFormat = format;
+ } else {
+ if (ma_get_format_priority_index(bestFormat) > ma_get_format_priority_index(format)) { /* <-- Lower = better. */
+ bestFormat = format;
+ }
+ }
+ }
+ }
+
+ return bestFormat;
}
-static ma_result ma_device_write__null(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten)
+static ma_uint32 ma_find_best_channels_from_sio_cap__sndio(struct ma_sio_cap* caps, ma_device_type deviceType, ma_format requiredFormat)
{
- ma_result result = MA_SUCCESS;
- ma_uint32 totalPCMFramesProcessed;
- ma_bool32 wasStartedOnEntry;
-
- if (pFramesWritten != NULL) {
- *pFramesWritten = 0;
- }
+ ma_uint32 maxChannels;
+ unsigned int iConfig;
- wasStartedOnEntry = c89atomic_load_32(&pDevice->null_device.isStarted);
+ MA_ASSERT(caps != NULL);
+ MA_ASSERT(requiredFormat != ma_format_unknown);
- /* Keep going until everything has been read. */
- totalPCMFramesProcessed = 0;
- while (totalPCMFramesProcessed < frameCount) {
- ma_uint64 targetFrame;
+ /* Just pick whatever configuration has the most channels. */
+ maxChannels = 0;
+ for (iConfig = 0; iConfig < caps->nconf; iConfig += 1) {
+ /* The encoding should be of requiredFormat. */
+ unsigned int iEncoding;
+ for (iEncoding = 0; iEncoding < MA_SIO_NENC; iEncoding += 1) {
+ unsigned int iChannel;
+ unsigned int bits;
+ unsigned int bps;
+ unsigned int sig;
+ unsigned int le;
+ unsigned int msb;
+ ma_format format;
- /* If there are any frames remaining in the current period, consume those first. */
- if (pDevice->null_device.currentPeriodFramesRemainingPlayback > 0) {
- ma_uint32 framesRemaining = (frameCount - totalPCMFramesProcessed);
- ma_uint32 framesToProcess = pDevice->null_device.currentPeriodFramesRemainingPlayback;
- if (framesToProcess > framesRemaining) {
- framesToProcess = framesRemaining;
+ if ((caps->confs[iConfig].enc & (1UL << iEncoding)) == 0) {
+ continue;
}
- /* We don't actually do anything with pPCMFrames, so just mark it as unused to prevent a warning. */
- (void)pPCMFrames;
-
- pDevice->null_device.currentPeriodFramesRemainingPlayback -= framesToProcess;
- totalPCMFramesProcessed += framesToProcess;
- }
+ bits = caps->enc[iEncoding].bits;
+ bps = caps->enc[iEncoding].bps;
+ sig = caps->enc[iEncoding].sig;
+ le = caps->enc[iEncoding].le;
+ msb = caps->enc[iEncoding].msb;
+ format = ma_format_from_sio_enc__sndio(bits, bps, sig, le, msb);
+ if (format != requiredFormat) {
+ continue;
+ }
- /* If we've consumed the current period we'll need to mark it as such an ensure the device is started if it's not already. */
- if (pDevice->null_device.currentPeriodFramesRemainingPlayback == 0) {
- pDevice->null_device.currentPeriodFramesRemainingPlayback = 0;
+ /* Getting here means the format is supported. Iterate over each channel count and grab the biggest one. */
+ for (iChannel = 0; iChannel < MA_SIO_NCHAN; iChannel += 1) {
+ unsigned int chan = 0;
+ unsigned int channels;
- if (!c89atomic_load_32(&pDevice->null_device.isStarted) && !wasStartedOnEntry) {
- result = ma_device_start__null(pDevice);
- if (result != MA_SUCCESS) {
- break;
+ if (deviceType == ma_device_type_playback) {
+ chan = caps->confs[iConfig].pchan;
+ } else {
+ chan = caps->confs[iConfig].rchan;
}
- }
- }
-
- /* If we've consumed the whole buffer we can return now. */
- MA_ASSERT(totalPCMFramesProcessed <= frameCount);
- if (totalPCMFramesProcessed == frameCount) {
- break;
- }
- /* Getting here means we've still got more frames to consume, we but need to wait for it to become available. */
- targetFrame = pDevice->null_device.lastProcessedFramePlayback;
- for (;;) {
- ma_uint64 currentFrame;
+ if ((chan & (1UL << iChannel)) == 0) {
+ continue;
+ }
- /* Stop waiting if the device has been stopped. */
- if (!c89atomic_load_32(&pDevice->null_device.isStarted)) {
- break;
- }
+ if (deviceType == ma_device_type_playback) {
+ channels = caps->pchan[iChannel];
+ } else {
+ channels = caps->rchan[iChannel];
+ }
- currentFrame = ma_device_get_total_run_time_in_frames__null(pDevice);
- if (currentFrame >= targetFrame) {
- break;
+ if (maxChannels < channels) {
+ maxChannels = channels;
+ }
}
-
- /* Getting here means we haven't yet reached the target sample, so continue waiting. */
- ma_sleep(10);
}
-
- pDevice->null_device.lastProcessedFramePlayback += pDevice->playback.internalPeriodSizeInFrames;
- pDevice->null_device.currentPeriodFramesRemainingPlayback = pDevice->playback.internalPeriodSizeInFrames;
- }
-
- if (pFramesWritten != NULL) {
- *pFramesWritten = totalPCMFramesProcessed;
}
- return result;
+ return maxChannels;
}
-static ma_result ma_device_read__null(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead)
+static ma_uint32 ma_find_best_sample_rate_from_sio_cap__sndio(struct ma_sio_cap* caps, ma_device_type deviceType, ma_format requiredFormat, ma_uint32 requiredChannels)
{
- ma_result result = MA_SUCCESS;
- ma_uint32 totalPCMFramesProcessed;
+ ma_uint32 firstSampleRate;
+ ma_uint32 bestSampleRate;
+ unsigned int iConfig;
- if (pFramesRead != NULL) {
- *pFramesRead = 0;
- }
+ MA_ASSERT(caps != NULL);
+ MA_ASSERT(requiredFormat != ma_format_unknown);
+ MA_ASSERT(requiredChannels > 0);
+ MA_ASSERT(requiredChannels <= MA_MAX_CHANNELS);
- /* Keep going until everything has been read. */
- totalPCMFramesProcessed = 0;
- while (totalPCMFramesProcessed < frameCount) {
- ma_uint64 targetFrame;
+ firstSampleRate = 0; /* <-- If the device does not support a standard rate we'll fall back to the first one that's found. */
+ bestSampleRate = 0;
- /* If there are any frames remaining in the current period, consume those first. */
- if (pDevice->null_device.currentPeriodFramesRemainingCapture > 0) {
- ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
- ma_uint32 framesRemaining = (frameCount - totalPCMFramesProcessed);
- ma_uint32 framesToProcess = pDevice->null_device.currentPeriodFramesRemainingCapture;
- if (framesToProcess > framesRemaining) {
- framesToProcess = framesRemaining;
+ for (iConfig = 0; iConfig < caps->nconf; iConfig += 1) {
+ /* The encoding should be of requiredFormat. */
+ unsigned int iEncoding;
+ for (iEncoding = 0; iEncoding < MA_SIO_NENC; iEncoding += 1) {
+ unsigned int iChannel;
+ unsigned int bits;
+ unsigned int bps;
+ unsigned int sig;
+ unsigned int le;
+ unsigned int msb;
+ ma_format format;
+
+ if ((caps->confs[iConfig].enc & (1UL << iEncoding)) == 0) {
+ continue;
}
- /* We need to ensure the output buffer is zeroed. */
- MA_ZERO_MEMORY(ma_offset_ptr(pPCMFrames, totalPCMFramesProcessed*bpf), framesToProcess*bpf);
+ bits = caps->enc[iEncoding].bits;
+ bps = caps->enc[iEncoding].bps;
+ sig = caps->enc[iEncoding].sig;
+ le = caps->enc[iEncoding].le;
+ msb = caps->enc[iEncoding].msb;
+ format = ma_format_from_sio_enc__sndio(bits, bps, sig, le, msb);
+ if (format != requiredFormat) {
+ continue;
+ }
- pDevice->null_device.currentPeriodFramesRemainingCapture -= framesToProcess;
- totalPCMFramesProcessed += framesToProcess;
- }
+ /* Getting here means the format is supported. Iterate over each channel count and grab the biggest one. */
+ for (iChannel = 0; iChannel < MA_SIO_NCHAN; iChannel += 1) {
+ unsigned int chan = 0;
+ unsigned int channels;
+ unsigned int iRate;
- /* If we've consumed the current period we'll need to mark it as such an ensure the device is started if it's not already. */
- if (pDevice->null_device.currentPeriodFramesRemainingCapture == 0) {
- pDevice->null_device.currentPeriodFramesRemainingCapture = 0;
- }
+ if (deviceType == ma_device_type_playback) {
+ chan = caps->confs[iConfig].pchan;
+ } else {
+ chan = caps->confs[iConfig].rchan;
+ }
- /* If we've consumed the whole buffer we can return now. */
- MA_ASSERT(totalPCMFramesProcessed <= frameCount);
- if (totalPCMFramesProcessed == frameCount) {
- break;
- }
+ if ((chan & (1UL << iChannel)) == 0) {
+ continue;
+ }
- /* Getting here means we've still got more frames to consume, we but need to wait for it to become available. */
- targetFrame = pDevice->null_device.lastProcessedFrameCapture + pDevice->capture.internalPeriodSizeInFrames;
- for (;;) {
- ma_uint64 currentFrame;
+ if (deviceType == ma_device_type_playback) {
+ channels = caps->pchan[iChannel];
+ } else {
+ channels = caps->rchan[iChannel];
+ }
- /* Stop waiting if the device has been stopped. */
- if (!c89atomic_load_32(&pDevice->null_device.isStarted)) {
- break;
- }
+ if (channels != requiredChannels) {
+ continue;
+ }
- currentFrame = ma_device_get_total_run_time_in_frames__null(pDevice);
- if (currentFrame >= targetFrame) {
- break;
- }
+ /* Getting here means we have found a compatible encoding/channel pair. */
+ for (iRate = 0; iRate < MA_SIO_NRATE; iRate += 1) {
+ ma_uint32 rate = (ma_uint32)caps->rate[iRate];
+ ma_uint32 ratePriority;
- /* Getting here means we haven't yet reached the target sample, so continue waiting. */
- ma_sleep(10);
- }
+ if (firstSampleRate == 0) {
+ firstSampleRate = rate;
+ }
- pDevice->null_device.lastProcessedFrameCapture += pDevice->capture.internalPeriodSizeInFrames;
- pDevice->null_device.currentPeriodFramesRemainingCapture = pDevice->capture.internalPeriodSizeInFrames;
+ /* Disregard this rate if it's not a standard one. */
+ ratePriority = ma_get_standard_sample_rate_priority_index__sndio(rate);
+ if (ratePriority == (ma_uint32)-1) {
+ continue;
+ }
+
+ if (ma_get_standard_sample_rate_priority_index__sndio(bestSampleRate) > ratePriority) { /* Lower = better. */
+ bestSampleRate = rate;
+ }
+ }
+ }
+ }
}
- if (pFramesRead != NULL) {
- *pFramesRead = totalPCMFramesProcessed;
+ /* If a standard sample rate was not found just fall back to the first one that was iterated. */
+ if (bestSampleRate == 0) {
+ bestSampleRate = firstSampleRate;
}
- return result;
+ return bestSampleRate;
}
-static ma_result ma_context_uninit__null(ma_context* pContext)
-{
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pContext->backend == ma_backend_null);
-
- (void)pContext;
- return MA_SUCCESS;
-}
-static ma_result ma_context_init__null(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+static ma_result ma_context_enumerate_devices__sndio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
{
- MA_ASSERT(pContext != NULL);
-
- (void)pConfig;
- (void)pContext;
-
- pCallbacks->onContextInit = ma_context_init__null;
- pCallbacks->onContextUninit = ma_context_uninit__null;
- pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__null;
- pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__null;
- pCallbacks->onDeviceInit = ma_device_init__null;
- pCallbacks->onDeviceUninit = ma_device_uninit__null;
- pCallbacks->onDeviceStart = ma_device_start__null;
- pCallbacks->onDeviceStop = ma_device_stop__null;
- pCallbacks->onDeviceRead = ma_device_read__null;
- pCallbacks->onDeviceWrite = ma_device_write__null;
- pCallbacks->onDeviceDataLoop = NULL; /* Our backend is asynchronous with a blocking read-write API which means we can get miniaudio to deal with the audio thread. */
-
- /* The null backend always works. */
- return MA_SUCCESS;
-}
-#endif
-
-
-
-/*******************************************************************************
-
-WIN32 COMMON
-
-*******************************************************************************/
-#if defined(MA_WIN32)
-#if defined(MA_WIN32_DESKTOP)
- #define ma_CoInitializeEx(pContext, pvReserved, dwCoInit) ((MA_PFN_CoInitializeEx)pContext->win32.CoInitializeEx)(pvReserved, dwCoInit)
- #define ma_CoUninitialize(pContext) ((MA_PFN_CoUninitialize)pContext->win32.CoUninitialize)()
- #define ma_CoCreateInstance(pContext, rclsid, pUnkOuter, dwClsContext, riid, ppv) ((MA_PFN_CoCreateInstance)pContext->win32.CoCreateInstance)(rclsid, pUnkOuter, dwClsContext, riid, ppv)
- #define ma_CoTaskMemFree(pContext, pv) ((MA_PFN_CoTaskMemFree)pContext->win32.CoTaskMemFree)(pv)
- #define ma_PropVariantClear(pContext, pvar) ((MA_PFN_PropVariantClear)pContext->win32.PropVariantClear)(pvar)
-#else
- #define ma_CoInitializeEx(pContext, pvReserved, dwCoInit) CoInitializeEx(pvReserved, dwCoInit)
- #define ma_CoUninitialize(pContext) CoUninitialize()
- #define ma_CoCreateInstance(pContext, rclsid, pUnkOuter, dwClsContext, riid, ppv) CoCreateInstance(rclsid, pUnkOuter, dwClsContext, riid, ppv)
- #define ma_CoTaskMemFree(pContext, pv) CoTaskMemFree(pv)
- #define ma_PropVariantClear(pContext, pvar) PropVariantClear(pvar)
-#endif
-
-#if !defined(MAXULONG_PTR) && !defined(__WATCOMC__)
-typedef size_t DWORD_PTR;
-#endif
+ ma_bool32 isTerminating = MA_FALSE;
+ struct ma_sio_hdl* handle;
-#if !defined(WAVE_FORMAT_44M08)
-#define WAVE_FORMAT_44M08 0x00000100
-#define WAVE_FORMAT_44S08 0x00000200
-#define WAVE_FORMAT_44M16 0x00000400
-#define WAVE_FORMAT_44S16 0x00000800
-#define WAVE_FORMAT_48M08 0x00001000
-#define WAVE_FORMAT_48S08 0x00002000
-#define WAVE_FORMAT_48M16 0x00004000
-#define WAVE_FORMAT_48S16 0x00008000
-#define WAVE_FORMAT_96M08 0x00010000
-#define WAVE_FORMAT_96S08 0x00020000
-#define WAVE_FORMAT_96M16 0x00040000
-#define WAVE_FORMAT_96S16 0x00080000
-#endif
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(callback != NULL);
-#ifndef SPEAKER_FRONT_LEFT
-#define SPEAKER_FRONT_LEFT 0x1
-#define SPEAKER_FRONT_RIGHT 0x2
-#define SPEAKER_FRONT_CENTER 0x4
-#define SPEAKER_LOW_FREQUENCY 0x8
-#define SPEAKER_BACK_LEFT 0x10
-#define SPEAKER_BACK_RIGHT 0x20
-#define SPEAKER_FRONT_LEFT_OF_CENTER 0x40
-#define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80
-#define SPEAKER_BACK_CENTER 0x100
-#define SPEAKER_SIDE_LEFT 0x200
-#define SPEAKER_SIDE_RIGHT 0x400
-#define SPEAKER_TOP_CENTER 0x800
-#define SPEAKER_TOP_FRONT_LEFT 0x1000
-#define SPEAKER_TOP_FRONT_CENTER 0x2000
-#define SPEAKER_TOP_FRONT_RIGHT 0x4000
-#define SPEAKER_TOP_BACK_LEFT 0x8000
-#define SPEAKER_TOP_BACK_CENTER 0x10000
-#define SPEAKER_TOP_BACK_RIGHT 0x20000
-#endif
+ /* sndio doesn't seem to have a good device enumeration API, so I'm therefore only enumerating over default devices for now. */
-/*
-The SDK that comes with old versions of MSVC (VC6, for example) does not appear to define WAVEFORMATEXTENSIBLE. We
-define our own implementation in this case.
-*/
-#if (defined(_MSC_VER) && !defined(_WAVEFORMATEXTENSIBLE_)) || defined(__DMC__)
-typedef struct
-{
- WAVEFORMATEX Format;
- union
- {
- WORD wValidBitsPerSample;
- WORD wSamplesPerBlock;
- WORD wReserved;
- } Samples;
- DWORD dwChannelMask;
- GUID SubFormat;
-} WAVEFORMATEXTENSIBLE;
-#endif
+ /* Playback. */
+ if (!isTerminating) {
+ handle = ((ma_sio_open_proc)pContext->sndio.sio_open)(MA_SIO_DEVANY, MA_SIO_PLAY, 0);
+ if (handle != NULL) {
+ /* Supports playback. */
+ ma_device_info deviceInfo;
+ MA_ZERO_OBJECT(&deviceInfo);
+ ma_strcpy_s(deviceInfo.id.sndio, sizeof(deviceInfo.id.sndio), MA_SIO_DEVANY);
+ ma_strcpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME);
-#ifndef WAVE_FORMAT_EXTENSIBLE
-#define WAVE_FORMAT_EXTENSIBLE 0xFFFE
-#endif
+ isTerminating = !callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
-#ifndef WAVE_FORMAT_IEEE_FLOAT
-#define WAVE_FORMAT_IEEE_FLOAT 0x0003
-#endif
+ ((ma_sio_close_proc)pContext->sndio.sio_close)(handle);
+ }
+ }
-/* Converts an individual Win32-style channel identifier (SPEAKER_FRONT_LEFT, etc.) to miniaudio. */
-static ma_uint8 ma_channel_id_to_ma__win32(DWORD id)
-{
- switch (id)
- {
- case SPEAKER_FRONT_LEFT: return MA_CHANNEL_FRONT_LEFT;
- case SPEAKER_FRONT_RIGHT: return MA_CHANNEL_FRONT_RIGHT;
- case SPEAKER_FRONT_CENTER: return MA_CHANNEL_FRONT_CENTER;
- case SPEAKER_LOW_FREQUENCY: return MA_CHANNEL_LFE;
- case SPEAKER_BACK_LEFT: return MA_CHANNEL_BACK_LEFT;
- case SPEAKER_BACK_RIGHT: return MA_CHANNEL_BACK_RIGHT;
- case SPEAKER_FRONT_LEFT_OF_CENTER: return MA_CHANNEL_FRONT_LEFT_CENTER;
- case SPEAKER_FRONT_RIGHT_OF_CENTER: return MA_CHANNEL_FRONT_RIGHT_CENTER;
- case SPEAKER_BACK_CENTER: return MA_CHANNEL_BACK_CENTER;
- case SPEAKER_SIDE_LEFT: return MA_CHANNEL_SIDE_LEFT;
- case SPEAKER_SIDE_RIGHT: return MA_CHANNEL_SIDE_RIGHT;
- case SPEAKER_TOP_CENTER: return MA_CHANNEL_TOP_CENTER;
- case SPEAKER_TOP_FRONT_LEFT: return MA_CHANNEL_TOP_FRONT_LEFT;
- case SPEAKER_TOP_FRONT_CENTER: return MA_CHANNEL_TOP_FRONT_CENTER;
- case SPEAKER_TOP_FRONT_RIGHT: return MA_CHANNEL_TOP_FRONT_RIGHT;
- case SPEAKER_TOP_BACK_LEFT: return MA_CHANNEL_TOP_BACK_LEFT;
- case SPEAKER_TOP_BACK_CENTER: return MA_CHANNEL_TOP_BACK_CENTER;
- case SPEAKER_TOP_BACK_RIGHT: return MA_CHANNEL_TOP_BACK_RIGHT;
- default: return 0;
+ /* Capture. */
+ if (!isTerminating) {
+ handle = ((ma_sio_open_proc)pContext->sndio.sio_open)(MA_SIO_DEVANY, MA_SIO_REC, 0);
+ if (handle != NULL) {
+ /* Supports capture. */
+ ma_device_info deviceInfo;
+ MA_ZERO_OBJECT(&deviceInfo);
+ ma_strcpy_s(deviceInfo.id.sndio, sizeof(deviceInfo.id.sndio), "default");
+ ma_strcpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME);
+
+ isTerminating = !callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
+
+ ((ma_sio_close_proc)pContext->sndio.sio_close)(handle);
+ }
}
+
+ return MA_SUCCESS;
}
-/* Converts an individual miniaudio channel identifier (MA_CHANNEL_FRONT_LEFT, etc.) to Win32-style. */
-static DWORD ma_channel_id_to_win32(DWORD id)
+static ma_result ma_context_get_device_info__sndio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
{
- switch (id)
- {
- case MA_CHANNEL_MONO: return SPEAKER_FRONT_CENTER;
- case MA_CHANNEL_FRONT_LEFT: return SPEAKER_FRONT_LEFT;
- case MA_CHANNEL_FRONT_RIGHT: return SPEAKER_FRONT_RIGHT;
- case MA_CHANNEL_FRONT_CENTER: return SPEAKER_FRONT_CENTER;
- case MA_CHANNEL_LFE: return SPEAKER_LOW_FREQUENCY;
- case MA_CHANNEL_BACK_LEFT: return SPEAKER_BACK_LEFT;
- case MA_CHANNEL_BACK_RIGHT: return SPEAKER_BACK_RIGHT;
- case MA_CHANNEL_FRONT_LEFT_CENTER: return SPEAKER_FRONT_LEFT_OF_CENTER;
- case MA_CHANNEL_FRONT_RIGHT_CENTER: return SPEAKER_FRONT_RIGHT_OF_CENTER;
- case MA_CHANNEL_BACK_CENTER: return SPEAKER_BACK_CENTER;
- case MA_CHANNEL_SIDE_LEFT: return SPEAKER_SIDE_LEFT;
- case MA_CHANNEL_SIDE_RIGHT: return SPEAKER_SIDE_RIGHT;
- case MA_CHANNEL_TOP_CENTER: return SPEAKER_TOP_CENTER;
- case MA_CHANNEL_TOP_FRONT_LEFT: return SPEAKER_TOP_FRONT_LEFT;
- case MA_CHANNEL_TOP_FRONT_CENTER: return SPEAKER_TOP_FRONT_CENTER;
- case MA_CHANNEL_TOP_FRONT_RIGHT: return SPEAKER_TOP_FRONT_RIGHT;
- case MA_CHANNEL_TOP_BACK_LEFT: return SPEAKER_TOP_BACK_LEFT;
- case MA_CHANNEL_TOP_BACK_CENTER: return SPEAKER_TOP_BACK_CENTER;
- case MA_CHANNEL_TOP_BACK_RIGHT: return SPEAKER_TOP_BACK_RIGHT;
- default: return 0;
+ char devid[256];
+ struct ma_sio_hdl* handle;
+ struct ma_sio_cap caps;
+ unsigned int iConfig;
+
+ MA_ASSERT(pContext != NULL);
+
+ /* We need to open the device before we can get information about it. */
+ if (pDeviceID == NULL) {
+ ma_strcpy_s(devid, sizeof(devid), MA_SIO_DEVANY);
+ ma_strcpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), (deviceType == ma_device_type_playback) ? MA_DEFAULT_PLAYBACK_DEVICE_NAME : MA_DEFAULT_CAPTURE_DEVICE_NAME);
+ } else {
+ ma_strcpy_s(devid, sizeof(devid), pDeviceID->sndio);
+ ma_strcpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), devid);
}
-}
-/* Converts a channel mapping to a Win32-style channel mask. */
-static DWORD ma_channel_map_to_channel_mask__win32(const ma_channel* pChannelMap, ma_uint32 channels)
-{
- DWORD dwChannelMask = 0;
- ma_uint32 iChannel;
+ handle = ((ma_sio_open_proc)pContext->sndio.sio_open)(devid, (deviceType == ma_device_type_playback) ? MA_SIO_PLAY : MA_SIO_REC, 0);
+ if (handle == NULL) {
+ return MA_NO_DEVICE;
+ }
- for (iChannel = 0; iChannel < channels; ++iChannel) {
- dwChannelMask |= ma_channel_id_to_win32(pChannelMap[iChannel]);
+ if (((ma_sio_getcap_proc)pContext->sndio.sio_getcap)(handle, &caps) == 0) {
+ return MA_ERROR;
}
- return dwChannelMask;
-}
+ pDeviceInfo->nativeDataFormatCount = 0;
-/* Converts a Win32-style channel mask to a miniaudio channel map. */
-static void ma_channel_mask_to_channel_map__win32(DWORD dwChannelMask, ma_uint32 channels, ma_channel* pChannelMap)
-{
- if (channels == 1 && dwChannelMask == 0) {
- pChannelMap[0] = MA_CHANNEL_MONO;
- } else if (channels == 2 && dwChannelMask == 0) {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- } else {
- if (channels == 1 && (dwChannelMask & SPEAKER_FRONT_CENTER) != 0) {
- pChannelMap[0] = MA_CHANNEL_MONO;
- } else {
- /* Just iterate over each bit. */
- ma_uint32 iChannel = 0;
- ma_uint32 iBit;
+ for (iConfig = 0; iConfig < caps.nconf; iConfig += 1) {
+ /*
+ The main thing we care about is that the encoding is supported by miniaudio. If it is, we want to give
+ preference to some formats over others.
+ */
+ unsigned int iEncoding;
+ unsigned int iChannel;
+ unsigned int iRate;
- for (iBit = 0; iBit < 32 && iChannel < channels; ++iBit) {
- DWORD bitValue = (dwChannelMask & (1UL << iBit));
- if (bitValue != 0) {
- /* The bit is set. */
- pChannelMap[iChannel] = ma_channel_id_to_ma__win32(bitValue);
- iChannel += 1;
+ for (iEncoding = 0; iEncoding < MA_SIO_NENC; iEncoding += 1) {
+ unsigned int bits;
+ unsigned int bps;
+ unsigned int sig;
+ unsigned int le;
+ unsigned int msb;
+ ma_format format;
+
+ if ((caps.confs[iConfig].enc & (1UL << iEncoding)) == 0) {
+ continue;
+ }
+
+ bits = caps.enc[iEncoding].bits;
+ bps = caps.enc[iEncoding].bps;
+ sig = caps.enc[iEncoding].sig;
+ le = caps.enc[iEncoding].le;
+ msb = caps.enc[iEncoding].msb;
+ format = ma_format_from_sio_enc__sndio(bits, bps, sig, le, msb);
+ if (format == ma_format_unknown) {
+ continue; /* Format not supported. */
+ }
+
+
+ /* Channels. */
+ for (iChannel = 0; iChannel < MA_SIO_NCHAN; iChannel += 1) {
+ unsigned int chan = 0;
+ unsigned int channels;
+
+ if (deviceType == ma_device_type_playback) {
+ chan = caps.confs[iConfig].pchan;
+ } else {
+ chan = caps.confs[iConfig].rchan;
+ }
+
+ if ((chan & (1UL << iChannel)) == 0) {
+ continue;
+ }
+
+ if (deviceType == ma_device_type_playback) {
+ channels = caps.pchan[iChannel];
+ } else {
+ channels = caps.rchan[iChannel];
+ }
+
+
+ /* Sample Rates. */
+ for (iRate = 0; iRate < MA_SIO_NRATE; iRate += 1) {
+ if ((caps.confs[iConfig].rate & (1UL << iRate)) != 0) {
+ ma_device_info_add_native_data_format(pDeviceInfo, format, channels, caps.rate[iRate], 0);
+ }
}
}
}
}
-}
-#ifdef __cplusplus
-static ma_bool32 ma_is_guid_equal(const void* a, const void* b)
-{
- return IsEqualGUID(*(const GUID*)a, *(const GUID*)b);
+ ((ma_sio_close_proc)pContext->sndio.sio_close)(handle);
+ return MA_SUCCESS;
}
-#else
-#define ma_is_guid_equal(a, b) IsEqualGUID((const GUID*)a, (const GUID*)b)
-#endif
-static MA_INLINE ma_bool32 ma_is_guid_null(const void* guid)
+static ma_result ma_device_uninit__sndio(ma_device* pDevice)
{
- static GUID nullguid = {0x00000000, 0x0000, 0x0000, {0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00}};
- return ma_is_guid_equal(guid, &nullguid);
+ MA_ASSERT(pDevice != NULL);
+
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ ((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)pDevice->sndio.handleCapture);
+ }
+
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ ((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback);
+ }
+
+ return MA_SUCCESS;
}
-static ma_format ma_format_from_WAVEFORMATEX(const WAVEFORMATEX* pWF)
+static ma_result ma_device_init_handle__sndio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptor, ma_device_type deviceType)
{
- MA_ASSERT(pWF != NULL);
+ const char* pDeviceName;
+ ma_ptr handle;
+ int openFlags = 0;
+ struct ma_sio_cap caps;
+ struct ma_sio_par par;
+ const ma_device_id* pDeviceID;
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ ma_format internalFormat;
+ ma_uint32 internalChannels;
+ ma_uint32 internalSampleRate;
+ ma_uint32 internalPeriodSizeInFrames;
+ ma_uint32 internalPeriods;
- if (pWF->wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
- const WAVEFORMATEXTENSIBLE* pWFEX = (const WAVEFORMATEXTENSIBLE*)pWF;
- if (ma_is_guid_equal(&pWFEX->SubFormat, &MA_GUID_KSDATAFORMAT_SUBTYPE_PCM)) {
- if (pWFEX->Samples.wValidBitsPerSample == 32) {
- return ma_format_s32;
- }
- if (pWFEX->Samples.wValidBitsPerSample == 24) {
- if (pWFEX->Format.wBitsPerSample == 32) {
- /*return ma_format_s24_32;*/
- }
- if (pWFEX->Format.wBitsPerSample == 24) {
- return ma_format_s24;
- }
- }
- if (pWFEX->Samples.wValidBitsPerSample == 16) {
- return ma_format_s16;
- }
- if (pWFEX->Samples.wValidBitsPerSample == 8) {
- return ma_format_u8;
- }
+ MA_ASSERT(pConfig != NULL);
+ MA_ASSERT(deviceType != ma_device_type_duplex);
+ MA_ASSERT(pDevice != NULL);
+
+ if (deviceType == ma_device_type_capture) {
+ openFlags = MA_SIO_REC;
+ } else {
+ openFlags = MA_SIO_PLAY;
+ }
+
+ pDeviceID = pDescriptor->pDeviceID;
+ format = pDescriptor->format;
+ channels = pDescriptor->channels;
+ sampleRate = pDescriptor->sampleRate;
+
+ pDeviceName = MA_SIO_DEVANY;
+ if (pDeviceID != NULL) {
+ pDeviceName = pDeviceID->sndio;
+ }
+
+ handle = (ma_ptr)((ma_sio_open_proc)pDevice->pContext->sndio.sio_open)(pDeviceName, openFlags, 0);
+ if (handle == NULL) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[sndio] Failed to open device.");
+ return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
+ }
+
+ /* We need to retrieve the device caps to determine the most appropriate format to use. */
+ if (((ma_sio_getcap_proc)pDevice->pContext->sndio.sio_getcap)((struct ma_sio_hdl*)handle, &caps) == 0) {
+ ((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)handle);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[sndio] Failed to retrieve device caps.");
+ return MA_ERROR;
+ }
+
+ /*
+ Note: sndio reports a huge range of available channels. This is inconvenient for us because there's no real
+ way, as far as I can tell, to get the _actual_ channel count of the device. I'm therefore restricting this
+ to the requested channels, regardless of whether or not the default channel count is requested.
+
+ For hardware devices, I'm suspecting only a single channel count will be reported and we can safely use the
+ value returned by ma_find_best_channels_from_sio_cap__sndio().
+ */
+ if (deviceType == ma_device_type_capture) {
+ if (format == ma_format_unknown) {
+ format = ma_find_best_format_from_sio_cap__sndio(&caps);
}
- if (ma_is_guid_equal(&pWFEX->SubFormat, &MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)) {
- if (pWFEX->Samples.wValidBitsPerSample == 32) {
- return ma_format_f32;
- }
- /*
- if (pWFEX->Samples.wValidBitsPerSample == 64) {
- return ma_format_f64;
+
+ if (channels == 0) {
+ if (strlen(pDeviceName) > strlen("rsnd/") && strncmp(pDeviceName, "rsnd/", strlen("rsnd/")) == 0) {
+ channels = ma_find_best_channels_from_sio_cap__sndio(&caps, deviceType, format);
+ } else {
+ channels = MA_DEFAULT_CHANNELS;
}
- */
}
} else {
- if (pWF->wFormatTag == WAVE_FORMAT_PCM) {
- if (pWF->wBitsPerSample == 32) {
- return ma_format_s32;
- }
- if (pWF->wBitsPerSample == 24) {
- return ma_format_s24;
- }
- if (pWF->wBitsPerSample == 16) {
- return ma_format_s16;
- }
- if (pWF->wBitsPerSample == 8) {
- return ma_format_u8;
- }
+ if (format == ma_format_unknown) {
+ format = ma_find_best_format_from_sio_cap__sndio(&caps);
}
- if (pWF->wFormatTag == WAVE_FORMAT_IEEE_FLOAT) {
- if (pWF->wBitsPerSample == 32) {
- return ma_format_f32;
- }
- if (pWF->wBitsPerSample == 64) {
- /*return ma_format_f64;*/
+
+ if (channels == 0) {
+ if (strlen(pDeviceName) > strlen("rsnd/") && strncmp(pDeviceName, "rsnd/", strlen("rsnd/")) == 0) {
+ channels = ma_find_best_channels_from_sio_cap__sndio(&caps, deviceType, format);
+ } else {
+ channels = MA_DEFAULT_CHANNELS;
}
}
}
- return ma_format_unknown;
-}
-#endif
+ if (sampleRate == 0) {
+ sampleRate = ma_find_best_sample_rate_from_sio_cap__sndio(&caps, pConfig->deviceType, format, channels);
+ }
-/*******************************************************************************
+ ((ma_sio_initpar_proc)pDevice->pContext->sndio.sio_initpar)(&par);
+ par.msb = 0;
+ par.le = ma_is_little_endian();
-WASAPI Backend
+ switch (format) {
+ case ma_format_u8:
+ {
+ par.bits = 8;
+ par.bps = 1;
+ par.sig = 0;
+ } break;
-*******************************************************************************/
-#ifdef MA_HAS_WASAPI
-#if 0
-#if defined(_MSC_VER)
- #pragma warning(push)
- #pragma warning(disable:4091) /* 'typedef ': ignored on left of '' when no variable is declared */
-#endif
-#include
-#include
-#if defined(_MSC_VER)
- #pragma warning(pop)
-#endif
-#endif /* 0 */
+ case ma_format_s24:
+ {
+ par.bits = 24;
+ par.bps = 3;
+ par.sig = 1;
+ } break;
-static ma_result ma_device_reroute__wasapi(ma_device* pDevice, ma_device_type deviceType);
+ case ma_format_s32:
+ {
+ par.bits = 32;
+ par.bps = 4;
+ par.sig = 1;
+ } break;
-/* Some compilers don't define VerifyVersionInfoW. Need to write this ourselves. */
-#define MA_WIN32_WINNT_VISTA 0x0600
-#define MA_VER_MINORVERSION 0x01
-#define MA_VER_MAJORVERSION 0x02
-#define MA_VER_SERVICEPACKMAJOR 0x20
-#define MA_VER_GREATER_EQUAL 0x03
+ case ma_format_s16:
+ case ma_format_f32:
+ case ma_format_unknown:
+ default:
+ {
+ par.bits = 16;
+ par.bps = 2;
+ par.sig = 1;
+ } break;
+ }
-typedef struct {
- DWORD dwOSVersionInfoSize;
- DWORD dwMajorVersion;
- DWORD dwMinorVersion;
- DWORD dwBuildNumber;
- DWORD dwPlatformId;
- WCHAR szCSDVersion[128];
- WORD wServicePackMajor;
- WORD wServicePackMinor;
- WORD wSuiteMask;
- BYTE wProductType;
- BYTE wReserved;
-} ma_OSVERSIONINFOEXW;
+ if (deviceType == ma_device_type_capture) {
+ par.rchan = channels;
+ } else {
+ par.pchan = channels;
+ }
-typedef BOOL (WINAPI * ma_PFNVerifyVersionInfoW) (ma_OSVERSIONINFOEXW* lpVersionInfo, DWORD dwTypeMask, DWORDLONG dwlConditionMask);
-typedef ULONGLONG (WINAPI * ma_PFNVerSetConditionMask)(ULONGLONG dwlConditionMask, DWORD dwTypeBitMask, BYTE dwConditionMask);
+ par.rate = sampleRate;
+ internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, par.rate, pConfig->performanceProfile);
-#ifndef PROPERTYKEY_DEFINED
-#define PROPERTYKEY_DEFINED
-#ifndef __WATCOMC__
-typedef struct
-{
- GUID fmtid;
- DWORD pid;
-} PROPERTYKEY;
-#endif
-#endif
+ par.round = internalPeriodSizeInFrames;
+ par.appbufsz = par.round * pDescriptor->periodCount;
-/* Some compilers don't define PropVariantInit(). We just do this ourselves since it's just a memset(). */
-static MA_INLINE void ma_PropVariantInit(PROPVARIANT* pProp)
+ if (((ma_sio_setpar_proc)pDevice->pContext->sndio.sio_setpar)((struct ma_sio_hdl*)handle, &par) == 0) {
+ ((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)handle);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[sndio] Failed to set buffer size.");
+ return MA_ERROR;
+ }
+
+ if (((ma_sio_getpar_proc)pDevice->pContext->sndio.sio_getpar)((struct ma_sio_hdl*)handle, &par) == 0) {
+ ((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)handle);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[sndio] Failed to retrieve buffer size.");
+ return MA_ERROR;
+ }
+
+ internalFormat = ma_format_from_sio_enc__sndio(par.bits, par.bps, par.sig, par.le, par.msb);
+ internalChannels = (deviceType == ma_device_type_capture) ? par.rchan : par.pchan;
+ internalSampleRate = par.rate;
+ internalPeriods = par.appbufsz / par.round;
+ internalPeriodSizeInFrames = par.round;
+
+ if (deviceType == ma_device_type_capture) {
+ pDevice->sndio.handleCapture = handle;
+ } else {
+ pDevice->sndio.handlePlayback = handle;
+ }
+
+ pDescriptor->format = internalFormat;
+ pDescriptor->channels = internalChannels;
+ pDescriptor->sampleRate = internalSampleRate;
+ ma_channel_map_init_standard(ma_standard_channel_map_sndio, pDescriptor->channelMap, ma_countof(pDescriptor->channelMap), internalChannels);
+ pDescriptor->periodSizeInFrames = internalPeriodSizeInFrames;
+ pDescriptor->periodCount = internalPeriods;
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_init__sndio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
{
- MA_ZERO_OBJECT(pProp);
+ MA_ASSERT(pDevice != NULL);
+
+ MA_ZERO_OBJECT(&pDevice->sndio);
+
+ if (pConfig->deviceType == ma_device_type_loopback) {
+ return MA_DEVICE_TYPE_NOT_SUPPORTED;
+ }
+
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
+ ma_result result = ma_device_init_handle__sndio(pDevice, pConfig, pDescriptorCapture, ma_device_type_capture);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ }
+
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ ma_result result = ma_device_init_handle__sndio(pDevice, pConfig, pDescriptorPlayback, ma_device_type_playback);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ }
+
+ return MA_SUCCESS;
}
+static ma_result ma_device_start__sndio(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
+
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ ((ma_sio_start_proc)pDevice->pContext->sndio.sio_start)((struct ma_sio_hdl*)pDevice->sndio.handleCapture);
+ }
-static const PROPERTYKEY MA_PKEY_Device_FriendlyName = {{0xA45C254E, 0xDF1C, 0x4EFD, {0x80, 0x20, 0x67, 0xD1, 0x46, 0xA8, 0x50, 0xE0}}, 14};
-static const PROPERTYKEY MA_PKEY_AudioEngine_DeviceFormat = {{0xF19F064D, 0x82C, 0x4E27, {0xBC, 0x73, 0x68, 0x82, 0xA1, 0xBB, 0x8E, 0x4C}}, 0};
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ ((ma_sio_start_proc)pDevice->pContext->sndio.sio_start)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback); /* <-- Doesn't actually playback until data is written. */
+ }
-static const IID MA_IID_IUnknown = {0x00000000, 0x0000, 0x0000, {0xC0, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x46}}; /* 00000000-0000-0000-C000-000000000046 */
-#ifndef MA_WIN32_DESKTOP
-static const IID MA_IID_IAgileObject = {0x94EA2B94, 0xE9CC, 0x49E0, {0xC0, 0xFF, 0xEE, 0x64, 0xCA, 0x8F, 0x5B, 0x90}}; /* 94EA2B94-E9CC-49E0-C0FF-EE64CA8F5B90 */
-#endif
+ return MA_SUCCESS;
+}
-static const IID MA_IID_IAudioClient = {0x1CB9AD4C, 0xDBFA, 0x4C32, {0xB1, 0x78, 0xC2, 0xF5, 0x68, 0xA7, 0x03, 0xB2}}; /* 1CB9AD4C-DBFA-4C32-B178-C2F568A703B2 = __uuidof(IAudioClient) */
-static const IID MA_IID_IAudioClient2 = {0x726778CD, 0xF60A, 0x4EDA, {0x82, 0xDE, 0xE4, 0x76, 0x10, 0xCD, 0x78, 0xAA}}; /* 726778CD-F60A-4EDA-82DE-E47610CD78AA = __uuidof(IAudioClient2) */
-static const IID MA_IID_IAudioClient3 = {0x7ED4EE07, 0x8E67, 0x4CD4, {0x8C, 0x1A, 0x2B, 0x7A, 0x59, 0x87, 0xAD, 0x42}}; /* 7ED4EE07-8E67-4CD4-8C1A-2B7A5987AD42 = __uuidof(IAudioClient3) */
-static const IID MA_IID_IAudioRenderClient = {0xF294ACFC, 0x3146, 0x4483, {0xA7, 0xBF, 0xAD, 0xDC, 0xA7, 0xC2, 0x60, 0xE2}}; /* F294ACFC-3146-4483-A7BF-ADDCA7C260E2 = __uuidof(IAudioRenderClient) */
-static const IID MA_IID_IAudioCaptureClient = {0xC8ADBD64, 0xE71E, 0x48A0, {0xA4, 0xDE, 0x18, 0x5C, 0x39, 0x5C, 0xD3, 0x17}}; /* C8ADBD64-E71E-48A0-A4DE-185C395CD317 = __uuidof(IAudioCaptureClient) */
-static const IID MA_IID_IMMNotificationClient = {0x7991EEC9, 0x7E89, 0x4D85, {0x83, 0x90, 0x6C, 0x70, 0x3C, 0xEC, 0x60, 0xC0}}; /* 7991EEC9-7E89-4D85-8390-6C703CEC60C0 = __uuidof(IMMNotificationClient) */
-#ifndef MA_WIN32_DESKTOP
-static const IID MA_IID_DEVINTERFACE_AUDIO_RENDER = {0xE6327CAD, 0xDCEC, 0x4949, {0xAE, 0x8A, 0x99, 0x1E, 0x97, 0x6A, 0x79, 0xD2}}; /* E6327CAD-DCEC-4949-AE8A-991E976A79D2 */
-static const IID MA_IID_DEVINTERFACE_AUDIO_CAPTURE = {0x2EEF81BE, 0x33FA, 0x4800, {0x96, 0x70, 0x1C, 0xD4, 0x74, 0x97, 0x2C, 0x3F}}; /* 2EEF81BE-33FA-4800-9670-1CD474972C3F */
-static const IID MA_IID_IActivateAudioInterfaceCompletionHandler = {0x41D949AB, 0x9862, 0x444A, {0x80, 0xF6, 0xC2, 0x61, 0x33, 0x4D, 0xA5, 0xEB}}; /* 41D949AB-9862-444A-80F6-C261334DA5EB */
-#endif
+static ma_result ma_device_stop__sndio(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
-static const IID MA_CLSID_MMDeviceEnumerator_Instance = {0xBCDE0395, 0xE52F, 0x467C, {0x8E, 0x3D, 0xC4, 0x57, 0x92, 0x91, 0x69, 0x2E}}; /* BCDE0395-E52F-467C-8E3D-C4579291692E = __uuidof(MMDeviceEnumerator) */
-static const IID MA_IID_IMMDeviceEnumerator_Instance = {0xA95664D2, 0x9614, 0x4F35, {0xA7, 0x46, 0xDE, 0x8D, 0xB6, 0x36, 0x17, 0xE6}}; /* A95664D2-9614-4F35-A746-DE8DB63617E6 = __uuidof(IMMDeviceEnumerator) */
-#ifdef __cplusplus
-#define MA_CLSID_MMDeviceEnumerator MA_CLSID_MMDeviceEnumerator_Instance
-#define MA_IID_IMMDeviceEnumerator MA_IID_IMMDeviceEnumerator_Instance
-#else
-#define MA_CLSID_MMDeviceEnumerator &MA_CLSID_MMDeviceEnumerator_Instance
-#define MA_IID_IMMDeviceEnumerator &MA_IID_IMMDeviceEnumerator_Instance
-#endif
+ /*
+ From the documentation:
-typedef struct ma_IUnknown ma_IUnknown;
-#ifdef MA_WIN32_DESKTOP
-#define MA_MM_DEVICE_STATE_ACTIVE 1
-#define MA_MM_DEVICE_STATE_DISABLED 2
-#define MA_MM_DEVICE_STATE_NOTPRESENT 4
-#define MA_MM_DEVICE_STATE_UNPLUGGED 8
+ The sio_stop() function puts the audio subsystem in the same state as before sio_start() is called. It stops recording, drains the play buffer and then
+ stops playback. If samples to play are queued but playback hasn't started yet then playback is forced immediately; playback will actually stop once the
+ buffer is drained. In no case are samples in the play buffer discarded.
-typedef struct ma_IMMDeviceEnumerator ma_IMMDeviceEnumerator;
-typedef struct ma_IMMDeviceCollection ma_IMMDeviceCollection;
-typedef struct ma_IMMDevice ma_IMMDevice;
-#else
-typedef struct ma_IActivateAudioInterfaceCompletionHandler ma_IActivateAudioInterfaceCompletionHandler;
-typedef struct ma_IActivateAudioInterfaceAsyncOperation ma_IActivateAudioInterfaceAsyncOperation;
-#endif
-typedef struct ma_IPropertyStore ma_IPropertyStore;
-typedef struct ma_IAudioClient ma_IAudioClient;
-typedef struct ma_IAudioClient2 ma_IAudioClient2;
-typedef struct ma_IAudioClient3 ma_IAudioClient3;
-typedef struct ma_IAudioRenderClient ma_IAudioRenderClient;
-typedef struct ma_IAudioCaptureClient ma_IAudioCaptureClient;
+ Therefore, sio_stop() performs all of the necessary draining for us.
+ */
-typedef ma_int64 MA_REFERENCE_TIME;
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ ((ma_sio_stop_proc)pDevice->pContext->sndio.sio_stop)((struct ma_sio_hdl*)pDevice->sndio.handleCapture);
+ }
-#define MA_AUDCLNT_STREAMFLAGS_CROSSPROCESS 0x00010000
-#define MA_AUDCLNT_STREAMFLAGS_LOOPBACK 0x00020000
-#define MA_AUDCLNT_STREAMFLAGS_EVENTCALLBACK 0x00040000
-#define MA_AUDCLNT_STREAMFLAGS_NOPERSIST 0x00080000
-#define MA_AUDCLNT_STREAMFLAGS_RATEADJUST 0x00100000
-#define MA_AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY 0x08000000
-#define MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM 0x80000000
-#define MA_AUDCLNT_SESSIONFLAGS_EXPIREWHENUNOWNED 0x10000000
-#define MA_AUDCLNT_SESSIONFLAGS_DISPLAY_HIDE 0x20000000
-#define MA_AUDCLNT_SESSIONFLAGS_DISPLAY_HIDEWHENEXPIRED 0x40000000
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ ((ma_sio_stop_proc)pDevice->pContext->sndio.sio_stop)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback);
+ }
-/* Buffer flags. */
-#define MA_AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY 1
-#define MA_AUDCLNT_BUFFERFLAGS_SILENT 2
-#define MA_AUDCLNT_BUFFERFLAGS_TIMESTAMP_ERROR 4
+ return MA_SUCCESS;
+}
-typedef enum
+static ma_result ma_device_write__sndio(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten)
{
- ma_eRender = 0,
- ma_eCapture = 1,
- ma_eAll = 2
-} ma_EDataFlow;
+ int result;
-typedef enum
-{
- ma_eConsole = 0,
- ma_eMultimedia = 1,
- ma_eCommunications = 2
-} ma_ERole;
+ if (pFramesWritten != NULL) {
+ *pFramesWritten = 0;
+ }
-typedef enum
-{
- MA_AUDCLNT_SHAREMODE_SHARED,
- MA_AUDCLNT_SHAREMODE_EXCLUSIVE
-} MA_AUDCLNT_SHAREMODE;
+ result = ((ma_sio_write_proc)pDevice->pContext->sndio.sio_write)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels));
+ if (result == 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[sndio] Failed to send data from the client to the device.");
+ return MA_IO_ERROR;
+ }
-typedef enum
-{
- MA_AudioCategory_Other = 0 /* <-- miniaudio is only caring about Other. */
-} MA_AUDIO_STREAM_CATEGORY;
+ if (pFramesWritten != NULL) {
+ *pFramesWritten = frameCount;
+ }
-typedef struct
-{
- ma_uint32 cbSize;
- BOOL bIsOffload;
- MA_AUDIO_STREAM_CATEGORY eCategory;
-} ma_AudioClientProperties;
+ return MA_SUCCESS;
+}
-/* IUnknown */
-typedef struct
-{
- /* IUnknown */
- HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IUnknown* pThis, const IID* const riid, void** ppObject);
- ULONG (STDMETHODCALLTYPE * AddRef) (ma_IUnknown* pThis);
- ULONG (STDMETHODCALLTYPE * Release) (ma_IUnknown* pThis);
-} ma_IUnknownVtbl;
-struct ma_IUnknown
+static ma_result ma_device_read__sndio(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead)
{
- ma_IUnknownVtbl* lpVtbl;
-};
-static MA_INLINE HRESULT ma_IUnknown_QueryInterface(ma_IUnknown* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
-static MA_INLINE ULONG ma_IUnknown_AddRef(ma_IUnknown* pThis) { return pThis->lpVtbl->AddRef(pThis); }
-static MA_INLINE ULONG ma_IUnknown_Release(ma_IUnknown* pThis) { return pThis->lpVtbl->Release(pThis); }
+ int result;
-#ifdef MA_WIN32_DESKTOP
- /* IMMNotificationClient */
- typedef struct
- {
- /* IUnknown */
- HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IMMNotificationClient* pThis, const IID* const riid, void** ppObject);
- ULONG (STDMETHODCALLTYPE * AddRef) (ma_IMMNotificationClient* pThis);
- ULONG (STDMETHODCALLTYPE * Release) (ma_IMMNotificationClient* pThis);
+ if (pFramesRead != NULL) {
+ *pFramesRead = 0;
+ }
- /* IMMNotificationClient */
- HRESULT (STDMETHODCALLTYPE * OnDeviceStateChanged) (ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID, DWORD dwNewState);
- HRESULT (STDMETHODCALLTYPE * OnDeviceAdded) (ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID);
- HRESULT (STDMETHODCALLTYPE * OnDeviceRemoved) (ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID);
- HRESULT (STDMETHODCALLTYPE * OnDefaultDeviceChanged)(ma_IMMNotificationClient* pThis, ma_EDataFlow dataFlow, ma_ERole role, LPCWSTR pDefaultDeviceID);
- HRESULT (STDMETHODCALLTYPE * OnPropertyValueChanged)(ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID, const PROPERTYKEY key);
- } ma_IMMNotificationClientVtbl;
+ result = ((ma_sio_read_proc)pDevice->pContext->sndio.sio_read)((struct ma_sio_hdl*)pDevice->sndio.handleCapture, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels));
+ if (result == 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[sndio] Failed to read data from the device to be sent to the device.");
+ return MA_IO_ERROR;
+ }
- /* IMMDeviceEnumerator */
- typedef struct
- {
- /* IUnknown */
- HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IMMDeviceEnumerator* pThis, const IID* const riid, void** ppObject);
- ULONG (STDMETHODCALLTYPE * AddRef) (ma_IMMDeviceEnumerator* pThis);
- ULONG (STDMETHODCALLTYPE * Release) (ma_IMMDeviceEnumerator* pThis);
+ if (pFramesRead != NULL) {
+ *pFramesRead = frameCount;
+ }
- /* IMMDeviceEnumerator */
- HRESULT (STDMETHODCALLTYPE * EnumAudioEndpoints) (ma_IMMDeviceEnumerator* pThis, ma_EDataFlow dataFlow, DWORD dwStateMask, ma_IMMDeviceCollection** ppDevices);
- HRESULT (STDMETHODCALLTYPE * GetDefaultAudioEndpoint) (ma_IMMDeviceEnumerator* pThis, ma_EDataFlow dataFlow, ma_ERole role, ma_IMMDevice** ppEndpoint);
- HRESULT (STDMETHODCALLTYPE * GetDevice) (ma_IMMDeviceEnumerator* pThis, LPCWSTR pID, ma_IMMDevice** ppDevice);
- HRESULT (STDMETHODCALLTYPE * RegisterEndpointNotificationCallback) (ma_IMMDeviceEnumerator* pThis, ma_IMMNotificationClient* pClient);
- HRESULT (STDMETHODCALLTYPE * UnregisterEndpointNotificationCallback)(ma_IMMDeviceEnumerator* pThis, ma_IMMNotificationClient* pClient);
- } ma_IMMDeviceEnumeratorVtbl;
- struct ma_IMMDeviceEnumerator
- {
- ma_IMMDeviceEnumeratorVtbl* lpVtbl;
- };
- static MA_INLINE HRESULT ma_IMMDeviceEnumerator_QueryInterface(ma_IMMDeviceEnumerator* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
- static MA_INLINE ULONG ma_IMMDeviceEnumerator_AddRef(ma_IMMDeviceEnumerator* pThis) { return pThis->lpVtbl->AddRef(pThis); }
- static MA_INLINE ULONG ma_IMMDeviceEnumerator_Release(ma_IMMDeviceEnumerator* pThis) { return pThis->lpVtbl->Release(pThis); }
- static MA_INLINE HRESULT ma_IMMDeviceEnumerator_EnumAudioEndpoints(ma_IMMDeviceEnumerator* pThis, ma_EDataFlow dataFlow, DWORD dwStateMask, ma_IMMDeviceCollection** ppDevices) { return pThis->lpVtbl->EnumAudioEndpoints(pThis, dataFlow, dwStateMask, ppDevices); }
- static MA_INLINE HRESULT ma_IMMDeviceEnumerator_GetDefaultAudioEndpoint(ma_IMMDeviceEnumerator* pThis, ma_EDataFlow dataFlow, ma_ERole role, ma_IMMDevice** ppEndpoint) { return pThis->lpVtbl->GetDefaultAudioEndpoint(pThis, dataFlow, role, ppEndpoint); }
- static MA_INLINE HRESULT ma_IMMDeviceEnumerator_GetDevice(ma_IMMDeviceEnumerator* pThis, LPCWSTR pID, ma_IMMDevice** ppDevice) { return pThis->lpVtbl->GetDevice(pThis, pID, ppDevice); }
- static MA_INLINE HRESULT ma_IMMDeviceEnumerator_RegisterEndpointNotificationCallback(ma_IMMDeviceEnumerator* pThis, ma_IMMNotificationClient* pClient) { return pThis->lpVtbl->RegisterEndpointNotificationCallback(pThis, pClient); }
- static MA_INLINE HRESULT ma_IMMDeviceEnumerator_UnregisterEndpointNotificationCallback(ma_IMMDeviceEnumerator* pThis, ma_IMMNotificationClient* pClient) { return pThis->lpVtbl->UnregisterEndpointNotificationCallback(pThis, pClient); }
+ return MA_SUCCESS;
+}
+static ma_result ma_context_uninit__sndio(ma_context* pContext)
+{
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pContext->backend == ma_backend_sndio);
- /* IMMDeviceCollection */
- typedef struct
- {
- /* IUnknown */
- HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IMMDeviceCollection* pThis, const IID* const riid, void** ppObject);
- ULONG (STDMETHODCALLTYPE * AddRef) (ma_IMMDeviceCollection* pThis);
- ULONG (STDMETHODCALLTYPE * Release) (ma_IMMDeviceCollection* pThis);
+ (void)pContext;
+ return MA_SUCCESS;
+}
- /* IMMDeviceCollection */
- HRESULT (STDMETHODCALLTYPE * GetCount)(ma_IMMDeviceCollection* pThis, UINT* pDevices);
- HRESULT (STDMETHODCALLTYPE * Item) (ma_IMMDeviceCollection* pThis, UINT nDevice, ma_IMMDevice** ppDevice);
- } ma_IMMDeviceCollectionVtbl;
- struct ma_IMMDeviceCollection
- {
- ma_IMMDeviceCollectionVtbl* lpVtbl;
+static ma_result ma_context_init__sndio(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+{
+#ifndef MA_NO_RUNTIME_LINKING
+ const char* libsndioNames[] = {
+ "libsndio.so"
};
- static MA_INLINE HRESULT ma_IMMDeviceCollection_QueryInterface(ma_IMMDeviceCollection* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
- static MA_INLINE ULONG ma_IMMDeviceCollection_AddRef(ma_IMMDeviceCollection* pThis) { return pThis->lpVtbl->AddRef(pThis); }
- static MA_INLINE ULONG ma_IMMDeviceCollection_Release(ma_IMMDeviceCollection* pThis) { return pThis->lpVtbl->Release(pThis); }
- static MA_INLINE HRESULT ma_IMMDeviceCollection_GetCount(ma_IMMDeviceCollection* pThis, UINT* pDevices) { return pThis->lpVtbl->GetCount(pThis, pDevices); }
- static MA_INLINE HRESULT ma_IMMDeviceCollection_Item(ma_IMMDeviceCollection* pThis, UINT nDevice, ma_IMMDevice** ppDevice) { return pThis->lpVtbl->Item(pThis, nDevice, ppDevice); }
+ size_t i;
+ for (i = 0; i < ma_countof(libsndioNames); ++i) {
+ pContext->sndio.sndioSO = ma_dlopen(pContext, libsndioNames[i]);
+ if (pContext->sndio.sndioSO != NULL) {
+ break;
+ }
+ }
- /* IMMDevice */
- typedef struct
- {
- /* IUnknown */
- HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IMMDevice* pThis, const IID* const riid, void** ppObject);
- ULONG (STDMETHODCALLTYPE * AddRef) (ma_IMMDevice* pThis);
- ULONG (STDMETHODCALLTYPE * Release) (ma_IMMDevice* pThis);
+ if (pContext->sndio.sndioSO == NULL) {
+ return MA_NO_BACKEND;
+ }
- /* IMMDevice */
- HRESULT (STDMETHODCALLTYPE * Activate) (ma_IMMDevice* pThis, const IID* const iid, DWORD dwClsCtx, PROPVARIANT* pActivationParams, void** ppInterface);
- HRESULT (STDMETHODCALLTYPE * OpenPropertyStore)(ma_IMMDevice* pThis, DWORD stgmAccess, ma_IPropertyStore** ppProperties);
- HRESULT (STDMETHODCALLTYPE * GetId) (ma_IMMDevice* pThis, LPWSTR *pID);
- HRESULT (STDMETHODCALLTYPE * GetState) (ma_IMMDevice* pThis, DWORD *pState);
- } ma_IMMDeviceVtbl;
- struct ma_IMMDevice
- {
- ma_IMMDeviceVtbl* lpVtbl;
- };
- static MA_INLINE HRESULT ma_IMMDevice_QueryInterface(ma_IMMDevice* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
- static MA_INLINE ULONG ma_IMMDevice_AddRef(ma_IMMDevice* pThis) { return pThis->lpVtbl->AddRef(pThis); }
- static MA_INLINE ULONG ma_IMMDevice_Release(ma_IMMDevice* pThis) { return pThis->lpVtbl->Release(pThis); }
- static MA_INLINE HRESULT ma_IMMDevice_Activate(ma_IMMDevice* pThis, const IID* const iid, DWORD dwClsCtx, PROPVARIANT* pActivationParams, void** ppInterface) { return pThis->lpVtbl->Activate(pThis, iid, dwClsCtx, pActivationParams, ppInterface); }
- static MA_INLINE HRESULT ma_IMMDevice_OpenPropertyStore(ma_IMMDevice* pThis, DWORD stgmAccess, ma_IPropertyStore** ppProperties) { return pThis->lpVtbl->OpenPropertyStore(pThis, stgmAccess, ppProperties); }
- static MA_INLINE HRESULT ma_IMMDevice_GetId(ma_IMMDevice* pThis, LPWSTR *pID) { return pThis->lpVtbl->GetId(pThis, pID); }
- static MA_INLINE HRESULT ma_IMMDevice_GetState(ma_IMMDevice* pThis, DWORD *pState) { return pThis->lpVtbl->GetState(pThis, pState); }
+ pContext->sndio.sio_open = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_open");
+ pContext->sndio.sio_close = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_close");
+ pContext->sndio.sio_setpar = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_setpar");
+ pContext->sndio.sio_getpar = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_getpar");
+ pContext->sndio.sio_getcap = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_getcap");
+ pContext->sndio.sio_write = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_write");
+ pContext->sndio.sio_read = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_read");
+ pContext->sndio.sio_start = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_start");
+ pContext->sndio.sio_stop = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_stop");
+ pContext->sndio.sio_initpar = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_initpar");
#else
- /* IActivateAudioInterfaceAsyncOperation */
- typedef struct
- {
- /* IUnknown */
- HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IActivateAudioInterfaceAsyncOperation* pThis, const IID* const riid, void** ppObject);
- ULONG (STDMETHODCALLTYPE * AddRef) (ma_IActivateAudioInterfaceAsyncOperation* pThis);
- ULONG (STDMETHODCALLTYPE * Release) (ma_IActivateAudioInterfaceAsyncOperation* pThis);
+ pContext->sndio.sio_open = sio_open;
+ pContext->sndio.sio_close = sio_close;
+ pContext->sndio.sio_setpar = sio_setpar;
+ pContext->sndio.sio_getpar = sio_getpar;
+ pContext->sndio.sio_getcap = sio_getcap;
+ pContext->sndio.sio_write = sio_write;
+ pContext->sndio.sio_read = sio_read;
+ pContext->sndio.sio_start = sio_start;
+ pContext->sndio.sio_stop = sio_stop;
+ pContext->sndio.sio_initpar = sio_initpar;
+#endif
- /* IActivateAudioInterfaceAsyncOperation */
- HRESULT (STDMETHODCALLTYPE * GetActivateResult)(ma_IActivateAudioInterfaceAsyncOperation* pThis, HRESULT *pActivateResult, ma_IUnknown** ppActivatedInterface);
- } ma_IActivateAudioInterfaceAsyncOperationVtbl;
- struct ma_IActivateAudioInterfaceAsyncOperation
- {
- ma_IActivateAudioInterfaceAsyncOperationVtbl* lpVtbl;
- };
- static MA_INLINE HRESULT ma_IActivateAudioInterfaceAsyncOperation_QueryInterface(ma_IActivateAudioInterfaceAsyncOperation* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
- static MA_INLINE ULONG ma_IActivateAudioInterfaceAsyncOperation_AddRef(ma_IActivateAudioInterfaceAsyncOperation* pThis) { return pThis->lpVtbl->AddRef(pThis); }
- static MA_INLINE ULONG ma_IActivateAudioInterfaceAsyncOperation_Release(ma_IActivateAudioInterfaceAsyncOperation* pThis) { return pThis->lpVtbl->Release(pThis); }
- static MA_INLINE HRESULT ma_IActivateAudioInterfaceAsyncOperation_GetActivateResult(ma_IActivateAudioInterfaceAsyncOperation* pThis, HRESULT *pActivateResult, ma_IUnknown** ppActivatedInterface) { return pThis->lpVtbl->GetActivateResult(pThis, pActivateResult, ppActivatedInterface); }
+ pCallbacks->onContextInit = ma_context_init__sndio;
+ pCallbacks->onContextUninit = ma_context_uninit__sndio;
+ pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__sndio;
+ pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__sndio;
+ pCallbacks->onDeviceInit = ma_device_init__sndio;
+ pCallbacks->onDeviceUninit = ma_device_uninit__sndio;
+ pCallbacks->onDeviceStart = ma_device_start__sndio;
+ pCallbacks->onDeviceStop = ma_device_stop__sndio;
+ pCallbacks->onDeviceRead = ma_device_read__sndio;
+ pCallbacks->onDeviceWrite = ma_device_write__sndio;
+ pCallbacks->onDeviceDataLoop = NULL;
+
+ (void)pConfig;
+ return MA_SUCCESS;
+}
+#endif /* sndio */
+
+
+
+/******************************************************************************
+
+audio(4) Backend
+
+******************************************************************************/
+#ifdef MA_HAS_AUDIO4
+#include
+#include
+#include
+#include
+#include
+#include
+#include
+
+#if defined(__OpenBSD__)
+ #include
+ #if defined(OpenBSD) && OpenBSD >= 201709
+ #define MA_AUDIO4_USE_NEW_API
+ #endif
#endif
-/* IPropertyStore */
-typedef struct
+static void ma_construct_device_id__audio4(char* id, size_t idSize, const char* base, int deviceIndex)
{
- /* IUnknown */
- HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IPropertyStore* pThis, const IID* const riid, void** ppObject);
- ULONG (STDMETHODCALLTYPE * AddRef) (ma_IPropertyStore* pThis);
- ULONG (STDMETHODCALLTYPE * Release) (ma_IPropertyStore* pThis);
+ size_t baseLen;
- /* IPropertyStore */
- HRESULT (STDMETHODCALLTYPE * GetCount)(ma_IPropertyStore* pThis, DWORD* pPropCount);
- HRESULT (STDMETHODCALLTYPE * GetAt) (ma_IPropertyStore* pThis, DWORD propIndex, PROPERTYKEY* pPropKey);
- HRESULT (STDMETHODCALLTYPE * GetValue)(ma_IPropertyStore* pThis, const PROPERTYKEY* const pKey, PROPVARIANT* pPropVar);
- HRESULT (STDMETHODCALLTYPE * SetValue)(ma_IPropertyStore* pThis, const PROPERTYKEY* const pKey, const PROPVARIANT* const pPropVar);
- HRESULT (STDMETHODCALLTYPE * Commit) (ma_IPropertyStore* pThis);
-} ma_IPropertyStoreVtbl;
-struct ma_IPropertyStore
-{
- ma_IPropertyStoreVtbl* lpVtbl;
-};
-static MA_INLINE HRESULT ma_IPropertyStore_QueryInterface(ma_IPropertyStore* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
-static MA_INLINE ULONG ma_IPropertyStore_AddRef(ma_IPropertyStore* pThis) { return pThis->lpVtbl->AddRef(pThis); }
-static MA_INLINE ULONG ma_IPropertyStore_Release(ma_IPropertyStore* pThis) { return pThis->lpVtbl->Release(pThis); }
-static MA_INLINE HRESULT ma_IPropertyStore_GetCount(ma_IPropertyStore* pThis, DWORD* pPropCount) { return pThis->lpVtbl->GetCount(pThis, pPropCount); }
-static MA_INLINE HRESULT ma_IPropertyStore_GetAt(ma_IPropertyStore* pThis, DWORD propIndex, PROPERTYKEY* pPropKey) { return pThis->lpVtbl->GetAt(pThis, propIndex, pPropKey); }
-static MA_INLINE HRESULT ma_IPropertyStore_GetValue(ma_IPropertyStore* pThis, const PROPERTYKEY* const pKey, PROPVARIANT* pPropVar) { return pThis->lpVtbl->GetValue(pThis, pKey, pPropVar); }
-static MA_INLINE HRESULT ma_IPropertyStore_SetValue(ma_IPropertyStore* pThis, const PROPERTYKEY* const pKey, const PROPVARIANT* const pPropVar) { return pThis->lpVtbl->SetValue(pThis, pKey, pPropVar); }
-static MA_INLINE HRESULT ma_IPropertyStore_Commit(ma_IPropertyStore* pThis) { return pThis->lpVtbl->Commit(pThis); }
+ MA_ASSERT(id != NULL);
+ MA_ASSERT(idSize > 0);
+ MA_ASSERT(deviceIndex >= 0);
+ baseLen = strlen(base);
+ MA_ASSERT(idSize > baseLen);
-/* IAudioClient */
-typedef struct
-{
- /* IUnknown */
- HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioClient* pThis, const IID* const riid, void** ppObject);
- ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioClient* pThis);
- ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioClient* pThis);
+ ma_strcpy_s(id, idSize, base);
+ ma_itoa_s(deviceIndex, id+baseLen, idSize-baseLen, 10);
+}
- /* IAudioClient */
- HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IAudioClient* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid);
- HRESULT (STDMETHODCALLTYPE * GetBufferSize) (ma_IAudioClient* pThis, ma_uint32* pNumBufferFrames);
- HRESULT (STDMETHODCALLTYPE * GetStreamLatency) (ma_IAudioClient* pThis, MA_REFERENCE_TIME* pLatency);
- HRESULT (STDMETHODCALLTYPE * GetCurrentPadding)(ma_IAudioClient* pThis, ma_uint32* pNumPaddingFrames);
- HRESULT (STDMETHODCALLTYPE * IsFormatSupported)(ma_IAudioClient* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch);
- HRESULT (STDMETHODCALLTYPE * GetMixFormat) (ma_IAudioClient* pThis, WAVEFORMATEX** ppDeviceFormat);
- HRESULT (STDMETHODCALLTYPE * GetDevicePeriod) (ma_IAudioClient* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod);
- HRESULT (STDMETHODCALLTYPE * Start) (ma_IAudioClient* pThis);
- HRESULT (STDMETHODCALLTYPE * Stop) (ma_IAudioClient* pThis);
- HRESULT (STDMETHODCALLTYPE * Reset) (ma_IAudioClient* pThis);
- HRESULT (STDMETHODCALLTYPE * SetEventHandle) (ma_IAudioClient* pThis, HANDLE eventHandle);
- HRESULT (STDMETHODCALLTYPE * GetService) (ma_IAudioClient* pThis, const IID* const riid, void** pp);
-} ma_IAudioClientVtbl;
-struct ma_IAudioClient
+static ma_result ma_extract_device_index_from_id__audio4(const char* id, const char* base, int* pIndexOut)
{
- ma_IAudioClientVtbl* lpVtbl;
-};
-static MA_INLINE HRESULT ma_IAudioClient_QueryInterface(ma_IAudioClient* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
-static MA_INLINE ULONG ma_IAudioClient_AddRef(ma_IAudioClient* pThis) { return pThis->lpVtbl->AddRef(pThis); }
-static MA_INLINE ULONG ma_IAudioClient_Release(ma_IAudioClient* pThis) { return pThis->lpVtbl->Release(pThis); }
-static MA_INLINE HRESULT ma_IAudioClient_Initialize(ma_IAudioClient* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid) { return pThis->lpVtbl->Initialize(pThis, shareMode, streamFlags, bufferDuration, periodicity, pFormat, pAudioSessionGuid); }
-static MA_INLINE HRESULT ma_IAudioClient_GetBufferSize(ma_IAudioClient* pThis, ma_uint32* pNumBufferFrames) { return pThis->lpVtbl->GetBufferSize(pThis, pNumBufferFrames); }
-static MA_INLINE HRESULT ma_IAudioClient_GetStreamLatency(ma_IAudioClient* pThis, MA_REFERENCE_TIME* pLatency) { return pThis->lpVtbl->GetStreamLatency(pThis, pLatency); }
-static MA_INLINE HRESULT ma_IAudioClient_GetCurrentPadding(ma_IAudioClient* pThis, ma_uint32* pNumPaddingFrames) { return pThis->lpVtbl->GetCurrentPadding(pThis, pNumPaddingFrames); }
-static MA_INLINE HRESULT ma_IAudioClient_IsFormatSupported(ma_IAudioClient* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch) { return pThis->lpVtbl->IsFormatSupported(pThis, shareMode, pFormat, ppClosestMatch); }
-static MA_INLINE HRESULT ma_IAudioClient_GetMixFormat(ma_IAudioClient* pThis, WAVEFORMATEX** ppDeviceFormat) { return pThis->lpVtbl->GetMixFormat(pThis, ppDeviceFormat); }
-static MA_INLINE HRESULT ma_IAudioClient_GetDevicePeriod(ma_IAudioClient* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod) { return pThis->lpVtbl->GetDevicePeriod(pThis, pDefaultDevicePeriod, pMinimumDevicePeriod); }
-static MA_INLINE HRESULT ma_IAudioClient_Start(ma_IAudioClient* pThis) { return pThis->lpVtbl->Start(pThis); }
-static MA_INLINE HRESULT ma_IAudioClient_Stop(ma_IAudioClient* pThis) { return pThis->lpVtbl->Stop(pThis); }
-static MA_INLINE HRESULT ma_IAudioClient_Reset(ma_IAudioClient* pThis) { return pThis->lpVtbl->Reset(pThis); }
-static MA_INLINE HRESULT ma_IAudioClient_SetEventHandle(ma_IAudioClient* pThis, HANDLE eventHandle) { return pThis->lpVtbl->SetEventHandle(pThis, eventHandle); }
-static MA_INLINE HRESULT ma_IAudioClient_GetService(ma_IAudioClient* pThis, const IID* const riid, void** pp) { return pThis->lpVtbl->GetService(pThis, riid, pp); }
+ size_t idLen;
+ size_t baseLen;
+ const char* deviceIndexStr;
-/* IAudioClient2 */
-typedef struct
-{
- /* IUnknown */
- HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioClient2* pThis, const IID* const riid, void** ppObject);
- ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioClient2* pThis);
- ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioClient2* pThis);
+ MA_ASSERT(id != NULL);
+ MA_ASSERT(base != NULL);
+ MA_ASSERT(pIndexOut != NULL);
- /* IAudioClient */
- HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IAudioClient2* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid);
- HRESULT (STDMETHODCALLTYPE * GetBufferSize) (ma_IAudioClient2* pThis, ma_uint32* pNumBufferFrames);
- HRESULT (STDMETHODCALLTYPE * GetStreamLatency) (ma_IAudioClient2* pThis, MA_REFERENCE_TIME* pLatency);
- HRESULT (STDMETHODCALLTYPE * GetCurrentPadding)(ma_IAudioClient2* pThis, ma_uint32* pNumPaddingFrames);
- HRESULT (STDMETHODCALLTYPE * IsFormatSupported)(ma_IAudioClient2* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch);
- HRESULT (STDMETHODCALLTYPE * GetMixFormat) (ma_IAudioClient2* pThis, WAVEFORMATEX** ppDeviceFormat);
- HRESULT (STDMETHODCALLTYPE * GetDevicePeriod) (ma_IAudioClient2* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod);
- HRESULT (STDMETHODCALLTYPE * Start) (ma_IAudioClient2* pThis);
- HRESULT (STDMETHODCALLTYPE * Stop) (ma_IAudioClient2* pThis);
- HRESULT (STDMETHODCALLTYPE * Reset) (ma_IAudioClient2* pThis);
- HRESULT (STDMETHODCALLTYPE * SetEventHandle) (ma_IAudioClient2* pThis, HANDLE eventHandle);
- HRESULT (STDMETHODCALLTYPE * GetService) (ma_IAudioClient2* pThis, const IID* const riid, void** pp);
+ idLen = strlen(id);
+ baseLen = strlen(base);
+ if (idLen <= baseLen) {
+ return MA_ERROR; /* Doesn't look like the id starts with the base. */
+ }
- /* IAudioClient2 */
- HRESULT (STDMETHODCALLTYPE * IsOffloadCapable) (ma_IAudioClient2* pThis, MA_AUDIO_STREAM_CATEGORY category, BOOL* pOffloadCapable);
- HRESULT (STDMETHODCALLTYPE * SetClientProperties)(ma_IAudioClient2* pThis, const ma_AudioClientProperties* pProperties);
- HRESULT (STDMETHODCALLTYPE * GetBufferSizeLimits)(ma_IAudioClient2* pThis, const WAVEFORMATEX* pFormat, BOOL eventDriven, MA_REFERENCE_TIME* pMinBufferDuration, MA_REFERENCE_TIME* pMaxBufferDuration);
-} ma_IAudioClient2Vtbl;
-struct ma_IAudioClient2
+ if (strncmp(id, base, baseLen) != 0) {
+ return MA_ERROR; /* ID does not begin with base. */
+ }
+
+ deviceIndexStr = id + baseLen;
+ if (deviceIndexStr[0] == '\0') {
+ return MA_ERROR; /* No index specified in the ID. */
+ }
+
+ if (pIndexOut) {
+ *pIndexOut = atoi(deviceIndexStr);
+ }
+
+ return MA_SUCCESS;
+}
+
+
+#if !defined(MA_AUDIO4_USE_NEW_API) /* Old API */
+static ma_format ma_format_from_encoding__audio4(unsigned int encoding, unsigned int precision)
{
- ma_IAudioClient2Vtbl* lpVtbl;
-};
-static MA_INLINE HRESULT ma_IAudioClient2_QueryInterface(ma_IAudioClient2* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
-static MA_INLINE ULONG ma_IAudioClient2_AddRef(ma_IAudioClient2* pThis) { return pThis->lpVtbl->AddRef(pThis); }
-static MA_INLINE ULONG ma_IAudioClient2_Release(ma_IAudioClient2* pThis) { return pThis->lpVtbl->Release(pThis); }
-static MA_INLINE HRESULT ma_IAudioClient2_Initialize(ma_IAudioClient2* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid) { return pThis->lpVtbl->Initialize(pThis, shareMode, streamFlags, bufferDuration, periodicity, pFormat, pAudioSessionGuid); }
-static MA_INLINE HRESULT ma_IAudioClient2_GetBufferSize(ma_IAudioClient2* pThis, ma_uint32* pNumBufferFrames) { return pThis->lpVtbl->GetBufferSize(pThis, pNumBufferFrames); }
-static MA_INLINE HRESULT ma_IAudioClient2_GetStreamLatency(ma_IAudioClient2* pThis, MA_REFERENCE_TIME* pLatency) { return pThis->lpVtbl->GetStreamLatency(pThis, pLatency); }
-static MA_INLINE HRESULT ma_IAudioClient2_GetCurrentPadding(ma_IAudioClient2* pThis, ma_uint32* pNumPaddingFrames) { return pThis->lpVtbl->GetCurrentPadding(pThis, pNumPaddingFrames); }
-static MA_INLINE HRESULT ma_IAudioClient2_IsFormatSupported(ma_IAudioClient2* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch) { return pThis->lpVtbl->IsFormatSupported(pThis, shareMode, pFormat, ppClosestMatch); }
-static MA_INLINE HRESULT ma_IAudioClient2_GetMixFormat(ma_IAudioClient2* pThis, WAVEFORMATEX** ppDeviceFormat) { return pThis->lpVtbl->GetMixFormat(pThis, ppDeviceFormat); }
-static MA_INLINE HRESULT ma_IAudioClient2_GetDevicePeriod(ma_IAudioClient2* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod) { return pThis->lpVtbl->GetDevicePeriod(pThis, pDefaultDevicePeriod, pMinimumDevicePeriod); }
-static MA_INLINE HRESULT ma_IAudioClient2_Start(ma_IAudioClient2* pThis) { return pThis->lpVtbl->Start(pThis); }
-static MA_INLINE HRESULT ma_IAudioClient2_Stop(ma_IAudioClient2* pThis) { return pThis->lpVtbl->Stop(pThis); }
-static MA_INLINE HRESULT ma_IAudioClient2_Reset(ma_IAudioClient2* pThis) { return pThis->lpVtbl->Reset(pThis); }
-static MA_INLINE HRESULT ma_IAudioClient2_SetEventHandle(ma_IAudioClient2* pThis, HANDLE eventHandle) { return pThis->lpVtbl->SetEventHandle(pThis, eventHandle); }
-static MA_INLINE HRESULT ma_IAudioClient2_GetService(ma_IAudioClient2* pThis, const IID* const riid, void** pp) { return pThis->lpVtbl->GetService(pThis, riid, pp); }
-static MA_INLINE HRESULT ma_IAudioClient2_IsOffloadCapable(ma_IAudioClient2* pThis, MA_AUDIO_STREAM_CATEGORY category, BOOL* pOffloadCapable) { return pThis->lpVtbl->IsOffloadCapable(pThis, category, pOffloadCapable); }
-static MA_INLINE HRESULT ma_IAudioClient2_SetClientProperties(ma_IAudioClient2* pThis, const ma_AudioClientProperties* pProperties) { return pThis->lpVtbl->SetClientProperties(pThis, pProperties); }
-static MA_INLINE HRESULT ma_IAudioClient2_GetBufferSizeLimits(ma_IAudioClient2* pThis, const WAVEFORMATEX* pFormat, BOOL eventDriven, MA_REFERENCE_TIME* pMinBufferDuration, MA_REFERENCE_TIME* pMaxBufferDuration) { return pThis->lpVtbl->GetBufferSizeLimits(pThis, pFormat, eventDriven, pMinBufferDuration, pMaxBufferDuration); }
+ if (precision == 8 && (encoding == AUDIO_ENCODING_ULINEAR || encoding == AUDIO_ENCODING_ULINEAR || encoding == AUDIO_ENCODING_ULINEAR_LE || encoding == AUDIO_ENCODING_ULINEAR_BE)) {
+ return ma_format_u8;
+ } else {
+ if (ma_is_little_endian() && encoding == AUDIO_ENCODING_SLINEAR_LE) {
+ if (precision == 16) {
+ return ma_format_s16;
+ } else if (precision == 24) {
+ return ma_format_s24;
+ } else if (precision == 32) {
+ return ma_format_s32;
+ }
+ } else if (ma_is_big_endian() && encoding == AUDIO_ENCODING_SLINEAR_BE) {
+ if (precision == 16) {
+ return ma_format_s16;
+ } else if (precision == 24) {
+ return ma_format_s24;
+ } else if (precision == 32) {
+ return ma_format_s32;
+ }
+ }
+ }
+ return ma_format_unknown; /* Encoding not supported. */
+}
-/* IAudioClient3 */
-typedef struct
+static void ma_encoding_from_format__audio4(ma_format format, unsigned int* pEncoding, unsigned int* pPrecision)
{
- /* IUnknown */
- HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioClient3* pThis, const IID* const riid, void** ppObject);
- ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioClient3* pThis);
- ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioClient3* pThis);
+ MA_ASSERT(pEncoding != NULL);
+ MA_ASSERT(pPrecision != NULL);
- /* IAudioClient */
- HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IAudioClient3* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid);
- HRESULT (STDMETHODCALLTYPE * GetBufferSize) (ma_IAudioClient3* pThis, ma_uint32* pNumBufferFrames);
- HRESULT (STDMETHODCALLTYPE * GetStreamLatency) (ma_IAudioClient3* pThis, MA_REFERENCE_TIME* pLatency);
- HRESULT (STDMETHODCALLTYPE * GetCurrentPadding)(ma_IAudioClient3* pThis, ma_uint32* pNumPaddingFrames);
- HRESULT (STDMETHODCALLTYPE * IsFormatSupported)(ma_IAudioClient3* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch);
- HRESULT (STDMETHODCALLTYPE * GetMixFormat) (ma_IAudioClient3* pThis, WAVEFORMATEX** ppDeviceFormat);
- HRESULT (STDMETHODCALLTYPE * GetDevicePeriod) (ma_IAudioClient3* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod);
- HRESULT (STDMETHODCALLTYPE * Start) (ma_IAudioClient3* pThis);
- HRESULT (STDMETHODCALLTYPE * Stop) (ma_IAudioClient3* pThis);
- HRESULT (STDMETHODCALLTYPE * Reset) (ma_IAudioClient3* pThis);
- HRESULT (STDMETHODCALLTYPE * SetEventHandle) (ma_IAudioClient3* pThis, HANDLE eventHandle);
- HRESULT (STDMETHODCALLTYPE * GetService) (ma_IAudioClient3* pThis, const IID* const riid, void** pp);
+ switch (format)
+ {
+ case ma_format_u8:
+ {
+ *pEncoding = AUDIO_ENCODING_ULINEAR;
+ *pPrecision = 8;
+ } break;
- /* IAudioClient2 */
- HRESULT (STDMETHODCALLTYPE * IsOffloadCapable) (ma_IAudioClient3* pThis, MA_AUDIO_STREAM_CATEGORY category, BOOL* pOffloadCapable);
- HRESULT (STDMETHODCALLTYPE * SetClientProperties)(ma_IAudioClient3* pThis, const ma_AudioClientProperties* pProperties);
- HRESULT (STDMETHODCALLTYPE * GetBufferSizeLimits)(ma_IAudioClient3* pThis, const WAVEFORMATEX* pFormat, BOOL eventDriven, MA_REFERENCE_TIME* pMinBufferDuration, MA_REFERENCE_TIME* pMaxBufferDuration);
+ case ma_format_s24:
+ {
+ *pEncoding = (ma_is_little_endian()) ? AUDIO_ENCODING_SLINEAR_LE : AUDIO_ENCODING_SLINEAR_BE;
+ *pPrecision = 24;
+ } break;
- /* IAudioClient3 */
- HRESULT (STDMETHODCALLTYPE * GetSharedModeEnginePeriod) (ma_IAudioClient3* pThis, const WAVEFORMATEX* pFormat, ma_uint32* pDefaultPeriodInFrames, ma_uint32* pFundamentalPeriodInFrames, ma_uint32* pMinPeriodInFrames, ma_uint32* pMaxPeriodInFrames);
- HRESULT (STDMETHODCALLTYPE * GetCurrentSharedModeEnginePeriod)(ma_IAudioClient3* pThis, WAVEFORMATEX** ppFormat, ma_uint32* pCurrentPeriodInFrames);
- HRESULT (STDMETHODCALLTYPE * InitializeSharedAudioStream) (ma_IAudioClient3* pThis, DWORD streamFlags, ma_uint32 periodInFrames, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid);
-} ma_IAudioClient3Vtbl;
-struct ma_IAudioClient3
-{
- ma_IAudioClient3Vtbl* lpVtbl;
-};
-static MA_INLINE HRESULT ma_IAudioClient3_QueryInterface(ma_IAudioClient3* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
-static MA_INLINE ULONG ma_IAudioClient3_AddRef(ma_IAudioClient3* pThis) { return pThis->lpVtbl->AddRef(pThis); }
-static MA_INLINE ULONG ma_IAudioClient3_Release(ma_IAudioClient3* pThis) { return pThis->lpVtbl->Release(pThis); }
-static MA_INLINE HRESULT ma_IAudioClient3_Initialize(ma_IAudioClient3* pThis, MA_AUDCLNT_SHAREMODE shareMode, DWORD streamFlags, MA_REFERENCE_TIME bufferDuration, MA_REFERENCE_TIME periodicity, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGuid) { return pThis->lpVtbl->Initialize(pThis, shareMode, streamFlags, bufferDuration, periodicity, pFormat, pAudioSessionGuid); }
-static MA_INLINE HRESULT ma_IAudioClient3_GetBufferSize(ma_IAudioClient3* pThis, ma_uint32* pNumBufferFrames) { return pThis->lpVtbl->GetBufferSize(pThis, pNumBufferFrames); }
-static MA_INLINE HRESULT ma_IAudioClient3_GetStreamLatency(ma_IAudioClient3* pThis, MA_REFERENCE_TIME* pLatency) { return pThis->lpVtbl->GetStreamLatency(pThis, pLatency); }
-static MA_INLINE HRESULT ma_IAudioClient3_GetCurrentPadding(ma_IAudioClient3* pThis, ma_uint32* pNumPaddingFrames) { return pThis->lpVtbl->GetCurrentPadding(pThis, pNumPaddingFrames); }
-static MA_INLINE HRESULT ma_IAudioClient3_IsFormatSupported(ma_IAudioClient3* pThis, MA_AUDCLNT_SHAREMODE shareMode, const WAVEFORMATEX* pFormat, WAVEFORMATEX** ppClosestMatch) { return pThis->lpVtbl->IsFormatSupported(pThis, shareMode, pFormat, ppClosestMatch); }
-static MA_INLINE HRESULT ma_IAudioClient3_GetMixFormat(ma_IAudioClient3* pThis, WAVEFORMATEX** ppDeviceFormat) { return pThis->lpVtbl->GetMixFormat(pThis, ppDeviceFormat); }
-static MA_INLINE HRESULT ma_IAudioClient3_GetDevicePeriod(ma_IAudioClient3* pThis, MA_REFERENCE_TIME* pDefaultDevicePeriod, MA_REFERENCE_TIME* pMinimumDevicePeriod) { return pThis->lpVtbl->GetDevicePeriod(pThis, pDefaultDevicePeriod, pMinimumDevicePeriod); }
-static MA_INLINE HRESULT ma_IAudioClient3_Start(ma_IAudioClient3* pThis) { return pThis->lpVtbl->Start(pThis); }
-static MA_INLINE HRESULT ma_IAudioClient3_Stop(ma_IAudioClient3* pThis) { return pThis->lpVtbl->Stop(pThis); }
-static MA_INLINE HRESULT ma_IAudioClient3_Reset(ma_IAudioClient3* pThis) { return pThis->lpVtbl->Reset(pThis); }
-static MA_INLINE HRESULT ma_IAudioClient3_SetEventHandle(ma_IAudioClient3* pThis, HANDLE eventHandle) { return pThis->lpVtbl->SetEventHandle(pThis, eventHandle); }
-static MA_INLINE HRESULT ma_IAudioClient3_GetService(ma_IAudioClient3* pThis, const IID* const riid, void** pp) { return pThis->lpVtbl->GetService(pThis, riid, pp); }
-static MA_INLINE HRESULT ma_IAudioClient3_IsOffloadCapable(ma_IAudioClient3* pThis, MA_AUDIO_STREAM_CATEGORY category, BOOL* pOffloadCapable) { return pThis->lpVtbl->IsOffloadCapable(pThis, category, pOffloadCapable); }
-static MA_INLINE HRESULT ma_IAudioClient3_SetClientProperties(ma_IAudioClient3* pThis, const ma_AudioClientProperties* pProperties) { return pThis->lpVtbl->SetClientProperties(pThis, pProperties); }
-static MA_INLINE HRESULT ma_IAudioClient3_GetBufferSizeLimits(ma_IAudioClient3* pThis, const WAVEFORMATEX* pFormat, BOOL eventDriven, MA_REFERENCE_TIME* pMinBufferDuration, MA_REFERENCE_TIME* pMaxBufferDuration) { return pThis->lpVtbl->GetBufferSizeLimits(pThis, pFormat, eventDriven, pMinBufferDuration, pMaxBufferDuration); }
-static MA_INLINE HRESULT ma_IAudioClient3_GetSharedModeEnginePeriod(ma_IAudioClient3* pThis, const WAVEFORMATEX* pFormat, ma_uint32* pDefaultPeriodInFrames, ma_uint32* pFundamentalPeriodInFrames, ma_uint32* pMinPeriodInFrames, ma_uint32* pMaxPeriodInFrames) { return pThis->lpVtbl->GetSharedModeEnginePeriod(pThis, pFormat, pDefaultPeriodInFrames, pFundamentalPeriodInFrames, pMinPeriodInFrames, pMaxPeriodInFrames); }
-static MA_INLINE HRESULT ma_IAudioClient3_GetCurrentSharedModeEnginePeriod(ma_IAudioClient3* pThis, WAVEFORMATEX** ppFormat, ma_uint32* pCurrentPeriodInFrames) { return pThis->lpVtbl->GetCurrentSharedModeEnginePeriod(pThis, ppFormat, pCurrentPeriodInFrames); }
-static MA_INLINE HRESULT ma_IAudioClient3_InitializeSharedAudioStream(ma_IAudioClient3* pThis, DWORD streamFlags, ma_uint32 periodInFrames, const WAVEFORMATEX* pFormat, const GUID* pAudioSessionGUID) { return pThis->lpVtbl->InitializeSharedAudioStream(pThis, streamFlags, periodInFrames, pFormat, pAudioSessionGUID); }
+ case ma_format_s32:
+ {
+ *pEncoding = (ma_is_little_endian()) ? AUDIO_ENCODING_SLINEAR_LE : AUDIO_ENCODING_SLINEAR_BE;
+ *pPrecision = 32;
+ } break;
+ case ma_format_s16:
+ case ma_format_f32:
+ case ma_format_unknown:
+ default:
+ {
+ *pEncoding = (ma_is_little_endian()) ? AUDIO_ENCODING_SLINEAR_LE : AUDIO_ENCODING_SLINEAR_BE;
+ *pPrecision = 16;
+ } break;
+ }
+}
-/* IAudioRenderClient */
-typedef struct
+static ma_format ma_format_from_prinfo__audio4(struct audio_prinfo* prinfo)
{
- /* IUnknown */
- HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioRenderClient* pThis, const IID* const riid, void** ppObject);
- ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioRenderClient* pThis);
- ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioRenderClient* pThis);
+ return ma_format_from_encoding__audio4(prinfo->encoding, prinfo->precision);
+}
- /* IAudioRenderClient */
- HRESULT (STDMETHODCALLTYPE * GetBuffer) (ma_IAudioRenderClient* pThis, ma_uint32 numFramesRequested, BYTE** ppData);
- HRESULT (STDMETHODCALLTYPE * ReleaseBuffer)(ma_IAudioRenderClient* pThis, ma_uint32 numFramesWritten, DWORD dwFlags);
-} ma_IAudioRenderClientVtbl;
-struct ma_IAudioRenderClient
+static ma_format ma_best_format_from_fd__audio4(int fd, ma_format preferredFormat)
{
- ma_IAudioRenderClientVtbl* lpVtbl;
-};
-static MA_INLINE HRESULT ma_IAudioRenderClient_QueryInterface(ma_IAudioRenderClient* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
-static MA_INLINE ULONG ma_IAudioRenderClient_AddRef(ma_IAudioRenderClient* pThis) { return pThis->lpVtbl->AddRef(pThis); }
-static MA_INLINE ULONG ma_IAudioRenderClient_Release(ma_IAudioRenderClient* pThis) { return pThis->lpVtbl->Release(pThis); }
-static MA_INLINE HRESULT ma_IAudioRenderClient_GetBuffer(ma_IAudioRenderClient* pThis, ma_uint32 numFramesRequested, BYTE** ppData) { return pThis->lpVtbl->GetBuffer(pThis, numFramesRequested, ppData); }
-static MA_INLINE HRESULT ma_IAudioRenderClient_ReleaseBuffer(ma_IAudioRenderClient* pThis, ma_uint32 numFramesWritten, DWORD dwFlags) { return pThis->lpVtbl->ReleaseBuffer(pThis, numFramesWritten, dwFlags); }
+ audio_encoding_t encoding;
+ ma_uint32 iFormat;
+ int counter = 0;
+ /* First check to see if the preferred format is supported. */
+ if (preferredFormat != ma_format_unknown) {
+ counter = 0;
+ for (;;) {
+ MA_ZERO_OBJECT(&encoding);
+ encoding.index = counter;
+ if (ioctl(fd, AUDIO_GETENC, &encoding) < 0) {
+ break;
+ }
-/* IAudioCaptureClient */
-typedef struct
-{
- /* IUnknown */
- HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IAudioCaptureClient* pThis, const IID* const riid, void** ppObject);
- ULONG (STDMETHODCALLTYPE * AddRef) (ma_IAudioCaptureClient* pThis);
- ULONG (STDMETHODCALLTYPE * Release) (ma_IAudioCaptureClient* pThis);
+ if (preferredFormat == ma_format_from_encoding__audio4(encoding.encoding, encoding.precision)) {
+ return preferredFormat; /* Found the preferred format. */
+ }
- /* IAudioRenderClient */
- HRESULT (STDMETHODCALLTYPE * GetBuffer) (ma_IAudioCaptureClient* pThis, BYTE** ppData, ma_uint32* pNumFramesToRead, DWORD* pFlags, ma_uint64* pDevicePosition, ma_uint64* pQPCPosition);
- HRESULT (STDMETHODCALLTYPE * ReleaseBuffer) (ma_IAudioCaptureClient* pThis, ma_uint32 numFramesRead);
- HRESULT (STDMETHODCALLTYPE * GetNextPacketSize)(ma_IAudioCaptureClient* pThis, ma_uint32* pNumFramesInNextPacket);
-} ma_IAudioCaptureClientVtbl;
-struct ma_IAudioCaptureClient
-{
- ma_IAudioCaptureClientVtbl* lpVtbl;
-};
-static MA_INLINE HRESULT ma_IAudioCaptureClient_QueryInterface(ma_IAudioCaptureClient* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
-static MA_INLINE ULONG ma_IAudioCaptureClient_AddRef(ma_IAudioCaptureClient* pThis) { return pThis->lpVtbl->AddRef(pThis); }
-static MA_INLINE ULONG ma_IAudioCaptureClient_Release(ma_IAudioCaptureClient* pThis) { return pThis->lpVtbl->Release(pThis); }
-static MA_INLINE HRESULT ma_IAudioCaptureClient_GetBuffer(ma_IAudioCaptureClient* pThis, BYTE** ppData, ma_uint32* pNumFramesToRead, DWORD* pFlags, ma_uint64* pDevicePosition, ma_uint64* pQPCPosition) { return pThis->lpVtbl->GetBuffer(pThis, ppData, pNumFramesToRead, pFlags, pDevicePosition, pQPCPosition); }
-static MA_INLINE HRESULT ma_IAudioCaptureClient_ReleaseBuffer(ma_IAudioCaptureClient* pThis, ma_uint32 numFramesRead) { return pThis->lpVtbl->ReleaseBuffer(pThis, numFramesRead); }
-static MA_INLINE HRESULT ma_IAudioCaptureClient_GetNextPacketSize(ma_IAudioCaptureClient* pThis, ma_uint32* pNumFramesInNextPacket) { return pThis->lpVtbl->GetNextPacketSize(pThis, pNumFramesInNextPacket); }
+ /* Getting here means this encoding does not match our preferred format so we need to more on to the next encoding. */
+ counter += 1;
+ }
+ }
-#ifndef MA_WIN32_DESKTOP
-#include
-typedef struct ma_completion_handler_uwp ma_completion_handler_uwp;
+ /* Getting here means our preferred format is not supported, so fall back to our standard priorities. */
+ for (iFormat = 0; iFormat < ma_countof(g_maFormatPriorities); iFormat += 1) {
+ ma_format format = g_maFormatPriorities[iFormat];
-typedef struct
-{
- /* IUnknown */
- HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_completion_handler_uwp* pThis, const IID* const riid, void** ppObject);
- ULONG (STDMETHODCALLTYPE * AddRef) (ma_completion_handler_uwp* pThis);
- ULONG (STDMETHODCALLTYPE * Release) (ma_completion_handler_uwp* pThis);
+ counter = 0;
+ for (;;) {
+ MA_ZERO_OBJECT(&encoding);
+ encoding.index = counter;
+ if (ioctl(fd, AUDIO_GETENC, &encoding) < 0) {
+ break;
+ }
- /* IActivateAudioInterfaceCompletionHandler */
- HRESULT (STDMETHODCALLTYPE * ActivateCompleted)(ma_completion_handler_uwp* pThis, ma_IActivateAudioInterfaceAsyncOperation* pActivateOperation);
-} ma_completion_handler_uwp_vtbl;
-struct ma_completion_handler_uwp
-{
- ma_completion_handler_uwp_vtbl* lpVtbl;
- MA_ATOMIC ma_uint32 counter;
- HANDLE hEvent;
-};
+ if (format == ma_format_from_encoding__audio4(encoding.encoding, encoding.precision)) {
+ return format; /* Found a workable format. */
+ }
-static HRESULT STDMETHODCALLTYPE ma_completion_handler_uwp_QueryInterface(ma_completion_handler_uwp* pThis, const IID* const riid, void** ppObject)
-{
- /*
- We need to "implement" IAgileObject which is just an indicator that's used internally by WASAPI for some multithreading management. To
- "implement" this, we just make sure we return pThis when the IAgileObject is requested.
- */
- if (!ma_is_guid_equal(riid, &MA_IID_IUnknown) && !ma_is_guid_equal(riid, &MA_IID_IActivateAudioInterfaceCompletionHandler) && !ma_is_guid_equal(riid, &MA_IID_IAgileObject)) {
- *ppObject = NULL;
- return E_NOINTERFACE;
+ /* Getting here means this encoding does not match our preferred format so we need to more on to the next encoding. */
+ counter += 1;
+ }
}
- /* Getting here means the IID is IUnknown or IMMNotificationClient. */
- *ppObject = (void*)pThis;
- ((ma_completion_handler_uwp_vtbl*)pThis->lpVtbl)->AddRef(pThis);
- return S_OK;
+ /* Getting here means not appropriate format was found. */
+ return ma_format_unknown;
}
-
-static ULONG STDMETHODCALLTYPE ma_completion_handler_uwp_AddRef(ma_completion_handler_uwp* pThis)
+#else
+static ma_format ma_format_from_swpar__audio4(struct audio_swpar* par)
{
- return (ULONG)c89atomic_fetch_add_32(&pThis->counter, 1) + 1;
+ if (par->bits == 8 && par->bps == 1 && par->sig == 0) {
+ return ma_format_u8;
+ }
+ if (par->bits == 16 && par->bps == 2 && par->sig == 1 && par->le == ma_is_little_endian()) {
+ return ma_format_s16;
+ }
+ if (par->bits == 24 && par->bps == 3 && par->sig == 1 && par->le == ma_is_little_endian()) {
+ return ma_format_s24;
+ }
+ if (par->bits == 32 && par->bps == 4 && par->sig == 1 && par->le == ma_is_little_endian()) {
+ return ma_format_f32;
+ }
+
+ /* Format not supported. */
+ return ma_format_unknown;
}
+#endif
-static ULONG STDMETHODCALLTYPE ma_completion_handler_uwp_Release(ma_completion_handler_uwp* pThis)
+static ma_result ma_context_get_device_info_from_fd__audio4(ma_context* pContext, ma_device_type deviceType, int fd, ma_device_info* pDeviceInfo)
{
- ma_uint32 newRefCount = c89atomic_fetch_sub_32(&pThis->counter, 1) - 1;
- if (newRefCount == 0) {
- return 0; /* We don't free anything here because we never allocate the object on the heap. */
+ audio_device_t fdDevice;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(fd >= 0);
+ MA_ASSERT(pDeviceInfo != NULL);
+
+ (void)pContext;
+ (void)deviceType;
+
+ if (ioctl(fd, AUDIO_GETDEV, &fdDevice) < 0) {
+ return MA_ERROR; /* Failed to retrieve device info. */
}
- return (ULONG)newRefCount;
-}
+ /* Name. */
+ ma_strcpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), fdDevice.name);
-static HRESULT STDMETHODCALLTYPE ma_completion_handler_uwp_ActivateCompleted(ma_completion_handler_uwp* pThis, ma_IActivateAudioInterfaceAsyncOperation* pActivateOperation)
-{
- (void)pActivateOperation;
- SetEvent(pThis->hEvent);
- return S_OK;
-}
+ #if !defined(MA_AUDIO4_USE_NEW_API)
+ {
+ audio_info_t fdInfo;
+ int counter = 0;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ if (ioctl(fd, AUDIO_GETINFO, &fdInfo) < 0) {
+ return MA_ERROR;
+ }
-static ma_completion_handler_uwp_vtbl g_maCompletionHandlerVtblInstance = {
- ma_completion_handler_uwp_QueryInterface,
- ma_completion_handler_uwp_AddRef,
- ma_completion_handler_uwp_Release,
- ma_completion_handler_uwp_ActivateCompleted
-};
+ if (deviceType == ma_device_type_playback) {
+ channels = fdInfo.play.channels;
+ sampleRate = fdInfo.play.sample_rate;
+ } else {
+ channels = fdInfo.record.channels;
+ sampleRate = fdInfo.record.sample_rate;
+ }
-static ma_result ma_completion_handler_uwp_init(ma_completion_handler_uwp* pHandler)
-{
- MA_ASSERT(pHandler != NULL);
- MA_ZERO_OBJECT(pHandler);
+ /* Supported formats. We get this by looking at the encodings. */
+ pDeviceInfo->nativeDataFormatCount = 0;
+ for (;;) {
+ audio_encoding_t encoding;
+ ma_format format;
- pHandler->lpVtbl = &g_maCompletionHandlerVtblInstance;
- pHandler->counter = 1;
- pHandler->hEvent = CreateEventW(NULL, FALSE, FALSE, NULL);
- if (pHandler->hEvent == NULL) {
- return ma_result_from_GetLastError(GetLastError());
+ MA_ZERO_OBJECT(&encoding);
+ encoding.index = counter;
+ if (ioctl(fd, AUDIO_GETENC, &encoding) < 0) {
+ break;
+ }
+
+ format = ma_format_from_encoding__audio4(encoding.encoding, encoding.precision);
+ if (format != ma_format_unknown) {
+ ma_device_info_add_native_data_format(pDeviceInfo, format, channels, sampleRate, 0);
+ }
+
+ counter += 1;
+ }
+ }
+ #else
+ {
+ struct audio_swpar fdPar;
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+
+ if (ioctl(fd, AUDIO_GETPAR, &fdPar) < 0) {
+ return MA_ERROR;
+ }
+
+ format = ma_format_from_swpar__audio4(&fdPar);
+ if (format == ma_format_unknown) {
+ return MA_FORMAT_NOT_SUPPORTED;
+ }
+
+ if (deviceType == ma_device_type_playback) {
+ channels = fdPar.pchan;
+ } else {
+ channels = fdPar.rchan;
+ }
+
+ sampleRate = fdPar.rate;
+
+ pDeviceInfo->nativeDataFormatCount = 0;
+ ma_device_info_add_native_data_format(pDeviceInfo, format, channels, sampleRate, 0);
}
+ #endif
return MA_SUCCESS;
}
-static void ma_completion_handler_uwp_uninit(ma_completion_handler_uwp* pHandler)
+static ma_result ma_context_enumerate_devices__audio4(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
{
- if (pHandler->hEvent != NULL) {
- CloseHandle(pHandler->hEvent);
+ const int maxDevices = 64;
+ char devpath[256];
+ int iDevice;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(callback != NULL);
+
+ /*
+ Every device will be named "/dev/audioN", with a "/dev/audioctlN" equivalent. We use the "/dev/audioctlN"
+ version here since we can open it even when another process has control of the "/dev/audioN" device.
+ */
+ for (iDevice = 0; iDevice < maxDevices; ++iDevice) {
+ struct stat st;
+ int fd;
+ ma_bool32 isTerminating = MA_FALSE;
+
+ ma_strcpy_s(devpath, sizeof(devpath), "/dev/audioctl");
+ ma_itoa_s(iDevice, devpath+strlen(devpath), sizeof(devpath)-strlen(devpath), 10);
+
+ if (stat(devpath, &st) < 0) {
+ break;
+ }
+
+ /* The device exists, but we need to check if it's usable as playback and/or capture. */
+
+ /* Playback. */
+ if (!isTerminating) {
+ fd = open(devpath, O_RDONLY, 0);
+ if (fd >= 0) {
+ /* Supports playback. */
+ ma_device_info deviceInfo;
+ MA_ZERO_OBJECT(&deviceInfo);
+ ma_construct_device_id__audio4(deviceInfo.id.audio4, sizeof(deviceInfo.id.audio4), "/dev/audio", iDevice);
+ if (ma_context_get_device_info_from_fd__audio4(pContext, ma_device_type_playback, fd, &deviceInfo) == MA_SUCCESS) {
+ isTerminating = !callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
+ }
+
+ close(fd);
+ }
+ }
+
+ /* Capture. */
+ if (!isTerminating) {
+ fd = open(devpath, O_WRONLY, 0);
+ if (fd >= 0) {
+ /* Supports capture. */
+ ma_device_info deviceInfo;
+ MA_ZERO_OBJECT(&deviceInfo);
+ ma_construct_device_id__audio4(deviceInfo.id.audio4, sizeof(deviceInfo.id.audio4), "/dev/audio", iDevice);
+ if (ma_context_get_device_info_from_fd__audio4(pContext, ma_device_type_capture, fd, &deviceInfo) == MA_SUCCESS) {
+ isTerminating = !callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
+ }
+
+ close(fd);
+ }
+ }
+
+ if (isTerminating) {
+ break;
+ }
}
-}
-static void ma_completion_handler_uwp_wait(ma_completion_handler_uwp* pHandler)
-{
- WaitForSingleObject(pHandler->hEvent, INFINITE);
+ return MA_SUCCESS;
}
-#endif /* !MA_WIN32_DESKTOP */
-/* We need a virtual table for our notification client object that's used for detecting changes to the default device. */
-#ifdef MA_WIN32_DESKTOP
-static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_QueryInterface(ma_IMMNotificationClient* pThis, const IID* const riid, void** ppObject)
+static ma_result ma_context_get_device_info__audio4(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
{
+ int fd = -1;
+ int deviceIndex = -1;
+ char ctlid[256];
+ ma_result result;
+
+ MA_ASSERT(pContext != NULL);
+
/*
- We care about two interfaces - IUnknown and IMMNotificationClient. If the requested IID is something else
- we just return E_NOINTERFACE. Otherwise we need to increment the reference counter and return S_OK.
+ We need to open the "/dev/audioctlN" device to get the info. To do this we need to extract the number
+ from the device ID which will be in "/dev/audioN" format.
*/
- if (!ma_is_guid_equal(riid, &MA_IID_IUnknown) && !ma_is_guid_equal(riid, &MA_IID_IMMNotificationClient)) {
- *ppObject = NULL;
- return E_NOINTERFACE;
+ if (pDeviceID == NULL) {
+ /* Default device. */
+ ma_strcpy_s(ctlid, sizeof(ctlid), "/dev/audioctl");
+ } else {
+ /* Specific device. We need to convert from "/dev/audioN" to "/dev/audioctlN". */
+ result = ma_extract_device_index_from_id__audio4(pDeviceID->audio4, "/dev/audio", &deviceIndex);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ ma_construct_device_id__audio4(ctlid, sizeof(ctlid), "/dev/audioctl", deviceIndex);
}
- /* Getting here means the IID is IUnknown or IMMNotificationClient. */
- *ppObject = (void*)pThis;
- ((ma_IMMNotificationClientVtbl*)pThis->lpVtbl)->AddRef(pThis);
- return S_OK;
-}
+ fd = open(ctlid, (deviceType == ma_device_type_playback) ? O_WRONLY : O_RDONLY, 0);
+ if (fd == -1) {
+ return MA_NO_DEVICE;
+ }
-static ULONG STDMETHODCALLTYPE ma_IMMNotificationClient_AddRef(ma_IMMNotificationClient* pThis)
-{
- return (ULONG)c89atomic_fetch_add_32(&pThis->counter, 1) + 1;
+ if (deviceIndex == -1) {
+ ma_strcpy_s(pDeviceInfo->id.audio4, sizeof(pDeviceInfo->id.audio4), "/dev/audio");
+ } else {
+ ma_construct_device_id__audio4(pDeviceInfo->id.audio4, sizeof(pDeviceInfo->id.audio4), "/dev/audio", deviceIndex);
+ }
+
+ result = ma_context_get_device_info_from_fd__audio4(pContext, deviceType, fd, pDeviceInfo);
+
+ close(fd);
+ return result;
}
-static ULONG STDMETHODCALLTYPE ma_IMMNotificationClient_Release(ma_IMMNotificationClient* pThis)
+static ma_result ma_device_uninit__audio4(ma_device* pDevice)
{
- ma_uint32 newRefCount = c89atomic_fetch_sub_32(&pThis->counter, 1) - 1;
- if (newRefCount == 0) {
- return 0; /* We don't free anything here because we never allocate the object on the heap. */
+ MA_ASSERT(pDevice != NULL);
+
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ close(pDevice->audio4.fdCapture);
}
- return (ULONG)newRefCount;
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ close(pDevice->audio4.fdPlayback);
+ }
+
+ return MA_SUCCESS;
}
-static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDeviceStateChanged(ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID, DWORD dwNewState)
+static ma_result ma_device_init_fd__audio4(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptor, ma_device_type deviceType)
{
- ma_bool32 isThisDevice = MA_FALSE;
- ma_bool32 isCapture = MA_FALSE;
- ma_bool32 isPlayback = MA_FALSE;
+ const char* pDefaultDeviceNames[] = {
+ "/dev/audio",
+ "/dev/audio0"
+ };
+ int fd;
+ int fdFlags = 0;
+ ma_format internalFormat;
+ ma_uint32 internalChannels;
+ ma_uint32 internalSampleRate;
+ ma_uint32 internalPeriodSizeInFrames;
+ ma_uint32 internalPeriods;
+
+ MA_ASSERT(pConfig != NULL);
+ MA_ASSERT(deviceType != ma_device_type_duplex);
+ MA_ASSERT(pDevice != NULL);
+
+ /* The first thing to do is open the file. */
+ if (deviceType == ma_device_type_capture) {
+ fdFlags = O_RDONLY;
+ } else {
+ fdFlags = O_WRONLY;
+ }
+ /*fdFlags |= O_NONBLOCK;*/
+
+ if (pDescriptor->pDeviceID == NULL) {
+ /* Default device. */
+ size_t iDevice;
+ for (iDevice = 0; iDevice < ma_countof(pDefaultDeviceNames); ++iDevice) {
+ fd = open(pDefaultDeviceNames[iDevice], fdFlags, 0);
+ if (fd != -1) {
+ break;
+ }
+ }
+ } else {
+ /* Specific device. */
+ fd = open(pDescriptor->pDeviceID->audio4, fdFlags, 0);
+ }
+
+ if (fd == -1) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to open device.");
+ return ma_result_from_errno(errno);
+ }
+
+ #if !defined(MA_AUDIO4_USE_NEW_API) /* Old API */
+ {
+ audio_info_t fdInfo;
+
+ /*
+ The documentation is a little bit unclear to me as to how it handles formats. It says the
+ following:
+
+ Regardless of formats supported by underlying driver, the audio driver accepts the
+ following formats.
+
+ By then the next sentence says this:
+
+ `encoding` and `precision` are one of the values obtained by AUDIO_GETENC.
+
+ It sounds like a direct contradiction to me. I'm going to play this safe any only use the
+ best sample format returned by AUDIO_GETENC. If the requested format is supported we'll
+ use that, but otherwise we'll just use our standard format priorities to pick an
+ appropriate one.
+ */
+ AUDIO_INITINFO(&fdInfo);
+
+ /* We get the driver to do as much of the data conversion as possible. */
+ if (deviceType == ma_device_type_capture) {
+ fdInfo.mode = AUMODE_RECORD;
+ ma_encoding_from_format__audio4(ma_best_format_from_fd__audio4(fd, pDescriptor->format), &fdInfo.record.encoding, &fdInfo.record.precision);
+
+ if (pDescriptor->channels != 0) {
+ fdInfo.record.channels = ma_clamp(pDescriptor->channels, 1, 12); /* From the documentation: `channels` ranges from 1 to 12. */
+ }
+
+ if (pDescriptor->sampleRate != 0) {
+ fdInfo.record.sample_rate = ma_clamp(pDescriptor->sampleRate, 1000, 192000); /* From the documentation: `frequency` ranges from 1000Hz to 192000Hz. (They mean `sample_rate` instead of `frequency`.) */
+ }
+ } else {
+ fdInfo.mode = AUMODE_PLAY;
+ ma_encoding_from_format__audio4(ma_best_format_from_fd__audio4(fd, pDescriptor->format), &fdInfo.play.encoding, &fdInfo.play.precision);
+
+ if (pDescriptor->channels != 0) {
+ fdInfo.play.channels = ma_clamp(pDescriptor->channels, 1, 12); /* From the documentation: `channels` ranges from 1 to 12. */
+ }
+
+ if (pDescriptor->sampleRate != 0) {
+ fdInfo.play.sample_rate = ma_clamp(pDescriptor->sampleRate, 1000, 192000); /* From the documentation: `frequency` ranges from 1000Hz to 192000Hz. (They mean `sample_rate` instead of `frequency`.) */
+ }
+ }
+
+ if (ioctl(fd, AUDIO_SETINFO, &fdInfo) < 0) {
+ close(fd);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to set device format. AUDIO_SETINFO failed.");
+ return ma_result_from_errno(errno);
+ }
+
+ if (ioctl(fd, AUDIO_GETINFO, &fdInfo) < 0) {
+ close(fd);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] AUDIO_GETINFO failed.");
+ return ma_result_from_errno(errno);
+ }
+
+ if (deviceType == ma_device_type_capture) {
+ internalFormat = ma_format_from_prinfo__audio4(&fdInfo.record);
+ internalChannels = fdInfo.record.channels;
+ internalSampleRate = fdInfo.record.sample_rate;
+ } else {
+ internalFormat = ma_format_from_prinfo__audio4(&fdInfo.play);
+ internalChannels = fdInfo.play.channels;
+ internalSampleRate = fdInfo.play.sample_rate;
+ }
+
+ if (internalFormat == ma_format_unknown) {
+ close(fd);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] The device's internal device format is not supported by miniaudio. The device is unusable.");
+ return MA_FORMAT_NOT_SUPPORTED;
+ }
+
+ /* Buffer. */
+ {
+ ma_uint32 internalPeriodSizeInBytes;
+
+ internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, internalSampleRate, pConfig->performanceProfile);
+
+ internalPeriodSizeInBytes = internalPeriodSizeInFrames * ma_get_bytes_per_frame(internalFormat, internalChannels);
+ if (internalPeriodSizeInBytes < 16) {
+ internalPeriodSizeInBytes = 16;
+ }
-#ifdef MA_DEBUG_OUTPUT
- /*ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "IMMNotificationClient_OnDeviceStateChanged(pDeviceID=%S, dwNewState=%u)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)", (unsigned int)dwNewState);*/
-#endif
+ internalPeriods = pDescriptor->periodCount;
+ if (internalPeriods < 2) {
+ internalPeriods = 2;
+ }
- /*
- There have been reports of a hang when a playback device is disconnected. The idea with this code is to explicitly stop the device if we detect
- that the device is disabled or has been unplugged.
- */
- if (pThis->pDevice->wasapi.allowCaptureAutoStreamRouting && (pThis->pDevice->type == ma_device_type_capture || pThis->pDevice->type == ma_device_type_duplex || pThis->pDevice->type == ma_device_type_loopback)) {
- isCapture = MA_TRUE;
- if (wcscmp(pThis->pDevice->capture.id.wasapi, pDeviceID) == 0) {
- isThisDevice = MA_TRUE;
- }
- }
+ /* What miniaudio calls a period, audio4 calls a block. */
+ AUDIO_INITINFO(&fdInfo);
+ fdInfo.hiwat = internalPeriods;
+ fdInfo.lowat = internalPeriods-1;
+ fdInfo.blocksize = internalPeriodSizeInBytes;
+ if (ioctl(fd, AUDIO_SETINFO, &fdInfo) < 0) {
+ close(fd);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to set internal buffer size. AUDIO_SETINFO failed.");
+ return ma_result_from_errno(errno);
+ }
- if (pThis->pDevice->wasapi.allowPlaybackAutoStreamRouting && (pThis->pDevice->type == ma_device_type_playback || pThis->pDevice->type == ma_device_type_duplex)) {
- isPlayback = MA_TRUE;
- if (wcscmp(pThis->pDevice->playback.id.wasapi, pDeviceID) == 0) {
- isThisDevice = MA_TRUE;
+ internalPeriods = fdInfo.hiwat;
+ internalPeriodSizeInFrames = fdInfo.blocksize / ma_get_bytes_per_frame(internalFormat, internalChannels);
}
}
+ #else
+ {
+ struct audio_swpar fdPar;
+ /* We need to retrieve the format of the device so we can know the channel count and sample rate. Then we can calculate the buffer size. */
+ if (ioctl(fd, AUDIO_GETPAR, &fdPar) < 0) {
+ close(fd);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to retrieve initial device parameters.");
+ return ma_result_from_errno(errno);
+ }
- /*
- If the device ID matches our device we need to mark our device as detached and stop it. When a
- device is added in OnDeviceAdded(), we'll restart it. We only mark it as detached if the device
- was started at the time of being removed.
- */
- if (isThisDevice) {
- if ((dwNewState & MA_MM_DEVICE_STATE_ACTIVE) == 0) {
- /*
- Unplugged or otherwise unavailable. Mark as detached if we were in a playing state. We'll
- use this to determine whether or not we need to automatically start the device when it's
- plugged back in again.
- */
- if (ma_device_get_state(pThis->pDevice) == MA_STATE_STARTED) {
- if (isPlayback) {
- pThis->pDevice->wasapi.isDetachedPlayback = MA_TRUE;
- }
- if (isCapture) {
- pThis->pDevice->wasapi.isDetachedCapture = MA_TRUE;
- }
+ internalFormat = ma_format_from_swpar__audio4(&fdPar);
+ internalChannels = (deviceType == ma_device_type_capture) ? fdPar.rchan : fdPar.pchan;
+ internalSampleRate = fdPar.rate;
- ma_device_stop(pThis->pDevice);
- }
+ if (internalFormat == ma_format_unknown) {
+ close(fd);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] The device's internal device format is not supported by miniaudio. The device is unusable.");
+ return MA_FORMAT_NOT_SUPPORTED;
}
- if ((dwNewState & MA_MM_DEVICE_STATE_ACTIVE) != 0) {
- /* The device was activated. If we were detached, we need to start it again. */
- ma_bool8 tryRestartingDevice = MA_FALSE;
+ /* Buffer. */
+ {
+ ma_uint32 internalPeriodSizeInBytes;
- if (isPlayback) {
- if (pThis->pDevice->wasapi.isDetachedPlayback) {
- pThis->pDevice->wasapi.isDetachedPlayback = MA_FALSE;
- ma_device_reroute__wasapi(pThis->pDevice, ma_device_type_playback);
- tryRestartingDevice = MA_TRUE;
- }
+ internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, internalSampleRate, pConfig->performanceProfile);
+
+ /* What miniaudio calls a period, audio4 calls a block. */
+ internalPeriodSizeInBytes = internalPeriodSizeInFrames * ma_get_bytes_per_frame(internalFormat, internalChannels);
+ if (internalPeriodSizeInBytes < 16) {
+ internalPeriodSizeInBytes = 16;
}
- if (isCapture) {
- if (pThis->pDevice->wasapi.isDetachedCapture) {
- pThis->pDevice->wasapi.isDetachedCapture = MA_FALSE;
- ma_device_reroute__wasapi(pThis->pDevice, (pThis->pDevice->type == ma_device_type_loopback) ? ma_device_type_loopback : ma_device_type_capture);
- tryRestartingDevice = MA_TRUE;
- }
+ fdPar.nblks = pDescriptor->periodCount;
+ fdPar.round = internalPeriodSizeInBytes;
+
+ if (ioctl(fd, AUDIO_SETPAR, &fdPar) < 0) {
+ close(fd);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to set device parameters.");
+ return ma_result_from_errno(errno);
}
- if (tryRestartingDevice) {
- if (pThis->pDevice->wasapi.isDetachedPlayback == MA_FALSE && pThis->pDevice->wasapi.isDetachedCapture == MA_FALSE) {
- ma_device_start(pThis->pDevice);
- }
+ if (ioctl(fd, AUDIO_GETPAR, &fdPar) < 0) {
+ close(fd);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to retrieve actual device parameters.");
+ return ma_result_from_errno(errno);
}
}
- }
- return S_OK;
-}
+ internalFormat = ma_format_from_swpar__audio4(&fdPar);
+ internalChannels = (deviceType == ma_device_type_capture) ? fdPar.rchan : fdPar.pchan;
+ internalSampleRate = fdPar.rate;
+ internalPeriods = fdPar.nblks;
+ internalPeriodSizeInFrames = fdPar.round / ma_get_bytes_per_frame(internalFormat, internalChannels);
+ }
+ #endif
-static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDeviceAdded(ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID)
-{
-#ifdef MA_DEBUG_OUTPUT
- /*ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "IMMNotificationClient_OnDeviceAdded(pDeviceID=%S)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)");*/
-#endif
+ if (internalFormat == ma_format_unknown) {
+ close(fd);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] The device's internal device format is not supported by miniaudio. The device is unusable.");
+ return MA_FORMAT_NOT_SUPPORTED;
+ }
- /* We don't need to worry about this event for our purposes. */
- (void)pThis;
- (void)pDeviceID;
- return S_OK;
-}
+ if (deviceType == ma_device_type_capture) {
+ pDevice->audio4.fdCapture = fd;
+ } else {
+ pDevice->audio4.fdPlayback = fd;
+ }
-static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDeviceRemoved(ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID)
-{
-#ifdef MA_DEBUG_OUTPUT
- /*ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "IMMNotificationClient_OnDeviceRemoved(pDeviceID=%S)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)");*/
-#endif
+ pDescriptor->format = internalFormat;
+ pDescriptor->channels = internalChannels;
+ pDescriptor->sampleRate = internalSampleRate;
+ ma_channel_map_init_standard(ma_standard_channel_map_sound4, pDescriptor->channelMap, ma_countof(pDescriptor->channelMap), internalChannels);
+ pDescriptor->periodSizeInFrames = internalPeriodSizeInFrames;
+ pDescriptor->periodCount = internalPeriods;
- /* We don't need to worry about this event for our purposes. */
- (void)pThis;
- (void)pDeviceID;
- return S_OK;
+ return MA_SUCCESS;
}
-static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDefaultDeviceChanged(ma_IMMNotificationClient* pThis, ma_EDataFlow dataFlow, ma_ERole role, LPCWSTR pDefaultDeviceID)
+static ma_result ma_device_init__audio4(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
{
-#ifdef MA_DEBUG_OUTPUT
- /*ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "IMMNotificationClient_OnDefaultDeviceChanged(dataFlow=%d, role=%d, pDefaultDeviceID=%S)\n", dataFlow, role, (pDefaultDeviceID != NULL) ? pDefaultDeviceID : L"(NULL)");*/
-#endif
+ MA_ASSERT(pDevice != NULL);
- /* We only ever use the eConsole role in miniaudio. */
- if (role != ma_eConsole) {
- ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Stream rerouting: role != eConsole\n");
- return S_OK;
- }
+ MA_ZERO_OBJECT(&pDevice->audio4);
- /* We only care about devices with the same data flow and role as the current device. */
- if ((pThis->pDevice->type == ma_device_type_playback && dataFlow != ma_eRender) ||
- (pThis->pDevice->type == ma_device_type_capture && dataFlow != ma_eCapture)) {
- ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Stream rerouting abandoned because dataFlow does match device type.\n");
- return S_OK;
+ if (pConfig->deviceType == ma_device_type_loopback) {
+ return MA_DEVICE_TYPE_NOT_SUPPORTED;
}
- /* Don't do automatic stream routing if we're not allowed. */
- if ((dataFlow == ma_eRender && pThis->pDevice->wasapi.allowPlaybackAutoStreamRouting == MA_FALSE) ||
- (dataFlow == ma_eCapture && pThis->pDevice->wasapi.allowCaptureAutoStreamRouting == MA_FALSE)) {
- ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Stream rerouting abandoned because automatic stream routing has been disabled by the device config.\n");
- return S_OK;
- }
+ pDevice->audio4.fdCapture = -1;
+ pDevice->audio4.fdPlayback = -1;
/*
- Not currently supporting automatic stream routing in exclusive mode. This is not working correctly on my machine due to
- AUDCLNT_E_DEVICE_IN_USE errors when reinitializing the device. If this is a bug in miniaudio, we can try re-enabling this once
- it's fixed.
+ The version of the operating system dictates whether or not the device is exclusive or shared. NetBSD
+ introduced in-kernel mixing which means it's shared. All other BSD flavours are exclusive as far as
+ I'm aware.
*/
- if ((dataFlow == ma_eRender && pThis->pDevice->playback.shareMode == ma_share_mode_exclusive) ||
- (dataFlow == ma_eCapture && pThis->pDevice->capture.shareMode == ma_share_mode_exclusive)) {
- ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Stream rerouting abandoned because the device shared mode is exclusive.\n");
- return S_OK;
+#if defined(__NetBSD_Version__) && __NetBSD_Version__ >= 800000000
+ /* NetBSD 8.0+ */
+ if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) ||
+ ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) {
+ return MA_SHARE_MODE_NOT_SUPPORTED;
}
+#else
+ /* All other flavors. */
+#endif
-
-
-
- /*
- Second attempt at device rerouting. We're going to retrieve the device's state at the time of
- the route change. We're then going to stop the device, reinitialize the device, and then start
- it again if the state before stopping was MA_STATE_STARTED.
- */
- {
- ma_uint32 previousState = ma_device_get_state(pThis->pDevice);
- ma_bool8 restartDevice = MA_FALSE;
-
- if (previousState == MA_STATE_STARTED) {
- ma_device_stop(pThis->pDevice);
- restartDevice = MA_TRUE;
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
+ ma_result result = ma_device_init_fd__audio4(pDevice, pConfig, pDescriptorCapture, ma_device_type_capture);
+ if (result != MA_SUCCESS) {
+ return result;
}
+ }
- if (pDefaultDeviceID != NULL) { /* <-- The input device ID will be null if there's no other device available. */
- if (dataFlow == ma_eRender) {
- ma_device_reroute__wasapi(pThis->pDevice, ma_device_type_playback);
-
- if (pThis->pDevice->wasapi.isDetachedPlayback) {
- pThis->pDevice->wasapi.isDetachedPlayback = MA_FALSE;
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ ma_result result = ma_device_init_fd__audio4(pDevice, pConfig, pDescriptorPlayback, ma_device_type_playback);
+ if (result != MA_SUCCESS) {
+ if (pConfig->deviceType == ma_device_type_duplex) {
+ close(pDevice->audio4.fdCapture);
+ }
+ return result;
+ }
+ }
- if (pThis->pDevice->type == ma_device_type_duplex && pThis->pDevice->wasapi.isDetachedCapture) {
- restartDevice = MA_FALSE; /* It's a duplex device and the capture side is detached. We cannot be restarting the device just yet. */
- } else {
- restartDevice = MA_TRUE; /* It's not a duplex device, or the capture side is also attached so we can go ahead and restart the device. */
- }
- }
- } else {
- ma_device_reroute__wasapi(pThis->pDevice, (pThis->pDevice->type == ma_device_type_loopback) ? ma_device_type_loopback : ma_device_type_capture);
+ return MA_SUCCESS;
+}
- if (pThis->pDevice->wasapi.isDetachedCapture) {
- pThis->pDevice->wasapi.isDetachedCapture = MA_FALSE;
+static ma_result ma_device_start__audio4(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
- if (pThis->pDevice->type == ma_device_type_duplex && pThis->pDevice->wasapi.isDetachedPlayback) {
- restartDevice = MA_FALSE; /* It's a duplex device and the playback side is detached. We cannot be restarting the device just yet. */
- } else {
- restartDevice = MA_TRUE; /* It's not a duplex device, or the playback side is also attached so we can go ahead and restart the device. */
- }
- }
- }
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ if (pDevice->audio4.fdCapture == -1) {
+ return MA_INVALID_ARGS;
+ }
+ }
- if (restartDevice) {
- ma_device_start(pThis->pDevice);
- }
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ if (pDevice->audio4.fdPlayback == -1) {
+ return MA_INVALID_ARGS;
}
}
- return S_OK;
+ return MA_SUCCESS;
}
-static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnPropertyValueChanged(ma_IMMNotificationClient* pThis, LPCWSTR pDeviceID, const PROPERTYKEY key)
+static ma_result ma_device_stop_fd__audio4(ma_device* pDevice, int fd)
{
-#ifdef MA_DEBUG_OUTPUT
- /*ma_log_postf(ma_device_get_log(pThis->pDevice), MA_LOG_LEVEL_DEBUG, "IMMNotificationClient_OnPropertyValueChanged(pDeviceID=%S)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)");*/
+ if (fd == -1) {
+ return MA_INVALID_ARGS;
+ }
+
+#if !defined(MA_AUDIO4_USE_NEW_API)
+ if (ioctl(fd, AUDIO_FLUSH, 0) < 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to stop device. AUDIO_FLUSH failed.");
+ return ma_result_from_errno(errno);
+ }
+#else
+ if (ioctl(fd, AUDIO_STOP, 0) < 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to stop device. AUDIO_STOP failed.");
+ return ma_result_from_errno(errno);
+ }
#endif
- (void)pThis;
- (void)pDeviceID;
- (void)key;
- return S_OK;
+ return MA_SUCCESS;
}
-static ma_IMMNotificationClientVtbl g_maNotificationCientVtbl = {
- ma_IMMNotificationClient_QueryInterface,
- ma_IMMNotificationClient_AddRef,
- ma_IMMNotificationClient_Release,
- ma_IMMNotificationClient_OnDeviceStateChanged,
- ma_IMMNotificationClient_OnDeviceAdded,
- ma_IMMNotificationClient_OnDeviceRemoved,
- ma_IMMNotificationClient_OnDefaultDeviceChanged,
- ma_IMMNotificationClient_OnPropertyValueChanged
-};
-#endif /* MA_WIN32_DESKTOP */
+static ma_result ma_device_stop__audio4(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
-#ifdef MA_WIN32_DESKTOP
-typedef ma_IMMDevice ma_WASAPIDeviceInterface;
-#else
-typedef ma_IUnknown ma_WASAPIDeviceInterface;
-#endif
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ ma_result result;
+ result = ma_device_stop_fd__audio4(pDevice, pDevice->audio4.fdCapture);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ }
-#define MA_CONTEXT_COMMAND_QUIT__WASAPI 1
-#define MA_CONTEXT_COMMAND_CREATE_IAUDIOCLIENT__WASAPI 2
-#define MA_CONTEXT_COMMAND_RELEASE_IAUDIOCLIENT__WASAPI 3
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ ma_result result;
-static ma_context_command__wasapi ma_context_init_command__wasapi(int code)
-{
- ma_context_command__wasapi cmd;
+ /* Drain the device first. If this fails we'll just need to flush without draining. Unfortunately draining isn't available on newer version of OpenBSD. */
+ #if !defined(MA_AUDIO4_USE_NEW_API)
+ ioctl(pDevice->audio4.fdPlayback, AUDIO_DRAIN, 0);
+ #endif
- MA_ZERO_OBJECT(&cmd);
- cmd.code = code;
+ /* Here is where the device is stopped immediately. */
+ result = ma_device_stop_fd__audio4(pDevice, pDevice->audio4.fdPlayback);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ }
- return cmd;
+ return MA_SUCCESS;
}
-static ma_result ma_context_post_command__wasapi(ma_context* pContext, const ma_context_command__wasapi* pCmd)
+static ma_result ma_device_write__audio4(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten)
{
- /* For now we are doing everything synchronously, but I might relax this later if the need arises. */
- ma_result result;
- ma_bool32 isUsingLocalEvent = MA_FALSE;
- ma_event localEvent;
+ int result;
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pCmd != NULL);
+ if (pFramesWritten != NULL) {
+ *pFramesWritten = 0;
+ }
- if (pCmd->pEvent == NULL) {
- isUsingLocalEvent = MA_TRUE;
+ result = write(pDevice->audio4.fdPlayback, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels));
+ if (result < 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to write data to the device.");
+ return ma_result_from_errno(errno);
+ }
- result = ma_event_init(&localEvent);
- if (result != MA_SUCCESS) {
- return result; /* Failed to create the event for this command. */
- }
+ if (pFramesWritten != NULL) {
+ *pFramesWritten = (ma_uint32)result / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
}
- /* Here is where we add the command to the list. If there's not enough room we'll spin until there is. */
- ma_mutex_lock(&pContext->wasapi.commandLock);
- {
- ma_uint32 index;
+ return MA_SUCCESS;
+}
- /* Spin until we've got some space available. */
- while (pContext->wasapi.commandCount == ma_countof(pContext->wasapi.commands)) {
- ma_yield();
- }
+static ma_result ma_device_read__audio4(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead)
+{
+ int result;
- /* Space is now available. Can safely add to the list. */
- index = (pContext->wasapi.commandIndex + pContext->wasapi.commandCount) % ma_countof(pContext->wasapi.commands);
- pContext->wasapi.commands[index] = *pCmd;
- pContext->wasapi.commands[index].pEvent = &localEvent;
- pContext->wasapi.commandCount += 1;
+ if (pFramesRead != NULL) {
+ *pFramesRead = 0;
+ }
- /* Now that the command has been added, release the semaphore so ma_context_next_command__wasapi() can return. */
- ma_semaphore_release(&pContext->wasapi.commandSem);
+ result = read(pDevice->audio4.fdCapture, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels));
+ if (result < 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to read data from the device.");
+ return ma_result_from_errno(errno);
}
- ma_mutex_unlock(&pContext->wasapi.commandLock);
- if (isUsingLocalEvent) {
- ma_event_wait(&localEvent);
- ma_event_uninit(&localEvent);
+ if (pFramesRead != NULL) {
+ *pFramesRead = (ma_uint32)result / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
}
return MA_SUCCESS;
}
-static ma_result ma_context_next_command__wasapi(ma_context* pContext, ma_context_command__wasapi* pCmd)
+static ma_result ma_context_uninit__audio4(ma_context* pContext)
{
- ma_result result = MA_SUCCESS;
-
MA_ASSERT(pContext != NULL);
- MA_ASSERT(pCmd != NULL);
-
- result = ma_semaphore_wait(&pContext->wasapi.commandSem);
- if (result == MA_SUCCESS) {
- ma_mutex_lock(&pContext->wasapi.commandLock);
- {
- *pCmd = pContext->wasapi.commands[pContext->wasapi.commandIndex];
- pContext->wasapi.commandIndex = (pContext->wasapi.commandIndex + 1) % ma_countof(pContext->wasapi.commands);
- pContext->wasapi.commandCount -= 1;
- }
- ma_mutex_unlock(&pContext->wasapi.commandLock);
- }
+ MA_ASSERT(pContext->backend == ma_backend_audio4);
- return result;
+ (void)pContext;
+ return MA_SUCCESS;
}
-static ma_thread_result MA_THREADCALL ma_context_command_thread__wasapi(void* pUserData)
+static ma_result ma_context_init__audio4(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
{
- ma_result result;
- ma_context* pContext = (ma_context*)pUserData;
MA_ASSERT(pContext != NULL);
- for (;;) {
- ma_context_command__wasapi cmd;
- result = ma_context_next_command__wasapi(pContext, &cmd);
- if (result != MA_SUCCESS) {
- break;
- }
+ (void)pConfig;
- switch (cmd.code)
- {
- case MA_CONTEXT_COMMAND_QUIT__WASAPI:
- {
- /* Do nothing. Handled after the switch. */
- } break;
+ pCallbacks->onContextInit = ma_context_init__audio4;
+ pCallbacks->onContextUninit = ma_context_uninit__audio4;
+ pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__audio4;
+ pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__audio4;
+ pCallbacks->onDeviceInit = ma_device_init__audio4;
+ pCallbacks->onDeviceUninit = ma_device_uninit__audio4;
+ pCallbacks->onDeviceStart = ma_device_start__audio4;
+ pCallbacks->onDeviceStop = ma_device_stop__audio4;
+ pCallbacks->onDeviceRead = ma_device_read__audio4;
+ pCallbacks->onDeviceWrite = ma_device_write__audio4;
+ pCallbacks->onDeviceDataLoop = NULL;
- case MA_CONTEXT_COMMAND_CREATE_IAUDIOCLIENT__WASAPI:
- {
- if (cmd.data.createAudioClient.deviceType == ma_device_type_playback) {
- *cmd.data.createAudioClient.pResult = ma_result_from_HRESULT(ma_IAudioClient_GetService((ma_IAudioClient*)cmd.data.createAudioClient.pAudioClient, &MA_IID_IAudioRenderClient, cmd.data.createAudioClient.ppAudioClientService));
- } else {
- *cmd.data.createAudioClient.pResult = ma_result_from_HRESULT(ma_IAudioClient_GetService((ma_IAudioClient*)cmd.data.createAudioClient.pAudioClient, &MA_IID_IAudioCaptureClient, cmd.data.createAudioClient.ppAudioClientService));
- }
- } break;
+ return MA_SUCCESS;
+}
+#endif /* audio4 */
- case MA_CONTEXT_COMMAND_RELEASE_IAUDIOCLIENT__WASAPI:
- {
- if (cmd.data.releaseAudioClient.deviceType == ma_device_type_playback) {
- if (cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientPlayback != NULL) {
- ma_IAudioClient_Release((ma_IAudioClient*)cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientPlayback);
- cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientPlayback = NULL;
- }
- }
- if (cmd.data.releaseAudioClient.deviceType == ma_device_type_capture) {
- if (cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientCapture != NULL) {
- ma_IAudioClient_Release((ma_IAudioClient*)cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientCapture);
- cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientCapture = NULL;
- }
- }
- } break;
+/******************************************************************************
- default:
- {
- /* Unknown command. Ignore it, but trigger an assert in debug mode so we're aware of it. */
- MA_ASSERT(MA_FALSE);
- } break;
- }
+OSS Backend
- if (cmd.pEvent != NULL) {
- ma_event_signal(cmd.pEvent);
- }
+******************************************************************************/
+#ifdef MA_HAS_OSS
+#include
+#include
+#include
+#include
- if (cmd.code == MA_CONTEXT_COMMAND_QUIT__WASAPI) {
- break; /* Received a quit message. Get out of here. */
- }
+#ifndef SNDCTL_DSP_HALT
+#define SNDCTL_DSP_HALT SNDCTL_DSP_RESET
+#endif
+
+#define MA_OSS_DEFAULT_DEVICE_NAME "/dev/dsp"
+
+static int ma_open_temp_device__oss()
+{
+ /* The OSS sample code uses "/dev/mixer" as the device for getting system properties so I'm going to do the same. */
+ int fd = open("/dev/mixer", O_RDONLY, 0);
+ if (fd >= 0) {
+ return fd;
+ }
+
+ return -1;
+}
+
+static ma_result ma_context_open_device__oss(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, int* pfd)
+{
+ const char* deviceName;
+ int flags;
+
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pfd != NULL);
+ (void)pContext;
+
+ *pfd = -1;
+
+ /* This function should only be called for playback or capture, not duplex. */
+ if (deviceType == ma_device_type_duplex) {
+ return MA_INVALID_ARGS;
+ }
+
+ deviceName = MA_OSS_DEFAULT_DEVICE_NAME;
+ if (pDeviceID != NULL) {
+ deviceName = pDeviceID->oss;
}
- return (ma_thread_result)0;
-}
-
-static ma_result ma_device_create_IAudioClient_service__wasapi(ma_context* pContext, ma_device_type deviceType, ma_IAudioClient* pAudioClient, void** ppAudioClientService)
-{
- ma_result result;
- ma_result cmdResult;
- ma_context_command__wasapi cmd = ma_context_init_command__wasapi(MA_CONTEXT_COMMAND_CREATE_IAUDIOCLIENT__WASAPI);
- cmd.data.createAudioClient.deviceType = deviceType;
- cmd.data.createAudioClient.pAudioClient = (void*)pAudioClient;
- cmd.data.createAudioClient.ppAudioClientService = ppAudioClientService;
- cmd.data.createAudioClient.pResult = &cmdResult; /* Declared locally, but won't be dereferenced after this function returns since execution of the command will wait here. */
+ flags = (deviceType == ma_device_type_playback) ? O_WRONLY : O_RDONLY;
+ if (shareMode == ma_share_mode_exclusive) {
+ flags |= O_EXCL;
+ }
- result = ma_context_post_command__wasapi(pContext, &cmd); /* This will not return until the command has actually been run. */
- if (result != MA_SUCCESS) {
- return result;
+ *pfd = open(deviceName, flags, 0);
+ if (*pfd == -1) {
+ return ma_result_from_errno(errno);
}
- return *cmd.data.createAudioClient.pResult;
+ return MA_SUCCESS;
}
-#if 0 /* Not used at the moment, but leaving here for future use. */
-static ma_result ma_device_release_IAudioClient_service__wasapi(ma_device* pDevice, ma_device_type deviceType)
+static ma_result ma_context_enumerate_devices__oss(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
{
- ma_result result;
- ma_context_command__wasapi cmd = ma_context_init_command__wasapi(MA_CONTEXT_COMMAND_RELEASE_IAUDIOCLIENT__WASAPI);
- cmd.data.releaseAudioClient.pDevice = pDevice;
- cmd.data.releaseAudioClient.deviceType = deviceType;
+ int fd;
+ oss_sysinfo si;
+ int result;
- result = ma_context_post_command__wasapi(pDevice->pContext, &cmd); /* This will not return until the command has actually been run. */
- if (result != MA_SUCCESS) {
- return result;
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(callback != NULL);
+
+ fd = ma_open_temp_device__oss();
+ if (fd == -1) {
+ ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[OSS] Failed to open a temporary device for retrieving system information used for device enumeration.");
+ return MA_NO_BACKEND;
}
- return MA_SUCCESS;
-}
-#endif
+ result = ioctl(fd, SNDCTL_SYSINFO, &si);
+ if (result != -1) {
+ int iAudioDevice;
+ for (iAudioDevice = 0; iAudioDevice < si.numaudios; ++iAudioDevice) {
+ oss_audioinfo ai;
+ ai.dev = iAudioDevice;
+ result = ioctl(fd, SNDCTL_AUDIOINFO, &ai);
+ if (result != -1) {
+ if (ai.devnode[0] != '\0') { /* <-- Can be blank, according to documentation. */
+ ma_device_info deviceInfo;
+ ma_bool32 isTerminating = MA_FALSE;
+ MA_ZERO_OBJECT(&deviceInfo);
-static void ma_add_native_data_format_to_device_info_from_WAVEFORMATEX(const WAVEFORMATEX* pWF, ma_share_mode shareMode, ma_device_info* pInfo)
-{
- MA_ASSERT(pWF != NULL);
- MA_ASSERT(pInfo != NULL);
+ /* ID */
+ ma_strncpy_s(deviceInfo.id.oss, sizeof(deviceInfo.id.oss), ai.devnode, (size_t)-1);
- if (pInfo->nativeDataFormatCount >= ma_countof(pInfo->nativeDataFormats)) {
- return; /* Too many data formats. Need to ignore this one. Don't think this should ever happen with WASAPI. */
+ /*
+ The human readable device name should be in the "ai.handle" variable, but it can
+ sometimes be empty in which case we just fall back to "ai.name" which is less user
+ friendly, but usually has a value.
+ */
+ if (ai.handle[0] != '\0') {
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), ai.handle, (size_t)-1);
+ } else {
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), ai.name, (size_t)-1);
+ }
+
+ /* The device can be both playback and capture. */
+ if (!isTerminating && (ai.caps & PCM_CAP_OUTPUT) != 0) {
+ isTerminating = !callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
+ }
+ if (!isTerminating && (ai.caps & PCM_CAP_INPUT) != 0) {
+ isTerminating = !callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
+ }
+
+ if (isTerminating) {
+ break;
+ }
+ }
+ }
+ }
+ } else {
+ close(fd);
+ ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[OSS] Failed to retrieve system information for device enumeration.");
+ return MA_NO_BACKEND;
}
- pInfo->nativeDataFormats[pInfo->nativeDataFormatCount].format = ma_format_from_WAVEFORMATEX(pWF);
- pInfo->nativeDataFormats[pInfo->nativeDataFormatCount].channels = pWF->nChannels;
- pInfo->nativeDataFormats[pInfo->nativeDataFormatCount].sampleRate = pWF->nSamplesPerSec;
- pInfo->nativeDataFormats[pInfo->nativeDataFormatCount].flags = (shareMode == ma_share_mode_exclusive) ? MA_DATA_FORMAT_FLAG_EXCLUSIVE_MODE : 0;
- pInfo->nativeDataFormatCount += 1;
+ close(fd);
+ return MA_SUCCESS;
}
-static ma_result ma_context_get_device_info_from_IAudioClient__wasapi(ma_context* pContext, /*ma_IMMDevice**/void* pMMDevice, ma_IAudioClient* pAudioClient, ma_device_info* pInfo)
+static void ma_context_add_native_data_format__oss(ma_context* pContext, oss_audioinfo* pAudioInfo, ma_format format, ma_device_info* pDeviceInfo)
{
- HRESULT hr;
- WAVEFORMATEX* pWF = NULL;
+ unsigned int minChannels;
+ unsigned int maxChannels;
+ unsigned int iRate;
- MA_ASSERT(pAudioClient != NULL);
- MA_ASSERT(pInfo != NULL);
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pAudioInfo != NULL);
+ MA_ASSERT(pDeviceInfo != NULL);
- /* Shared Mode. We use GetMixFormat() here. */
- hr = ma_IAudioClient_GetMixFormat((ma_IAudioClient*)pAudioClient, (WAVEFORMATEX**)&pWF);
- if (SUCCEEDED(hr)) {
- ma_add_native_data_format_to_device_info_from_WAVEFORMATEX(pWF, ma_share_mode_shared, pInfo);
- } else {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve mix format for device info retrieval.", ma_result_from_HRESULT(hr));
- }
+ /* If we support all channels we just report 0. */
+ minChannels = ma_clamp(pAudioInfo->min_channels, MA_MIN_CHANNELS, MA_MAX_CHANNELS);
+ maxChannels = ma_clamp(pAudioInfo->max_channels, MA_MIN_CHANNELS, MA_MAX_CHANNELS);
/*
- Exlcusive Mode. We repeatedly call IsFormatSupported() here. This is not currently supported on
- UWP. Failure to retrieve the exclusive mode format is not considered an error, so from here on
- out, MA_SUCCESS is guaranteed to be returned.
+ OSS has this annoying thing where sample rates can be reported in two ways. We prefer explicitness,
+ which OSS has in the form of nrates/rates, however there are times where nrates can be 0, in which
+ case we'll need to use min_rate and max_rate and report only standard rates.
*/
- #ifdef MA_WIN32_DESKTOP
- {
- ma_IPropertyStore *pProperties;
-
- /*
- The first thing to do is get the format from PKEY_AudioEngine_DeviceFormat. This should give us a channel count we assume is
- correct which will simplify our searching.
- */
- hr = ma_IMMDevice_OpenPropertyStore((ma_IMMDevice*)pMMDevice, STGM_READ, &pProperties);
- if (SUCCEEDED(hr)) {
- PROPVARIANT var;
- ma_PropVariantInit(&var);
+ if (pAudioInfo->nrates > 0) {
+ for (iRate = 0; iRate < pAudioInfo->nrates; iRate += 1) {
+ unsigned int rate = pAudioInfo->rates[iRate];
- hr = ma_IPropertyStore_GetValue(pProperties, &MA_PKEY_AudioEngine_DeviceFormat, &var);
- if (SUCCEEDED(hr)) {
- pWF = (WAVEFORMATEX*)var.blob.pBlobData;
+ if (minChannels == MA_MIN_CHANNELS && maxChannels == MA_MAX_CHANNELS) {
+ ma_device_info_add_native_data_format(pDeviceInfo, format, 0, rate, 0); /* Set the channel count to 0 to indicate that all channel counts are supported. */
+ } else {
+ unsigned int iChannel;
+ for (iChannel = minChannels; iChannel <= maxChannels; iChannel += 1) {
+ ma_device_info_add_native_data_format(pDeviceInfo, format, iChannel, rate, 0);
+ }
+ }
+ }
+ } else {
+ for (iRate = 0; iRate < ma_countof(g_maStandardSampleRatePriorities); iRate += 1) {
+ ma_uint32 standardRate = g_maStandardSampleRatePriorities[iRate];
- /*
- In my testing, the format returned by PKEY_AudioEngine_DeviceFormat is suitable for exclusive mode so we check this format
- first. If this fails, fall back to a search.
- */
- hr = ma_IAudioClient_IsFormatSupported((ma_IAudioClient*)pAudioClient, MA_AUDCLNT_SHAREMODE_EXCLUSIVE, pWF, NULL);
- if (SUCCEEDED(hr)) {
- /* The format returned by PKEY_AudioEngine_DeviceFormat is supported. */
- ma_add_native_data_format_to_device_info_from_WAVEFORMATEX(pWF, ma_share_mode_exclusive, pInfo);
+ if (standardRate >= (ma_uint32)pAudioInfo->min_rate && standardRate <= (ma_uint32)pAudioInfo->max_rate) {
+ if (minChannels == MA_MIN_CHANNELS && maxChannels == MA_MAX_CHANNELS) {
+ ma_device_info_add_native_data_format(pDeviceInfo, format, 0, standardRate, 0); /* Set the channel count to 0 to indicate that all channel counts are supported. */
} else {
- /*
- The format returned by PKEY_AudioEngine_DeviceFormat is not supported, so fall back to a search. We assume the channel
- count returned by MA_PKEY_AudioEngine_DeviceFormat is valid and correct. For simplicity we're only returning one format.
- */
- ma_uint32 channels = pInfo->minChannels;
- ma_channel defaultChannelMap[MA_MAX_CHANNELS];
- WAVEFORMATEXTENSIBLE wf;
- ma_bool32 found;
- ma_uint32 iFormat;
-
- /* Make sure we don't overflow the channel map. */
- if (channels > MA_MAX_CHANNELS) {
- channels = MA_MAX_CHANNELS;
+ unsigned int iChannel;
+ for (iChannel = minChannels; iChannel <= maxChannels; iChannel += 1) {
+ ma_device_info_add_native_data_format(pDeviceInfo, format, iChannel, standardRate, 0);
}
+ }
+ }
+ }
+ }
+}
+
+static ma_result ma_context_get_device_info__oss(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+{
+ ma_bool32 foundDevice;
+ int fdTemp;
+ oss_sysinfo si;
+ int result;
- ma_get_standard_channel_map(ma_standard_channel_map_microsoft, channels, defaultChannelMap);
+ MA_ASSERT(pContext != NULL);
- MA_ZERO_OBJECT(&wf);
- wf.Format.cbSize = sizeof(wf);
- wf.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
- wf.Format.nChannels = (WORD)channels;
- wf.dwChannelMask = ma_channel_map_to_channel_mask__win32(defaultChannelMap, channels);
+ /* Handle the default device a little differently. */
+ if (pDeviceID == NULL) {
+ if (deviceType == ma_device_type_playback) {
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
+ } else {
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
+ }
- found = MA_FALSE;
- for (iFormat = 0; iFormat < ma_countof(g_maFormatPriorities); ++iFormat) {
- ma_format format = g_maFormatPriorities[iFormat];
- ma_uint32 iSampleRate;
+ return MA_SUCCESS;
+ }
- wf.Format.wBitsPerSample = (WORD)(ma_get_bytes_per_sample(format)*8);
- wf.Format.nBlockAlign = (WORD)(wf.Format.nChannels * wf.Format.wBitsPerSample / 8);
- wf.Format.nAvgBytesPerSec = wf.Format.nBlockAlign * wf.Format.nSamplesPerSec;
- wf.Samples.wValidBitsPerSample = /*(format == ma_format_s24_32) ? 24 :*/ wf.Format.wBitsPerSample;
- if (format == ma_format_f32) {
- wf.SubFormat = MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
+
+ /* If we get here it means we are _not_ using the default device. */
+ foundDevice = MA_FALSE;
+
+ fdTemp = ma_open_temp_device__oss();
+ if (fdTemp == -1) {
+ ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[OSS] Failed to open a temporary device for retrieving system information used for device enumeration.");
+ return MA_NO_BACKEND;
+ }
+
+ result = ioctl(fdTemp, SNDCTL_SYSINFO, &si);
+ if (result != -1) {
+ int iAudioDevice;
+ for (iAudioDevice = 0; iAudioDevice < si.numaudios; ++iAudioDevice) {
+ oss_audioinfo ai;
+ ai.dev = iAudioDevice;
+ result = ioctl(fdTemp, SNDCTL_AUDIOINFO, &ai);
+ if (result != -1) {
+ if (ma_strcmp(ai.devnode, pDeviceID->oss) == 0) {
+ /* It has the same name, so now just confirm the type. */
+ if ((deviceType == ma_device_type_playback && ((ai.caps & PCM_CAP_OUTPUT) != 0)) ||
+ (deviceType == ma_device_type_capture && ((ai.caps & PCM_CAP_INPUT) != 0))) {
+ unsigned int formatMask;
+
+ /* ID */
+ ma_strncpy_s(pDeviceInfo->id.oss, sizeof(pDeviceInfo->id.oss), ai.devnode, (size_t)-1);
+
+ /*
+ The human readable device name should be in the "ai.handle" variable, but it can
+ sometimes be empty in which case we just fall back to "ai.name" which is less user
+ friendly, but usually has a value.
+ */
+ if (ai.handle[0] != '\0') {
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), ai.handle, (size_t)-1);
} else {
- wf.SubFormat = MA_GUID_KSDATAFORMAT_SUBTYPE_PCM;
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), ai.name, (size_t)-1);
}
- for (iSampleRate = 0; iSampleRate < ma_countof(g_maStandardSampleRatePriorities); ++iSampleRate) {
- wf.Format.nSamplesPerSec = g_maStandardSampleRatePriorities[iSampleRate];
- hr = ma_IAudioClient_IsFormatSupported((ma_IAudioClient*)pAudioClient, MA_AUDCLNT_SHAREMODE_EXCLUSIVE, (WAVEFORMATEX*)&wf, NULL);
- if (SUCCEEDED(hr)) {
- ma_add_native_data_format_to_device_info_from_WAVEFORMATEX((WAVEFORMATEX*)&wf, ma_share_mode_exclusive, pInfo);
- found = MA_TRUE;
- break;
- }
- }
+ pDeviceInfo->nativeDataFormatCount = 0;
- if (found) {
- break;
+ if (deviceType == ma_device_type_playback) {
+ formatMask = ai.oformats;
+ } else {
+ formatMask = ai.iformats;
}
- }
- ma_PropVariantClear(pContext, &var);
+ if (((formatMask & AFMT_S16_LE) != 0 && ma_is_little_endian()) || (AFMT_S16_BE && ma_is_big_endian())) {
+ ma_context_add_native_data_format__oss(pContext, &ai, ma_format_s16, pDeviceInfo);
+ }
+ if (((formatMask & AFMT_S32_LE) != 0 && ma_is_little_endian()) || (AFMT_S32_BE && ma_is_big_endian())) {
+ ma_context_add_native_data_format__oss(pContext, &ai, ma_format_s32, pDeviceInfo);
+ }
+ if ((formatMask & AFMT_U8) != 0) {
+ ma_context_add_native_data_format__oss(pContext, &ai, ma_format_u8, pDeviceInfo);
+ }
- if (!found) {
- ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_WARNING, "[WASAPI] Failed to find suitable device format for device info retrieval.", MA_FORMAT_NOT_SUPPORTED);
+ foundDevice = MA_TRUE;
+ break;
}
}
- } else {
- ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_WARNING, "[WASAPI] Failed to retrieve device format for device info retrieval.", ma_result_from_HRESULT(hr));
}
-
- ma_IPropertyStore_Release(pProperties);
- } else {
- ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_WARNING, "[WASAPI] Failed to open property store for device info retrieval.", ma_result_from_HRESULT(hr));
}
- }
- #endif
-
- return MA_SUCCESS;
-}
-
-#ifdef MA_WIN32_DESKTOP
-static ma_EDataFlow ma_device_type_to_EDataFlow(ma_device_type deviceType)
-{
- if (deviceType == ma_device_type_playback) {
- return ma_eRender;
- } else if (deviceType == ma_device_type_capture) {
- return ma_eCapture;
} else {
- MA_ASSERT(MA_FALSE);
- return ma_eRender; /* Should never hit this. */
+ close(fdTemp);
+ ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[OSS] Failed to retrieve system information for device enumeration.");
+ return MA_NO_BACKEND;
}
-}
-
-static ma_result ma_context_create_IMMDeviceEnumerator__wasapi(ma_context* pContext, ma_IMMDeviceEnumerator** ppDeviceEnumerator)
-{
- HRESULT hr;
- ma_IMMDeviceEnumerator* pDeviceEnumerator;
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(ppDeviceEnumerator != NULL);
- *ppDeviceEnumerator = NULL; /* Safety. */
+ close(fdTemp);
- hr = ma_CoCreateInstance(pContext, MA_CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, MA_IID_IMMDeviceEnumerator, (void**)&pDeviceEnumerator);
- if (FAILED(hr)) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create device enumerator.", ma_result_from_HRESULT(hr));
+ if (!foundDevice) {
+ return MA_NO_DEVICE;
}
- *ppDeviceEnumerator = pDeviceEnumerator;
-
return MA_SUCCESS;
}
-static LPWSTR ma_context_get_default_device_id_from_IMMDeviceEnumerator__wasapi(ma_context* pContext, ma_IMMDeviceEnumerator* pDeviceEnumerator, ma_device_type deviceType)
+static ma_result ma_device_uninit__oss(ma_device* pDevice)
{
- HRESULT hr;
- ma_IMMDevice* pMMDefaultDevice = NULL;
- LPWSTR pDefaultDeviceID = NULL;
- ma_EDataFlow dataFlow;
- ma_ERole role;
-
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pDeviceEnumerator != NULL);
+ MA_ASSERT(pDevice != NULL);
- (void)pContext;
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ close(pDevice->oss.fdCapture);
+ }
- /* Grab the EDataFlow type from the device type. */
- dataFlow = ma_device_type_to_EDataFlow(deviceType);
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ close(pDevice->oss.fdPlayback);
+ }
- /* The role is always eConsole, but we may make this configurable later. */
- role = ma_eConsole;
+ return MA_SUCCESS;
+}
- hr = ma_IMMDeviceEnumerator_GetDefaultAudioEndpoint(pDeviceEnumerator, dataFlow, role, &pMMDefaultDevice);
- if (FAILED(hr)) {
- return NULL;
+static int ma_format_to_oss(ma_format format)
+{
+ int ossFormat = AFMT_U8;
+ switch (format) {
+ case ma_format_s16: ossFormat = (ma_is_little_endian()) ? AFMT_S16_LE : AFMT_S16_BE; break;
+ case ma_format_s24: ossFormat = (ma_is_little_endian()) ? AFMT_S32_LE : AFMT_S32_BE; break;
+ case ma_format_s32: ossFormat = (ma_is_little_endian()) ? AFMT_S32_LE : AFMT_S32_BE; break;
+ case ma_format_f32: ossFormat = (ma_is_little_endian()) ? AFMT_S16_LE : AFMT_S16_BE; break;
+ case ma_format_u8:
+ default: ossFormat = AFMT_U8; break;
}
- hr = ma_IMMDevice_GetId(pMMDefaultDevice, &pDefaultDeviceID);
-
- ma_IMMDevice_Release(pMMDefaultDevice);
- pMMDefaultDevice = NULL;
+ return ossFormat;
+}
- if (FAILED(hr)) {
- return NULL;
+static ma_format ma_format_from_oss(int ossFormat)
+{
+ if (ossFormat == AFMT_U8) {
+ return ma_format_u8;
+ } else {
+ if (ma_is_little_endian()) {
+ switch (ossFormat) {
+ case AFMT_S16_LE: return ma_format_s16;
+ case AFMT_S32_LE: return ma_format_s32;
+ default: return ma_format_unknown;
+ }
+ } else {
+ switch (ossFormat) {
+ case AFMT_S16_BE: return ma_format_s16;
+ case AFMT_S32_BE: return ma_format_s32;
+ default: return ma_format_unknown;
+ }
+ }
}
- return pDefaultDeviceID;
+ return ma_format_unknown;
}
-static LPWSTR ma_context_get_default_device_id__wasapi(ma_context* pContext, ma_device_type deviceType) /* Free the returned pointer with ma_CoTaskMemFree() */
+static ma_result ma_device_init_fd__oss(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptor, ma_device_type deviceType)
{
ma_result result;
- ma_IMMDeviceEnumerator* pDeviceEnumerator;
- LPWSTR pDefaultDeviceID = NULL;
+ int ossResult;
+ int fd;
+ const ma_device_id* pDeviceID = NULL;
+ ma_share_mode shareMode;
+ int ossFormat;
+ int ossChannels;
+ int ossSampleRate;
+ int ossFragment;
- MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(pConfig != NULL);
+ MA_ASSERT(deviceType != ma_device_type_duplex);
- result = ma_context_create_IMMDeviceEnumerator__wasapi(pContext, &pDeviceEnumerator);
+ pDeviceID = pDescriptor->pDeviceID;
+ shareMode = pDescriptor->shareMode;
+ ossFormat = ma_format_to_oss((pDescriptor->format != ma_format_unknown) ? pDescriptor->format : ma_format_s16); /* Use s16 by default because OSS doesn't like floating point. */
+ ossChannels = (int)(pDescriptor->channels > 0) ? pDescriptor->channels : MA_DEFAULT_CHANNELS;
+ ossSampleRate = (int)(pDescriptor->sampleRate > 0) ? pDescriptor->sampleRate : MA_DEFAULT_SAMPLE_RATE;
+
+ result = ma_context_open_device__oss(pDevice->pContext, deviceType, pDeviceID, shareMode, &fd);
if (result != MA_SUCCESS) {
- return NULL;
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to open device.");
+ return result;
}
- pDefaultDeviceID = ma_context_get_default_device_id_from_IMMDeviceEnumerator__wasapi(pContext, pDeviceEnumerator, deviceType);
-
- ma_IMMDeviceEnumerator_Release(pDeviceEnumerator);
- return pDefaultDeviceID;
-}
-
-static ma_result ma_context_get_MMDevice__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_IMMDevice** ppMMDevice)
-{
- ma_IMMDeviceEnumerator* pDeviceEnumerator;
- HRESULT hr;
-
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(ppMMDevice != NULL);
+ /*
+ The OSS documantation is very clear about the order we should be initializing the device's properties:
+ 1) Format
+ 2) Channels
+ 3) Sample rate.
+ */
- hr = ma_CoCreateInstance(pContext, MA_CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, MA_IID_IMMDeviceEnumerator, (void**)&pDeviceEnumerator);
- if (FAILED(hr)) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create IMMDeviceEnumerator.", ma_result_from_HRESULT(hr));
+ /* Format. */
+ ossResult = ioctl(fd, SNDCTL_DSP_SETFMT, &ossFormat);
+ if (ossResult == -1) {
+ close(fd);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to set format.");
+ return ma_result_from_errno(errno);
}
- if (pDeviceID == NULL) {
- hr = ma_IMMDeviceEnumerator_GetDefaultAudioEndpoint(pDeviceEnumerator, (deviceType == ma_device_type_capture) ? ma_eCapture : ma_eRender, ma_eConsole, ppMMDevice);
- } else {
- hr = ma_IMMDeviceEnumerator_GetDevice(pDeviceEnumerator, pDeviceID->wasapi, ppMMDevice);
+ /* Channels. */
+ ossResult = ioctl(fd, SNDCTL_DSP_CHANNELS, &ossChannels);
+ if (ossResult == -1) {
+ close(fd);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to set channel count.");
+ return ma_result_from_errno(errno);
}
- ma_IMMDeviceEnumerator_Release(pDeviceEnumerator);
- if (FAILED(hr)) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve IMMDevice.", ma_result_from_HRESULT(hr));
+ /* Sample Rate. */
+ ossResult = ioctl(fd, SNDCTL_DSP_SPEED, &ossSampleRate);
+ if (ossResult == -1) {
+ close(fd);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to set sample rate.");
+ return ma_result_from_errno(errno);
}
- return MA_SUCCESS;
-}
+ /*
+ Buffer.
-static ma_result ma_context_get_device_id_from_MMDevice__wasapi(ma_context* pContext, ma_IMMDevice* pMMDevice, ma_device_id* pDeviceID)
-{
- LPWSTR pDeviceIDString;
- HRESULT hr;
+ The documentation says that the fragment settings should be set as soon as possible, but I'm not sure if
+ it should be done before or after format/channels/rate.
- MA_ASSERT(pDeviceID != NULL);
+ OSS wants the fragment size in bytes and a power of 2. When setting, we specify the power, not the actual
+ value.
+ */
+ {
+ ma_uint32 periodSizeInFrames;
+ ma_uint32 periodSizeInBytes;
+ ma_uint32 ossFragmentSizePower;
- hr = ma_IMMDevice_GetId(pMMDevice, &pDeviceIDString);
- if (SUCCEEDED(hr)) {
- size_t idlen = wcslen(pDeviceIDString);
- if (idlen+1 > ma_countof(pDeviceID->wasapi)) {
- ma_CoTaskMemFree(pContext, pDeviceIDString);
- MA_ASSERT(MA_FALSE); /* NOTE: If this is triggered, please report it. It means the format of the ID must haved change and is too long to fit in our fixed sized buffer. */
- return MA_ERROR;
+ periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, (ma_uint32)ossSampleRate, pConfig->performanceProfile);
+
+ periodSizeInBytes = ma_round_to_power_of_2(periodSizeInFrames * ma_get_bytes_per_frame(ma_format_from_oss(ossFormat), ossChannels));
+ if (periodSizeInBytes < 16) {
+ periodSizeInBytes = 16;
}
- MA_COPY_MEMORY(pDeviceID->wasapi, pDeviceIDString, idlen * sizeof(wchar_t));
- pDeviceID->wasapi[idlen] = '\0';
+ ossFragmentSizePower = 4;
+ periodSizeInBytes >>= 4;
+ while (periodSizeInBytes >>= 1) {
+ ossFragmentSizePower += 1;
+ }
- ma_CoTaskMemFree(pContext, pDeviceIDString);
+ ossFragment = (int)((pConfig->periods << 16) | ossFragmentSizePower);
+ ossResult = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &ossFragment);
+ if (ossResult == -1) {
+ close(fd);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to set fragment size and period count.");
+ return ma_result_from_errno(errno);
+ }
+ }
+
+ /* Internal settings. */
+ if (deviceType == ma_device_type_capture) {
+ pDevice->oss.fdCapture = fd;
+ } else {
+ pDevice->oss.fdPlayback = fd;
+ }
+
+ pDescriptor->format = ma_format_from_oss(ossFormat);
+ pDescriptor->channels = ossChannels;
+ pDescriptor->sampleRate = ossSampleRate;
+ ma_channel_map_init_standard(ma_standard_channel_map_sound4, pDescriptor->channelMap, ma_countof(pDescriptor->channelMap), pDescriptor->channels);
+ pDescriptor->periodCount = (ma_uint32)(ossFragment >> 16);
+ pDescriptor->periodSizeInFrames = (ma_uint32)(1 << (ossFragment & 0xFFFF)) / ma_get_bytes_per_frame(pDescriptor->format, pDescriptor->channels);
- return MA_SUCCESS;
+ if (pDescriptor->format == ma_format_unknown) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] The device's internal format is not supported by miniaudio.");
+ return MA_FORMAT_NOT_SUPPORTED;
}
- return MA_ERROR;
+ return MA_SUCCESS;
}
-static ma_result ma_context_get_device_info_from_MMDevice__wasapi(ma_context* pContext, ma_IMMDevice* pMMDevice, LPWSTR pDefaultDeviceID, ma_bool32 onlySimpleInfo, ma_device_info* pInfo)
+static ma_result ma_device_init__oss(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
{
- ma_result result;
- HRESULT hr;
+ MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(pConfig != NULL);
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pMMDevice != NULL);
- MA_ASSERT(pInfo != NULL);
+ MA_ZERO_OBJECT(&pDevice->oss);
- /* ID. */
- result = ma_context_get_device_id_from_MMDevice__wasapi(pContext, pMMDevice, &pInfo->id);
- if (result == MA_SUCCESS) {
- if (pDefaultDeviceID != NULL) {
- if (wcscmp(pInfo->id.wasapi, pDefaultDeviceID) == 0) {
- pInfo->isDefault = MA_TRUE;
- }
- }
+ if (pConfig->deviceType == ma_device_type_loopback) {
+ return MA_DEVICE_TYPE_NOT_SUPPORTED;
}
- /* Description / Friendly Name */
- {
- ma_IPropertyStore *pProperties;
- hr = ma_IMMDevice_OpenPropertyStore(pMMDevice, STGM_READ, &pProperties);
- if (SUCCEEDED(hr)) {
- PROPVARIANT var;
-
- ma_PropVariantInit(&var);
- hr = ma_IPropertyStore_GetValue(pProperties, &MA_PKEY_Device_FriendlyName, &var);
- if (SUCCEEDED(hr)) {
- WideCharToMultiByte(CP_UTF8, 0, var.pwszVal, -1, pInfo->name, sizeof(pInfo->name), 0, FALSE);
- ma_PropVariantClear(pContext, &var);
- }
-
- ma_IPropertyStore_Release(pProperties);
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
+ ma_result result = ma_device_init_fd__oss(pDevice, pConfig, pDescriptorCapture, ma_device_type_capture);
+ if (result != MA_SUCCESS) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to open device.");
+ return result;
}
}
- /* Format */
- if (!onlySimpleInfo) {
- ma_IAudioClient* pAudioClient;
- hr = ma_IMMDevice_Activate(pMMDevice, &MA_IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&pAudioClient);
- if (SUCCEEDED(hr)) {
- result = ma_context_get_device_info_from_IAudioClient__wasapi(pContext, pMMDevice, pAudioClient, pInfo);
-
- ma_IAudioClient_Release(pAudioClient);
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ ma_result result = ma_device_init_fd__oss(pDevice, pConfig, pDescriptorPlayback, ma_device_type_playback);
+ if (result != MA_SUCCESS) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to open device.");
return result;
- } else {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to activate audio client for device info retrieval.", ma_result_from_HRESULT(hr));
}
}
return MA_SUCCESS;
}
-static ma_result ma_context_enumerate_devices_by_type__wasapi(ma_context* pContext, ma_IMMDeviceEnumerator* pDeviceEnumerator, ma_device_type deviceType, ma_enum_devices_callback_proc callback, void* pUserData)
-{
- ma_result result = MA_SUCCESS;
- UINT deviceCount;
- HRESULT hr;
- ma_uint32 iDevice;
- LPWSTR pDefaultDeviceID = NULL;
- ma_IMMDeviceCollection* pDeviceCollection = NULL;
-
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(callback != NULL);
-
- /* Grab the default device. We use this to know whether or not flag the returned device info as being the default. */
- pDefaultDeviceID = ma_context_get_default_device_id_from_IMMDeviceEnumerator__wasapi(pContext, pDeviceEnumerator, deviceType);
-
- /* We need to enumerate the devices which returns a device collection. */
- hr = ma_IMMDeviceEnumerator_EnumAudioEndpoints(pDeviceEnumerator, ma_device_type_to_EDataFlow(deviceType), MA_MM_DEVICE_STATE_ACTIVE, &pDeviceCollection);
- if (SUCCEEDED(hr)) {
- hr = ma_IMMDeviceCollection_GetCount(pDeviceCollection, &deviceCount);
- if (FAILED(hr)) {
- result = ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to get device count.", ma_result_from_HRESULT(hr));
- goto done;
- }
-
- for (iDevice = 0; iDevice < deviceCount; ++iDevice) {
- ma_device_info deviceInfo;
- ma_IMMDevice* pMMDevice;
-
- MA_ZERO_OBJECT(&deviceInfo);
+/*
+Note on Starting and Stopping
+=============================
+In the past I was using SNDCTL_DSP_HALT to stop the device, however this results in issues when
+trying to resume the device again. If we use SNDCTL_DSP_HALT, the next write() or read() will
+fail. Instead what we need to do is just not write or read to and from the device when the
+device is not running.
- hr = ma_IMMDeviceCollection_Item(pDeviceCollection, iDevice, &pMMDevice);
- if (SUCCEEDED(hr)) {
- result = ma_context_get_device_info_from_MMDevice__wasapi(pContext, pMMDevice, pDefaultDeviceID, MA_TRUE, &deviceInfo); /* MA_TRUE = onlySimpleInfo. */
+As a result, both the start and stop functions for OSS are just empty stubs. The starting and
+stopping logic is handled by ma_device_write__oss() and ma_device_read__oss(). These will check
+the device state, and if the device is stopped they will simply not do any kind of processing.
- ma_IMMDevice_Release(pMMDevice);
- if (result == MA_SUCCESS) {
- ma_bool32 cbResult = callback(pContext, deviceType, &deviceInfo, pUserData);
- if (cbResult == MA_FALSE) {
- break;
- }
- }
- }
- }
- }
+The downside to this technique is that I've noticed a fairly lengthy delay in stopping the
+device, up to a second. This is on a virtual machine, and as such might just be due to the
+virtual drivers, but I'm not fully sure. I am not sure how to work around this problem so for
+the moment that's just how it's going to have to be.
-done:
- if (pDefaultDeviceID != NULL) {
- ma_CoTaskMemFree(pContext, pDefaultDeviceID);
- pDefaultDeviceID = NULL;
- }
+When starting the device, OSS will automatically start it when write() or read() is called.
+*/
+static ma_result ma_device_start__oss(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
- if (pDeviceCollection != NULL) {
- ma_IMMDeviceCollection_Release(pDeviceCollection);
- pDeviceCollection = NULL;
- }
+ /* The device is automatically started with reading and writing. */
+ (void)pDevice;
- return result;
+ return MA_SUCCESS;
}
-static ma_result ma_context_get_IAudioClient_Desktop__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_IAudioClient** ppAudioClient, ma_IMMDevice** ppMMDevice)
+static ma_result ma_device_stop__oss(ma_device* pDevice)
{
- ma_result result;
- HRESULT hr;
-
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(ppAudioClient != NULL);
- MA_ASSERT(ppMMDevice != NULL);
-
- result = ma_context_get_MMDevice__wasapi(pContext, deviceType, pDeviceID, ppMMDevice);
- if (result != MA_SUCCESS) {
- return result;
- }
+ MA_ASSERT(pDevice != NULL);
- hr = ma_IMMDevice_Activate(*ppMMDevice, &MA_IID_IAudioClient, CLSCTX_ALL, NULL, (void**)ppAudioClient);
- if (FAILED(hr)) {
- return ma_result_from_HRESULT(hr);
- }
+ /* See note above on why this is empty. */
+ (void)pDevice;
return MA_SUCCESS;
}
-#else
-static ma_result ma_context_get_IAudioClient_UWP__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_IAudioClient** ppAudioClient, ma_IUnknown** ppActivatedInterface)
-{
- ma_IActivateAudioInterfaceAsyncOperation *pAsyncOp = NULL;
- ma_completion_handler_uwp completionHandler;
- IID iid;
- LPOLESTR iidStr;
- HRESULT hr;
- ma_result result;
- HRESULT activateResult;
- ma_IUnknown* pActivatedInterface;
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(ppAudioClient != NULL);
+static ma_result ma_device_write__oss(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten)
+{
+ int resultOSS;
+ ma_uint32 deviceState;
- if (pDeviceID != NULL) {
- MA_COPY_MEMORY(&iid, pDeviceID->wasapi, sizeof(iid));
- } else {
- if (deviceType == ma_device_type_playback) {
- iid = MA_IID_DEVINTERFACE_AUDIO_RENDER;
- } else {
- iid = MA_IID_DEVINTERFACE_AUDIO_CAPTURE;
- }
+ if (pFramesWritten != NULL) {
+ *pFramesWritten = 0;
}
-#if defined(__cplusplus)
- hr = StringFromIID(iid, &iidStr);
-#else
- hr = StringFromIID(&iid, &iidStr);
-#endif
- if (FAILED(hr)) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to convert device IID to string for ActivateAudioInterfaceAsync(). Out of memory.", ma_result_from_HRESULT(hr));
+ /* Don't do any processing if the device is stopped. */
+ deviceState = ma_device_get_state(pDevice);
+ if (deviceState != ma_device_state_started && deviceState != ma_device_state_starting) {
+ return MA_SUCCESS;
}
- result = ma_completion_handler_uwp_init(&completionHandler);
- if (result != MA_SUCCESS) {
- ma_CoTaskMemFree(pContext, iidStr);
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create event for waiting for ActivateAudioInterfaceAsync().", result);
+ resultOSS = write(pDevice->oss.fdPlayback, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels));
+ if (resultOSS < 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to send data from the client to the device.");
+ return ma_result_from_errno(errno);
}
-#if defined(__cplusplus)
- hr = ActivateAudioInterfaceAsync(iidStr, MA_IID_IAudioClient, NULL, (IActivateAudioInterfaceCompletionHandler*)&completionHandler, (IActivateAudioInterfaceAsyncOperation**)&pAsyncOp);
-#else
- hr = ActivateAudioInterfaceAsync(iidStr, &MA_IID_IAudioClient, NULL, (IActivateAudioInterfaceCompletionHandler*)&completionHandler, (IActivateAudioInterfaceAsyncOperation**)&pAsyncOp);
-#endif
- if (FAILED(hr)) {
- ma_completion_handler_uwp_uninit(&completionHandler);
- ma_CoTaskMemFree(pContext, iidStr);
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] ActivateAudioInterfaceAsync() failed.", ma_result_from_HRESULT(hr));
+ if (pFramesWritten != NULL) {
+ *pFramesWritten = (ma_uint32)resultOSS / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
}
- ma_CoTaskMemFree(pContext, iidStr);
+ return MA_SUCCESS;
+}
- /* Wait for the async operation for finish. */
- ma_completion_handler_uwp_wait(&completionHandler);
- ma_completion_handler_uwp_uninit(&completionHandler);
+static ma_result ma_device_read__oss(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead)
+{
+ int resultOSS;
+ ma_uint32 deviceState;
- hr = ma_IActivateAudioInterfaceAsyncOperation_GetActivateResult(pAsyncOp, &activateResult, &pActivatedInterface);
- ma_IActivateAudioInterfaceAsyncOperation_Release(pAsyncOp);
+ if (pFramesRead != NULL) {
+ *pFramesRead = 0;
+ }
- if (FAILED(hr) || FAILED(activateResult)) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to activate device.", FAILED(hr) ? ma_result_from_HRESULT(hr) : ma_result_from_HRESULT(activateResult));
+ /* Don't do any processing if the device is stopped. */
+ deviceState = ma_device_get_state(pDevice);
+ if (deviceState != ma_device_state_started && deviceState != ma_device_state_starting) {
+ return MA_SUCCESS;
}
- /* Here is where we grab the IAudioClient interface. */
- hr = ma_IUnknown_QueryInterface(pActivatedInterface, &MA_IID_IAudioClient, (void**)ppAudioClient);
- if (FAILED(hr)) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to query IAudioClient interface.", ma_result_from_HRESULT(hr));
+ resultOSS = read(pDevice->oss.fdCapture, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels));
+ if (resultOSS < 0) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to read data from the device to be sent to the client.");
+ return ma_result_from_errno(errno);
}
- if (ppActivatedInterface) {
- *ppActivatedInterface = pActivatedInterface;
- } else {
- ma_IUnknown_Release(pActivatedInterface);
+ if (pFramesRead != NULL) {
+ *pFramesRead = (ma_uint32)resultOSS / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
}
return MA_SUCCESS;
}
-#endif
-static ma_result ma_context_get_IAudioClient__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_IAudioClient** ppAudioClient, ma_WASAPIDeviceInterface** ppDeviceInterface)
+static ma_result ma_context_uninit__oss(ma_context* pContext)
{
-#ifdef MA_WIN32_DESKTOP
- return ma_context_get_IAudioClient_Desktop__wasapi(pContext, deviceType, pDeviceID, ppAudioClient, ppDeviceInterface);
-#else
- return ma_context_get_IAudioClient_UWP__wasapi(pContext, deviceType, pDeviceID, ppAudioClient, ppDeviceInterface);
-#endif
-}
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pContext->backend == ma_backend_oss);
+ (void)pContext;
+ return MA_SUCCESS;
+}
-static ma_result ma_context_enumerate_devices__wasapi(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+static ma_result ma_context_init__oss(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
{
- /* Different enumeration for desktop and UWP. */
-#ifdef MA_WIN32_DESKTOP
- /* Desktop */
- HRESULT hr;
- ma_IMMDeviceEnumerator* pDeviceEnumerator;
-
- hr = ma_CoCreateInstance(pContext, MA_CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, MA_IID_IMMDeviceEnumerator, (void**)&pDeviceEnumerator);
- if (FAILED(hr)) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create device enumerator.", ma_result_from_HRESULT(hr));
- }
-
- ma_context_enumerate_devices_by_type__wasapi(pContext, pDeviceEnumerator, ma_device_type_playback, callback, pUserData);
- ma_context_enumerate_devices_by_type__wasapi(pContext, pDeviceEnumerator, ma_device_type_capture, callback, pUserData);
-
- ma_IMMDeviceEnumerator_Release(pDeviceEnumerator);
-#else
- /*
- UWP
-
- The MMDevice API is only supported on desktop applications. For now, while I'm still figuring out how to properly enumerate
- over devices without using MMDevice, I'm restricting devices to defaults.
+ int fd;
+ int ossVersion;
+ int result;
- Hint: DeviceInformation::FindAllAsync() with DeviceClass.AudioCapture/AudioRender. https://blogs.windows.com/buildingapps/2014/05/15/real-time-audio-in-windows-store-and-windows-phone-apps/
- */
- if (callback) {
- ma_bool32 cbResult = MA_TRUE;
+ MA_ASSERT(pContext != NULL);
- /* Playback. */
- if (cbResult) {
- ma_device_info deviceInfo;
- MA_ZERO_OBJECT(&deviceInfo);
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
- deviceInfo.isDefault = MA_TRUE;
- cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
- }
+ (void)pConfig;
- /* Capture. */
- if (cbResult) {
- ma_device_info deviceInfo;
- MA_ZERO_OBJECT(&deviceInfo);
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
- deviceInfo.isDefault = MA_TRUE;
- cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
- }
+ /* Try opening a temporary device first so we can get version information. This is closed at the end. */
+ fd = ma_open_temp_device__oss();
+ if (fd == -1) {
+ ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[OSS] Failed to open temporary device for retrieving system properties."); /* Looks liks OSS isn't installed, or there are no available devices. */
+ return MA_NO_BACKEND;
}
-#endif
-
- return MA_SUCCESS;
-}
-static ma_result ma_context_get_device_info__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
-{
-#ifdef MA_WIN32_DESKTOP
- ma_result result;
- ma_IMMDevice* pMMDevice = NULL;
- LPWSTR pDefaultDeviceID = NULL;
-
- result = ma_context_get_MMDevice__wasapi(pContext, deviceType, pDeviceID, &pMMDevice);
- if (result != MA_SUCCESS) {
- return result;
+ /* Grab the OSS version. */
+ ossVersion = 0;
+ result = ioctl(fd, OSS_GETVERSION, &ossVersion);
+ if (result == -1) {
+ close(fd);
+ ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[OSS] Failed to retrieve OSS version.");
+ return MA_NO_BACKEND;
}
- /* We need the default device ID so we can set the isDefault flag in the device info. */
- pDefaultDeviceID = ma_context_get_default_device_id__wasapi(pContext, deviceType);
+ /* The file handle to temp device is no longer needed. Close ASAP. */
+ close(fd);
- result = ma_context_get_device_info_from_MMDevice__wasapi(pContext, pMMDevice, pDefaultDeviceID, MA_FALSE, pDeviceInfo); /* MA_FALSE = !onlySimpleInfo. */
+ pContext->oss.versionMajor = ((ossVersion & 0xFF0000) >> 16);
+ pContext->oss.versionMinor = ((ossVersion & 0x00FF00) >> 8);
- if (pDefaultDeviceID != NULL) {
- ma_CoTaskMemFree(pContext, pDefaultDeviceID);
- pDefaultDeviceID = NULL;
- }
+ pCallbacks->onContextInit = ma_context_init__oss;
+ pCallbacks->onContextUninit = ma_context_uninit__oss;
+ pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__oss;
+ pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__oss;
+ pCallbacks->onDeviceInit = ma_device_init__oss;
+ pCallbacks->onDeviceUninit = ma_device_uninit__oss;
+ pCallbacks->onDeviceStart = ma_device_start__oss;
+ pCallbacks->onDeviceStop = ma_device_stop__oss;
+ pCallbacks->onDeviceRead = ma_device_read__oss;
+ pCallbacks->onDeviceWrite = ma_device_write__oss;
+ pCallbacks->onDeviceDataLoop = NULL;
- ma_IMMDevice_Release(pMMDevice);
+ return MA_SUCCESS;
+}
+#endif /* OSS */
- return result;
-#else
- ma_IAudioClient* pAudioClient;
- ma_result result;
- /* UWP currently only uses default devices. */
- if (deviceType == ma_device_type_playback) {
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
- } else {
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
- }
+/******************************************************************************
- result = ma_context_get_IAudioClient_UWP__wasapi(pContext, deviceType, pDeviceID, &pAudioClient, NULL);
- if (result != MA_SUCCESS) {
- return result;
- }
+AAudio Backend
- result = ma_context_get_device_info_from_IAudioClient__wasapi(pContext, NULL, pAudioClient, pDeviceInfo);
+******************************************************************************/
+#ifdef MA_HAS_AAUDIO
- pDeviceInfo->isDefault = MA_TRUE; /* UWP only supports default devices. */
+/*#include */
- ma_IAudioClient_Release(pAudioClient);
- return result;
-#endif
-}
+typedef int32_t ma_aaudio_result_t;
+typedef int32_t ma_aaudio_direction_t;
+typedef int32_t ma_aaudio_sharing_mode_t;
+typedef int32_t ma_aaudio_format_t;
+typedef int32_t ma_aaudio_stream_state_t;
+typedef int32_t ma_aaudio_performance_mode_t;
+typedef int32_t ma_aaudio_usage_t;
+typedef int32_t ma_aaudio_content_type_t;
+typedef int32_t ma_aaudio_input_preset_t;
+typedef int32_t ma_aaudio_data_callback_result_t;
+typedef struct ma_AAudioStreamBuilder_t* ma_AAudioStreamBuilder;
+typedef struct ma_AAudioStream_t* ma_AAudioStream;
-static ma_result ma_device_uninit__wasapi(ma_device* pDevice)
-{
- MA_ASSERT(pDevice != NULL);
+#define MA_AAUDIO_UNSPECIFIED 0
-#ifdef MA_WIN32_DESKTOP
- if (pDevice->wasapi.pDeviceEnumerator) {
- ((ma_IMMDeviceEnumerator*)pDevice->wasapi.pDeviceEnumerator)->lpVtbl->UnregisterEndpointNotificationCallback((ma_IMMDeviceEnumerator*)pDevice->wasapi.pDeviceEnumerator, &pDevice->wasapi.notificationClient);
- ma_IMMDeviceEnumerator_Release((ma_IMMDeviceEnumerator*)pDevice->wasapi.pDeviceEnumerator);
- }
-#endif
+/* Result codes. miniaudio only cares about the success code. */
+#define MA_AAUDIO_OK 0
- if (pDevice->wasapi.pRenderClient) {
- ma_IAudioRenderClient_Release((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient);
- }
- if (pDevice->wasapi.pCaptureClient) {
- ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient);
- }
+/* Directions. */
+#define MA_AAUDIO_DIRECTION_OUTPUT 0
+#define MA_AAUDIO_DIRECTION_INPUT 1
- if (pDevice->wasapi.pAudioClientPlayback) {
- ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback);
- }
- if (pDevice->wasapi.pAudioClientCapture) {
- ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture);
- }
+/* Sharing modes. */
+#define MA_AAUDIO_SHARING_MODE_EXCLUSIVE 0
+#define MA_AAUDIO_SHARING_MODE_SHARED 1
- if (pDevice->wasapi.hEventPlayback) {
- CloseHandle(pDevice->wasapi.hEventPlayback);
- }
- if (pDevice->wasapi.hEventCapture) {
- CloseHandle(pDevice->wasapi.hEventCapture);
- }
+/* Formats. */
+#define MA_AAUDIO_FORMAT_PCM_I16 1
+#define MA_AAUDIO_FORMAT_PCM_FLOAT 2
- return MA_SUCCESS;
-}
+/* Stream states. */
+#define MA_AAUDIO_STREAM_STATE_UNINITIALIZED 0
+#define MA_AAUDIO_STREAM_STATE_UNKNOWN 1
+#define MA_AAUDIO_STREAM_STATE_OPEN 2
+#define MA_AAUDIO_STREAM_STATE_STARTING 3
+#define MA_AAUDIO_STREAM_STATE_STARTED 4
+#define MA_AAUDIO_STREAM_STATE_PAUSING 5
+#define MA_AAUDIO_STREAM_STATE_PAUSED 6
+#define MA_AAUDIO_STREAM_STATE_FLUSHING 7
+#define MA_AAUDIO_STREAM_STATE_FLUSHED 8
+#define MA_AAUDIO_STREAM_STATE_STOPPING 9
+#define MA_AAUDIO_STREAM_STATE_STOPPED 10
+#define MA_AAUDIO_STREAM_STATE_CLOSING 11
+#define MA_AAUDIO_STREAM_STATE_CLOSED 12
+#define MA_AAUDIO_STREAM_STATE_DISCONNECTED 13
+/* Performance modes. */
+#define MA_AAUDIO_PERFORMANCE_MODE_NONE 10
+#define MA_AAUDIO_PERFORMANCE_MODE_POWER_SAVING 11
+#define MA_AAUDIO_PERFORMANCE_MODE_LOW_LATENCY 12
-typedef struct
-{
- /* Input. */
- ma_format formatIn;
- ma_uint32 channelsIn;
- ma_uint32 sampleRateIn;
- ma_channel channelMapIn[MA_MAX_CHANNELS];
- ma_uint32 periodSizeInFramesIn;
- ma_uint32 periodSizeInMillisecondsIn;
- ma_uint32 periodsIn;
- ma_share_mode shareMode;
- ma_performance_profile performanceProfile;
- ma_bool32 noAutoConvertSRC;
- ma_bool32 noDefaultQualitySRC;
- ma_bool32 noHardwareOffloading;
+/* Usage types. */
+#define MA_AAUDIO_USAGE_MEDIA 1
+#define MA_AAUDIO_USAGE_VOICE_COMMUNICATION 2
+#define MA_AAUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING 3
+#define MA_AAUDIO_USAGE_ALARM 4
+#define MA_AAUDIO_USAGE_NOTIFICATION 5
+#define MA_AAUDIO_USAGE_NOTIFICATION_RINGTONE 6
+#define MA_AAUDIO_USAGE_NOTIFICATION_EVENT 10
+#define MA_AAUDIO_USAGE_ASSISTANCE_ACCESSIBILITY 11
+#define MA_AAUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE 12
+#define MA_AAUDIO_USAGE_ASSISTANCE_SONIFICATION 13
+#define MA_AAUDIO_USAGE_GAME 14
+#define MA_AAUDIO_USAGE_ASSISTANT 16
+#define MA_AAUDIO_SYSTEM_USAGE_EMERGENCY 1000
+#define MA_AAUDIO_SYSTEM_USAGE_SAFETY 1001
+#define MA_AAUDIO_SYSTEM_USAGE_VEHICLE_STATUS 1002
+#define MA_AAUDIO_SYSTEM_USAGE_ANNOUNCEMENT 1003
- /* Output. */
- ma_IAudioClient* pAudioClient;
- ma_IAudioRenderClient* pRenderClient;
- ma_IAudioCaptureClient* pCaptureClient;
- ma_format formatOut;
- ma_uint32 channelsOut;
- ma_uint32 sampleRateOut;
- ma_channel channelMapOut[MA_MAX_CHANNELS];
- ma_uint32 periodSizeInFramesOut;
- ma_uint32 periodsOut;
- ma_bool32 usingAudioClient3;
- char deviceName[256];
- ma_device_id id;
-} ma_device_init_internal_data__wasapi;
+/* Content types. */
+#define MA_AAUDIO_CONTENT_TYPE_SPEECH 1
+#define MA_AAUDIO_CONTENT_TYPE_MUSIC 2
+#define MA_AAUDIO_CONTENT_TYPE_MOVIE 3
+#define MA_AAUDIO_CONTENT_TYPE_SONIFICATION 4
-static ma_result ma_device_init_internal__wasapi(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_init_internal_data__wasapi* pData)
-{
- HRESULT hr;
- ma_result result = MA_SUCCESS;
- const char* errorMsg = "";
- MA_AUDCLNT_SHAREMODE shareMode = MA_AUDCLNT_SHAREMODE_SHARED;
- DWORD streamFlags = 0;
- MA_REFERENCE_TIME periodDurationInMicroseconds;
- ma_bool32 wasInitializedUsingIAudioClient3 = MA_FALSE;
- WAVEFORMATEXTENSIBLE wf;
- ma_WASAPIDeviceInterface* pDeviceInterface = NULL;
- ma_IAudioClient2* pAudioClient2;
- ma_uint32 nativeSampleRate;
+/* Input presets. */
+#define MA_AAUDIO_INPUT_PRESET_GENERIC 1
+#define MA_AAUDIO_INPUT_PRESET_CAMCORDER 5
+#define MA_AAUDIO_INPUT_PRESET_VOICE_RECOGNITION 6
+#define MA_AAUDIO_INPUT_PRESET_VOICE_COMMUNICATION 7
+#define MA_AAUDIO_INPUT_PRESET_UNPROCESSED 9
+#define MA_AAUDIO_INPUT_PRESET_VOICE_PERFORMANCE 10
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pData != NULL);
+/* Callback results. */
+#define MA_AAUDIO_CALLBACK_RESULT_CONTINUE 0
+#define MA_AAUDIO_CALLBACK_RESULT_STOP 1
- /* This function is only used to initialize one device type: either playback, capture or loopback. Never full-duplex. */
- if (deviceType == ma_device_type_duplex) {
- return MA_INVALID_ARGS;
- }
- pData->pAudioClient = NULL;
- pData->pRenderClient = NULL;
- pData->pCaptureClient = NULL;
+typedef ma_aaudio_data_callback_result_t (* ma_AAudioStream_dataCallback) (ma_AAudioStream* pStream, void* pUserData, void* pAudioData, int32_t numFrames);
+typedef void (* ma_AAudioStream_errorCallback)(ma_AAudioStream *pStream, void *pUserData, ma_aaudio_result_t error);
- streamFlags = MA_AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
- if (!pData->noAutoConvertSRC && pData->sampleRateIn != 0 && pData->shareMode != ma_share_mode_exclusive) { /* <-- Exclusive streams must use the native sample rate. */
- streamFlags |= MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM;
- }
- if (!pData->noDefaultQualitySRC && pData->sampleRateIn != 0 && (streamFlags & MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM) != 0) {
- streamFlags |= MA_AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY;
- }
- if (deviceType == ma_device_type_loopback) {
- streamFlags |= MA_AUDCLNT_STREAMFLAGS_LOOPBACK;
- }
+typedef ma_aaudio_result_t (* MA_PFN_AAudio_createStreamBuilder) (ma_AAudioStreamBuilder** ppBuilder);
+typedef ma_aaudio_result_t (* MA_PFN_AAudioStreamBuilder_delete) (ma_AAudioStreamBuilder* pBuilder);
+typedef void (* MA_PFN_AAudioStreamBuilder_setDeviceId) (ma_AAudioStreamBuilder* pBuilder, int32_t deviceId);
+typedef void (* MA_PFN_AAudioStreamBuilder_setDirection) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_direction_t direction);
+typedef void (* MA_PFN_AAudioStreamBuilder_setSharingMode) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_sharing_mode_t sharingMode);
+typedef void (* MA_PFN_AAudioStreamBuilder_setFormat) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_format_t format);
+typedef void (* MA_PFN_AAudioStreamBuilder_setChannelCount) (ma_AAudioStreamBuilder* pBuilder, int32_t channelCount);
+typedef void (* MA_PFN_AAudioStreamBuilder_setSampleRate) (ma_AAudioStreamBuilder* pBuilder, int32_t sampleRate);
+typedef void (* MA_PFN_AAudioStreamBuilder_setBufferCapacityInFrames)(ma_AAudioStreamBuilder* pBuilder, int32_t numFrames);
+typedef void (* MA_PFN_AAudioStreamBuilder_setFramesPerDataCallback) (ma_AAudioStreamBuilder* pBuilder, int32_t numFrames);
+typedef void (* MA_PFN_AAudioStreamBuilder_setDataCallback) (ma_AAudioStreamBuilder* pBuilder, ma_AAudioStream_dataCallback callback, void* pUserData);
+typedef void (* MA_PFN_AAudioStreamBuilder_setErrorCallback) (ma_AAudioStreamBuilder* pBuilder, ma_AAudioStream_errorCallback callback, void* pUserData);
+typedef void (* MA_PFN_AAudioStreamBuilder_setPerformanceMode) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_performance_mode_t mode);
+typedef void (* MA_PFN_AAudioStreamBuilder_setUsage) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_usage_t contentType);
+typedef void (* MA_PFN_AAudioStreamBuilder_setContentType) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_content_type_t contentType);
+typedef void (* MA_PFN_AAudioStreamBuilder_setInputPreset) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_input_preset_t inputPreset);
+typedef ma_aaudio_result_t (* MA_PFN_AAudioStreamBuilder_openStream) (ma_AAudioStreamBuilder* pBuilder, ma_AAudioStream** ppStream);
+typedef ma_aaudio_result_t (* MA_PFN_AAudioStream_close) (ma_AAudioStream* pStream);
+typedef ma_aaudio_stream_state_t (* MA_PFN_AAudioStream_getState) (ma_AAudioStream* pStream);
+typedef ma_aaudio_result_t (* MA_PFN_AAudioStream_waitForStateChange) (ma_AAudioStream* pStream, ma_aaudio_stream_state_t inputState, ma_aaudio_stream_state_t* pNextState, int64_t timeoutInNanoseconds);
+typedef ma_aaudio_format_t (* MA_PFN_AAudioStream_getFormat) (ma_AAudioStream* pStream);
+typedef int32_t (* MA_PFN_AAudioStream_getChannelCount) (ma_AAudioStream* pStream);
+typedef int32_t (* MA_PFN_AAudioStream_getSampleRate) (ma_AAudioStream* pStream);
+typedef int32_t (* MA_PFN_AAudioStream_getBufferCapacityInFrames) (ma_AAudioStream* pStream);
+typedef int32_t (* MA_PFN_AAudioStream_getFramesPerDataCallback) (ma_AAudioStream* pStream);
+typedef int32_t (* MA_PFN_AAudioStream_getFramesPerBurst) (ma_AAudioStream* pStream);
+typedef ma_aaudio_result_t (* MA_PFN_AAudioStream_requestStart) (ma_AAudioStream* pStream);
+typedef ma_aaudio_result_t (* MA_PFN_AAudioStream_requestStop) (ma_AAudioStream* pStream);
- result = ma_context_get_IAudioClient__wasapi(pContext, deviceType, pDeviceID, &pData->pAudioClient, &pDeviceInterface);
- if (result != MA_SUCCESS) {
- goto done;
+static ma_result ma_result_from_aaudio(ma_aaudio_result_t resultAA)
+{
+ switch (resultAA)
+ {
+ case MA_AAUDIO_OK: return MA_SUCCESS;
+ default: break;
}
- MA_ZERO_OBJECT(&wf);
-
- /* Try enabling hardware offloading. */
- if (!pData->noHardwareOffloading) {
- hr = ma_IAudioClient_QueryInterface(pData->pAudioClient, &MA_IID_IAudioClient2, (void**)&pAudioClient2);
- if (SUCCEEDED(hr)) {
- BOOL isHardwareOffloadingSupported = 0;
- hr = ma_IAudioClient2_IsOffloadCapable(pAudioClient2, MA_AudioCategory_Other, &isHardwareOffloadingSupported);
- if (SUCCEEDED(hr) && isHardwareOffloadingSupported) {
- ma_AudioClientProperties clientProperties;
- MA_ZERO_OBJECT(&clientProperties);
- clientProperties.cbSize = sizeof(clientProperties);
- clientProperties.bIsOffload = 1;
- clientProperties.eCategory = MA_AudioCategory_Other;
- ma_IAudioClient2_SetClientProperties(pAudioClient2, &clientProperties);
- }
+ return MA_ERROR;
+}
- pAudioClient2->lpVtbl->Release(pAudioClient2);
- }
+static ma_aaudio_usage_t ma_to_usage__aaudio(ma_aaudio_usage usage)
+{
+ switch (usage) {
+ case ma_aaudio_usage_announcement: return MA_AAUDIO_USAGE_MEDIA;
+ case ma_aaudio_usage_emergency: return MA_AAUDIO_USAGE_VOICE_COMMUNICATION;
+ case ma_aaudio_usage_safety: return MA_AAUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING;
+ case ma_aaudio_usage_vehicle_status: return MA_AAUDIO_USAGE_ALARM;
+ case ma_aaudio_usage_alarm: return MA_AAUDIO_USAGE_NOTIFICATION;
+ case ma_aaudio_usage_assistance_accessibility: return MA_AAUDIO_USAGE_NOTIFICATION_RINGTONE;
+ case ma_aaudio_usage_assistance_navigation_guidance: return MA_AAUDIO_USAGE_NOTIFICATION_EVENT;
+ case ma_aaudio_usage_assistance_sonification: return MA_AAUDIO_USAGE_ASSISTANCE_ACCESSIBILITY;
+ case ma_aaudio_usage_assitant: return MA_AAUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE;
+ case ma_aaudio_usage_game: return MA_AAUDIO_USAGE_ASSISTANCE_SONIFICATION;
+ case ma_aaudio_usage_media: return MA_AAUDIO_USAGE_GAME;
+ case ma_aaudio_usage_notification: return MA_AAUDIO_USAGE_ASSISTANT;
+ case ma_aaudio_usage_notification_event: return MA_AAUDIO_SYSTEM_USAGE_EMERGENCY;
+ case ma_aaudio_usage_notification_ringtone: return MA_AAUDIO_SYSTEM_USAGE_SAFETY;
+ case ma_aaudio_usage_voice_communication: return MA_AAUDIO_SYSTEM_USAGE_VEHICLE_STATUS;
+ case ma_aaudio_usage_voice_communication_signalling: return MA_AAUDIO_SYSTEM_USAGE_ANNOUNCEMENT;
+ default: break;
}
- /* Here is where we try to determine the best format to use with the device. If the client if wanting exclusive mode, first try finding the best format for that. If this fails, fall back to shared mode. */
- result = MA_FORMAT_NOT_SUPPORTED;
- if (pData->shareMode == ma_share_mode_exclusive) {
- #ifdef MA_WIN32_DESKTOP
- /* In exclusive mode on desktop we always use the backend's native format. */
- ma_IPropertyStore* pStore = NULL;
- hr = ma_IMMDevice_OpenPropertyStore(pDeviceInterface, STGM_READ, &pStore);
- if (SUCCEEDED(hr)) {
- PROPVARIANT prop;
- ma_PropVariantInit(&prop);
- hr = ma_IPropertyStore_GetValue(pStore, &MA_PKEY_AudioEngine_DeviceFormat, &prop);
- if (SUCCEEDED(hr)) {
- WAVEFORMATEX* pActualFormat = (WAVEFORMATEX*)prop.blob.pBlobData;
- hr = ma_IAudioClient_IsFormatSupported((ma_IAudioClient*)pData->pAudioClient, MA_AUDCLNT_SHAREMODE_EXCLUSIVE, pActualFormat, NULL);
- if (SUCCEEDED(hr)) {
- MA_COPY_MEMORY(&wf, pActualFormat, sizeof(WAVEFORMATEXTENSIBLE));
- }
+ return MA_AAUDIO_USAGE_MEDIA;
+}
- ma_PropVariantClear(pContext, &prop);
- }
+static ma_aaudio_content_type_t ma_to_content_type__aaudio(ma_aaudio_content_type contentType)
+{
+ switch (contentType) {
+ case ma_aaudio_content_type_movie: return MA_AAUDIO_CONTENT_TYPE_MOVIE;
+ case ma_aaudio_content_type_music: return MA_AAUDIO_CONTENT_TYPE_MUSIC;
+ case ma_aaudio_content_type_sonification: return MA_AAUDIO_CONTENT_TYPE_SONIFICATION;
+ case ma_aaudio_content_type_speech: return MA_AAUDIO_CONTENT_TYPE_SPEECH;
+ default: break;
+ }
- ma_IPropertyStore_Release(pStore);
- }
- #else
- /*
- I do not know how to query the device's native format on UWP so for now I'm just disabling support for
- exclusive mode. The alternative is to enumerate over different formats and check IsFormatSupported()
- until you find one that works.
+ return MA_AAUDIO_CONTENT_TYPE_SPEECH;
+}
- TODO: Add support for exclusive mode to UWP.
- */
- hr = S_FALSE;
- #endif
+static ma_aaudio_input_preset_t ma_to_input_preset__aaudio(ma_aaudio_input_preset inputPreset)
+{
+ switch (inputPreset) {
+ case ma_aaudio_input_preset_generic: return MA_AAUDIO_INPUT_PRESET_GENERIC;
+ case ma_aaudio_input_preset_camcorder: return MA_AAUDIO_INPUT_PRESET_CAMCORDER;
+ case ma_aaudio_input_preset_unprocessed: return MA_AAUDIO_INPUT_PRESET_UNPROCESSED;
+ case ma_aaudio_input_preset_voice_recognition: return MA_AAUDIO_INPUT_PRESET_VOICE_RECOGNITION;
+ case ma_aaudio_input_preset_voice_communication: return MA_AAUDIO_INPUT_PRESET_VOICE_COMMUNICATION;
+ case ma_aaudio_input_preset_voice_performance: return MA_AAUDIO_INPUT_PRESET_VOICE_PERFORMANCE;
+ default: break;
+ }
- if (hr == S_OK) {
- shareMode = MA_AUDCLNT_SHAREMODE_EXCLUSIVE;
- result = MA_SUCCESS;
- } else {
- result = MA_SHARE_MODE_NOT_SUPPORTED;
- }
- } else {
- /* In shared mode we are always using the format reported by the operating system. */
- WAVEFORMATEXTENSIBLE* pNativeFormat = NULL;
- hr = ma_IAudioClient_GetMixFormat((ma_IAudioClient*)pData->pAudioClient, (WAVEFORMATEX**)&pNativeFormat);
- if (hr != S_OK) {
- result = MA_FORMAT_NOT_SUPPORTED;
- } else {
- MA_COPY_MEMORY(&wf, pNativeFormat, sizeof(wf));
- result = MA_SUCCESS;
- }
+ return MA_AAUDIO_INPUT_PRESET_GENERIC;
+}
- ma_CoTaskMemFree(pContext, pNativeFormat);
+static void ma_stream_error_callback__aaudio(ma_AAudioStream* pStream, void* pUserData, ma_aaudio_result_t error)
+{
+ ma_device* pDevice = (ma_device*)pUserData;
+ MA_ASSERT(pDevice != NULL);
- shareMode = MA_AUDCLNT_SHAREMODE_SHARED;
- }
+ (void)error;
- /* Return an error if we still haven't found a format. */
- if (result != MA_SUCCESS) {
- errorMsg = "[WASAPI] Failed to find best device mix format.";
- goto done;
- }
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[AAudio] ERROR CALLBACK: error=%d, AAudioStream_getState()=%d\n", error, ((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream));
/*
- Override the native sample rate with the one requested by the caller, but only if we're not using the default sample rate. We'll use
- WASAPI to perform the sample rate conversion.
+ From the documentation for AAudio, when a device is disconnected all we can do is stop it. However, we cannot stop it from the callback - we need
+ to do it from another thread. Therefore we are going to use an event thread for the AAudio backend to do this cleanly and safely.
*/
- nativeSampleRate = wf.Format.nSamplesPerSec;
- if (streamFlags & MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM) {
- wf.Format.nSamplesPerSec = (pData->sampleRateIn != 0) ? pData->sampleRateIn : MA_DEFAULT_SAMPLE_RATE;
- wf.Format.nAvgBytesPerSec = wf.Format.nSamplesPerSec * wf.Format.nBlockAlign;
- }
+ if (((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream) == MA_AAUDIO_STREAM_STATE_DISCONNECTED) {
+ /* We need to post a job to the job thread for processing. This will reroute the device by reinitializing the stream. */
+ ma_result result;
+ ma_job job = ma_job_init(MA_JOB_TYPE_DEVICE_AAUDIO_REROUTE);
+ job.data.device.aaudio.reroute.pDevice = pDevice;
- pData->formatOut = ma_format_from_WAVEFORMATEX((WAVEFORMATEX*)&wf);
- if (pData->formatOut == ma_format_unknown) {
- /*
- The format isn't supported. This is almost certainly because the exclusive mode format isn't supported by miniaudio. We need to return MA_SHARE_MODE_NOT_SUPPORTED
- in this case so that the caller can detect it and fall back to shared mode if desired. We should never get here if shared mode was requested, but just for
- completeness we'll check for it and return MA_FORMAT_NOT_SUPPORTED.
- */
- if (shareMode == MA_AUDCLNT_SHAREMODE_EXCLUSIVE) {
- result = MA_SHARE_MODE_NOT_SUPPORTED;
+ if (pStream == pDevice->aaudio.pStreamCapture) {
+ job.data.device.aaudio.reroute.deviceType = ma_device_type_capture;
} else {
- result = MA_FORMAT_NOT_SUPPORTED;
+ job.data.device.aaudio.reroute.deviceType = ma_device_type_playback;
}
- errorMsg = "[WASAPI] Native format not supported.";
- goto done;
- }
-
- pData->channelsOut = wf.Format.nChannels;
- pData->sampleRateOut = wf.Format.nSamplesPerSec;
-
- /* Get the internal channel map based on the channel mask. */
- ma_channel_mask_to_channel_map__win32(wf.dwChannelMask, pData->channelsOut, pData->channelMapOut);
-
- /* Period size. */
- pData->periodsOut = (pData->periodsIn != 0) ? pData->periodsIn : MA_DEFAULT_PERIODS;
- pData->periodSizeInFramesOut = pData->periodSizeInFramesIn;
- if (pData->periodSizeInFramesOut == 0) {
- if (pData->periodSizeInMillisecondsIn == 0) {
- if (pData->performanceProfile == ma_performance_profile_low_latency) {
- pData->periodSizeInFramesOut = ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY, wf.Format.nSamplesPerSec);
- } else {
- pData->periodSizeInFramesOut = ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE, wf.Format.nSamplesPerSec);
- }
- } else {
- pData->periodSizeInFramesOut = ma_calculate_buffer_size_in_frames_from_milliseconds(pData->periodSizeInMillisecondsIn, wf.Format.nSamplesPerSec);
+ result = ma_device_job_thread_post(&pDevice->pContext->aaudio.jobThread, &job);
+ if (result != MA_SUCCESS) {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[AAudio] Device Disconnected. Failed to post job for rerouting.\n");
+ return;
}
}
+}
- periodDurationInMicroseconds = ((ma_uint64)pData->periodSizeInFramesOut * 1000 * 1000) / wf.Format.nSamplesPerSec;
-
+static ma_aaudio_data_callback_result_t ma_stream_data_callback_capture__aaudio(ma_AAudioStream* pStream, void* pUserData, void* pAudioData, int32_t frameCount)
+{
+ ma_device* pDevice = (ma_device*)pUserData;
+ MA_ASSERT(pDevice != NULL);
- /* Slightly different initialization for shared and exclusive modes. We try exclusive mode first, and if it fails, fall back to shared mode. */
- if (shareMode == MA_AUDCLNT_SHAREMODE_EXCLUSIVE) {
- MA_REFERENCE_TIME bufferDuration = periodDurationInMicroseconds * 10;
+ ma_device_handle_backend_data_callback(pDevice, NULL, pAudioData, frameCount);
- /*
- If the periodicy is too small, Initialize() will fail with AUDCLNT_E_INVALID_DEVICE_PERIOD. In this case we should just keep increasing
- it and trying it again.
- */
- hr = E_FAIL;
- for (;;) {
- hr = ma_IAudioClient_Initialize((ma_IAudioClient*)pData->pAudioClient, shareMode, streamFlags, bufferDuration, bufferDuration, (WAVEFORMATEX*)&wf, NULL);
- if (hr == MA_AUDCLNT_E_INVALID_DEVICE_PERIOD) {
- if (bufferDuration > 500*10000) {
- break;
- } else {
- if (bufferDuration == 0) { /* <-- Just a sanity check to prevent an infinit loop. Should never happen, but it makes me feel better. */
- break;
- }
+ (void)pStream;
+ return MA_AAUDIO_CALLBACK_RESULT_CONTINUE;
+}
- bufferDuration = bufferDuration * 2;
- continue;
- }
- } else {
- break;
- }
- }
+static ma_aaudio_data_callback_result_t ma_stream_data_callback_playback__aaudio(ma_AAudioStream* pStream, void* pUserData, void* pAudioData, int32_t frameCount)
+{
+ ma_device* pDevice = (ma_device*)pUserData;
+ MA_ASSERT(pDevice != NULL);
- if (hr == MA_AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED) {
- ma_uint32 bufferSizeInFrames;
- hr = ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pData->pAudioClient, &bufferSizeInFrames);
- if (SUCCEEDED(hr)) {
- bufferDuration = (MA_REFERENCE_TIME)((10000.0 * 1000 / wf.Format.nSamplesPerSec * bufferSizeInFrames) + 0.5);
+ ma_device_handle_backend_data_callback(pDevice, pAudioData, NULL, frameCount);
- /* Unfortunately we need to release and re-acquire the audio client according to MSDN. Seems silly - why not just call IAudioClient_Initialize() again?! */
- ma_IAudioClient_Release((ma_IAudioClient*)pData->pAudioClient);
+ (void)pStream;
+ return MA_AAUDIO_CALLBACK_RESULT_CONTINUE;
+}
- #ifdef MA_WIN32_DESKTOP
- hr = ma_IMMDevice_Activate(pDeviceInterface, &MA_IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&pData->pAudioClient);
- #else
- hr = ma_IUnknown_QueryInterface(pDeviceInterface, &MA_IID_IAudioClient, (void**)&pData->pAudioClient);
- #endif
+static ma_result ma_create_and_configure_AAudioStreamBuilder__aaudio(ma_context* pContext, const ma_device_id* pDeviceID, ma_device_type deviceType, ma_share_mode shareMode, const ma_device_descriptor* pDescriptor, const ma_device_config* pConfig, ma_device* pDevice, ma_AAudioStreamBuilder** ppBuilder)
+{
+ ma_AAudioStreamBuilder* pBuilder;
+ ma_aaudio_result_t resultAA;
+ ma_uint32 bufferCapacityInFrames;
- if (SUCCEEDED(hr)) {
- hr = ma_IAudioClient_Initialize((ma_IAudioClient*)pData->pAudioClient, shareMode, streamFlags, bufferDuration, bufferDuration, (WAVEFORMATEX*)&wf, NULL);
- }
- }
- }
+ /* Safety. */
+ *ppBuilder = NULL;
- if (FAILED(hr)) {
- /* Failed to initialize in exclusive mode. Don't fall back to shared mode - instead tell the client about it. They can reinitialize in shared mode if they want. */
- if (hr == E_ACCESSDENIED) {
- errorMsg = "[WASAPI] Failed to initialize device in exclusive mode. Access denied.", result = MA_ACCESS_DENIED;
- } else if (hr == MA_AUDCLNT_E_DEVICE_IN_USE) {
- errorMsg = "[WASAPI] Failed to initialize device in exclusive mode. Device in use.", result = MA_BUSY;
- } else {
- errorMsg = "[WASAPI] Failed to initialize device in exclusive mode."; result = ma_result_from_HRESULT(hr);
- }
- goto done;
- }
+ resultAA = ((MA_PFN_AAudio_createStreamBuilder)pContext->aaudio.AAudio_createStreamBuilder)(&pBuilder);
+ if (resultAA != MA_AAUDIO_OK) {
+ return ma_result_from_aaudio(resultAA);
}
- if (shareMode == MA_AUDCLNT_SHAREMODE_SHARED) {
- /*
- Low latency shared mode via IAudioClient3.
-
- NOTE
- ====
- Contrary to the documentation on MSDN (https://docs.microsoft.com/en-us/windows/win32/api/audioclient/nf-audioclient-iaudioclient3-initializesharedaudiostream), the
- use of AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM and AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY with IAudioClient3_InitializeSharedAudioStream() absolutely does not work. Using
- any of these flags will result in HRESULT code 0x88890021. The other problem is that calling IAudioClient3_GetSharedModeEnginePeriod() with a sample rate different to
- that returned by IAudioClient_GetMixFormat() also results in an error. I'm therefore disabling low-latency shared mode with AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM.
- */
- #ifndef MA_WASAPI_NO_LOW_LATENCY_SHARED_MODE
- {
- if ((streamFlags & MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM) == 0 || nativeSampleRate == wf.Format.nSamplesPerSec) {
- ma_IAudioClient3* pAudioClient3 = NULL;
- hr = ma_IAudioClient_QueryInterface(pData->pAudioClient, &MA_IID_IAudioClient3, (void**)&pAudioClient3);
- if (SUCCEEDED(hr)) {
- ma_uint32 defaultPeriodInFrames;
- ma_uint32 fundamentalPeriodInFrames;
- ma_uint32 minPeriodInFrames;
- ma_uint32 maxPeriodInFrames;
- hr = ma_IAudioClient3_GetSharedModeEnginePeriod(pAudioClient3, (WAVEFORMATEX*)&wf, &defaultPeriodInFrames, &fundamentalPeriodInFrames, &minPeriodInFrames, &maxPeriodInFrames);
- if (SUCCEEDED(hr)) {
- ma_uint32 desiredPeriodInFrames = pData->periodSizeInFramesOut;
- ma_uint32 actualPeriodInFrames = desiredPeriodInFrames;
-
- /* Make sure the period size is a multiple of fundamentalPeriodInFrames. */
- actualPeriodInFrames = actualPeriodInFrames / fundamentalPeriodInFrames;
- actualPeriodInFrames = actualPeriodInFrames * fundamentalPeriodInFrames;
+ if (pDeviceID != NULL) {
+ ((MA_PFN_AAudioStreamBuilder_setDeviceId)pContext->aaudio.AAudioStreamBuilder_setDeviceId)(pBuilder, pDeviceID->aaudio);
+ }
- /* The period needs to be clamped between minPeriodInFrames and maxPeriodInFrames. */
- actualPeriodInFrames = ma_clamp(actualPeriodInFrames, minPeriodInFrames, maxPeriodInFrames);
+ ((MA_PFN_AAudioStreamBuilder_setDirection)pContext->aaudio.AAudioStreamBuilder_setDirection)(pBuilder, (deviceType == ma_device_type_playback) ? MA_AAUDIO_DIRECTION_OUTPUT : MA_AAUDIO_DIRECTION_INPUT);
+ ((MA_PFN_AAudioStreamBuilder_setSharingMode)pContext->aaudio.AAudioStreamBuilder_setSharingMode)(pBuilder, (shareMode == ma_share_mode_shared) ? MA_AAUDIO_SHARING_MODE_SHARED : MA_AAUDIO_SHARING_MODE_EXCLUSIVE);
- #if defined(MA_DEBUG_OUTPUT)
- {
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[WASAPI] Trying IAudioClient3_InitializeSharedAudioStream(actualPeriodInFrames=%d)\n", actualPeriodInFrames);
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, " defaultPeriodInFrames=%d\n", defaultPeriodInFrames);
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, " fundamentalPeriodInFrames=%d\n", fundamentalPeriodInFrames);
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, " minPeriodInFrames=%d\n", minPeriodInFrames);
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, " maxPeriodInFrames=%d\n", maxPeriodInFrames);
- }
- #endif
- /* If the client requested a largish buffer than we don't actually want to use low latency shared mode because it forces small buffers. */
- if (actualPeriodInFrames >= desiredPeriodInFrames) {
- /*
- MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM | MA_AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY must not be in the stream flags. If either of these are specified,
- IAudioClient3_InitializeSharedAudioStream() will fail.
- */
- hr = ma_IAudioClient3_InitializeSharedAudioStream(pAudioClient3, streamFlags & ~(MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM | MA_AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY), actualPeriodInFrames, (WAVEFORMATEX*)&wf, NULL);
- if (SUCCEEDED(hr)) {
- wasInitializedUsingIAudioClient3 = MA_TRUE;
- pData->periodSizeInFramesOut = actualPeriodInFrames;
- #if defined(MA_DEBUG_OUTPUT)
- {
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[WASAPI] Using IAudioClient3\n");
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, " periodSizeInFramesOut=%d\n", pData->periodSizeInFramesOut);
- }
- #endif
- } else {
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[WASAPI] IAudioClient3_InitializeSharedAudioStream failed. Falling back to IAudioClient.\n");
- }
- } else {
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[WASAPI] Not using IAudioClient3 because the desired period size is larger than the maximum supported by IAudioClient3.\n");
- }
- } else {
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[WASAPI] IAudioClient3_GetSharedModeEnginePeriod failed. Falling back to IAudioClient.\n");
- }
+ /* If we have a device descriptor make sure we configure the stream builder to take our requested parameters. */
+ if (pDescriptor != NULL) {
+ MA_ASSERT(pConfig != NULL); /* We must have a device config if we also have a descriptor. The config is required for AAudio specific configuration options. */
- ma_IAudioClient3_Release(pAudioClient3);
- pAudioClient3 = NULL;
- }
- }
- }
- #else
- {
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[WASAPI] Not using IAudioClient3 because MA_WASAPI_NO_LOW_LATENCY_SHARED_MODE is enabled.\n");
+ if (pDescriptor->sampleRate != 0) {
+ ((MA_PFN_AAudioStreamBuilder_setSampleRate)pContext->aaudio.AAudioStreamBuilder_setSampleRate)(pBuilder, pDescriptor->sampleRate);
}
- #endif
- /* If we don't have an IAudioClient3 then we need to use the normal initialization routine. */
- if (!wasInitializedUsingIAudioClient3) {
- MA_REFERENCE_TIME bufferDuration = periodDurationInMicroseconds * pData->periodsOut * 10; /* <-- Multiply by 10 for microseconds to 100-nanoseconds. */
- hr = ma_IAudioClient_Initialize((ma_IAudioClient*)pData->pAudioClient, shareMode, streamFlags, bufferDuration, 0, (WAVEFORMATEX*)&wf, NULL);
- if (FAILED(hr)) {
- if (hr == E_ACCESSDENIED) {
- errorMsg = "[WASAPI] Failed to initialize device. Access denied.", result = MA_ACCESS_DENIED;
- } else if (hr == MA_AUDCLNT_E_DEVICE_IN_USE) {
- errorMsg = "[WASAPI] Failed to initialize device. Device in use.", result = MA_BUSY;
- } else {
- errorMsg = "[WASAPI] Failed to initialize device.", result = ma_result_from_HRESULT(hr);
- }
-
- goto done;
+ if (deviceType == ma_device_type_capture) {
+ if (pDescriptor->channels != 0) {
+ ((MA_PFN_AAudioStreamBuilder_setChannelCount)pContext->aaudio.AAudioStreamBuilder_setChannelCount)(pBuilder, pDescriptor->channels);
+ }
+ if (pDescriptor->format != ma_format_unknown) {
+ ((MA_PFN_AAudioStreamBuilder_setFormat)pContext->aaudio.AAudioStreamBuilder_setFormat)(pBuilder, (pDescriptor->format == ma_format_s16) ? MA_AAUDIO_FORMAT_PCM_I16 : MA_AAUDIO_FORMAT_PCM_FLOAT);
+ }
+ } else {
+ if (pDescriptor->channels != 0) {
+ ((MA_PFN_AAudioStreamBuilder_setChannelCount)pContext->aaudio.AAudioStreamBuilder_setChannelCount)(pBuilder, pDescriptor->channels);
+ }
+ if (pDescriptor->format != ma_format_unknown) {
+ ((MA_PFN_AAudioStreamBuilder_setFormat)pContext->aaudio.AAudioStreamBuilder_setFormat)(pBuilder, (pDescriptor->format == ma_format_s16) ? MA_AAUDIO_FORMAT_PCM_I16 : MA_AAUDIO_FORMAT_PCM_FLOAT);
}
}
- }
-
- if (!wasInitializedUsingIAudioClient3) {
- ma_uint32 bufferSizeInFrames;
- hr = ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pData->pAudioClient, &bufferSizeInFrames);
- if (FAILED(hr)) {
- errorMsg = "[WASAPI] Failed to get audio client's actual buffer size.", result = ma_result_from_HRESULT(hr);
- goto done;
- }
-
- pData->periodSizeInFramesOut = bufferSizeInFrames / pData->periodsOut;
- }
-
- pData->usingAudioClient3 = wasInitializedUsingIAudioClient3;
-
- if (deviceType == ma_device_type_playback) {
- result = ma_device_create_IAudioClient_service__wasapi(pContext, deviceType, (ma_IAudioClient*)pData->pAudioClient, (void**)&pData->pRenderClient);
- } else {
- result = ma_device_create_IAudioClient_service__wasapi(pContext, deviceType, (ma_IAudioClient*)pData->pAudioClient, (void**)&pData->pCaptureClient);
- }
+ /*
+ AAudio is annoying when it comes to it's buffer calculation stuff because it doesn't let you
+ retrieve the actual sample rate until after you've opened the stream. But you need to configure
+ the buffer capacity before you open the stream... :/
- /*if (FAILED(hr)) {*/
- if (result != MA_SUCCESS) {
- errorMsg = "[WASAPI] Failed to get audio client service.";
- goto done;
- }
+ To solve, we're just going to assume MA_DEFAULT_SAMPLE_RATE (48000) and move on.
+ */
+ bufferCapacityInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, pDescriptor->sampleRate, pConfig->performanceProfile) * pDescriptor->periodCount;
+ ((MA_PFN_AAudioStreamBuilder_setBufferCapacityInFrames)pContext->aaudio.AAudioStreamBuilder_setBufferCapacityInFrames)(pBuilder, bufferCapacityInFrames);
+ ((MA_PFN_AAudioStreamBuilder_setFramesPerDataCallback)pContext->aaudio.AAudioStreamBuilder_setFramesPerDataCallback)(pBuilder, bufferCapacityInFrames / pDescriptor->periodCount);
- /* Grab the name of the device. */
- #ifdef MA_WIN32_DESKTOP
- {
- ma_IPropertyStore *pProperties;
- hr = ma_IMMDevice_OpenPropertyStore(pDeviceInterface, STGM_READ, &pProperties);
- if (SUCCEEDED(hr)) {
- PROPVARIANT varName;
- ma_PropVariantInit(&varName);
- hr = ma_IPropertyStore_GetValue(pProperties, &MA_PKEY_Device_FriendlyName, &varName);
- if (SUCCEEDED(hr)) {
- WideCharToMultiByte(CP_UTF8, 0, varName.pwszVal, -1, pData->deviceName, sizeof(pData->deviceName), 0, FALSE);
- ma_PropVariantClear(pContext, &varName);
+ if (deviceType == ma_device_type_capture) {
+ if (pConfig->aaudio.inputPreset != ma_aaudio_input_preset_default && pContext->aaudio.AAudioStreamBuilder_setInputPreset != NULL) {
+ ((MA_PFN_AAudioStreamBuilder_setInputPreset)pContext->aaudio.AAudioStreamBuilder_setInputPreset)(pBuilder, ma_to_input_preset__aaudio(pConfig->aaudio.inputPreset));
}
- ma_IPropertyStore_Release(pProperties);
- }
- }
- #endif
-
- /*
- For the WASAPI backend we need to know the actual IDs of the device in order to do automatic
- stream routing so that IDs can be compared and we can determine which device has been detached
- and whether or not it matches with our ma_device.
- */
- #ifdef MA_WIN32_DESKTOP
- {
- /* Desktop */
- ma_context_get_device_id_from_MMDevice__wasapi(pContext, pDeviceInterface, &pData->id);
- }
- #else
- {
- /* UWP */
- /* TODO: Implement me. Need to figure out how to get the ID of the default device. */
- }
- #endif
-
-done:
- /* Clean up. */
-#ifdef MA_WIN32_DESKTOP
- if (pDeviceInterface != NULL) {
- ma_IMMDevice_Release(pDeviceInterface);
- }
-#else
- if (pDeviceInterface != NULL) {
- ma_IUnknown_Release(pDeviceInterface);
- }
-#endif
+ ((MA_PFN_AAudioStreamBuilder_setDataCallback)pContext->aaudio.AAudioStreamBuilder_setDataCallback)(pBuilder, ma_stream_data_callback_capture__aaudio, (void*)pDevice);
+ } else {
+ if (pConfig->aaudio.usage != ma_aaudio_usage_default && pContext->aaudio.AAudioStreamBuilder_setUsage != NULL) {
+ ((MA_PFN_AAudioStreamBuilder_setUsage)pContext->aaudio.AAudioStreamBuilder_setUsage)(pBuilder, ma_to_usage__aaudio(pConfig->aaudio.usage));
+ }
- if (result != MA_SUCCESS) {
- if (pData->pRenderClient) {
- ma_IAudioRenderClient_Release((ma_IAudioRenderClient*)pData->pRenderClient);
- pData->pRenderClient = NULL;
- }
- if (pData->pCaptureClient) {
- ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pData->pCaptureClient);
- pData->pCaptureClient = NULL;
- }
- if (pData->pAudioClient) {
- ma_IAudioClient_Release((ma_IAudioClient*)pData->pAudioClient);
- pData->pAudioClient = NULL;
- }
+ if (pConfig->aaudio.contentType != ma_aaudio_content_type_default && pContext->aaudio.AAudioStreamBuilder_setContentType != NULL) {
+ ((MA_PFN_AAudioStreamBuilder_setContentType)pContext->aaudio.AAudioStreamBuilder_setContentType)(pBuilder, ma_to_content_type__aaudio(pConfig->aaudio.contentType));
+ }
- if (errorMsg != NULL && errorMsg[0] != '\0') {
- ma_post_log_message(pContext, NULL, MA_LOG_LEVEL_ERROR, errorMsg);
+ ((MA_PFN_AAudioStreamBuilder_setDataCallback)pContext->aaudio.AAudioStreamBuilder_setDataCallback)(pBuilder, ma_stream_data_callback_playback__aaudio, (void*)pDevice);
}
- return result;
- } else {
- return MA_SUCCESS;
+ /* Not sure how this affects things, but since there's a mapping between miniaudio's performance profiles and AAudio's performance modes, let go ahead and set it. */
+ ((MA_PFN_AAudioStreamBuilder_setPerformanceMode)pContext->aaudio.AAudioStreamBuilder_setPerformanceMode)(pBuilder, (pConfig->performanceProfile == ma_performance_profile_low_latency) ? MA_AAUDIO_PERFORMANCE_MODE_LOW_LATENCY : MA_AAUDIO_PERFORMANCE_MODE_NONE);
+
+ /* We need to set an error callback to detect device changes. */
+ if (pDevice != NULL) { /* <-- pDevice should never be null if pDescriptor is not null, which is always the case if we hit this branch. Check anyway for safety. */
+ ((MA_PFN_AAudioStreamBuilder_setErrorCallback)pContext->aaudio.AAudioStreamBuilder_setErrorCallback)(pBuilder, ma_stream_error_callback__aaudio, (void*)pDevice);
+ }
}
+
+ *ppBuilder = pBuilder;
+
+ return MA_SUCCESS;
}
-static ma_result ma_device_reinit__wasapi(ma_device* pDevice, ma_device_type deviceType)
+static ma_result ma_open_stream_and_close_builder__aaudio(ma_context* pContext, ma_AAudioStreamBuilder* pBuilder, ma_AAudioStream** ppStream)
{
- ma_device_init_internal_data__wasapi data;
ma_result result;
- MA_ASSERT(pDevice != NULL);
-
- /* We only re-initialize the playback or capture device. Never a full-duplex device. */
- if (deviceType == ma_device_type_duplex) {
- return MA_INVALID_ARGS;
- }
+ result = ma_result_from_aaudio(((MA_PFN_AAudioStreamBuilder_openStream)pContext->aaudio.AAudioStreamBuilder_openStream)(pBuilder, ppStream));
+ ((MA_PFN_AAudioStreamBuilder_delete)pContext->aaudio.AAudioStreamBuilder_delete)(pBuilder);
+ return result;
+}
- /*
- Before reinitializing the device we need to free the previous audio clients.
+static ma_result ma_open_stream_basic__aaudio(ma_context* pContext, const ma_device_id* pDeviceID, ma_device_type deviceType, ma_share_mode shareMode, ma_AAudioStream** ppStream)
+{
+ ma_result result;
+ ma_AAudioStreamBuilder* pBuilder;
- There's a known memory leak here. We will be calling this from the routing change callback that
- is fired by WASAPI. If we attempt to release the IAudioClient we will deadlock. In my opinion
- this is a bug. I'm not sure what I need to do to handle this cleanly, but I think we'll probably
- need some system where we post an event, but delay the execution of it until the callback has
- returned. I'm not sure how to do this reliably, however. I have set up some infrastructure for
- a command thread which might be useful for this.
- */
- if (deviceType == ma_device_type_capture || deviceType == ma_device_type_loopback) {
- if (pDevice->wasapi.pCaptureClient) {
- ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient);
- pDevice->wasapi.pCaptureClient = NULL;
- }
+ *ppStream = NULL;
- if (pDevice->wasapi.pAudioClientCapture) {
- /*ma_device_release_IAudioClient_service__wasapi(pDevice, ma_device_type_capture);*/
- pDevice->wasapi.pAudioClientCapture = NULL;
- }
+ result = ma_create_and_configure_AAudioStreamBuilder__aaudio(pContext, pDeviceID, deviceType, shareMode, NULL, NULL, NULL, &pBuilder);
+ if (result != MA_SUCCESS) {
+ return result;
}
- if (deviceType == ma_device_type_playback) {
- if (pDevice->wasapi.pRenderClient) {
- ma_IAudioRenderClient_Release((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient);
- pDevice->wasapi.pRenderClient = NULL;
- }
+ return ma_open_stream_and_close_builder__aaudio(pContext, pBuilder, ppStream);
+}
- if (pDevice->wasapi.pAudioClientPlayback) {
- /*ma_device_release_IAudioClient_service__wasapi(pDevice, ma_device_type_playback);*/
- pDevice->wasapi.pAudioClientPlayback = NULL;
- }
- }
+static ma_result ma_open_stream__aaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_type deviceType, const ma_device_descriptor* pDescriptor, ma_AAudioStream** ppStream)
+{
+ ma_result result;
+ ma_AAudioStreamBuilder* pBuilder;
+ MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(pDescriptor != NULL);
+ MA_ASSERT(deviceType != ma_device_type_duplex); /* This function should not be called for a full-duplex device type. */
- if (deviceType == ma_device_type_playback) {
- data.formatIn = pDevice->playback.format;
- data.channelsIn = pDevice->playback.channels;
- MA_COPY_MEMORY(data.channelMapIn, pDevice->playback.channelMap, sizeof(pDevice->playback.channelMap));
- data.shareMode = pDevice->playback.shareMode;
- } else {
- data.formatIn = pDevice->capture.format;
- data.channelsIn = pDevice->capture.channels;
- MA_COPY_MEMORY(data.channelMapIn, pDevice->capture.channelMap, sizeof(pDevice->capture.channelMap));
- data.shareMode = pDevice->capture.shareMode;
- }
+ *ppStream = NULL;
- data.sampleRateIn = pDevice->sampleRate;
- data.periodSizeInFramesIn = pDevice->wasapi.originalPeriodSizeInFrames;
- data.periodSizeInMillisecondsIn = pDevice->wasapi.originalPeriodSizeInMilliseconds;
- data.periodsIn = pDevice->wasapi.originalPeriods;
- data.performanceProfile = pDevice->wasapi.originalPerformanceProfile;
- data.noAutoConvertSRC = pDevice->wasapi.noAutoConvertSRC;
- data.noDefaultQualitySRC = pDevice->wasapi.noDefaultQualitySRC;
- data.noHardwareOffloading = pDevice->wasapi.noHardwareOffloading;
- result = ma_device_init_internal__wasapi(pDevice->pContext, deviceType, NULL, &data);
+ result = ma_create_and_configure_AAudioStreamBuilder__aaudio(pDevice->pContext, pDescriptor->pDeviceID, deviceType, pDescriptor->shareMode, pDescriptor, pConfig, pDevice, &pBuilder);
if (result != MA_SUCCESS) {
return result;
}
- /* At this point we have some new objects ready to go. We need to uninitialize the previous ones and then set the new ones. */
- if (deviceType == ma_device_type_capture || deviceType == ma_device_type_loopback) {
- pDevice->wasapi.pAudioClientCapture = data.pAudioClient;
- pDevice->wasapi.pCaptureClient = data.pCaptureClient;
-
- pDevice->capture.internalFormat = data.formatOut;
- pDevice->capture.internalChannels = data.channelsOut;
- pDevice->capture.internalSampleRate = data.sampleRateOut;
- MA_COPY_MEMORY(pDevice->capture.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut));
- pDevice->capture.internalPeriodSizeInFrames = data.periodSizeInFramesOut;
- pDevice->capture.internalPeriods = data.periodsOut;
- ma_strcpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), data.deviceName);
-
- ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, pDevice->wasapi.hEventCapture);
+ return ma_open_stream_and_close_builder__aaudio(pDevice->pContext, pBuilder, ppStream);
+}
- pDevice->wasapi.periodSizeInFramesCapture = data.periodSizeInFramesOut;
- ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, &pDevice->wasapi.actualPeriodSizeInFramesCapture);
+static ma_result ma_close_stream__aaudio(ma_context* pContext, ma_AAudioStream* pStream)
+{
+ return ma_result_from_aaudio(((MA_PFN_AAudioStream_close)pContext->aaudio.AAudioStream_close)(pStream));
+}
- /* We must always have a valid ID. */
- ma_wcscpy_s(pDevice->capture.id.wasapi, sizeof(pDevice->capture.id.wasapi), data.id.wasapi);
+static ma_bool32 ma_has_default_device__aaudio(ma_context* pContext, ma_device_type deviceType)
+{
+ /* The only way to know this is to try creating a stream. */
+ ma_AAudioStream* pStream;
+ ma_result result = ma_open_stream_basic__aaudio(pContext, NULL, deviceType, ma_share_mode_shared, &pStream);
+ if (result != MA_SUCCESS) {
+ return MA_FALSE;
}
- if (deviceType == ma_device_type_playback) {
- pDevice->wasapi.pAudioClientPlayback = data.pAudioClient;
- pDevice->wasapi.pRenderClient = data.pRenderClient;
-
- pDevice->playback.internalFormat = data.formatOut;
- pDevice->playback.internalChannels = data.channelsOut;
- pDevice->playback.internalSampleRate = data.sampleRateOut;
- MA_COPY_MEMORY(pDevice->playback.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut));
- pDevice->playback.internalPeriodSizeInFrames = data.periodSizeInFramesOut;
- pDevice->playback.internalPeriods = data.periodsOut;
- ma_strcpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), data.deviceName);
-
- ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, pDevice->wasapi.hEventPlayback);
+ ma_close_stream__aaudio(pContext, pStream);
+ return MA_TRUE;
+}
- pDevice->wasapi.periodSizeInFramesPlayback = data.periodSizeInFramesOut;
- ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &pDevice->wasapi.actualPeriodSizeInFramesPlayback);
+static ma_result ma_wait_for_simple_state_transition__aaudio(ma_context* pContext, ma_AAudioStream* pStream, ma_aaudio_stream_state_t oldState, ma_aaudio_stream_state_t newState)
+{
+ ma_aaudio_stream_state_t actualNewState;
+ ma_aaudio_result_t resultAA = ((MA_PFN_AAudioStream_waitForStateChange)pContext->aaudio.AAudioStream_waitForStateChange)(pStream, oldState, &actualNewState, 5000000000); /* 5 second timeout. */
+ if (resultAA != MA_AAUDIO_OK) {
+ return ma_result_from_aaudio(resultAA);
+ }
- /* We must always have a valid ID. */
- ma_wcscpy_s(pDevice->playback.id.wasapi, sizeof(pDevice->playback.id.wasapi), data.id.wasapi);
+ if (newState != actualNewState) {
+ return MA_ERROR; /* Failed to transition into the expected state. */
}
return MA_SUCCESS;
}
-static ma_result ma_device_init__wasapi(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+
+static ma_result ma_context_enumerate_devices__aaudio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
{
- ma_result result = MA_SUCCESS;
+ ma_bool32 cbResult = MA_TRUE;
-#ifdef MA_WIN32_DESKTOP
- HRESULT hr;
- ma_IMMDeviceEnumerator* pDeviceEnumerator;
-#endif
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(callback != NULL);
- MA_ASSERT(pDevice != NULL);
+ /* Unfortunately AAudio does not have an enumeration API. Therefore I'm only going to report default devices, but only if it can instantiate a stream. */
- MA_ZERO_OBJECT(&pDevice->wasapi);
- pDevice->wasapi.noAutoConvertSRC = pConfig->wasapi.noAutoConvertSRC;
- pDevice->wasapi.noDefaultQualitySRC = pConfig->wasapi.noDefaultQualitySRC;
- pDevice->wasapi.noHardwareOffloading = pConfig->wasapi.noHardwareOffloading;
+ /* Playback. */
+ if (cbResult) {
+ ma_device_info deviceInfo;
+ MA_ZERO_OBJECT(&deviceInfo);
+ deviceInfo.id.aaudio = MA_AAUDIO_UNSPECIFIED;
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
- /* Exclusive mode is not allowed with loopback. */
- if (pConfig->deviceType == ma_device_type_loopback && pConfig->playback.shareMode == ma_share_mode_exclusive) {
- return MA_INVALID_DEVICE_CONFIG;
+ if (ma_has_default_device__aaudio(pContext, ma_device_type_playback)) {
+ cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
+ }
}
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex || pConfig->deviceType == ma_device_type_loopback) {
- ma_device_init_internal_data__wasapi data;
- data.formatIn = pDescriptorCapture->format;
- data.channelsIn = pDescriptorCapture->channels;
- data.sampleRateIn = pDescriptorCapture->sampleRate;
- MA_COPY_MEMORY(data.channelMapIn, pDescriptorCapture->channelMap, sizeof(pDescriptorCapture->channelMap));
- data.periodSizeInFramesIn = pDescriptorCapture->periodSizeInFrames;
- data.periodSizeInMillisecondsIn = pDescriptorCapture->periodSizeInMilliseconds;
- data.periodsIn = pDescriptorCapture->periodCount;
- data.shareMode = pDescriptorCapture->shareMode;
- data.performanceProfile = pConfig->performanceProfile;
- data.noAutoConvertSRC = pConfig->wasapi.noAutoConvertSRC;
- data.noDefaultQualitySRC = pConfig->wasapi.noDefaultQualitySRC;
- data.noHardwareOffloading = pConfig->wasapi.noHardwareOffloading;
+ /* Capture. */
+ if (cbResult) {
+ ma_device_info deviceInfo;
+ MA_ZERO_OBJECT(&deviceInfo);
+ deviceInfo.id.aaudio = MA_AAUDIO_UNSPECIFIED;
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
- result = ma_device_init_internal__wasapi(pDevice->pContext, (pConfig->deviceType == ma_device_type_loopback) ? ma_device_type_loopback : ma_device_type_capture, pDescriptorCapture->pDeviceID, &data);
- if (result != MA_SUCCESS) {
- return result;
+ if (ma_has_default_device__aaudio(pContext, ma_device_type_capture)) {
+ cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
}
+ }
- pDevice->wasapi.pAudioClientCapture = data.pAudioClient;
- pDevice->wasapi.pCaptureClient = data.pCaptureClient;
- pDevice->wasapi.originalPeriodSizeInMilliseconds = pDescriptorCapture->periodSizeInMilliseconds;
- pDevice->wasapi.originalPeriodSizeInFrames = pDescriptorCapture->periodSizeInFrames;
- pDevice->wasapi.originalPeriods = pDescriptorCapture->periodCount;
- pDevice->wasapi.originalPerformanceProfile = pConfig->performanceProfile;
+ return MA_SUCCESS;
+}
- /*
- The event for capture needs to be manual reset for the same reason as playback. We keep the initial state set to unsignaled,
- however, because we want to block until we actually have something for the first call to ma_device_read().
- */
- pDevice->wasapi.hEventCapture = CreateEventW(NULL, FALSE, FALSE, NULL); /* Auto reset, unsignaled by default. */
- if (pDevice->wasapi.hEventCapture == NULL) {
- result = ma_result_from_GetLastError(GetLastError());
+static void ma_context_add_native_data_format_from_AAudioStream_ex__aaudio(ma_context* pContext, ma_AAudioStream* pStream, ma_format format, ma_uint32 flags, ma_device_info* pDeviceInfo)
+{
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pStream != NULL);
+ MA_ASSERT(pDeviceInfo != NULL);
- if (pDevice->wasapi.pCaptureClient != NULL) {
- ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient);
- pDevice->wasapi.pCaptureClient = NULL;
- }
- if (pDevice->wasapi.pAudioClientCapture != NULL) {
- ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture);
- pDevice->wasapi.pAudioClientCapture = NULL;
- }
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = format;
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = ((MA_PFN_AAudioStream_getChannelCount)pContext->aaudio.AAudioStream_getChannelCount)(pStream);
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = ((MA_PFN_AAudioStream_getSampleRate)pContext->aaudio.AAudioStream_getSampleRate)(pStream);
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = flags;
+ pDeviceInfo->nativeDataFormatCount += 1;
+}
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create event for capture.", result);
- }
- ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, pDevice->wasapi.hEventCapture);
+static void ma_context_add_native_data_format_from_AAudioStream__aaudio(ma_context* pContext, ma_AAudioStream* pStream, ma_uint32 flags, ma_device_info* pDeviceInfo)
+{
+ /* AAudio supports s16 and f32. */
+ ma_context_add_native_data_format_from_AAudioStream_ex__aaudio(pContext, pStream, ma_format_f32, flags, pDeviceInfo);
+ ma_context_add_native_data_format_from_AAudioStream_ex__aaudio(pContext, pStream, ma_format_s16, flags, pDeviceInfo);
+}
- pDevice->wasapi.periodSizeInFramesCapture = data.periodSizeInFramesOut;
- ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, &pDevice->wasapi.actualPeriodSizeInFramesCapture);
+static ma_result ma_context_get_device_info__aaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+{
+ ma_AAudioStream* pStream;
+ ma_result result;
- /* We must always have a valid ID. */
- ma_wcscpy_s(pDevice->capture.id.wasapi, sizeof(pDevice->capture.id.wasapi), data.id.wasapi);
+ MA_ASSERT(pContext != NULL);
- /* The descriptor needs to be updated with actual values. */
- pDescriptorCapture->format = data.formatOut;
- pDescriptorCapture->channels = data.channelsOut;
- pDescriptorCapture->sampleRate = data.sampleRateOut;
- MA_COPY_MEMORY(pDescriptorCapture->channelMap, data.channelMapOut, sizeof(data.channelMapOut));
- pDescriptorCapture->periodSizeInFrames = data.periodSizeInFramesOut;
- pDescriptorCapture->periodCount = data.periodsOut;
+ /* ID */
+ if (pDeviceID != NULL) {
+ pDeviceInfo->id.aaudio = pDeviceID->aaudio;
+ } else {
+ pDeviceInfo->id.aaudio = MA_AAUDIO_UNSPECIFIED;
}
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- ma_device_init_internal_data__wasapi data;
- data.formatIn = pDescriptorPlayback->format;
- data.channelsIn = pDescriptorPlayback->channels;
- data.sampleRateIn = pDescriptorPlayback->sampleRate;
- MA_COPY_MEMORY(data.channelMapIn, pDescriptorPlayback->channelMap, sizeof(pDescriptorPlayback->channelMap));
- data.periodSizeInFramesIn = pDescriptorPlayback->periodSizeInFrames;
- data.periodSizeInMillisecondsIn = pDescriptorPlayback->periodSizeInMilliseconds;
- data.periodsIn = pDescriptorPlayback->periodCount;
- data.shareMode = pDescriptorPlayback->shareMode;
- data.performanceProfile = pConfig->performanceProfile;
- data.noAutoConvertSRC = pConfig->wasapi.noAutoConvertSRC;
- data.noDefaultQualitySRC = pConfig->wasapi.noDefaultQualitySRC;
- data.noHardwareOffloading = pConfig->wasapi.noHardwareOffloading;
+ /* Name */
+ if (deviceType == ma_device_type_playback) {
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
+ } else {
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
+ }
- result = ma_device_init_internal__wasapi(pDevice->pContext, ma_device_type_playback, pDescriptorPlayback->pDeviceID, &data);
- if (result != MA_SUCCESS) {
- if (pConfig->deviceType == ma_device_type_duplex) {
- if (pDevice->wasapi.pCaptureClient != NULL) {
- ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient);
- pDevice->wasapi.pCaptureClient = NULL;
- }
- if (pDevice->wasapi.pAudioClientCapture != NULL) {
- ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture);
- pDevice->wasapi.pAudioClientCapture = NULL;
- }
- CloseHandle(pDevice->wasapi.hEventCapture);
- pDevice->wasapi.hEventCapture = NULL;
- }
- return result;
- }
+ pDeviceInfo->nativeDataFormatCount = 0;
- pDevice->wasapi.pAudioClientPlayback = data.pAudioClient;
- pDevice->wasapi.pRenderClient = data.pRenderClient;
- pDevice->wasapi.originalPeriodSizeInMilliseconds = pDescriptorPlayback->periodSizeInMilliseconds;
- pDevice->wasapi.originalPeriodSizeInFrames = pDescriptorPlayback->periodSizeInFrames;
- pDevice->wasapi.originalPeriods = pDescriptorPlayback->periodCount;
- pDevice->wasapi.originalPerformanceProfile = pConfig->performanceProfile;
+ /* We'll need to open the device to get accurate sample rate and channel count information. */
+ result = ma_open_stream_basic__aaudio(pContext, pDeviceID, deviceType, ma_share_mode_shared, &pStream);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- /*
- The event for playback is needs to be manual reset because we want to explicitly control the fact that it becomes signalled
- only after the whole available space has been filled, never before.
+ ma_context_add_native_data_format_from_AAudioStream__aaudio(pContext, pStream, 0, pDeviceInfo);
- The playback event also needs to be initially set to a signaled state so that the first call to ma_device_write() is able
- to get passed WaitForMultipleObjects().
- */
- pDevice->wasapi.hEventPlayback = CreateEventW(NULL, FALSE, TRUE, NULL); /* Auto reset, signaled by default. */
- if (pDevice->wasapi.hEventPlayback == NULL) {
- result = ma_result_from_GetLastError(GetLastError());
+ ma_close_stream__aaudio(pContext, pStream);
+ pStream = NULL;
- if (pConfig->deviceType == ma_device_type_duplex) {
- if (pDevice->wasapi.pCaptureClient != NULL) {
- ma_IAudioCaptureClient_Release((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient);
- pDevice->wasapi.pCaptureClient = NULL;
- }
- if (pDevice->wasapi.pAudioClientCapture != NULL) {
- ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture);
- pDevice->wasapi.pAudioClientCapture = NULL;
- }
+ return MA_SUCCESS;
+}
- CloseHandle(pDevice->wasapi.hEventCapture);
- pDevice->wasapi.hEventCapture = NULL;
- }
- if (pDevice->wasapi.pRenderClient != NULL) {
- ma_IAudioRenderClient_Release((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient);
- pDevice->wasapi.pRenderClient = NULL;
- }
- if (pDevice->wasapi.pAudioClientPlayback != NULL) {
- ma_IAudioClient_Release((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback);
- pDevice->wasapi.pAudioClientPlayback = NULL;
- }
+static ma_result ma_device_uninit__aaudio(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create event for playback.", result);
- }
- ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, pDevice->wasapi.hEventPlayback);
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ ma_close_stream__aaudio(pDevice->pContext, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture);
+ pDevice->aaudio.pStreamCapture = NULL;
+ }
- pDevice->wasapi.periodSizeInFramesPlayback = data.periodSizeInFramesOut;
- ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &pDevice->wasapi.actualPeriodSizeInFramesPlayback);
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ ma_close_stream__aaudio(pDevice->pContext, (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback);
+ pDevice->aaudio.pStreamPlayback = NULL;
+ }
- /* We must always have a valid ID. */
- ma_wcscpy_s(pDevice->playback.id.wasapi, sizeof(pDevice->playback.id.wasapi), data.id.wasapi);
+ return MA_SUCCESS;
+}
- /* The descriptor needs to be updated with actual values. */
- pDescriptorPlayback->format = data.formatOut;
- pDescriptorPlayback->channels = data.channelsOut;
- pDescriptorPlayback->sampleRate = data.sampleRateOut;
- MA_COPY_MEMORY(pDescriptorPlayback->channelMap, data.channelMapOut, sizeof(data.channelMapOut));
- pDescriptorPlayback->periodSizeInFrames = data.periodSizeInFramesOut;
- pDescriptorPlayback->periodCount = data.periodsOut;
- }
+static ma_result ma_device_init_by_type__aaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_type deviceType, ma_device_descriptor* pDescriptor, ma_AAudioStream** ppStream)
+{
+ ma_result result;
+ int32_t bufferCapacityInFrames;
+ int32_t framesPerDataCallback;
+ ma_AAudioStream* pStream;
- /*
- We need to register a notification client to detect when the device has been disabled, unplugged or re-routed (when the default device changes). When
- we are connecting to the default device we want to do automatic stream routing when the device is disabled or unplugged. Otherwise we want to just
- stop the device outright and let the application handle it.
- */
-#ifdef MA_WIN32_DESKTOP
- if (pConfig->wasapi.noAutoStreamRouting == MA_FALSE) {
- if ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.pDeviceID == NULL) {
- pDevice->wasapi.allowCaptureAutoStreamRouting = MA_TRUE;
- }
- if ((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.pDeviceID == NULL) {
- pDevice->wasapi.allowPlaybackAutoStreamRouting = MA_TRUE;
- }
+ MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(pConfig != NULL);
+ MA_ASSERT(pDescriptor != NULL);
+
+ *ppStream = NULL; /* Safety. */
+
+ /* First step is to open the stream. From there we'll be able to extract the internal configuration. */
+ result = ma_open_stream__aaudio(pDevice, pConfig, deviceType, pDescriptor, &pStream);
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to open the AAudio stream. */
}
- hr = ma_CoCreateInstance(pDevice->pContext, MA_CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, MA_IID_IMMDeviceEnumerator, (void**)&pDeviceEnumerator);
- if (FAILED(hr)) {
- ma_device_uninit__wasapi(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create device enumerator.", ma_result_from_HRESULT(hr));
+ /* Now extract the internal configuration. */
+ pDescriptor->format = (((MA_PFN_AAudioStream_getFormat)pDevice->pContext->aaudio.AAudioStream_getFormat)(pStream) == MA_AAUDIO_FORMAT_PCM_I16) ? ma_format_s16 : ma_format_f32;
+ pDescriptor->channels = ((MA_PFN_AAudioStream_getChannelCount)pDevice->pContext->aaudio.AAudioStream_getChannelCount)(pStream);
+ pDescriptor->sampleRate = ((MA_PFN_AAudioStream_getSampleRate)pDevice->pContext->aaudio.AAudioStream_getSampleRate)(pStream);
+
+ /* For the channel map we need to be sure we don't overflow any buffers. */
+ if (pDescriptor->channels <= MA_MAX_CHANNELS) {
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pDescriptor->channelMap, ma_countof(pDescriptor->channelMap), pDescriptor->channels); /* <-- Cannot find info on channel order, so assuming a default. */
+ } else {
+ ma_channel_map_init_blank(pDescriptor->channelMap, MA_MAX_CHANNELS); /* Too many channels. Use a blank channel map. */
}
- pDevice->wasapi.notificationClient.lpVtbl = (void*)&g_maNotificationCientVtbl;
- pDevice->wasapi.notificationClient.counter = 1;
- pDevice->wasapi.notificationClient.pDevice = pDevice;
+ bufferCapacityInFrames = ((MA_PFN_AAudioStream_getBufferCapacityInFrames)pDevice->pContext->aaudio.AAudioStream_getBufferCapacityInFrames)(pStream);
+ framesPerDataCallback = ((MA_PFN_AAudioStream_getFramesPerDataCallback)pDevice->pContext->aaudio.AAudioStream_getFramesPerDataCallback)(pStream);
- hr = pDeviceEnumerator->lpVtbl->RegisterEndpointNotificationCallback(pDeviceEnumerator, &pDevice->wasapi.notificationClient);
- if (SUCCEEDED(hr)) {
- pDevice->wasapi.pDeviceEnumerator = (ma_ptr)pDeviceEnumerator;
+ if (framesPerDataCallback > 0) {
+ pDescriptor->periodSizeInFrames = framesPerDataCallback;
+ pDescriptor->periodCount = bufferCapacityInFrames / framesPerDataCallback;
} else {
- /* Not the end of the world if we fail to register the notification callback. We just won't support automatic stream routing. */
- ma_IMMDeviceEnumerator_Release(pDeviceEnumerator);
+ pDescriptor->periodSizeInFrames = bufferCapacityInFrames;
+ pDescriptor->periodCount = 1;
}
-#endif
- c89atomic_exchange_32(&pDevice->wasapi.isStartedCapture, MA_FALSE);
- c89atomic_exchange_32(&pDevice->wasapi.isStartedPlayback, MA_FALSE);
+ *ppStream = pStream;
return MA_SUCCESS;
}
-static ma_result ma_device__get_available_frames__wasapi(ma_device* pDevice, ma_IAudioClient* pAudioClient, ma_uint32* pFrameCount)
+static ma_result ma_device_init__aaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
{
- ma_uint32 paddingFramesCount;
- HRESULT hr;
- ma_share_mode shareMode;
+ ma_result result;
MA_ASSERT(pDevice != NULL);
- MA_ASSERT(pFrameCount != NULL);
- *pFrameCount = 0;
+ if (pConfig->deviceType == ma_device_type_loopback) {
+ return MA_DEVICE_TYPE_NOT_SUPPORTED;
+ }
- if ((ma_ptr)pAudioClient != pDevice->wasapi.pAudioClientPlayback && (ma_ptr)pAudioClient != pDevice->wasapi.pAudioClientCapture) {
- return MA_INVALID_OPERATION;
+ pDevice->aaudio.usage = pConfig->aaudio.usage;
+ pDevice->aaudio.contentType = pConfig->aaudio.contentType;
+ pDevice->aaudio.inputPreset = pConfig->aaudio.inputPreset;
+ pDevice->aaudio.noAutoStartAfterReroute = pConfig->aaudio.noAutoStartAfterReroute;
+
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
+ result = ma_device_init_by_type__aaudio(pDevice, pConfig, ma_device_type_capture, pDescriptorCapture, (ma_AAudioStream**)&pDevice->aaudio.pStreamCapture);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
}
- /*
- I've had a report that GetCurrentPadding() is returning a frame count of 0 which is preventing
- higher level function calls from doing anything because it thinks nothing is available. I have
- taken a look at the documentation and it looks like this is unnecessary in exclusive mode.
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ result = ma_device_init_by_type__aaudio(pDevice, pConfig, ma_device_type_playback, pDescriptorPlayback, (ma_AAudioStream**)&pDevice->aaudio.pStreamPlayback);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ }
- From Microsoft's documentation:
+ return MA_SUCCESS;
+}
- For an exclusive-mode rendering or capture stream that was initialized with the
- AUDCLNT_STREAMFLAGS_EVENTCALLBACK flag, the client typically has no use for the padding
- value reported by GetCurrentPadding. Instead, the client accesses an entire buffer during
- each processing pass.
+static ma_result ma_device_start_stream__aaudio(ma_device* pDevice, ma_AAudioStream* pStream)
+{
+ ma_aaudio_result_t resultAA;
+ ma_aaudio_stream_state_t currentState;
- Considering this, I'm going to skip GetCurrentPadding() for exclusive mode and just report the
- entire buffer. This depends on the caller making sure they wait on the event handler.
- */
- shareMode = ((ma_ptr)pAudioClient == pDevice->wasapi.pAudioClientPlayback) ? pDevice->playback.shareMode : pDevice->capture.shareMode;
- if (shareMode == ma_share_mode_shared) {
- /* Shared mode. */
- hr = ma_IAudioClient_GetCurrentPadding(pAudioClient, &paddingFramesCount);
- if (FAILED(hr)) {
- return ma_result_from_HRESULT(hr);
- }
+ MA_ASSERT(pDevice != NULL);
- if ((ma_ptr)pAudioClient == pDevice->wasapi.pAudioClientPlayback) {
- *pFrameCount = pDevice->wasapi.actualPeriodSizeInFramesPlayback - paddingFramesCount;
- } else {
- *pFrameCount = paddingFramesCount;
+ resultAA = ((MA_PFN_AAudioStream_requestStart)pDevice->pContext->aaudio.AAudioStream_requestStart)(pStream);
+ if (resultAA != MA_AAUDIO_OK) {
+ return ma_result_from_aaudio(resultAA);
+ }
+
+ /* Do we actually need to wait for the device to transition into it's started state? */
+
+ /* The device should be in either a starting or started state. If it's not set to started we need to wait for it to transition. It should go from starting to started. */
+ currentState = ((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream);
+ if (currentState != MA_AAUDIO_STREAM_STATE_STARTED) {
+ ma_result result;
+
+ if (currentState != MA_AAUDIO_STREAM_STATE_STARTING) {
+ return MA_ERROR; /* Expecting the stream to be a starting or started state. */
}
- } else {
- /* Exclusive mode. */
- if ((ma_ptr)pAudioClient == pDevice->wasapi.pAudioClientPlayback) {
- *pFrameCount = pDevice->wasapi.actualPeriodSizeInFramesPlayback;
- } else {
- *pFrameCount = pDevice->wasapi.actualPeriodSizeInFramesCapture;
+
+ result = ma_wait_for_simple_state_transition__aaudio(pDevice->pContext, pStream, currentState, MA_AAUDIO_STREAM_STATE_STARTED);
+ if (result != MA_SUCCESS) {
+ return result;
}
}
return MA_SUCCESS;
}
-
-static ma_result ma_device_reroute__wasapi(ma_device* pDevice, ma_device_type deviceType)
+static ma_result ma_device_stop_stream__aaudio(ma_device* pDevice, ma_AAudioStream* pStream)
{
- ma_result result;
+ ma_aaudio_result_t resultAA;
+ ma_aaudio_stream_state_t currentState;
- if (deviceType == ma_device_type_duplex) {
- return MA_INVALID_ARGS;
- }
+ MA_ASSERT(pDevice != NULL);
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "=== CHANGING DEVICE ===\n");
+ /*
+ From the AAudio documentation:
- result = ma_device_reinit__wasapi(pDevice, deviceType);
- if (result != MA_SUCCESS) {
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Reinitializing device after route change failed.\n");
- return result;
+ The stream will stop after all of the data currently buffered has been played.
+
+ This maps with miniaudio's requirement that device's be drained which means we don't need to implement any draining logic.
+ */
+ currentState = ((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream);
+ if (currentState == MA_AAUDIO_STREAM_STATE_DISCONNECTED) {
+ return MA_SUCCESS; /* The device is disconnected. Don't try stopping it. */
}
- ma_device__post_init_setup(pDevice, deviceType);
+ resultAA = ((MA_PFN_AAudioStream_requestStop)pDevice->pContext->aaudio.AAudioStream_requestStop)(pStream);
+ if (resultAA != MA_AAUDIO_OK) {
+ return ma_result_from_aaudio(resultAA);
+ }
+
+ /* The device should be in either a stopping or stopped state. If it's not set to started we need to wait for it to transition. It should go from stopping to stopped. */
+ currentState = ((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream);
+ if (currentState != MA_AAUDIO_STREAM_STATE_STOPPED) {
+ ma_result result;
+
+ if (currentState != MA_AAUDIO_STREAM_STATE_STOPPING) {
+ return MA_ERROR; /* Expecting the stream to be a stopping or stopped state. */
+ }
+
+ result = ma_wait_for_simple_state_transition__aaudio(pDevice->pContext, pStream, currentState, MA_AAUDIO_STREAM_STATE_STOPPED);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ }
return MA_SUCCESS;
}
-static ma_result ma_device_start__wasapi(ma_device* pDevice)
+static ma_result ma_device_start__aaudio(ma_device* pDevice)
{
- HRESULT hr;
-
MA_ASSERT(pDevice != NULL);
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) {
- hr = ma_IAudioClient_Start((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture);
- if (FAILED(hr)) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to start internal capture device.", ma_result_from_HRESULT(hr));
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ ma_result result = ma_device_start_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture);
+ if (result != MA_SUCCESS) {
+ return result;
}
-
- c89atomic_exchange_32(&pDevice->wasapi.isStartedCapture, MA_TRUE);
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- /* No need to do anything for playback as that'll be started automatically in the data loop. */
+ ma_result result = ma_device_start_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback);
+ if (result != MA_SUCCESS) {
+ if (pDevice->type == ma_device_type_duplex) {
+ ma_device_stop_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture);
+ }
+ return result;
+ }
}
return MA_SUCCESS;
}
-static ma_result ma_device_stop__wasapi(ma_device* pDevice)
+static ma_result ma_device_stop__aaudio(ma_device* pDevice)
{
- ma_result result;
- HRESULT hr;
-
MA_ASSERT(pDevice != NULL);
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) {
- hr = ma_IAudioClient_Stop((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture);
- if (FAILED(hr)) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to stop internal capture device.", ma_result_from_HRESULT(hr));
- }
-
- /* The audio client needs to be reset otherwise restarting will fail. */
- hr = ma_IAudioClient_Reset((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture);
- if (FAILED(hr)) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to reset internal capture device.", ma_result_from_HRESULT(hr));
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ ma_result result = ma_device_stop_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture);
+ if (result != MA_SUCCESS) {
+ return result;
}
-
- c89atomic_exchange_32(&pDevice->wasapi.isStartedCapture, MA_FALSE);
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- /*
- The buffer needs to be drained before stopping the device. Not doing this will result in the last few frames not getting output to
- the speakers. This is a problem for very short sounds because it'll result in a significant portion of it not getting played.
- */
- if (c89atomic_load_32(&pDevice->wasapi.isStartedPlayback)) {
- /* We need to make sure we put a timeout here or else we'll risk getting stuck in a deadlock in some cases. */
- DWORD waitTime = pDevice->wasapi.actualPeriodSizeInFramesPlayback / pDevice->playback.internalSampleRate;
-
- if (pDevice->playback.shareMode == ma_share_mode_exclusive) {
- WaitForSingleObject(pDevice->wasapi.hEventPlayback, waitTime);
- } else {
- ma_uint32 prevFramesAvaialablePlayback = (ma_uint32)-1;
- ma_uint32 framesAvailablePlayback;
- for (;;) {
- result = ma_device__get_available_frames__wasapi(pDevice, (ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &framesAvailablePlayback);
- if (result != MA_SUCCESS) {
- break;
- }
-
- if (framesAvailablePlayback >= pDevice->wasapi.actualPeriodSizeInFramesPlayback) {
- break;
- }
-
- /*
- Just a safety check to avoid an infinite loop. If this iteration results in a situation where the number of available frames
- has not changed, get out of the loop. I don't think this should ever happen, but I think it's nice to have just in case.
- */
- if (framesAvailablePlayback == prevFramesAvaialablePlayback) {
- break;
- }
- prevFramesAvaialablePlayback = framesAvailablePlayback;
-
- WaitForSingleObject(pDevice->wasapi.hEventPlayback, waitTime);
- ResetEvent(pDevice->wasapi.hEventPlayback); /* Manual reset. */
- }
- }
- }
-
- hr = ma_IAudioClient_Stop((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback);
- if (FAILED(hr)) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to stop internal playback device.", ma_result_from_HRESULT(hr));
- }
-
- /* The audio client needs to be reset otherwise restarting will fail. */
- hr = ma_IAudioClient_Reset((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback);
- if (FAILED(hr)) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to reset internal playback device.", ma_result_from_HRESULT(hr));
+ ma_result result = ma_device_stop_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback);
+ if (result != MA_SUCCESS) {
+ return result;
}
-
- c89atomic_exchange_32(&pDevice->wasapi.isStartedPlayback, MA_FALSE);
}
+ ma_device__on_notification_stopped(pDevice);
+
return MA_SUCCESS;
}
-
-#ifndef MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS
-#define MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS 5000
-#endif
-
-static ma_result ma_device_data_loop__wasapi(ma_device* pDevice)
+static ma_result ma_device_reinit__aaudio(ma_device* pDevice, ma_device_type deviceType)
{
ma_result result;
- HRESULT hr;
- ma_bool32 exitLoop = MA_FALSE;
- ma_uint32 framesWrittenToPlaybackDevice = 0;
- ma_uint32 mappedDeviceBufferSizeInFramesCapture = 0;
- ma_uint32 mappedDeviceBufferSizeInFramesPlayback = 0;
- ma_uint32 mappedDeviceBufferFramesRemainingCapture = 0;
- ma_uint32 mappedDeviceBufferFramesRemainingPlayback = 0;
- BYTE* pMappedDeviceBufferCapture = NULL;
- BYTE* pMappedDeviceBufferPlayback = NULL;
- ma_uint32 bpfCaptureDevice = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
- ma_uint32 bpfPlaybackDevice = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
- ma_uint32 bpfCaptureClient = ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels);
- ma_uint32 bpfPlaybackClient = ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
- ma_uint8 inputDataInClientFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
- ma_uint32 inputDataInClientFormatCap = 0;
- ma_uint8 outputDataInClientFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
- ma_uint32 outputDataInClientFormatCap = 0;
- ma_uint32 outputDataInClientFormatCount = 0;
- ma_uint32 outputDataInClientFormatConsumed = 0;
- ma_uint32 periodSizeInFramesCapture = 0;
MA_ASSERT(pDevice != NULL);
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) {
- periodSizeInFramesCapture = pDevice->capture.internalPeriodSizeInFrames;
- inputDataInClientFormatCap = sizeof(inputDataInClientFormat) / bpfCaptureClient;
+ /* The first thing to do is close the streams. */
+ if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex) {
+ ma_close_stream__aaudio(pDevice->pContext, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture);
+ pDevice->aaudio.pStreamCapture = NULL;
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- outputDataInClientFormatCap = sizeof(outputDataInClientFormat) / bpfPlaybackClient;
+ if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) {
+ ma_close_stream__aaudio(pDevice->pContext, (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback);
+ pDevice->aaudio.pStreamPlayback = NULL;
}
- while (ma_device_get_state(pDevice) == MA_STATE_STARTED && !exitLoop) {
- switch (pDevice->type)
- {
- case ma_device_type_duplex:
- {
- ma_uint32 framesAvailableCapture;
- ma_uint32 framesAvailablePlayback;
- DWORD flagsCapture; /* Passed to IAudioCaptureClient_GetBuffer(). */
-
- /* The process is to map the playback buffer and fill it as quickly as possible from input data. */
- if (pMappedDeviceBufferPlayback == NULL) {
- result = ma_device__get_available_frames__wasapi(pDevice, (ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &framesAvailablePlayback);
- if (result != MA_SUCCESS) {
- return result;
- }
-
- /* In exclusive mode, the frame count needs to exactly match the value returned by GetCurrentPadding(). */
- if (pDevice->playback.shareMode != ma_share_mode_exclusive) {
- if (framesAvailablePlayback > pDevice->wasapi.periodSizeInFramesPlayback) {
- framesAvailablePlayback = pDevice->wasapi.periodSizeInFramesPlayback;
- }
- }
-
- /* We're ready to map the playback device's buffer. We don't release this until it's been entirely filled. */
- hr = ma_IAudioRenderClient_GetBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, framesAvailablePlayback, &pMappedDeviceBufferPlayback);
- if (FAILED(hr)) {
- ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from playback device in preparation for writing to the device.", ma_result_from_HRESULT(hr));
- exitLoop = MA_TRUE;
- break;
- }
-
- mappedDeviceBufferSizeInFramesPlayback = framesAvailablePlayback;
- mappedDeviceBufferFramesRemainingPlayback = framesAvailablePlayback;
- }
-
- if (mappedDeviceBufferFramesRemainingPlayback > 0) {
- /* At this point we should have a buffer available for output. We need to keep writing input samples to it. */
- for (;;) {
- /* Try grabbing some captured data if we haven't already got a mapped buffer. */
- if (pMappedDeviceBufferCapture == NULL) {
- if (pDevice->capture.shareMode == ma_share_mode_shared) {
- if (WaitForSingleObject(pDevice->wasapi.hEventCapture, MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS) != WAIT_OBJECT_0) {
- return MA_ERROR; /* Wait failed. */
- }
- }
-
- result = ma_device__get_available_frames__wasapi(pDevice, (ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, &framesAvailableCapture);
- if (result != MA_SUCCESS) {
- exitLoop = MA_TRUE;
- break;
- }
-
- /* Wait for more if nothing is available. */
- if (framesAvailableCapture == 0) {
- /* In exclusive mode we waited at the top. */
- if (pDevice->capture.shareMode != ma_share_mode_shared) {
- if (WaitForSingleObject(pDevice->wasapi.hEventCapture, MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS) != WAIT_OBJECT_0) {
- return MA_ERROR; /* Wait failed. */
- }
- }
-
- continue;
- }
-
- /* Getting here means there's data available for writing to the output device. */
- mappedDeviceBufferSizeInFramesCapture = ma_min(framesAvailableCapture, periodSizeInFramesCapture);
- hr = ma_IAudioCaptureClient_GetBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, (BYTE**)&pMappedDeviceBufferCapture, &mappedDeviceBufferSizeInFramesCapture, &flagsCapture, NULL, NULL);
- if (FAILED(hr)) {
- ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from capture device in preparation for writing to the device.", ma_result_from_HRESULT(hr));
- exitLoop = MA_TRUE;
- break;
- }
-
-
- /* Overrun detection. */
- if ((flagsCapture & MA_AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY) != 0) {
- /* Glitched. Probably due to an overrun. */
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Data discontinuity (possible overrun). framesAvailableCapture=%d, mappedBufferSizeInFramesCapture=%d\n", framesAvailableCapture, mappedDeviceBufferSizeInFramesCapture);
-
- /*
- Exeriment: If we get an overrun it probably means we're straddling the end of the buffer. In order to prevent a never-ending sequence of glitches let's experiment
- by dropping every frame until we're left with only a single period. To do this we just keep retrieving and immediately releasing buffers until we're down to the
- last period.
- */
- if (framesAvailableCapture >= pDevice->wasapi.actualPeriodSizeInFramesCapture) {
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Synchronizing capture stream. ");
- do
- {
- hr = ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, mappedDeviceBufferSizeInFramesCapture);
- if (FAILED(hr)) {
- break;
- }
-
- framesAvailableCapture -= mappedDeviceBufferSizeInFramesCapture;
-
- if (framesAvailableCapture > 0) {
- mappedDeviceBufferSizeInFramesCapture = ma_min(framesAvailableCapture, periodSizeInFramesCapture);
- hr = ma_IAudioCaptureClient_GetBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, (BYTE**)&pMappedDeviceBufferCapture, &mappedDeviceBufferSizeInFramesCapture, &flagsCapture, NULL, NULL);
- if (FAILED(hr)) {
- ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from capture device in preparation for writing to the device.", ma_result_from_HRESULT(hr));
- exitLoop = MA_TRUE;
- break;
- }
- } else {
- pMappedDeviceBufferCapture = NULL;
- mappedDeviceBufferSizeInFramesCapture = 0;
- }
- } while (framesAvailableCapture > periodSizeInFramesCapture);
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "framesAvailableCapture=%d, mappedBufferSizeInFramesCapture=%d\n", framesAvailableCapture, mappedDeviceBufferSizeInFramesCapture);
- }
- } else {
- #ifdef MA_DEBUG_OUTPUT
- if (flagsCapture != 0) {
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Capture Flags: %ld\n", flagsCapture);
- }
- #endif
- }
-
- mappedDeviceBufferFramesRemainingCapture = mappedDeviceBufferSizeInFramesCapture;
- }
-
-
- /* At this point we should have both input and output data available. We now need to convert the data and post it to the client. */
- for (;;) {
- BYTE* pRunningDeviceBufferCapture;
- BYTE* pRunningDeviceBufferPlayback;
- ma_uint32 framesToProcess;
- ma_uint32 framesProcessed;
-
- pRunningDeviceBufferCapture = pMappedDeviceBufferCapture + ((mappedDeviceBufferSizeInFramesCapture - mappedDeviceBufferFramesRemainingCapture ) * bpfCaptureDevice);
- pRunningDeviceBufferPlayback = pMappedDeviceBufferPlayback + ((mappedDeviceBufferSizeInFramesPlayback - mappedDeviceBufferFramesRemainingPlayback) * bpfPlaybackDevice);
-
- /* There may be some data sitting in the converter that needs to be processed first. Once this is exhaused, run the data callback again. */
- if (!pDevice->playback.converter.isPassthrough && outputDataInClientFormatConsumed < outputDataInClientFormatCount) {
- ma_uint64 convertedFrameCountClient = (outputDataInClientFormatCount - outputDataInClientFormatConsumed);
- ma_uint64 convertedFrameCountDevice = mappedDeviceBufferFramesRemainingPlayback;
- void* pConvertedFramesClient = outputDataInClientFormat + (outputDataInClientFormatConsumed * bpfPlaybackClient);
- void* pConvertedFramesDevice = pRunningDeviceBufferPlayback;
- result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, pConvertedFramesClient, &convertedFrameCountClient, pConvertedFramesDevice, &convertedFrameCountDevice);
- if (result != MA_SUCCESS) {
- break;
- }
-
- outputDataInClientFormatConsumed += (ma_uint32)convertedFrameCountClient; /* Safe cast. */
- mappedDeviceBufferFramesRemainingPlayback -= (ma_uint32)convertedFrameCountDevice; /* Safe cast. */
-
- if (mappedDeviceBufferFramesRemainingPlayback == 0) {
- break;
- }
- }
-
- /*
- Getting here means we need to fire the callback. If format conversion is unnecessary, we can optimize this by passing the pointers to the internal
- buffers directly to the callback.
- */
- if (pDevice->capture.converter.isPassthrough && pDevice->playback.converter.isPassthrough) {
- /* Optimal path. We can pass mapped pointers directly to the callback. */
- framesToProcess = ma_min(mappedDeviceBufferFramesRemainingCapture, mappedDeviceBufferFramesRemainingPlayback);
- framesProcessed = framesToProcess;
-
- ma_device__on_data(pDevice, pRunningDeviceBufferPlayback, pRunningDeviceBufferCapture, framesToProcess);
-
- mappedDeviceBufferFramesRemainingCapture -= framesProcessed;
- mappedDeviceBufferFramesRemainingPlayback -= framesProcessed;
-
- if (mappedDeviceBufferFramesRemainingCapture == 0) {
- break; /* Exhausted input data. */
- }
- if (mappedDeviceBufferFramesRemainingPlayback == 0) {
- break; /* Exhausted output data. */
- }
- } else if (pDevice->capture.converter.isPassthrough) {
- /* The input buffer is a passthrough, but the playback buffer requires a conversion. */
- framesToProcess = ma_min(mappedDeviceBufferFramesRemainingCapture, outputDataInClientFormatCap);
- framesProcessed = framesToProcess;
-
- ma_device__on_data(pDevice, outputDataInClientFormat, pRunningDeviceBufferCapture, framesToProcess);
- outputDataInClientFormatCount = framesProcessed;
- outputDataInClientFormatConsumed = 0;
-
- mappedDeviceBufferFramesRemainingCapture -= framesProcessed;
- if (mappedDeviceBufferFramesRemainingCapture == 0) {
- break; /* Exhausted input data. */
- }
- } else if (pDevice->playback.converter.isPassthrough) {
- /* The input buffer requires conversion, the playback buffer is passthrough. */
- ma_uint64 capturedDeviceFramesToProcess = mappedDeviceBufferFramesRemainingCapture;
- ma_uint64 capturedClientFramesToProcess = ma_min(inputDataInClientFormatCap, mappedDeviceBufferFramesRemainingPlayback);
-
- result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningDeviceBufferCapture, &capturedDeviceFramesToProcess, inputDataInClientFormat, &capturedClientFramesToProcess);
- if (result != MA_SUCCESS) {
- break;
- }
-
- if (capturedClientFramesToProcess == 0) {
- break;
- }
-
- ma_device__on_data(pDevice, pRunningDeviceBufferPlayback, inputDataInClientFormat, (ma_uint32)capturedClientFramesToProcess); /* Safe cast. */
-
- mappedDeviceBufferFramesRemainingCapture -= (ma_uint32)capturedDeviceFramesToProcess;
- mappedDeviceBufferFramesRemainingPlayback -= (ma_uint32)capturedClientFramesToProcess;
- } else {
- ma_uint64 capturedDeviceFramesToProcess = mappedDeviceBufferFramesRemainingCapture;
- ma_uint64 capturedClientFramesToProcess = ma_min(inputDataInClientFormatCap, outputDataInClientFormatCap);
-
- result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningDeviceBufferCapture, &capturedDeviceFramesToProcess, inputDataInClientFormat, &capturedClientFramesToProcess);
- if (result != MA_SUCCESS) {
- break;
- }
+ /* Now we need to reinitialize each streams. The hardest part with this is just filling output the config and descriptors. */
+ {
+ ma_device_config deviceConfig;
+ ma_device_descriptor descriptorPlayback;
+ ma_device_descriptor descriptorCapture;
+
+ deviceConfig = ma_device_config_init(deviceType);
+ deviceConfig.playback.pDeviceID = NULL; /* Only doing rerouting with default devices. */
+ deviceConfig.playback.shareMode = pDevice->playback.shareMode;
+ deviceConfig.playback.format = pDevice->playback.format;
+ deviceConfig.playback.channels = pDevice->playback.channels;
+ deviceConfig.capture.pDeviceID = NULL; /* Only doing rerouting with default devices. */
+ deviceConfig.capture.shareMode = pDevice->capture.shareMode;
+ deviceConfig.capture.format = pDevice->capture.format;
+ deviceConfig.capture.channels = pDevice->capture.channels;
+ deviceConfig.sampleRate = pDevice->sampleRate;
+ deviceConfig.aaudio.usage = pDevice->aaudio.usage;
+ deviceConfig.aaudio.contentType = pDevice->aaudio.contentType;
+ deviceConfig.aaudio.inputPreset = pDevice->aaudio.inputPreset;
+ deviceConfig.aaudio.noAutoStartAfterReroute = pDevice->aaudio.noAutoStartAfterReroute;
+ deviceConfig.periods = 1;
+
+ /* Try to get an accurate period size. */
+ if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) {
+ deviceConfig.periodSizeInFrames = pDevice->playback.internalPeriodSizeInFrames;
+ } else {
+ deviceConfig.periodSizeInFrames = pDevice->capture.internalPeriodSizeInFrames;
+ }
- if (capturedClientFramesToProcess == 0) {
- break;
- }
+ if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex || deviceType == ma_device_type_loopback) {
+ descriptorCapture.pDeviceID = deviceConfig.capture.pDeviceID;
+ descriptorCapture.shareMode = deviceConfig.capture.shareMode;
+ descriptorCapture.format = deviceConfig.capture.format;
+ descriptorCapture.channels = deviceConfig.capture.channels;
+ descriptorCapture.sampleRate = deviceConfig.sampleRate;
+ descriptorCapture.periodSizeInFrames = deviceConfig.periodSizeInFrames;
+ descriptorCapture.periodCount = deviceConfig.periods;
+ }
+
+ if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) {
+ descriptorPlayback.pDeviceID = deviceConfig.playback.pDeviceID;
+ descriptorPlayback.shareMode = deviceConfig.playback.shareMode;
+ descriptorPlayback.format = deviceConfig.playback.format;
+ descriptorPlayback.channels = deviceConfig.playback.channels;
+ descriptorPlayback.sampleRate = deviceConfig.sampleRate;
+ descriptorPlayback.periodSizeInFrames = deviceConfig.periodSizeInFrames;
+ descriptorPlayback.periodCount = deviceConfig.periods;
+ }
+
+ result = ma_device_init__aaudio(pDevice, &deviceConfig, &descriptorPlayback, &descriptorCapture);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- ma_device__on_data(pDevice, outputDataInClientFormat, inputDataInClientFormat, (ma_uint32)capturedClientFramesToProcess);
+ result = ma_device_post_init(pDevice, deviceType, &descriptorPlayback, &descriptorCapture);
+ if (result != MA_SUCCESS) {
+ ma_device_uninit__aaudio(pDevice);
+ return result;
+ }
- mappedDeviceBufferFramesRemainingCapture -= (ma_uint32)capturedDeviceFramesToProcess;
- outputDataInClientFormatCount = (ma_uint32)capturedClientFramesToProcess;
- outputDataInClientFormatConsumed = 0;
- }
- }
+ /* We'll only ever do this in response to a reroute. */
+ ma_device__on_notification_rerouted(pDevice);
+ /* If the device is started, start the streams. Maybe make this configurable? */
+ if (ma_device_get_state(pDevice) == ma_device_state_started) {
+ if (pDevice->aaudio.noAutoStartAfterReroute == MA_FALSE) {
+ ma_device_start__aaudio(pDevice);
+ } else {
+ ma_device_stop(pDevice); /* Do a full device stop so we set internal state correctly. */
+ }
+ }
- /* If at this point we've run out of capture data we need to release the buffer. */
- if (mappedDeviceBufferFramesRemainingCapture == 0 && pMappedDeviceBufferCapture != NULL) {
- hr = ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, mappedDeviceBufferSizeInFramesCapture);
- if (FAILED(hr)) {
- ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to release internal buffer from capture device after reading from the device.", ma_result_from_HRESULT(hr));
- exitLoop = MA_TRUE;
- break;
- }
+ return MA_SUCCESS;
+ }
+}
- pMappedDeviceBufferCapture = NULL;
- mappedDeviceBufferFramesRemainingCapture = 0;
- mappedDeviceBufferSizeInFramesCapture = 0;
- }
+static ma_result ma_device_get_info__aaudio(ma_device* pDevice, ma_device_type type, ma_device_info* pDeviceInfo)
+{
+ ma_AAudioStream* pStream = NULL;
- /* Get out of this loop if we're run out of room in the playback buffer. */
- if (mappedDeviceBufferFramesRemainingPlayback == 0) {
- break;
- }
- }
- }
+ MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(type != ma_device_type_duplex);
+ MA_ASSERT(pDeviceInfo != NULL);
+ if (type == ma_device_type_playback) {
+ pStream = (ma_AAudioStream*)pDevice->aaudio.pStreamCapture;
+ pDeviceInfo->id.aaudio = pDevice->capture.id.aaudio;
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1); /* Only supporting default devices. */
+ }
+ if (type == ma_device_type_capture) {
+ pStream = (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback;
+ pDeviceInfo->id.aaudio = pDevice->playback.id.aaudio;
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1); /* Only supporting default devices. */
+ }
- /* If at this point we've run out of data we need to release the buffer. */
- if (mappedDeviceBufferFramesRemainingPlayback == 0 && pMappedDeviceBufferPlayback != NULL) {
- hr = ma_IAudioRenderClient_ReleaseBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, mappedDeviceBufferSizeInFramesPlayback, 0);
- if (FAILED(hr)) {
- ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to release internal buffer from playback device after writing to the device.", ma_result_from_HRESULT(hr));
- exitLoop = MA_TRUE;
- break;
- }
+ /* Safety. Should never happen. */
+ if (pStream == NULL) {
+ return MA_INVALID_OPERATION;
+ }
- framesWrittenToPlaybackDevice += mappedDeviceBufferSizeInFramesPlayback;
+ pDeviceInfo->nativeDataFormatCount = 0;
+ ma_context_add_native_data_format_from_AAudioStream__aaudio(pDevice->pContext, pStream, 0, pDeviceInfo);
- pMappedDeviceBufferPlayback = NULL;
- mappedDeviceBufferFramesRemainingPlayback = 0;
- mappedDeviceBufferSizeInFramesPlayback = 0;
- }
+ return MA_SUCCESS;
+}
- if (!c89atomic_load_32(&pDevice->wasapi.isStartedPlayback)) {
- ma_uint32 startThreshold = pDevice->playback.internalPeriodSizeInFrames * 1;
- /* Prevent a deadlock. If we don't clamp against the actual buffer size we'll never end up starting the playback device which will result in a deadlock. */
- if (startThreshold > pDevice->wasapi.actualPeriodSizeInFramesPlayback) {
- startThreshold = pDevice->wasapi.actualPeriodSizeInFramesPlayback;
- }
+static ma_result ma_context_uninit__aaudio(ma_context* pContext)
+{
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pContext->backend == ma_backend_aaudio);
- if (pDevice->playback.shareMode == ma_share_mode_exclusive || framesWrittenToPlaybackDevice >= startThreshold) {
- hr = ma_IAudioClient_Start((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback);
- if (FAILED(hr)) {
- ma_IAudioClient_Stop((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture);
- ma_IAudioClient_Reset((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to start internal playback device.", ma_result_from_HRESULT(hr));
- }
+ ma_device_job_thread_uninit(&pContext->aaudio.jobThread, &pContext->allocationCallbacks);
- c89atomic_exchange_32(&pDevice->wasapi.isStartedPlayback, MA_TRUE);
- }
- }
+ ma_dlclose(pContext, pContext->aaudio.hAAudio);
+ pContext->aaudio.hAAudio = NULL;
- /* Make sure the device has started before waiting. */
- if (WaitForSingleObject(pDevice->wasapi.hEventPlayback, MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS) != WAIT_OBJECT_0) {
- return MA_ERROR; /* Wait failed. */
- }
- } break;
+ return MA_SUCCESS;
+}
+static ma_result ma_context_init__aaudio(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+{
+ size_t i;
+ const char* libNames[] = {
+ "libaaudio.so"
+ };
+ for (i = 0; i < ma_countof(libNames); ++i) {
+ pContext->aaudio.hAAudio = ma_dlopen(pContext, libNames[i]);
+ if (pContext->aaudio.hAAudio != NULL) {
+ break;
+ }
+ }
- case ma_device_type_capture:
- case ma_device_type_loopback:
- {
- ma_uint32 framesAvailableCapture;
- DWORD flagsCapture; /* Passed to IAudioCaptureClient_GetBuffer(). */
+ if (pContext->aaudio.hAAudio == NULL) {
+ return MA_FAILED_TO_INIT_BACKEND;
+ }
- /* Wait for data to become available first. */
- if (WaitForSingleObject(pDevice->wasapi.hEventCapture, MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS) != WAIT_OBJECT_0) {
- /*
- For capture we can terminate here because it probably means the microphone just isn't delivering data for whatever reason, but
- for loopback is most likely means nothing is actually playing. We want to keep trying in this situation.
- */
- if (pDevice->type == ma_device_type_loopback) {
- continue; /* Keep waiting in loopback mode. */
- } else {
- exitLoop = MA_TRUE;
- break; /* Wait failed. */
- }
- }
+ pContext->aaudio.AAudio_createStreamBuilder = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudio_createStreamBuilder");
+ pContext->aaudio.AAudioStreamBuilder_delete = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_delete");
+ pContext->aaudio.AAudioStreamBuilder_setDeviceId = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setDeviceId");
+ pContext->aaudio.AAudioStreamBuilder_setDirection = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setDirection");
+ pContext->aaudio.AAudioStreamBuilder_setSharingMode = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setSharingMode");
+ pContext->aaudio.AAudioStreamBuilder_setFormat = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setFormat");
+ pContext->aaudio.AAudioStreamBuilder_setChannelCount = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setChannelCount");
+ pContext->aaudio.AAudioStreamBuilder_setSampleRate = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setSampleRate");
+ pContext->aaudio.AAudioStreamBuilder_setBufferCapacityInFrames = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setBufferCapacityInFrames");
+ pContext->aaudio.AAudioStreamBuilder_setFramesPerDataCallback = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setFramesPerDataCallback");
+ pContext->aaudio.AAudioStreamBuilder_setDataCallback = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setDataCallback");
+ pContext->aaudio.AAudioStreamBuilder_setErrorCallback = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setErrorCallback");
+ pContext->aaudio.AAudioStreamBuilder_setPerformanceMode = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setPerformanceMode");
+ pContext->aaudio.AAudioStreamBuilder_setUsage = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setUsage");
+ pContext->aaudio.AAudioStreamBuilder_setContentType = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setContentType");
+ pContext->aaudio.AAudioStreamBuilder_setInputPreset = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setInputPreset");
+ pContext->aaudio.AAudioStreamBuilder_openStream = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_openStream");
+ pContext->aaudio.AAudioStream_close = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_close");
+ pContext->aaudio.AAudioStream_getState = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getState");
+ pContext->aaudio.AAudioStream_waitForStateChange = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_waitForStateChange");
+ pContext->aaudio.AAudioStream_getFormat = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getFormat");
+ pContext->aaudio.AAudioStream_getChannelCount = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getChannelCount");
+ pContext->aaudio.AAudioStream_getSampleRate = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getSampleRate");
+ pContext->aaudio.AAudioStream_getBufferCapacityInFrames = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getBufferCapacityInFrames");
+ pContext->aaudio.AAudioStream_getFramesPerDataCallback = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getFramesPerDataCallback");
+ pContext->aaudio.AAudioStream_getFramesPerBurst = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getFramesPerBurst");
+ pContext->aaudio.AAudioStream_requestStart = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_requestStart");
+ pContext->aaudio.AAudioStream_requestStop = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_requestStop");
- /* See how many frames are available. Since we waited at the top, I don't think this should ever return 0. I'm checking for this anyway. */
- result = ma_device__get_available_frames__wasapi(pDevice, (ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, &framesAvailableCapture);
- if (result != MA_SUCCESS) {
- exitLoop = MA_TRUE;
- break;
- }
- if (framesAvailableCapture < pDevice->wasapi.periodSizeInFramesCapture) {
- continue; /* Nothing available. Keep waiting. */
- }
+ pCallbacks->onContextInit = ma_context_init__aaudio;
+ pCallbacks->onContextUninit = ma_context_uninit__aaudio;
+ pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__aaudio;
+ pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__aaudio;
+ pCallbacks->onDeviceInit = ma_device_init__aaudio;
+ pCallbacks->onDeviceUninit = ma_device_uninit__aaudio;
+ pCallbacks->onDeviceStart = ma_device_start__aaudio;
+ pCallbacks->onDeviceStop = ma_device_stop__aaudio;
+ pCallbacks->onDeviceRead = NULL; /* Not used because AAudio is asynchronous. */
+ pCallbacks->onDeviceWrite = NULL; /* Not used because AAudio is asynchronous. */
+ pCallbacks->onDeviceDataLoop = NULL; /* Not used because AAudio is asynchronous. */
+ pCallbacks->onDeviceGetInfo = ma_device_get_info__aaudio;
- /* Map the data buffer in preparation for sending to the client. */
- mappedDeviceBufferSizeInFramesCapture = framesAvailableCapture;
- hr = ma_IAudioCaptureClient_GetBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, (BYTE**)&pMappedDeviceBufferCapture, &mappedDeviceBufferSizeInFramesCapture, &flagsCapture, NULL, NULL);
- if (FAILED(hr)) {
- ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from capture device in preparation for writing to the device.", ma_result_from_HRESULT(hr));
- exitLoop = MA_TRUE;
- break;
- }
- /* Overrun detection. */
- if ((flagsCapture & MA_AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY) != 0) {
- /* Glitched. Probably due to an overrun. */
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Data discontinuity (possible overrun). framesAvailableCapture=%d, mappedBufferSizeInFramesCapture=%d\n", framesAvailableCapture, mappedDeviceBufferSizeInFramesCapture);
+ /* We need a job thread so we can deal with rerouting. */
+ {
+ ma_result result;
+ ma_device_job_thread_config jobThreadConfig;
- /*
- Exeriment: If we get an overrun it probably means we're straddling the end of the buffer. In order to prevent a never-ending sequence of glitches let's experiment
- by dropping every frame until we're left with only a single period. To do this we just keep retrieving and immediately releasing buffers until we're down to the
- last period.
- */
- if (framesAvailableCapture >= pDevice->wasapi.actualPeriodSizeInFramesCapture) {
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Synchronizing capture stream. ");
- do
- {
- hr = ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, mappedDeviceBufferSizeInFramesCapture);
- if (FAILED(hr)) {
- break;
- }
+ jobThreadConfig = ma_device_job_thread_config_init();
- framesAvailableCapture -= mappedDeviceBufferSizeInFramesCapture;
+ result = ma_device_job_thread_init(&jobThreadConfig, &pContext->allocationCallbacks, &pContext->aaudio.jobThread);
+ if (result != MA_SUCCESS) {
+ ma_dlclose(pContext, pContext->aaudio.hAAudio);
+ pContext->aaudio.hAAudio = NULL;
+ return result;
+ }
+ }
+
- if (framesAvailableCapture > 0) {
- mappedDeviceBufferSizeInFramesCapture = ma_min(framesAvailableCapture, periodSizeInFramesCapture);
- hr = ma_IAudioCaptureClient_GetBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, (BYTE**)&pMappedDeviceBufferCapture, &mappedDeviceBufferSizeInFramesCapture, &flagsCapture, NULL, NULL);
- if (FAILED(hr)) {
- ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from capture device in preparation for writing to the device.", ma_result_from_HRESULT(hr));
- exitLoop = MA_TRUE;
- break;
- }
- } else {
- pMappedDeviceBufferCapture = NULL;
- mappedDeviceBufferSizeInFramesCapture = 0;
- }
- } while (framesAvailableCapture > periodSizeInFramesCapture);
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "framesAvailableCapture=%d, mappedBufferSizeInFramesCapture=%d\n", framesAvailableCapture, mappedDeviceBufferSizeInFramesCapture);
- }
- } else {
- #ifdef MA_DEBUG_OUTPUT
- if (flagsCapture != 0) {
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Capture Flags: %ld\n", flagsCapture);
- }
- #endif
- }
+ (void)pConfig;
+ return MA_SUCCESS;
+}
- /* We should have a buffer at this point, but let's just do a sanity check anyway. */
- if (mappedDeviceBufferSizeInFramesCapture > 0 && pMappedDeviceBufferCapture != NULL) {
- ma_device__send_frames_to_client(pDevice, mappedDeviceBufferSizeInFramesCapture, pMappedDeviceBufferCapture);
+static ma_result ma_job_process__device__aaudio_reroute(ma_job* pJob)
+{
+ ma_device* pDevice;
- /* At this point we're done with the buffer. */
- hr = ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, mappedDeviceBufferSizeInFramesCapture);
- pMappedDeviceBufferCapture = NULL; /* <-- Important. Not doing this can result in an error once we leave this loop because it will use this to know whether or not a final ReleaseBuffer() needs to be called. */
- mappedDeviceBufferSizeInFramesCapture = 0;
- if (FAILED(hr)) {
- ma_post_log_message(ma_device_get_context(pDevice), pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to release internal buffer from capture device after reading from the device.");
- exitLoop = MA_TRUE;
- break;
- }
- }
- } break;
+ MA_ASSERT(pJob != NULL);
+ pDevice = (ma_device*)pJob->data.device.aaudio.reroute.pDevice;
+ MA_ASSERT(pDevice != NULL);
+ /* Here is where we need to reroute the device. To do this we need to uninitialize the stream and reinitialize it. */
+ return ma_device_reinit__aaudio(pDevice, (ma_device_type)pJob->data.device.aaudio.reroute.deviceType);
+}
+#else
+/* Getting here means there is no AAudio backend so we need a no-op job implementation. */
+static ma_result ma_job_process__device__aaudio_reroute(ma_job* pJob)
+{
+ return ma_job_process__noop(pJob);
+}
+#endif /* AAudio */
- case ma_device_type_playback:
- {
- ma_uint32 framesAvailablePlayback;
- /* Check how much space is available. If this returns 0 we just keep waiting. */
- result = ma_device__get_available_frames__wasapi(pDevice, (ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &framesAvailablePlayback);
- if (result != MA_SUCCESS) {
- exitLoop = MA_TRUE;
- break;
- }
+/******************************************************************************
- if (framesAvailablePlayback >= pDevice->wasapi.periodSizeInFramesPlayback) {
- /* Map a the data buffer in preparation for the callback. */
- hr = ma_IAudioRenderClient_GetBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, framesAvailablePlayback, &pMappedDeviceBufferPlayback);
- if (FAILED(hr)) {
- ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from playback device in preparation for writing to the device.", ma_result_from_HRESULT(hr));
- exitLoop = MA_TRUE;
- break;
- }
+OpenSL|ES Backend
- /* We should have a buffer at this point. */
- ma_device__read_frames_from_client(pDevice, framesAvailablePlayback, pMappedDeviceBufferPlayback);
+******************************************************************************/
+#ifdef MA_HAS_OPENSL
+#include
+#ifdef MA_ANDROID
+#include
+#endif
- /* At this point we're done writing to the device and we just need to release the buffer. */
- hr = ma_IAudioRenderClient_ReleaseBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, framesAvailablePlayback, 0);
- pMappedDeviceBufferPlayback = NULL; /* <-- Important. Not doing this can result in an error once we leave this loop because it will use this to know whether or not a final ReleaseBuffer() needs to be called. */
- mappedDeviceBufferSizeInFramesPlayback = 0;
+typedef SLresult (SLAPIENTRY * ma_slCreateEngine_proc)(SLObjectItf* pEngine, SLuint32 numOptions, SLEngineOption* pEngineOptions, SLuint32 numInterfaces, SLInterfaceID* pInterfaceIds, SLboolean* pInterfaceRequired);
- if (FAILED(hr)) {
- ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to release internal buffer from playback device after writing to the device.", ma_result_from_HRESULT(hr));
- exitLoop = MA_TRUE;
- break;
- }
+/* OpenSL|ES has one-per-application objects :( */
+static SLObjectItf g_maEngineObjectSL = NULL;
+static SLEngineItf g_maEngineSL = NULL;
+static ma_uint32 g_maOpenSLInitCounter = 0;
+static ma_spinlock g_maOpenSLSpinlock = 0; /* For init/uninit. */
- framesWrittenToPlaybackDevice += framesAvailablePlayback;
- }
+#define MA_OPENSL_OBJ(p) (*((SLObjectItf)(p)))
+#define MA_OPENSL_OUTPUTMIX(p) (*((SLOutputMixItf)(p)))
+#define MA_OPENSL_PLAY(p) (*((SLPlayItf)(p)))
+#define MA_OPENSL_RECORD(p) (*((SLRecordItf)(p)))
- if (!c89atomic_load_32(&pDevice->wasapi.isStartedPlayback)) {
- hr = ma_IAudioClient_Start((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback);
- if (FAILED(hr)) {
- ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to start internal playback device.", ma_result_from_HRESULT(hr));
- exitLoop = MA_TRUE;
- break;
- }
+#ifdef MA_ANDROID
+#define MA_OPENSL_BUFFERQUEUE(p) (*((SLAndroidSimpleBufferQueueItf)(p)))
+#else
+#define MA_OPENSL_BUFFERQUEUE(p) (*((SLBufferQueueItf)(p)))
+#endif
- c89atomic_exchange_32(&pDevice->wasapi.isStartedPlayback, MA_TRUE);
- }
+static ma_result ma_result_from_OpenSL(SLuint32 result)
+{
+ switch (result)
+ {
+ case SL_RESULT_SUCCESS: return MA_SUCCESS;
+ case SL_RESULT_PRECONDITIONS_VIOLATED: return MA_ERROR;
+ case SL_RESULT_PARAMETER_INVALID: return MA_INVALID_ARGS;
+ case SL_RESULT_MEMORY_FAILURE: return MA_OUT_OF_MEMORY;
+ case SL_RESULT_RESOURCE_ERROR: return MA_INVALID_DATA;
+ case SL_RESULT_RESOURCE_LOST: return MA_ERROR;
+ case SL_RESULT_IO_ERROR: return MA_IO_ERROR;
+ case SL_RESULT_BUFFER_INSUFFICIENT: return MA_NO_SPACE;
+ case SL_RESULT_CONTENT_CORRUPTED: return MA_INVALID_DATA;
+ case SL_RESULT_CONTENT_UNSUPPORTED: return MA_FORMAT_NOT_SUPPORTED;
+ case SL_RESULT_CONTENT_NOT_FOUND: return MA_ERROR;
+ case SL_RESULT_PERMISSION_DENIED: return MA_ACCESS_DENIED;
+ case SL_RESULT_FEATURE_UNSUPPORTED: return MA_NOT_IMPLEMENTED;
+ case SL_RESULT_INTERNAL_ERROR: return MA_ERROR;
+ case SL_RESULT_UNKNOWN_ERROR: return MA_ERROR;
+ case SL_RESULT_OPERATION_ABORTED: return MA_ERROR;
+ case SL_RESULT_CONTROL_LOST: return MA_ERROR;
+ default: return MA_ERROR;
+ }
+}
- /* Make sure we don't wait on the event before we've started the device or we may end up deadlocking. */
- if (WaitForSingleObject(pDevice->wasapi.hEventPlayback, MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS) != WAIT_OBJECT_0) {
- exitLoop = MA_TRUE;
- break; /* Wait failed. Probably timed out. */
- }
- } break;
+/* Converts an individual OpenSL-style channel identifier (SL_SPEAKER_FRONT_LEFT, etc.) to miniaudio. */
+static ma_uint8 ma_channel_id_to_ma__opensl(SLuint32 id)
+{
+ switch (id)
+ {
+ case SL_SPEAKER_FRONT_LEFT: return MA_CHANNEL_FRONT_LEFT;
+ case SL_SPEAKER_FRONT_RIGHT: return MA_CHANNEL_FRONT_RIGHT;
+ case SL_SPEAKER_FRONT_CENTER: return MA_CHANNEL_FRONT_CENTER;
+ case SL_SPEAKER_LOW_FREQUENCY: return MA_CHANNEL_LFE;
+ case SL_SPEAKER_BACK_LEFT: return MA_CHANNEL_BACK_LEFT;
+ case SL_SPEAKER_BACK_RIGHT: return MA_CHANNEL_BACK_RIGHT;
+ case SL_SPEAKER_FRONT_LEFT_OF_CENTER: return MA_CHANNEL_FRONT_LEFT_CENTER;
+ case SL_SPEAKER_FRONT_RIGHT_OF_CENTER: return MA_CHANNEL_FRONT_RIGHT_CENTER;
+ case SL_SPEAKER_BACK_CENTER: return MA_CHANNEL_BACK_CENTER;
+ case SL_SPEAKER_SIDE_LEFT: return MA_CHANNEL_SIDE_LEFT;
+ case SL_SPEAKER_SIDE_RIGHT: return MA_CHANNEL_SIDE_RIGHT;
+ case SL_SPEAKER_TOP_CENTER: return MA_CHANNEL_TOP_CENTER;
+ case SL_SPEAKER_TOP_FRONT_LEFT: return MA_CHANNEL_TOP_FRONT_LEFT;
+ case SL_SPEAKER_TOP_FRONT_CENTER: return MA_CHANNEL_TOP_FRONT_CENTER;
+ case SL_SPEAKER_TOP_FRONT_RIGHT: return MA_CHANNEL_TOP_FRONT_RIGHT;
+ case SL_SPEAKER_TOP_BACK_LEFT: return MA_CHANNEL_TOP_BACK_LEFT;
+ case SL_SPEAKER_TOP_BACK_CENTER: return MA_CHANNEL_TOP_BACK_CENTER;
+ case SL_SPEAKER_TOP_BACK_RIGHT: return MA_CHANNEL_TOP_BACK_RIGHT;
+ default: return 0;
+ }
+}
- default: return MA_INVALID_ARGS;
- }
+/* Converts an individual miniaudio channel identifier (MA_CHANNEL_FRONT_LEFT, etc.) to OpenSL-style. */
+static SLuint32 ma_channel_id_to_opensl(ma_uint8 id)
+{
+ switch (id)
+ {
+ case MA_CHANNEL_MONO: return SL_SPEAKER_FRONT_CENTER;
+ case MA_CHANNEL_FRONT_LEFT: return SL_SPEAKER_FRONT_LEFT;
+ case MA_CHANNEL_FRONT_RIGHT: return SL_SPEAKER_FRONT_RIGHT;
+ case MA_CHANNEL_FRONT_CENTER: return SL_SPEAKER_FRONT_CENTER;
+ case MA_CHANNEL_LFE: return SL_SPEAKER_LOW_FREQUENCY;
+ case MA_CHANNEL_BACK_LEFT: return SL_SPEAKER_BACK_LEFT;
+ case MA_CHANNEL_BACK_RIGHT: return SL_SPEAKER_BACK_RIGHT;
+ case MA_CHANNEL_FRONT_LEFT_CENTER: return SL_SPEAKER_FRONT_LEFT_OF_CENTER;
+ case MA_CHANNEL_FRONT_RIGHT_CENTER: return SL_SPEAKER_FRONT_RIGHT_OF_CENTER;
+ case MA_CHANNEL_BACK_CENTER: return SL_SPEAKER_BACK_CENTER;
+ case MA_CHANNEL_SIDE_LEFT: return SL_SPEAKER_SIDE_LEFT;
+ case MA_CHANNEL_SIDE_RIGHT: return SL_SPEAKER_SIDE_RIGHT;
+ case MA_CHANNEL_TOP_CENTER: return SL_SPEAKER_TOP_CENTER;
+ case MA_CHANNEL_TOP_FRONT_LEFT: return SL_SPEAKER_TOP_FRONT_LEFT;
+ case MA_CHANNEL_TOP_FRONT_CENTER: return SL_SPEAKER_TOP_FRONT_CENTER;
+ case MA_CHANNEL_TOP_FRONT_RIGHT: return SL_SPEAKER_TOP_FRONT_RIGHT;
+ case MA_CHANNEL_TOP_BACK_LEFT: return SL_SPEAKER_TOP_BACK_LEFT;
+ case MA_CHANNEL_TOP_BACK_CENTER: return SL_SPEAKER_TOP_BACK_CENTER;
+ case MA_CHANNEL_TOP_BACK_RIGHT: return SL_SPEAKER_TOP_BACK_RIGHT;
+ default: return 0;
}
+}
- /* Here is where the device needs to be stopped. */
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) {
- /* Any mapped buffers need to be released. */
- if (pMappedDeviceBufferCapture != NULL) {
- hr = ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, mappedDeviceBufferSizeInFramesCapture);
- }
+/* Converts a channel mapping to an OpenSL-style channel mask. */
+static SLuint32 ma_channel_map_to_channel_mask__opensl(const ma_channel* pChannelMap, ma_uint32 channels)
+{
+ SLuint32 channelMask = 0;
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < channels; ++iChannel) {
+ channelMask |= ma_channel_id_to_opensl(pChannelMap[iChannel]);
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- /* Any mapped buffers need to be released. */
- if (pMappedDeviceBufferPlayback != NULL) {
- hr = ma_IAudioRenderClient_ReleaseBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, mappedDeviceBufferSizeInFramesPlayback, 0);
+ return channelMask;
+}
+
+/* Converts an OpenSL-style channel mask to a miniaudio channel map. */
+static void ma_channel_mask_to_channel_map__opensl(SLuint32 channelMask, ma_uint32 channels, ma_channel* pChannelMap)
+{
+ if (channels == 1 && channelMask == 0) {
+ pChannelMap[0] = MA_CHANNEL_MONO;
+ } else if (channels == 2 && channelMask == 0) {
+ pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
+ pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
+ } else {
+ if (channels == 1 && (channelMask & SL_SPEAKER_FRONT_CENTER) != 0) {
+ pChannelMap[0] = MA_CHANNEL_MONO;
+ } else {
+ /* Just iterate over each bit. */
+ ma_uint32 iChannel = 0;
+ ma_uint32 iBit;
+ for (iBit = 0; iBit < 32 && iChannel < channels; ++iBit) {
+ SLuint32 bitValue = (channelMask & (1UL << iBit));
+ if (bitValue != 0) {
+ /* The bit is set. */
+ pChannelMap[iChannel] = ma_channel_id_to_ma__opensl(bitValue);
+ iChannel += 1;
+ }
+ }
}
}
-
- return MA_SUCCESS;
}
-static ma_result ma_device_data_loop_wakeup__wasapi(ma_device* pDevice)
+static SLuint32 ma_round_to_standard_sample_rate__opensl(SLuint32 samplesPerSec)
{
- MA_ASSERT(pDevice != NULL);
-
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) {
- SetEvent((HANDLE)pDevice->wasapi.hEventCapture);
+ if (samplesPerSec <= SL_SAMPLINGRATE_8) {
+ return SL_SAMPLINGRATE_8;
+ }
+ if (samplesPerSec <= SL_SAMPLINGRATE_11_025) {
+ return SL_SAMPLINGRATE_11_025;
+ }
+ if (samplesPerSec <= SL_SAMPLINGRATE_12) {
+ return SL_SAMPLINGRATE_12;
+ }
+ if (samplesPerSec <= SL_SAMPLINGRATE_16) {
+ return SL_SAMPLINGRATE_16;
+ }
+ if (samplesPerSec <= SL_SAMPLINGRATE_22_05) {
+ return SL_SAMPLINGRATE_22_05;
+ }
+ if (samplesPerSec <= SL_SAMPLINGRATE_24) {
+ return SL_SAMPLINGRATE_24;
+ }
+ if (samplesPerSec <= SL_SAMPLINGRATE_32) {
+ return SL_SAMPLINGRATE_32;
+ }
+ if (samplesPerSec <= SL_SAMPLINGRATE_44_1) {
+ return SL_SAMPLINGRATE_44_1;
+ }
+ if (samplesPerSec <= SL_SAMPLINGRATE_48) {
+ return SL_SAMPLINGRATE_48;
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- SetEvent((HANDLE)pDevice->wasapi.hEventPlayback);
+ /* Android doesn't support more than 48000. */
+#ifndef MA_ANDROID
+ if (samplesPerSec <= SL_SAMPLINGRATE_64) {
+ return SL_SAMPLINGRATE_64;
+ }
+ if (samplesPerSec <= SL_SAMPLINGRATE_88_2) {
+ return SL_SAMPLINGRATE_88_2;
+ }
+ if (samplesPerSec <= SL_SAMPLINGRATE_96) {
+ return SL_SAMPLINGRATE_96;
}
+ if (samplesPerSec <= SL_SAMPLINGRATE_192) {
+ return SL_SAMPLINGRATE_192;
+ }
+#endif
- return MA_SUCCESS;
+ return SL_SAMPLINGRATE_16;
}
-static ma_result ma_context_uninit__wasapi(ma_context* pContext)
+static SLint32 ma_to_stream_type__opensl(ma_opensl_stream_type streamType)
{
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pContext->backend == ma_backend_wasapi);
+ switch (streamType) {
+ case ma_opensl_stream_type_voice: return SL_ANDROID_STREAM_VOICE;
+ case ma_opensl_stream_type_system: return SL_ANDROID_STREAM_SYSTEM;
+ case ma_opensl_stream_type_ring: return SL_ANDROID_STREAM_RING;
+ case ma_opensl_stream_type_media: return SL_ANDROID_STREAM_MEDIA;
+ case ma_opensl_stream_type_alarm: return SL_ANDROID_STREAM_ALARM;
+ case ma_opensl_stream_type_notification: return SL_ANDROID_STREAM_NOTIFICATION;
+ default: break;
+ }
- if (pContext->wasapi.commandThread != NULL) {
- ma_context_command__wasapi cmd = ma_context_init_command__wasapi(MA_CONTEXT_COMMAND_QUIT__WASAPI);
- ma_context_post_command__wasapi(pContext, &cmd);
- ma_thread_wait(&pContext->wasapi.commandThread);
+ return SL_ANDROID_STREAM_VOICE;
+}
- /* Only after the thread has been terminated can we uninitialize the sync objects for the command thread. */
- ma_semaphore_uninit(&pContext->wasapi.commandSem);
- ma_mutex_uninit(&pContext->wasapi.commandLock);
+static SLint32 ma_to_recording_preset__opensl(ma_opensl_recording_preset recordingPreset)
+{
+ switch (recordingPreset) {
+ case ma_opensl_recording_preset_generic: return SL_ANDROID_RECORDING_PRESET_GENERIC;
+ case ma_opensl_recording_preset_camcorder: return SL_ANDROID_RECORDING_PRESET_CAMCORDER;
+ case ma_opensl_recording_preset_voice_recognition: return SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION;
+ case ma_opensl_recording_preset_voice_communication: return SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION;
+ case ma_opensl_recording_preset_voice_unprocessed: return SL_ANDROID_RECORDING_PRESET_UNPROCESSED;
+ default: break;
}
- return MA_SUCCESS;
+ return SL_ANDROID_RECORDING_PRESET_NONE;
}
-static ma_result ma_context_init__wasapi(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+
+static ma_result ma_context_enumerate_devices__opensl(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
{
- ma_result result = MA_SUCCESS;
+ ma_bool32 cbResult;
MA_ASSERT(pContext != NULL);
+ MA_ASSERT(callback != NULL);
- (void)pConfig;
+ MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to enumerate devices. */
+ if (g_maOpenSLInitCounter == 0) {
+ return MA_INVALID_OPERATION;
+ }
-#ifdef MA_WIN32_DESKTOP
/*
- WASAPI is only supported in Vista SP1 and newer. The reason for SP1 and not the base version of Vista is that event-driven
- exclusive mode does not work until SP1.
+ TODO: Test Me.
- Unfortunately older compilers don't define these functions so we need to dynamically load them in order to avoid a link error.
+ This is currently untested, so for now we are just returning default devices.
*/
- {
- ma_OSVERSIONINFOEXW osvi;
- ma_handle kernel32DLL;
- ma_PFNVerifyVersionInfoW _VerifyVersionInfoW;
- ma_PFNVerSetConditionMask _VerSetConditionMask;
-
- kernel32DLL = ma_dlopen(pContext, "kernel32.dll");
- if (kernel32DLL == NULL) {
- return MA_NO_BACKEND;
- }
-
- _VerifyVersionInfoW = (ma_PFNVerifyVersionInfoW )ma_dlsym(pContext, kernel32DLL, "VerifyVersionInfoW");
- _VerSetConditionMask = (ma_PFNVerSetConditionMask)ma_dlsym(pContext, kernel32DLL, "VerSetConditionMask");
- if (_VerifyVersionInfoW == NULL || _VerSetConditionMask == NULL) {
- ma_dlclose(pContext, kernel32DLL);
- return MA_NO_BACKEND;
- }
-
- MA_ZERO_OBJECT(&osvi);
- osvi.dwOSVersionInfoSize = sizeof(osvi);
- osvi.dwMajorVersion = ((MA_WIN32_WINNT_VISTA >> 8) & 0xFF);
- osvi.dwMinorVersion = ((MA_WIN32_WINNT_VISTA >> 0) & 0xFF);
- osvi.wServicePackMajor = 1;
- if (_VerifyVersionInfoW(&osvi, MA_VER_MAJORVERSION | MA_VER_MINORVERSION | MA_VER_SERVICEPACKMAJOR, _VerSetConditionMask(_VerSetConditionMask(_VerSetConditionMask(0, MA_VER_MAJORVERSION, MA_VER_GREATER_EQUAL), MA_VER_MINORVERSION, MA_VER_GREATER_EQUAL), MA_VER_SERVICEPACKMAJOR, MA_VER_GREATER_EQUAL))) {
- result = MA_SUCCESS;
- } else {
- result = MA_NO_BACKEND;
- }
+#if 0 && !defined(MA_ANDROID)
+ ma_bool32 isTerminated = MA_FALSE;
- ma_dlclose(pContext, kernel32DLL);
- }
-#endif
+ SLuint32 pDeviceIDs[128];
+ SLint32 deviceCount = sizeof(pDeviceIDs) / sizeof(pDeviceIDs[0]);
- if (result != MA_SUCCESS) {
- return result;
+ SLAudioIODeviceCapabilitiesItf deviceCaps;
+ SLresult resultSL = (*g_maEngineObjectSL)->GetInterface(g_maEngineObjectSL, (SLInterfaceID)pContext->opensl.SL_IID_AUDIOIODEVICECAPABILITIES, &deviceCaps);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ /* The interface may not be supported so just report a default device. */
+ goto return_default_device;
}
- MA_ZERO_OBJECT(&pContext->wasapi);
-
- /*
- Annoyingly, WASAPI does not allow you to release an IAudioClient object from a different thread
- than the one that retrieved it with GetService(). This can result in a deadlock in two
- situations:
-
- 1) When calling ma_device_uninit() from a different thread to ma_device_init(); and
- 2) When uninitializing and reinitializing the internal IAudioClient object in response to
- automatic stream routing.
-
- We could define ma_device_uninit() such that it must be called on the same thread as
- ma_device_init(). We could also just not release the IAudioClient when performing automatic
- stream routing to avoid the deadlock. Neither of these are acceptable solutions in my view so
- we're going to have to work around this with a worker thread. This is not ideal, but I can't
- think of a better way to do this.
-
- More information about this can be found here:
-
- https://docs.microsoft.com/en-us/windows/win32/api/audioclient/nn-audioclient-iaudiorenderclient
-
- Note this section:
-
- When releasing an IAudioRenderClient interface instance, the client must call the interface's
- Release method from the same thread as the call to IAudioClient::GetService that created the
- object.
- */
- {
- result = ma_mutex_init(&pContext->wasapi.commandLock);
- if (result != MA_SUCCESS) {
- return result;
+ /* Playback */
+ if (!isTerminated) {
+ resultSL = (*deviceCaps)->GetAvailableAudioOutputs(deviceCaps, &deviceCount, pDeviceIDs);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ return ma_result_from_OpenSL(resultSL);
}
- result = ma_semaphore_init(0, &pContext->wasapi.commandSem);
- if (result != MA_SUCCESS) {
- ma_mutex_uninit(&pContext->wasapi.commandLock);
- return result;
- }
+ for (SLint32 iDevice = 0; iDevice < deviceCount; ++iDevice) {
+ ma_device_info deviceInfo;
+ MA_ZERO_OBJECT(&deviceInfo);
+ deviceInfo.id.opensl = pDeviceIDs[iDevice];
- result = ma_thread_create(&pContext->wasapi.commandThread, ma_thread_priority_normal, 0, ma_context_command_thread__wasapi, pContext, &pContext->allocationCallbacks);
- if (result != MA_SUCCESS) {
- ma_semaphore_uninit(&pContext->wasapi.commandSem);
- ma_mutex_uninit(&pContext->wasapi.commandLock);
- return result;
+ SLAudioOutputDescriptor desc;
+ resultSL = (*deviceCaps)->QueryAudioOutputCapabilities(deviceCaps, deviceInfo.id.opensl, &desc);
+ if (resultSL == SL_RESULT_SUCCESS) {
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), (const char*)desc.pDeviceName, (size_t)-1);
+
+ ma_bool32 cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
+ if (cbResult == MA_FALSE) {
+ isTerminated = MA_TRUE;
+ break;
+ }
+ }
}
}
+ /* Capture */
+ if (!isTerminated) {
+ resultSL = (*deviceCaps)->GetAvailableAudioInputs(deviceCaps, &deviceCount, pDeviceIDs);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ return ma_result_from_OpenSL(resultSL);
+ }
- pCallbacks->onContextInit = ma_context_init__wasapi;
- pCallbacks->onContextUninit = ma_context_uninit__wasapi;
- pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__wasapi;
- pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__wasapi;
- pCallbacks->onDeviceInit = ma_device_init__wasapi;
- pCallbacks->onDeviceUninit = ma_device_uninit__wasapi;
- pCallbacks->onDeviceStart = ma_device_start__wasapi;
- pCallbacks->onDeviceStop = ma_device_stop__wasapi;
- pCallbacks->onDeviceRead = NULL; /* Not used. Reading is done manually in the audio thread. */
- pCallbacks->onDeviceWrite = NULL; /* Not used. Writing is done manually in the audio thread. */
- pCallbacks->onDeviceDataLoop = ma_device_data_loop__wasapi;
- pCallbacks->onDeviceDataLoopWakeup = ma_device_data_loop_wakeup__wasapi;
-
- return MA_SUCCESS;
-}
-#endif
-
-/******************************************************************************
-
-DirectSound Backend
-
-******************************************************************************/
-#ifdef MA_HAS_DSOUND
-/*#include */
-
-/*static const GUID MA_GUID_IID_DirectSoundNotify = {0xb0210783, 0x89cd, 0x11d0, {0xaf, 0x08, 0x00, 0xa0, 0xc9, 0x25, 0xcd, 0x16}};*/
-
-/* miniaudio only uses priority or exclusive modes. */
-#define MA_DSSCL_NORMAL 1
-#define MA_DSSCL_PRIORITY 2
-#define MA_DSSCL_EXCLUSIVE 3
-#define MA_DSSCL_WRITEPRIMARY 4
-
-#define MA_DSCAPS_PRIMARYMONO 0x00000001
-#define MA_DSCAPS_PRIMARYSTEREO 0x00000002
-#define MA_DSCAPS_PRIMARY8BIT 0x00000004
-#define MA_DSCAPS_PRIMARY16BIT 0x00000008
-#define MA_DSCAPS_CONTINUOUSRATE 0x00000010
-#define MA_DSCAPS_EMULDRIVER 0x00000020
-#define MA_DSCAPS_CERTIFIED 0x00000040
-#define MA_DSCAPS_SECONDARYMONO 0x00000100
-#define MA_DSCAPS_SECONDARYSTEREO 0x00000200
-#define MA_DSCAPS_SECONDARY8BIT 0x00000400
-#define MA_DSCAPS_SECONDARY16BIT 0x00000800
-
-#define MA_DSBCAPS_PRIMARYBUFFER 0x00000001
-#define MA_DSBCAPS_STATIC 0x00000002
-#define MA_DSBCAPS_LOCHARDWARE 0x00000004
-#define MA_DSBCAPS_LOCSOFTWARE 0x00000008
-#define MA_DSBCAPS_CTRL3D 0x00000010
-#define MA_DSBCAPS_CTRLFREQUENCY 0x00000020
-#define MA_DSBCAPS_CTRLPAN 0x00000040
-#define MA_DSBCAPS_CTRLVOLUME 0x00000080
-#define MA_DSBCAPS_CTRLPOSITIONNOTIFY 0x00000100
-#define MA_DSBCAPS_CTRLFX 0x00000200
-#define MA_DSBCAPS_STICKYFOCUS 0x00004000
-#define MA_DSBCAPS_GLOBALFOCUS 0x00008000
-#define MA_DSBCAPS_GETCURRENTPOSITION2 0x00010000
-#define MA_DSBCAPS_MUTE3DATMAXDISTANCE 0x00020000
-#define MA_DSBCAPS_LOCDEFER 0x00040000
-#define MA_DSBCAPS_TRUEPLAYPOSITION 0x00080000
-
-#define MA_DSBPLAY_LOOPING 0x00000001
-#define MA_DSBPLAY_LOCHARDWARE 0x00000002
-#define MA_DSBPLAY_LOCSOFTWARE 0x00000004
-#define MA_DSBPLAY_TERMINATEBY_TIME 0x00000008
-#define MA_DSBPLAY_TERMINATEBY_DISTANCE 0x00000010
-#define MA_DSBPLAY_TERMINATEBY_PRIORITY 0x00000020
-
-#define MA_DSCBSTART_LOOPING 0x00000001
-
-typedef struct
-{
- DWORD dwSize;
- DWORD dwFlags;
- DWORD dwBufferBytes;
- DWORD dwReserved;
- WAVEFORMATEX* lpwfxFormat;
- GUID guid3DAlgorithm;
-} MA_DSBUFFERDESC;
-
-typedef struct
-{
- DWORD dwSize;
- DWORD dwFlags;
- DWORD dwBufferBytes;
- DWORD dwReserved;
- WAVEFORMATEX* lpwfxFormat;
- DWORD dwFXCount;
- void* lpDSCFXDesc; /* <-- miniaudio doesn't use this, so set to void*. */
-} MA_DSCBUFFERDESC;
-
-typedef struct
-{
- DWORD dwSize;
- DWORD dwFlags;
- DWORD dwMinSecondarySampleRate;
- DWORD dwMaxSecondarySampleRate;
- DWORD dwPrimaryBuffers;
- DWORD dwMaxHwMixingAllBuffers;
- DWORD dwMaxHwMixingStaticBuffers;
- DWORD dwMaxHwMixingStreamingBuffers;
- DWORD dwFreeHwMixingAllBuffers;
- DWORD dwFreeHwMixingStaticBuffers;
- DWORD dwFreeHwMixingStreamingBuffers;
- DWORD dwMaxHw3DAllBuffers;
- DWORD dwMaxHw3DStaticBuffers;
- DWORD dwMaxHw3DStreamingBuffers;
- DWORD dwFreeHw3DAllBuffers;
- DWORD dwFreeHw3DStaticBuffers;
- DWORD dwFreeHw3DStreamingBuffers;
- DWORD dwTotalHwMemBytes;
- DWORD dwFreeHwMemBytes;
- DWORD dwMaxContigFreeHwMemBytes;
- DWORD dwUnlockTransferRateHwBuffers;
- DWORD dwPlayCpuOverheadSwBuffers;
- DWORD dwReserved1;
- DWORD dwReserved2;
-} MA_DSCAPS;
-
-typedef struct
-{
- DWORD dwSize;
- DWORD dwFlags;
- DWORD dwBufferBytes;
- DWORD dwUnlockTransferRate;
- DWORD dwPlayCpuOverhead;
-} MA_DSBCAPS;
-
-typedef struct
-{
- DWORD dwSize;
- DWORD dwFlags;
- DWORD dwFormats;
- DWORD dwChannels;
-} MA_DSCCAPS;
-
-typedef struct
-{
- DWORD dwSize;
- DWORD dwFlags;
- DWORD dwBufferBytes;
- DWORD dwReserved;
-} MA_DSCBCAPS;
-
-typedef struct
-{
- DWORD dwOffset;
- HANDLE hEventNotify;
-} MA_DSBPOSITIONNOTIFY;
-
-typedef struct ma_IDirectSound ma_IDirectSound;
-typedef struct ma_IDirectSoundBuffer ma_IDirectSoundBuffer;
-typedef struct ma_IDirectSoundCapture ma_IDirectSoundCapture;
-typedef struct ma_IDirectSoundCaptureBuffer ma_IDirectSoundCaptureBuffer;
-typedef struct ma_IDirectSoundNotify ma_IDirectSoundNotify;
-
-
-/*
-COM objects. The way these work is that you have a vtable (a list of function pointers, kind of
-like how C++ works internally), and then you have a structure with a single member, which is a
-pointer to the vtable. The vtable is where the methods of the object are defined. Methods need
-to be in a specific order, and parent classes need to have their methods declared first.
-*/
-
-/* IDirectSound */
-typedef struct
-{
- /* IUnknown */
- HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSound* pThis, const IID* const riid, void** ppObject);
- ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSound* pThis);
- ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSound* pThis);
-
- /* IDirectSound */
- HRESULT (STDMETHODCALLTYPE * CreateSoundBuffer) (ma_IDirectSound* pThis, const MA_DSBUFFERDESC* pDSBufferDesc, ma_IDirectSoundBuffer** ppDSBuffer, void* pUnkOuter);
- HRESULT (STDMETHODCALLTYPE * GetCaps) (ma_IDirectSound* pThis, MA_DSCAPS* pDSCaps);
- HRESULT (STDMETHODCALLTYPE * DuplicateSoundBuffer)(ma_IDirectSound* pThis, ma_IDirectSoundBuffer* pDSBufferOriginal, ma_IDirectSoundBuffer** ppDSBufferDuplicate);
- HRESULT (STDMETHODCALLTYPE * SetCooperativeLevel) (ma_IDirectSound* pThis, HWND hwnd, DWORD dwLevel);
- HRESULT (STDMETHODCALLTYPE * Compact) (ma_IDirectSound* pThis);
- HRESULT (STDMETHODCALLTYPE * GetSpeakerConfig) (ma_IDirectSound* pThis, DWORD* pSpeakerConfig);
- HRESULT (STDMETHODCALLTYPE * SetSpeakerConfig) (ma_IDirectSound* pThis, DWORD dwSpeakerConfig);
- HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IDirectSound* pThis, const GUID* pGuidDevice);
-} ma_IDirectSoundVtbl;
-struct ma_IDirectSound
-{
- ma_IDirectSoundVtbl* lpVtbl;
-};
-static MA_INLINE HRESULT ma_IDirectSound_QueryInterface(ma_IDirectSound* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
-static MA_INLINE ULONG ma_IDirectSound_AddRef(ma_IDirectSound* pThis) { return pThis->lpVtbl->AddRef(pThis); }
-static MA_INLINE ULONG ma_IDirectSound_Release(ma_IDirectSound* pThis) { return pThis->lpVtbl->Release(pThis); }
-static MA_INLINE HRESULT ma_IDirectSound_CreateSoundBuffer(ma_IDirectSound* pThis, const MA_DSBUFFERDESC* pDSBufferDesc, ma_IDirectSoundBuffer** ppDSBuffer, void* pUnkOuter) { return pThis->lpVtbl->CreateSoundBuffer(pThis, pDSBufferDesc, ppDSBuffer, pUnkOuter); }
-static MA_INLINE HRESULT ma_IDirectSound_GetCaps(ma_IDirectSound* pThis, MA_DSCAPS* pDSCaps) { return pThis->lpVtbl->GetCaps(pThis, pDSCaps); }
-static MA_INLINE HRESULT ma_IDirectSound_DuplicateSoundBuffer(ma_IDirectSound* pThis, ma_IDirectSoundBuffer* pDSBufferOriginal, ma_IDirectSoundBuffer** ppDSBufferDuplicate) { return pThis->lpVtbl->DuplicateSoundBuffer(pThis, pDSBufferOriginal, ppDSBufferDuplicate); }
-static MA_INLINE HRESULT ma_IDirectSound_SetCooperativeLevel(ma_IDirectSound* pThis, HWND hwnd, DWORD dwLevel) { return pThis->lpVtbl->SetCooperativeLevel(pThis, hwnd, dwLevel); }
-static MA_INLINE HRESULT ma_IDirectSound_Compact(ma_IDirectSound* pThis) { return pThis->lpVtbl->Compact(pThis); }
-static MA_INLINE HRESULT ma_IDirectSound_GetSpeakerConfig(ma_IDirectSound* pThis, DWORD* pSpeakerConfig) { return pThis->lpVtbl->GetSpeakerConfig(pThis, pSpeakerConfig); }
-static MA_INLINE HRESULT ma_IDirectSound_SetSpeakerConfig(ma_IDirectSound* pThis, DWORD dwSpeakerConfig) { return pThis->lpVtbl->SetSpeakerConfig(pThis, dwSpeakerConfig); }
-static MA_INLINE HRESULT ma_IDirectSound_Initialize(ma_IDirectSound* pThis, const GUID* pGuidDevice) { return pThis->lpVtbl->Initialize(pThis, pGuidDevice); }
-
+ for (SLint32 iDevice = 0; iDevice < deviceCount; ++iDevice) {
+ ma_device_info deviceInfo;
+ MA_ZERO_OBJECT(&deviceInfo);
+ deviceInfo.id.opensl = pDeviceIDs[iDevice];
-/* IDirectSoundBuffer */
-typedef struct
-{
- /* IUnknown */
- HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSoundBuffer* pThis, const IID* const riid, void** ppObject);
- ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSoundBuffer* pThis);
- ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSoundBuffer* pThis);
+ SLAudioInputDescriptor desc;
+ resultSL = (*deviceCaps)->QueryAudioInputCapabilities(deviceCaps, deviceInfo.id.opensl, &desc);
+ if (resultSL == SL_RESULT_SUCCESS) {
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), (const char*)desc.deviceName, (size_t)-1);
- /* IDirectSoundBuffer */
- HRESULT (STDMETHODCALLTYPE * GetCaps) (ma_IDirectSoundBuffer* pThis, MA_DSBCAPS* pDSBufferCaps);
- HRESULT (STDMETHODCALLTYPE * GetCurrentPosition)(ma_IDirectSoundBuffer* pThis, DWORD* pCurrentPlayCursor, DWORD* pCurrentWriteCursor);
- HRESULT (STDMETHODCALLTYPE * GetFormat) (ma_IDirectSoundBuffer* pThis, WAVEFORMATEX* pFormat, DWORD dwSizeAllocated, DWORD* pSizeWritten);
- HRESULT (STDMETHODCALLTYPE * GetVolume) (ma_IDirectSoundBuffer* pThis, LONG* pVolume);
- HRESULT (STDMETHODCALLTYPE * GetPan) (ma_IDirectSoundBuffer* pThis, LONG* pPan);
- HRESULT (STDMETHODCALLTYPE * GetFrequency) (ma_IDirectSoundBuffer* pThis, DWORD* pFrequency);
- HRESULT (STDMETHODCALLTYPE * GetStatus) (ma_IDirectSoundBuffer* pThis, DWORD* pStatus);
- HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IDirectSoundBuffer* pThis, ma_IDirectSound* pDirectSound, const MA_DSBUFFERDESC* pDSBufferDesc);
- HRESULT (STDMETHODCALLTYPE * Lock) (ma_IDirectSoundBuffer* pThis, DWORD dwOffset, DWORD dwBytes, void** ppAudioPtr1, DWORD* pAudioBytes1, void** ppAudioPtr2, DWORD* pAudioBytes2, DWORD dwFlags);
- HRESULT (STDMETHODCALLTYPE * Play) (ma_IDirectSoundBuffer* pThis, DWORD dwReserved1, DWORD dwPriority, DWORD dwFlags);
- HRESULT (STDMETHODCALLTYPE * SetCurrentPosition)(ma_IDirectSoundBuffer* pThis, DWORD dwNewPosition);
- HRESULT (STDMETHODCALLTYPE * SetFormat) (ma_IDirectSoundBuffer* pThis, const WAVEFORMATEX* pFormat);
- HRESULT (STDMETHODCALLTYPE * SetVolume) (ma_IDirectSoundBuffer* pThis, LONG volume);
- HRESULT (STDMETHODCALLTYPE * SetPan) (ma_IDirectSoundBuffer* pThis, LONG pan);
- HRESULT (STDMETHODCALLTYPE * SetFrequency) (ma_IDirectSoundBuffer* pThis, DWORD dwFrequency);
- HRESULT (STDMETHODCALLTYPE * Stop) (ma_IDirectSoundBuffer* pThis);
- HRESULT (STDMETHODCALLTYPE * Unlock) (ma_IDirectSoundBuffer* pThis, void* pAudioPtr1, DWORD dwAudioBytes1, void* pAudioPtr2, DWORD dwAudioBytes2);
- HRESULT (STDMETHODCALLTYPE * Restore) (ma_IDirectSoundBuffer* pThis);
-} ma_IDirectSoundBufferVtbl;
-struct ma_IDirectSoundBuffer
-{
- ma_IDirectSoundBufferVtbl* lpVtbl;
-};
-static MA_INLINE HRESULT ma_IDirectSoundBuffer_QueryInterface(ma_IDirectSoundBuffer* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
-static MA_INLINE ULONG ma_IDirectSoundBuffer_AddRef(ma_IDirectSoundBuffer* pThis) { return pThis->lpVtbl->AddRef(pThis); }
-static MA_INLINE ULONG ma_IDirectSoundBuffer_Release(ma_IDirectSoundBuffer* pThis) { return pThis->lpVtbl->Release(pThis); }
-static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetCaps(ma_IDirectSoundBuffer* pThis, MA_DSBCAPS* pDSBufferCaps) { return pThis->lpVtbl->GetCaps(pThis, pDSBufferCaps); }
-static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetCurrentPosition(ma_IDirectSoundBuffer* pThis, DWORD* pCurrentPlayCursor, DWORD* pCurrentWriteCursor) { return pThis->lpVtbl->GetCurrentPosition(pThis, pCurrentPlayCursor, pCurrentWriteCursor); }
-static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetFormat(ma_IDirectSoundBuffer* pThis, WAVEFORMATEX* pFormat, DWORD dwSizeAllocated, DWORD* pSizeWritten) { return pThis->lpVtbl->GetFormat(pThis, pFormat, dwSizeAllocated, pSizeWritten); }
-static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetVolume(ma_IDirectSoundBuffer* pThis, LONG* pVolume) { return pThis->lpVtbl->GetVolume(pThis, pVolume); }
-static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetPan(ma_IDirectSoundBuffer* pThis, LONG* pPan) { return pThis->lpVtbl->GetPan(pThis, pPan); }
-static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetFrequency(ma_IDirectSoundBuffer* pThis, DWORD* pFrequency) { return pThis->lpVtbl->GetFrequency(pThis, pFrequency); }
-static MA_INLINE HRESULT ma_IDirectSoundBuffer_GetStatus(ma_IDirectSoundBuffer* pThis, DWORD* pStatus) { return pThis->lpVtbl->GetStatus(pThis, pStatus); }
-static MA_INLINE HRESULT ma_IDirectSoundBuffer_Initialize(ma_IDirectSoundBuffer* pThis, ma_IDirectSound* pDirectSound, const MA_DSBUFFERDESC* pDSBufferDesc) { return pThis->lpVtbl->Initialize(pThis, pDirectSound, pDSBufferDesc); }
-static MA_INLINE HRESULT ma_IDirectSoundBuffer_Lock(ma_IDirectSoundBuffer* pThis, DWORD dwOffset, DWORD dwBytes, void** ppAudioPtr1, DWORD* pAudioBytes1, void** ppAudioPtr2, DWORD* pAudioBytes2, DWORD dwFlags) { return pThis->lpVtbl->Lock(pThis, dwOffset, dwBytes, ppAudioPtr1, pAudioBytes1, ppAudioPtr2, pAudioBytes2, dwFlags); }
-static MA_INLINE HRESULT ma_IDirectSoundBuffer_Play(ma_IDirectSoundBuffer* pThis, DWORD dwReserved1, DWORD dwPriority, DWORD dwFlags) { return pThis->lpVtbl->Play(pThis, dwReserved1, dwPriority, dwFlags); }
-static MA_INLINE HRESULT ma_IDirectSoundBuffer_SetCurrentPosition(ma_IDirectSoundBuffer* pThis, DWORD dwNewPosition) { return pThis->lpVtbl->SetCurrentPosition(pThis, dwNewPosition); }
-static MA_INLINE HRESULT ma_IDirectSoundBuffer_SetFormat(ma_IDirectSoundBuffer* pThis, const WAVEFORMATEX* pFormat) { return pThis->lpVtbl->SetFormat(pThis, pFormat); }
-static MA_INLINE HRESULT ma_IDirectSoundBuffer_SetVolume(ma_IDirectSoundBuffer* pThis, LONG volume) { return pThis->lpVtbl->SetVolume(pThis, volume); }
-static MA_INLINE HRESULT ma_IDirectSoundBuffer_SetPan(ma_IDirectSoundBuffer* pThis, LONG pan) { return pThis->lpVtbl->SetPan(pThis, pan); }
-static MA_INLINE HRESULT ma_IDirectSoundBuffer_SetFrequency(ma_IDirectSoundBuffer* pThis, DWORD dwFrequency) { return pThis->lpVtbl->SetFrequency(pThis, dwFrequency); }
-static MA_INLINE HRESULT ma_IDirectSoundBuffer_Stop(ma_IDirectSoundBuffer* pThis) { return pThis->lpVtbl->Stop(pThis); }
-static MA_INLINE HRESULT ma_IDirectSoundBuffer_Unlock(ma_IDirectSoundBuffer* pThis, void* pAudioPtr1, DWORD dwAudioBytes1, void* pAudioPtr2, DWORD dwAudioBytes2) { return pThis->lpVtbl->Unlock(pThis, pAudioPtr1, dwAudioBytes1, pAudioPtr2, dwAudioBytes2); }
-static MA_INLINE HRESULT ma_IDirectSoundBuffer_Restore(ma_IDirectSoundBuffer* pThis) { return pThis->lpVtbl->Restore(pThis); }
+ ma_bool32 cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
+ if (cbResult == MA_FALSE) {
+ isTerminated = MA_TRUE;
+ break;
+ }
+ }
+ }
+ }
+ return MA_SUCCESS;
+#else
+ goto return_default_device;
+#endif
-/* IDirectSoundCapture */
-typedef struct
-{
- /* IUnknown */
- HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSoundCapture* pThis, const IID* const riid, void** ppObject);
- ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSoundCapture* pThis);
- ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSoundCapture* pThis);
+return_default_device:;
+ cbResult = MA_TRUE;
- /* IDirectSoundCapture */
- HRESULT (STDMETHODCALLTYPE * CreateCaptureBuffer)(ma_IDirectSoundCapture* pThis, const MA_DSCBUFFERDESC* pDSCBufferDesc, ma_IDirectSoundCaptureBuffer** ppDSCBuffer, void* pUnkOuter);
- HRESULT (STDMETHODCALLTYPE * GetCaps) (ma_IDirectSoundCapture* pThis, MA_DSCCAPS* pDSCCaps);
- HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IDirectSoundCapture* pThis, const GUID* pGuidDevice);
-} ma_IDirectSoundCaptureVtbl;
-struct ma_IDirectSoundCapture
-{
- ma_IDirectSoundCaptureVtbl* lpVtbl;
-};
-static MA_INLINE HRESULT ma_IDirectSoundCapture_QueryInterface(ma_IDirectSoundCapture* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
-static MA_INLINE ULONG ma_IDirectSoundCapture_AddRef(ma_IDirectSoundCapture* pThis) { return pThis->lpVtbl->AddRef(pThis); }
-static MA_INLINE ULONG ma_IDirectSoundCapture_Release(ma_IDirectSoundCapture* pThis) { return pThis->lpVtbl->Release(pThis); }
-static MA_INLINE HRESULT ma_IDirectSoundCapture_CreateCaptureBuffer(ma_IDirectSoundCapture* pThis, const MA_DSCBUFFERDESC* pDSCBufferDesc, ma_IDirectSoundCaptureBuffer** ppDSCBuffer, void* pUnkOuter) { return pThis->lpVtbl->CreateCaptureBuffer(pThis, pDSCBufferDesc, ppDSCBuffer, pUnkOuter); }
-static MA_INLINE HRESULT ma_IDirectSoundCapture_GetCaps (ma_IDirectSoundCapture* pThis, MA_DSCCAPS* pDSCCaps) { return pThis->lpVtbl->GetCaps(pThis, pDSCCaps); }
-static MA_INLINE HRESULT ma_IDirectSoundCapture_Initialize (ma_IDirectSoundCapture* pThis, const GUID* pGuidDevice) { return pThis->lpVtbl->Initialize(pThis, pGuidDevice); }
+ /* Playback. */
+ if (cbResult) {
+ ma_device_info deviceInfo;
+ MA_ZERO_OBJECT(&deviceInfo);
+ deviceInfo.id.opensl = SL_DEFAULTDEVICEID_AUDIOOUTPUT;
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
+ cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
+ }
+ /* Capture. */
+ if (cbResult) {
+ ma_device_info deviceInfo;
+ MA_ZERO_OBJECT(&deviceInfo);
+ deviceInfo.id.opensl = SL_DEFAULTDEVICEID_AUDIOINPUT;
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
+ cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
+ }
-/* IDirectSoundCaptureBuffer */
-typedef struct
-{
- /* IUnknown */
- HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSoundCaptureBuffer* pThis, const IID* const riid, void** ppObject);
- ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSoundCaptureBuffer* pThis);
- ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSoundCaptureBuffer* pThis);
+ return MA_SUCCESS;
+}
- /* IDirectSoundCaptureBuffer */
- HRESULT (STDMETHODCALLTYPE * GetCaps) (ma_IDirectSoundCaptureBuffer* pThis, MA_DSCBCAPS* pDSCBCaps);
- HRESULT (STDMETHODCALLTYPE * GetCurrentPosition)(ma_IDirectSoundCaptureBuffer* pThis, DWORD* pCapturePosition, DWORD* pReadPosition);
- HRESULT (STDMETHODCALLTYPE * GetFormat) (ma_IDirectSoundCaptureBuffer* pThis, WAVEFORMATEX* pFormat, DWORD dwSizeAllocated, DWORD* pSizeWritten);
- HRESULT (STDMETHODCALLTYPE * GetStatus) (ma_IDirectSoundCaptureBuffer* pThis, DWORD* pStatus);
- HRESULT (STDMETHODCALLTYPE * Initialize) (ma_IDirectSoundCaptureBuffer* pThis, ma_IDirectSoundCapture* pDirectSoundCapture, const MA_DSCBUFFERDESC* pDSCBufferDesc);
- HRESULT (STDMETHODCALLTYPE * Lock) (ma_IDirectSoundCaptureBuffer* pThis, DWORD dwOffset, DWORD dwBytes, void** ppAudioPtr1, DWORD* pAudioBytes1, void** ppAudioPtr2, DWORD* pAudioBytes2, DWORD dwFlags);
- HRESULT (STDMETHODCALLTYPE * Start) (ma_IDirectSoundCaptureBuffer* pThis, DWORD dwFlags);
- HRESULT (STDMETHODCALLTYPE * Stop) (ma_IDirectSoundCaptureBuffer* pThis);
- HRESULT (STDMETHODCALLTYPE * Unlock) (ma_IDirectSoundCaptureBuffer* pThis, void* pAudioPtr1, DWORD dwAudioBytes1, void* pAudioPtr2, DWORD dwAudioBytes2);
-} ma_IDirectSoundCaptureBufferVtbl;
-struct ma_IDirectSoundCaptureBuffer
+static void ma_context_add_data_format_ex__opensl(ma_context* pContext, ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_device_info* pDeviceInfo)
{
- ma_IDirectSoundCaptureBufferVtbl* lpVtbl;
-};
-static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_QueryInterface(ma_IDirectSoundCaptureBuffer* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
-static MA_INLINE ULONG ma_IDirectSoundCaptureBuffer_AddRef(ma_IDirectSoundCaptureBuffer* pThis) { return pThis->lpVtbl->AddRef(pThis); }
-static MA_INLINE ULONG ma_IDirectSoundCaptureBuffer_Release(ma_IDirectSoundCaptureBuffer* pThis) { return pThis->lpVtbl->Release(pThis); }
-static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_GetCaps(ma_IDirectSoundCaptureBuffer* pThis, MA_DSCBCAPS* pDSCBCaps) { return pThis->lpVtbl->GetCaps(pThis, pDSCBCaps); }
-static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_GetCurrentPosition(ma_IDirectSoundCaptureBuffer* pThis, DWORD* pCapturePosition, DWORD* pReadPosition) { return pThis->lpVtbl->GetCurrentPosition(pThis, pCapturePosition, pReadPosition); }
-static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_GetFormat(ma_IDirectSoundCaptureBuffer* pThis, WAVEFORMATEX* pFormat, DWORD dwSizeAllocated, DWORD* pSizeWritten) { return pThis->lpVtbl->GetFormat(pThis, pFormat, dwSizeAllocated, pSizeWritten); }
-static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_GetStatus(ma_IDirectSoundCaptureBuffer* pThis, DWORD* pStatus) { return pThis->lpVtbl->GetStatus(pThis, pStatus); }
-static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_Initialize(ma_IDirectSoundCaptureBuffer* pThis, ma_IDirectSoundCapture* pDirectSoundCapture, const MA_DSCBUFFERDESC* pDSCBufferDesc) { return pThis->lpVtbl->Initialize(pThis, pDirectSoundCapture, pDSCBufferDesc); }
-static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_Lock(ma_IDirectSoundCaptureBuffer* pThis, DWORD dwOffset, DWORD dwBytes, void** ppAudioPtr1, DWORD* pAudioBytes1, void** ppAudioPtr2, DWORD* pAudioBytes2, DWORD dwFlags) { return pThis->lpVtbl->Lock(pThis, dwOffset, dwBytes, ppAudioPtr1, pAudioBytes1, ppAudioPtr2, pAudioBytes2, dwFlags); }
-static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_Start(ma_IDirectSoundCaptureBuffer* pThis, DWORD dwFlags) { return pThis->lpVtbl->Start(pThis, dwFlags); }
-static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_Stop(ma_IDirectSoundCaptureBuffer* pThis) { return pThis->lpVtbl->Stop(pThis); }
-static MA_INLINE HRESULT ma_IDirectSoundCaptureBuffer_Unlock(ma_IDirectSoundCaptureBuffer* pThis, void* pAudioPtr1, DWORD dwAudioBytes1, void* pAudioPtr2, DWORD dwAudioBytes2) { return pThis->lpVtbl->Unlock(pThis, pAudioPtr1, dwAudioBytes1, pAudioPtr2, dwAudioBytes2); }
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pDeviceInfo != NULL);
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = format;
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = channels;
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = sampleRate;
+ pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = 0;
+ pDeviceInfo->nativeDataFormatCount += 1;
+}
-/* IDirectSoundNotify */
-typedef struct
+static void ma_context_add_data_format__opensl(ma_context* pContext, ma_format format, ma_device_info* pDeviceInfo)
{
- /* IUnknown */
- HRESULT (STDMETHODCALLTYPE * QueryInterface)(ma_IDirectSoundNotify* pThis, const IID* const riid, void** ppObject);
- ULONG (STDMETHODCALLTYPE * AddRef) (ma_IDirectSoundNotify* pThis);
- ULONG (STDMETHODCALLTYPE * Release) (ma_IDirectSoundNotify* pThis);
+ ma_uint32 minChannels = 1;
+ ma_uint32 maxChannels = 2;
+ ma_uint32 minSampleRate = (ma_uint32)ma_standard_sample_rate_8000;
+ ma_uint32 maxSampleRate = (ma_uint32)ma_standard_sample_rate_48000;
+ ma_uint32 iChannel;
+ ma_uint32 iSampleRate;
- /* IDirectSoundNotify */
- HRESULT (STDMETHODCALLTYPE * SetNotificationPositions)(ma_IDirectSoundNotify* pThis, DWORD dwPositionNotifies, const MA_DSBPOSITIONNOTIFY* pPositionNotifies);
-} ma_IDirectSoundNotifyVtbl;
-struct ma_IDirectSoundNotify
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pDeviceInfo != NULL);
+
+ /*
+ Each sample format can support mono and stereo, and we'll support a small subset of standard
+ rates (up to 48000). A better solution would be to somehow find a native sample rate.
+ */
+ for (iChannel = minChannels; iChannel < maxChannels; iChannel += 1) {
+ for (iSampleRate = 0; iSampleRate < ma_countof(g_maStandardSampleRatePriorities); iSampleRate += 1) {
+ ma_uint32 standardSampleRate = g_maStandardSampleRatePriorities[iSampleRate];
+ if (standardSampleRate >= minSampleRate && standardSampleRate <= maxSampleRate) {
+ ma_context_add_data_format_ex__opensl(pContext, format, iChannel, standardSampleRate, pDeviceInfo);
+ }
+ }
+ }
+}
+
+static ma_result ma_context_get_device_info__opensl(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
{
- ma_IDirectSoundNotifyVtbl* lpVtbl;
-};
-static MA_INLINE HRESULT ma_IDirectSoundNotify_QueryInterface(ma_IDirectSoundNotify* pThis, const IID* const riid, void** ppObject) { return pThis->lpVtbl->QueryInterface(pThis, riid, ppObject); }
-static MA_INLINE ULONG ma_IDirectSoundNotify_AddRef(ma_IDirectSoundNotify* pThis) { return pThis->lpVtbl->AddRef(pThis); }
-static MA_INLINE ULONG ma_IDirectSoundNotify_Release(ma_IDirectSoundNotify* pThis) { return pThis->lpVtbl->Release(pThis); }
-static MA_INLINE HRESULT ma_IDirectSoundNotify_SetNotificationPositions(ma_IDirectSoundNotify* pThis, DWORD dwPositionNotifies, const MA_DSBPOSITIONNOTIFY* pPositionNotifies) { return pThis->lpVtbl->SetNotificationPositions(pThis, dwPositionNotifies, pPositionNotifies); }
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to get device info. */
+ if (g_maOpenSLInitCounter == 0) {
+ return MA_INVALID_OPERATION;
+ }
-typedef BOOL (CALLBACK * ma_DSEnumCallbackAProc) (LPGUID pDeviceGUID, LPCSTR pDeviceDescription, LPCSTR pModule, LPVOID pContext);
-typedef HRESULT (WINAPI * ma_DirectSoundCreateProc) (const GUID* pcGuidDevice, ma_IDirectSound** ppDS8, LPUNKNOWN pUnkOuter);
-typedef HRESULT (WINAPI * ma_DirectSoundEnumerateAProc) (ma_DSEnumCallbackAProc pDSEnumCallback, LPVOID pContext);
-typedef HRESULT (WINAPI * ma_DirectSoundCaptureCreateProc) (const GUID* pcGuidDevice, ma_IDirectSoundCapture** ppDSC8, LPUNKNOWN pUnkOuter);
-typedef HRESULT (WINAPI * ma_DirectSoundCaptureEnumerateAProc)(ma_DSEnumCallbackAProc pDSEnumCallback, LPVOID pContext);
+ /*
+ TODO: Test Me.
-static ma_uint32 ma_get_best_sample_rate_within_range(ma_uint32 sampleRateMin, ma_uint32 sampleRateMax)
-{
- /* Normalize the range in case we were given something stupid. */
- if (sampleRateMin < (ma_uint32)ma_standard_sample_rate_min) {
- sampleRateMin = (ma_uint32)ma_standard_sample_rate_min;
+ This is currently untested, so for now we are just returning default devices.
+ */
+#if 0 && !defined(MA_ANDROID)
+ SLAudioIODeviceCapabilitiesItf deviceCaps;
+ SLresult resultSL = (*g_maEngineObjectSL)->GetInterface(g_maEngineObjectSL, (SLInterfaceID)pContext->opensl.SL_IID_AUDIOIODEVICECAPABILITIES, &deviceCaps);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ /* The interface may not be supported so just report a default device. */
+ goto return_default_device;
}
- if (sampleRateMax > (ma_uint32)ma_standard_sample_rate_max) {
- sampleRateMax = (ma_uint32)ma_standard_sample_rate_max;
+
+ if (deviceType == ma_device_type_playback) {
+ SLAudioOutputDescriptor desc;
+ resultSL = (*deviceCaps)->QueryAudioOutputCapabilities(deviceCaps, pDeviceID->opensl, &desc);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ return ma_result_from_OpenSL(resultSL);
+ }
+
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), (const char*)desc.pDeviceName, (size_t)-1);
+ } else {
+ SLAudioInputDescriptor desc;
+ resultSL = (*deviceCaps)->QueryAudioInputCapabilities(deviceCaps, pDeviceID->opensl, &desc);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ return ma_result_from_OpenSL(resultSL);
+ }
+
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), (const char*)desc.deviceName, (size_t)-1);
}
- if (sampleRateMin > sampleRateMax) {
- sampleRateMin = sampleRateMax;
+
+ goto return_detailed_info;
+#else
+ goto return_default_device;
+#endif
+
+return_default_device:
+ if (pDeviceID != NULL) {
+ if ((deviceType == ma_device_type_playback && pDeviceID->opensl != SL_DEFAULTDEVICEID_AUDIOOUTPUT) ||
+ (deviceType == ma_device_type_capture && pDeviceID->opensl != SL_DEFAULTDEVICEID_AUDIOINPUT)) {
+ return MA_NO_DEVICE; /* Don't know the device. */
+ }
}
- if (sampleRateMin == sampleRateMax) {
- return sampleRateMax;
+ /* ID and Name / Description */
+ if (deviceType == ma_device_type_playback) {
+ pDeviceInfo->id.opensl = SL_DEFAULTDEVICEID_AUDIOOUTPUT;
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
} else {
- size_t iStandardRate;
- for (iStandardRate = 0; iStandardRate < ma_countof(g_maStandardSampleRatePriorities); ++iStandardRate) {
- ma_uint32 standardRate = g_maStandardSampleRatePriorities[iStandardRate];
- if (standardRate >= sampleRateMin && standardRate <= sampleRateMax) {
- return standardRate;
- }
- }
+ pDeviceInfo->id.opensl = SL_DEFAULTDEVICEID_AUDIOINPUT;
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
}
- /* Should never get here. */
- MA_ASSERT(MA_FALSE);
- return 0;
+ pDeviceInfo->isDefault = MA_TRUE;
+
+ goto return_detailed_info;
+
+
+return_detailed_info:
+
+ /*
+ For now we're just outputting a set of values that are supported by the API but not necessarily supported
+ by the device natively. Later on we should work on this so that it more closely reflects the device's
+ actual native format.
+ */
+ pDeviceInfo->nativeDataFormatCount = 0;
+#if defined(MA_ANDROID) && __ANDROID_API__ >= 21
+ ma_context_add_data_format__opensl(pContext, ma_format_f32, pDeviceInfo);
+#endif
+ ma_context_add_data_format__opensl(pContext, ma_format_s16, pDeviceInfo);
+ ma_context_add_data_format__opensl(pContext, ma_format_u8, pDeviceInfo);
+
+ return MA_SUCCESS;
}
-/*
-Retrieves the channel count and channel map for the given speaker configuration. If the speaker configuration is unknown,
-the channel count and channel map will be left unmodified.
-*/
-static void ma_get_channels_from_speaker_config__dsound(DWORD speakerConfig, WORD* pChannelsOut, DWORD* pChannelMapOut)
+
+#ifdef MA_ANDROID
+/*void ma_buffer_queue_callback_capture__opensl_android(SLAndroidSimpleBufferQueueItf pBufferQueue, SLuint32 eventFlags, const void* pBuffer, SLuint32 bufferSize, SLuint32 dataUsed, void* pContext)*/
+static void ma_buffer_queue_callback_capture__opensl_android(SLAndroidSimpleBufferQueueItf pBufferQueue, void* pUserData)
{
- WORD channels;
- DWORD channelMap;
+ ma_device* pDevice = (ma_device*)pUserData;
+ size_t periodSizeInBytes;
+ ma_uint8* pBuffer;
+ SLresult resultSL;
- channels = 0;
- if (pChannelsOut != NULL) {
- channels = *pChannelsOut;
- }
+ MA_ASSERT(pDevice != NULL);
- channelMap = 0;
- if (pChannelMapOut != NULL) {
- channelMap = *pChannelMapOut;
- }
+ (void)pBufferQueue;
/*
- The speaker configuration is a combination of speaker config and speaker geometry. The lower 8 bits is what we care about. The upper
- 16 bits is for the geometry.
+ For now, don't do anything unless the buffer was fully processed. From what I can tell, it looks like
+ OpenSL|ES 1.1 improves on buffer queues to the point that we could much more intelligently handle this,
+ but unfortunately it looks like Android is only supporting OpenSL|ES 1.0.1 for now :(
*/
- switch ((BYTE)(speakerConfig)) {
- case 1 /*DSSPEAKER_HEADPHONE*/: channels = 2; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT; break;
- case 2 /*DSSPEAKER_MONO*/: channels = 1; channelMap = SPEAKER_FRONT_CENTER; break;
- case 3 /*DSSPEAKER_QUAD*/: channels = 4; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT; break;
- case 4 /*DSSPEAKER_STEREO*/: channels = 2; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT; break;
- case 5 /*DSSPEAKER_SURROUND*/: channels = 4; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_BACK_CENTER; break;
- case 6 /*DSSPEAKER_5POINT1_BACK*/ /*DSSPEAKER_5POINT1*/: channels = 6; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT; break;
- case 7 /*DSSPEAKER_7POINT1_WIDE*/ /*DSSPEAKER_7POINT1*/: channels = 8; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_FRONT_LEFT_OF_CENTER | SPEAKER_FRONT_RIGHT_OF_CENTER; break;
- case 8 /*DSSPEAKER_7POINT1_SURROUND*/: channels = 8; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT; break;
- case 9 /*DSSPEAKER_5POINT1_SURROUND*/: channels = 6; channelMap = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT; break;
- default: break;
+
+ /* Don't do anything if the device is not started. */
+ if (ma_device_get_state(pDevice) != ma_device_state_started) {
+ return;
}
- if (pChannelsOut != NULL) {
- *pChannelsOut = channels;
+ /* Don't do anything if the device is being drained. */
+ if (pDevice->opensl.isDrainingCapture) {
+ return;
}
- if (pChannelMapOut != NULL) {
- *pChannelMapOut = channelMap;
+ periodSizeInBytes = pDevice->capture.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
+ pBuffer = pDevice->opensl.pBufferCapture + (pDevice->opensl.currentBufferIndexCapture * periodSizeInBytes);
+
+ ma_device_handle_backend_data_callback(pDevice, NULL, pBuffer, pDevice->capture.internalPeriodSizeInFrames);
+
+ resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture, pBuffer, periodSizeInBytes);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ return;
}
-}
+ pDevice->opensl.currentBufferIndexCapture = (pDevice->opensl.currentBufferIndexCapture + 1) % pDevice->capture.internalPeriods;
+}
-static ma_result ma_context_create_IDirectSound__dsound(ma_context* pContext, ma_share_mode shareMode, const ma_device_id* pDeviceID, ma_IDirectSound** ppDirectSound)
+static void ma_buffer_queue_callback_playback__opensl_android(SLAndroidSimpleBufferQueueItf pBufferQueue, void* pUserData)
{
- ma_IDirectSound* pDirectSound;
- HWND hWnd;
- HRESULT hr;
+ ma_device* pDevice = (ma_device*)pUserData;
+ size_t periodSizeInBytes;
+ ma_uint8* pBuffer;
+ SLresult resultSL;
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(ppDirectSound != NULL);
+ MA_ASSERT(pDevice != NULL);
- *ppDirectSound = NULL;
- pDirectSound = NULL;
+ (void)pBufferQueue;
- if (FAILED(((ma_DirectSoundCreateProc)pContext->dsound.DirectSoundCreate)((pDeviceID == NULL) ? NULL : (const GUID*)pDeviceID->dsound, &pDirectSound, NULL))) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[DirectSound] DirectSoundCreate() failed for playback device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE);
+ /* Don't do anything if the device is not started. */
+ if (ma_device_get_state(pDevice) != ma_device_state_started) {
+ return;
}
- /* The cooperative level must be set before doing anything else. */
- hWnd = ((MA_PFN_GetForegroundWindow)pContext->win32.GetForegroundWindow)();
- if (hWnd == NULL) {
- hWnd = ((MA_PFN_GetDesktopWindow)pContext->win32.GetDesktopWindow)();
+ /* Don't do anything if the device is being drained. */
+ if (pDevice->opensl.isDrainingPlayback) {
+ return;
}
- hr = ma_IDirectSound_SetCooperativeLevel(pDirectSound, hWnd, (shareMode == ma_share_mode_exclusive) ? MA_DSSCL_EXCLUSIVE : MA_DSSCL_PRIORITY);
- if (FAILED(hr)) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_SetCooperateiveLevel() failed for playback device.", ma_result_from_HRESULT(hr));
+ periodSizeInBytes = pDevice->playback.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
+ pBuffer = pDevice->opensl.pBufferPlayback + (pDevice->opensl.currentBufferIndexPlayback * periodSizeInBytes);
+
+ ma_device_handle_backend_data_callback(pDevice, pBuffer, NULL, pDevice->playback.internalPeriodSizeInFrames);
+
+ resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback, pBuffer, periodSizeInBytes);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ return;
}
- *ppDirectSound = pDirectSound;
- return MA_SUCCESS;
+ pDevice->opensl.currentBufferIndexPlayback = (pDevice->opensl.currentBufferIndexPlayback + 1) % pDevice->playback.internalPeriods;
}
+#endif
-static ma_result ma_context_create_IDirectSoundCapture__dsound(ma_context* pContext, ma_share_mode shareMode, const ma_device_id* pDeviceID, ma_IDirectSoundCapture** ppDirectSoundCapture)
+static ma_result ma_device_uninit__opensl(ma_device* pDevice)
{
- ma_IDirectSoundCapture* pDirectSoundCapture;
- HRESULT hr;
+ MA_ASSERT(pDevice != NULL);
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(ppDirectSoundCapture != NULL);
+ MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it before uninitializing the device. */
+ if (g_maOpenSLInitCounter == 0) {
+ return MA_INVALID_OPERATION;
+ }
- /* DirectSound does not support exclusive mode for capture. */
- if (shareMode == ma_share_mode_exclusive) {
- return MA_SHARE_MODE_NOT_SUPPORTED;
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ if (pDevice->opensl.pAudioRecorderObj) {
+ MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->Destroy((SLObjectItf)pDevice->opensl.pAudioRecorderObj);
+ }
+
+ ma_free(pDevice->opensl.pBufferCapture, &pDevice->pContext->allocationCallbacks);
}
- *ppDirectSoundCapture = NULL;
- pDirectSoundCapture = NULL;
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ if (pDevice->opensl.pAudioPlayerObj) {
+ MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->Destroy((SLObjectItf)pDevice->opensl.pAudioPlayerObj);
+ }
+ if (pDevice->opensl.pOutputMixObj) {
+ MA_OPENSL_OBJ(pDevice->opensl.pOutputMixObj)->Destroy((SLObjectItf)pDevice->opensl.pOutputMixObj);
+ }
- hr = ((ma_DirectSoundCaptureCreateProc)pContext->dsound.DirectSoundCaptureCreate)((pDeviceID == NULL) ? NULL : (const GUID*)pDeviceID->dsound, &pDirectSoundCapture, NULL);
- if (FAILED(hr)) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[DirectSound] DirectSoundCaptureCreate() failed for capture device.", ma_result_from_HRESULT(hr));
+ ma_free(pDevice->opensl.pBufferPlayback, &pDevice->pContext->allocationCallbacks);
}
- *ppDirectSoundCapture = pDirectSoundCapture;
return MA_SUCCESS;
}
-static ma_result ma_context_get_format_info_for_IDirectSoundCapture__dsound(ma_context* pContext, ma_IDirectSoundCapture* pDirectSoundCapture, WORD* pChannels, WORD* pBitsPerSample, DWORD* pSampleRate)
-{
- HRESULT hr;
- MA_DSCCAPS caps;
- WORD bitsPerSample;
- DWORD sampleRate;
-
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pDirectSoundCapture != NULL);
+#if defined(MA_ANDROID) && __ANDROID_API__ >= 21
+typedef SLAndroidDataFormat_PCM_EX ma_SLDataFormat_PCM;
+#else
+typedef SLDataFormat_PCM ma_SLDataFormat_PCM;
+#endif
- if (pChannels) {
- *pChannels = 0;
+static ma_result ma_SLDataFormat_PCM_init__opensl(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, const ma_channel* channelMap, ma_SLDataFormat_PCM* pDataFormat)
+{
+ /* We need to convert our format/channels/rate so that they aren't set to default. */
+ if (format == ma_format_unknown) {
+ format = MA_DEFAULT_FORMAT;
}
- if (pBitsPerSample) {
- *pBitsPerSample = 0;
+ if (channels == 0) {
+ channels = MA_DEFAULT_CHANNELS;
}
- if (pSampleRate) {
- *pSampleRate = 0;
+ if (sampleRate == 0) {
+ sampleRate = MA_DEFAULT_SAMPLE_RATE;
}
- MA_ZERO_OBJECT(&caps);
- caps.dwSize = sizeof(caps);
- hr = ma_IDirectSoundCapture_GetCaps(pDirectSoundCapture, &caps);
- if (FAILED(hr)) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCapture_GetCaps() failed for capture device.", ma_result_from_HRESULT(hr));
+#if defined(MA_ANDROID) && __ANDROID_API__ >= 21
+ if (format == ma_format_f32) {
+ pDataFormat->formatType = SL_ANDROID_DATAFORMAT_PCM_EX;
+ pDataFormat->representation = SL_ANDROID_PCM_REPRESENTATION_FLOAT;
+ } else {
+ pDataFormat->formatType = SL_DATAFORMAT_PCM;
}
+#else
+ pDataFormat->formatType = SL_DATAFORMAT_PCM;
+#endif
- if (pChannels) {
- *pChannels = (WORD)caps.dwChannels;
+ pDataFormat->numChannels = channels;
+ ((SLDataFormat_PCM*)pDataFormat)->samplesPerSec = ma_round_to_standard_sample_rate__opensl(sampleRate * 1000); /* In millihertz. Annoyingly, the sample rate variable is named differently between SLAndroidDataFormat_PCM_EX and SLDataFormat_PCM */
+ pDataFormat->bitsPerSample = ma_get_bytes_per_sample(format) * 8;
+ pDataFormat->channelMask = ma_channel_map_to_channel_mask__opensl(channelMap, channels);
+ pDataFormat->endianness = (ma_is_little_endian()) ? SL_BYTEORDER_LITTLEENDIAN : SL_BYTEORDER_BIGENDIAN;
+
+ /*
+ Android has a few restrictions on the format as documented here: https://developer.android.com/ndk/guides/audio/opensl-for-android.html
+ - Only mono and stereo is supported.
+ - Only u8 and s16 formats are supported.
+ - Maximum sample rate of 48000.
+ */
+#ifdef MA_ANDROID
+ if (pDataFormat->numChannels > 2) {
+ pDataFormat->numChannels = 2;
+ }
+#if __ANDROID_API__ >= 21
+ if (pDataFormat->formatType == SL_ANDROID_DATAFORMAT_PCM_EX) {
+ /* It's floating point. */
+ MA_ASSERT(pDataFormat->representation == SL_ANDROID_PCM_REPRESENTATION_FLOAT);
+ if (pDataFormat->bitsPerSample > 32) {
+ pDataFormat->bitsPerSample = 32;
+ }
+ } else {
+ if (pDataFormat->bitsPerSample > 16) {
+ pDataFormat->bitsPerSample = 16;
+ }
+ }
+#else
+ if (pDataFormat->bitsPerSample > 16) {
+ pDataFormat->bitsPerSample = 16;
+ }
+#endif
+ if (((SLDataFormat_PCM*)pDataFormat)->samplesPerSec > SL_SAMPLINGRATE_48) {
+ ((SLDataFormat_PCM*)pDataFormat)->samplesPerSec = SL_SAMPLINGRATE_48;
}
+#endif
- /* The device can support multiple formats. We just go through the different formats in order of priority and pick the first one. This the same type of system as the WinMM backend. */
- bitsPerSample = 16;
- sampleRate = 48000;
+ pDataFormat->containerSize = pDataFormat->bitsPerSample; /* Always tightly packed for now. */
- if (caps.dwChannels == 1) {
- if ((caps.dwFormats & WAVE_FORMAT_48M16) != 0) {
- sampleRate = 48000;
- } else if ((caps.dwFormats & WAVE_FORMAT_44M16) != 0) {
- sampleRate = 44100;
- } else if ((caps.dwFormats & WAVE_FORMAT_2M16) != 0) {
- sampleRate = 22050;
- } else if ((caps.dwFormats & WAVE_FORMAT_1M16) != 0) {
- sampleRate = 11025;
- } else if ((caps.dwFormats & WAVE_FORMAT_96M16) != 0) {
- sampleRate = 96000;
- } else {
- bitsPerSample = 8;
- if ((caps.dwFormats & WAVE_FORMAT_48M08) != 0) {
- sampleRate = 48000;
- } else if ((caps.dwFormats & WAVE_FORMAT_44M08) != 0) {
- sampleRate = 44100;
- } else if ((caps.dwFormats & WAVE_FORMAT_2M08) != 0) {
- sampleRate = 22050;
- } else if ((caps.dwFormats & WAVE_FORMAT_1M08) != 0) {
- sampleRate = 11025;
- } else if ((caps.dwFormats & WAVE_FORMAT_96M08) != 0) {
- sampleRate = 96000;
- } else {
- bitsPerSample = 16; /* Didn't find it. Just fall back to 16-bit. */
- }
+ return MA_SUCCESS;
+}
+
+static ma_result ma_deconstruct_SLDataFormat_PCM__opensl(ma_SLDataFormat_PCM* pDataFormat, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
+{
+ ma_bool32 isFloatingPoint = MA_FALSE;
+#if defined(MA_ANDROID) && __ANDROID_API__ >= 21
+ if (pDataFormat->formatType == SL_ANDROID_DATAFORMAT_PCM_EX) {
+ MA_ASSERT(pDataFormat->representation == SL_ANDROID_PCM_REPRESENTATION_FLOAT);
+ isFloatingPoint = MA_TRUE;
+ }
+#endif
+ if (isFloatingPoint) {
+ if (pDataFormat->bitsPerSample == 32) {
+ *pFormat = ma_format_f32;
}
- } else if (caps.dwChannels == 2) {
- if ((caps.dwFormats & WAVE_FORMAT_48S16) != 0) {
- sampleRate = 48000;
- } else if ((caps.dwFormats & WAVE_FORMAT_44S16) != 0) {
- sampleRate = 44100;
- } else if ((caps.dwFormats & WAVE_FORMAT_2S16) != 0) {
- sampleRate = 22050;
- } else if ((caps.dwFormats & WAVE_FORMAT_1S16) != 0) {
- sampleRate = 11025;
- } else if ((caps.dwFormats & WAVE_FORMAT_96S16) != 0) {
- sampleRate = 96000;
- } else {
- bitsPerSample = 8;
- if ((caps.dwFormats & WAVE_FORMAT_48S08) != 0) {
- sampleRate = 48000;
- } else if ((caps.dwFormats & WAVE_FORMAT_44S08) != 0) {
- sampleRate = 44100;
- } else if ((caps.dwFormats & WAVE_FORMAT_2S08) != 0) {
- sampleRate = 22050;
- } else if ((caps.dwFormats & WAVE_FORMAT_1S08) != 0) {
- sampleRate = 11025;
- } else if ((caps.dwFormats & WAVE_FORMAT_96S08) != 0) {
- sampleRate = 96000;
- } else {
- bitsPerSample = 16; /* Didn't find it. Just fall back to 16-bit. */
- }
+ } else {
+ if (pDataFormat->bitsPerSample == 8) {
+ *pFormat = ma_format_u8;
+ } else if (pDataFormat->bitsPerSample == 16) {
+ *pFormat = ma_format_s16;
+ } else if (pDataFormat->bitsPerSample == 24) {
+ *pFormat = ma_format_s24;
+ } else if (pDataFormat->bitsPerSample == 32) {
+ *pFormat = ma_format_s32;
}
}
- if (pBitsPerSample) {
- *pBitsPerSample = bitsPerSample;
- }
- if (pSampleRate) {
- *pSampleRate = sampleRate;
- }
+ *pChannels = pDataFormat->numChannels;
+ *pSampleRate = ((SLDataFormat_PCM*)pDataFormat)->samplesPerSec / 1000;
+ ma_channel_mask_to_channel_map__opensl(pDataFormat->channelMask, ma_min(pDataFormat->numChannels, channelMapCap), pChannelMap);
return MA_SUCCESS;
}
-
-typedef struct
-{
- ma_context* pContext;
- ma_device_type deviceType;
- ma_enum_devices_callback_proc callback;
- void* pUserData;
- ma_bool32 terminated;
-} ma_context_enumerate_devices_callback_data__dsound;
-
-static BOOL CALLBACK ma_context_enumerate_devices_callback__dsound(LPGUID lpGuid, LPCSTR lpcstrDescription, LPCSTR lpcstrModule, LPVOID lpContext)
+static ma_result ma_device_init__opensl(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
{
- ma_context_enumerate_devices_callback_data__dsound* pData = (ma_context_enumerate_devices_callback_data__dsound*)lpContext;
- ma_device_info deviceInfo;
-
- (void)lpcstrModule;
-
- MA_ZERO_OBJECT(&deviceInfo);
+#ifdef MA_ANDROID
+ SLDataLocator_AndroidSimpleBufferQueue queue;
+ SLresult resultSL;
+ size_t bufferSizeInBytes;
+ SLInterfaceID itfIDs[2];
+ const SLboolean itfIDsRequired[] = {
+ SL_BOOLEAN_TRUE, /* SL_IID_ANDROIDSIMPLEBUFFERQUEUE */
+ SL_BOOLEAN_FALSE /* SL_IID_ANDROIDCONFIGURATION */
+ };
+#endif
- /* ID. */
- if (lpGuid != NULL) {
- MA_COPY_MEMORY(deviceInfo.id.dsound, lpGuid, 16);
- } else {
- MA_ZERO_MEMORY(deviceInfo.id.dsound, 16);
- deviceInfo.isDefault = MA_TRUE;
+ MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to initialize a new device. */
+ if (g_maOpenSLInitCounter == 0) {
+ return MA_INVALID_OPERATION;
}
- /* Name / Description */
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), lpcstrDescription, (size_t)-1);
+ if (pConfig->deviceType == ma_device_type_loopback) {
+ return MA_DEVICE_TYPE_NOT_SUPPORTED;
+ }
+ /*
+ For now, only supporting Android implementations of OpenSL|ES since that's the only one I've
+ been able to test with and I currently depend on Android-specific extensions (simple buffer
+ queues).
+ */
+#ifdef MA_ANDROID
+ itfIDs[0] = (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDSIMPLEBUFFERQUEUE;
+ itfIDs[1] = (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDCONFIGURATION;
- /* Call the callback function, but make sure we stop enumerating if the callee requested so. */
- MA_ASSERT(pData != NULL);
- pData->terminated = !pData->callback(pData->pContext, pData->deviceType, &deviceInfo, pData->pUserData);
- if (pData->terminated) {
- return FALSE; /* Stop enumeration. */
- } else {
- return TRUE; /* Continue enumeration. */
+ /* No exclusive mode with OpenSL|ES. */
+ if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) ||
+ ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) {
+ return MA_SHARE_MODE_NOT_SUPPORTED;
}
-}
-static ma_result ma_context_enumerate_devices__dsound(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
-{
- ma_context_enumerate_devices_callback_data__dsound data;
+ /* Now we can start initializing the device properly. */
+ MA_ASSERT(pDevice != NULL);
+ MA_ZERO_OBJECT(&pDevice->opensl);
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(callback != NULL);
+ queue.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
- data.pContext = pContext;
- data.callback = callback;
- data.pUserData = pUserData;
- data.terminated = MA_FALSE;
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
+ ma_SLDataFormat_PCM pcm;
+ SLDataLocator_IODevice locatorDevice;
+ SLDataSource source;
+ SLDataSink sink;
+ SLAndroidConfigurationItf pRecorderConfig;
- /* Playback. */
- if (!data.terminated) {
- data.deviceType = ma_device_type_playback;
- ((ma_DirectSoundEnumerateAProc)pContext->dsound.DirectSoundEnumerateA)(ma_context_enumerate_devices_callback__dsound, &data);
- }
+ ma_SLDataFormat_PCM_init__opensl(pDescriptorCapture->format, pDescriptorCapture->channels, pDescriptorCapture->sampleRate, pDescriptorCapture->channelMap, &pcm);
- /* Capture. */
- if (!data.terminated) {
- data.deviceType = ma_device_type_capture;
- ((ma_DirectSoundCaptureEnumerateAProc)pContext->dsound.DirectSoundCaptureEnumerateA)(ma_context_enumerate_devices_callback__dsound, &data);
- }
+ locatorDevice.locatorType = SL_DATALOCATOR_IODEVICE;
+ locatorDevice.deviceType = SL_IODEVICE_AUDIOINPUT;
+ locatorDevice.deviceID = SL_DEFAULTDEVICEID_AUDIOINPUT; /* Must always use the default device with Android. */
+ locatorDevice.device = NULL;
- return MA_SUCCESS;
-}
+ source.pLocator = &locatorDevice;
+ source.pFormat = NULL;
+ queue.numBuffers = pDescriptorCapture->periodCount;
-typedef struct
-{
- const ma_device_id* pDeviceID;
- ma_device_info* pDeviceInfo;
- ma_bool32 found;
-} ma_context_get_device_info_callback_data__dsound;
+ sink.pLocator = &queue;
+ sink.pFormat = (SLDataFormat_PCM*)&pcm;
-static BOOL CALLBACK ma_context_get_device_info_callback__dsound(LPGUID lpGuid, LPCSTR lpcstrDescription, LPCSTR lpcstrModule, LPVOID lpContext)
-{
- ma_context_get_device_info_callback_data__dsound* pData = (ma_context_get_device_info_callback_data__dsound*)lpContext;
- MA_ASSERT(pData != NULL);
+ resultSL = (*g_maEngineSL)->CreateAudioRecorder(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioRecorderObj, &source, &sink, ma_countof(itfIDs), itfIDs, itfIDsRequired);
+ if (resultSL == SL_RESULT_CONTENT_UNSUPPORTED || resultSL == SL_RESULT_PARAMETER_INVALID) {
+ /* Unsupported format. Fall back to something safer and try again. If this fails, just abort. */
+ pcm.formatType = SL_DATAFORMAT_PCM;
+ pcm.numChannels = 1;
+ ((SLDataFormat_PCM*)&pcm)->samplesPerSec = SL_SAMPLINGRATE_16; /* The name of the sample rate variable is different between SLAndroidDataFormat_PCM_EX and SLDataFormat_PCM. */
+ pcm.bitsPerSample = 16;
+ pcm.containerSize = pcm.bitsPerSample; /* Always tightly packed for now. */
+ pcm.channelMask = 0;
+ resultSL = (*g_maEngineSL)->CreateAudioRecorder(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioRecorderObj, &source, &sink, ma_countof(itfIDs), itfIDs, itfIDsRequired);
+ }
- if ((pData->pDeviceID == NULL || ma_is_guid_null(pData->pDeviceID->dsound)) && (lpGuid == NULL || ma_is_guid_null(lpGuid))) {
- /* Default device. */
- ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), lpcstrDescription, (size_t)-1);
- pData->pDeviceInfo->isDefault = MA_TRUE;
- pData->found = MA_TRUE;
- return FALSE; /* Stop enumeration. */
- } else {
- /* Not the default device. */
- if (lpGuid != NULL && pData->pDeviceID != NULL) {
- if (memcmp(pData->pDeviceID->dsound, lpGuid, sizeof(pData->pDeviceID->dsound)) == 0) {
- ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), lpcstrDescription, (size_t)-1);
- pData->found = MA_TRUE;
- return FALSE; /* Stop enumeration. */
- }
+ if (resultSL != SL_RESULT_SUCCESS) {
+ ma_device_uninit__opensl(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to create audio recorder.");
+ return ma_result_from_OpenSL(resultSL);
}
- }
- (void)lpcstrModule;
- return TRUE;
-}
-static ma_result ma_context_get_device_info__dsound(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
-{
- ma_result result;
- HRESULT hr;
+ /* Set the recording preset before realizing the player. */
+ if (pConfig->opensl.recordingPreset != ma_opensl_recording_preset_default) {
+ resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioRecorderObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDCONFIGURATION, &pRecorderConfig);
+ if (resultSL == SL_RESULT_SUCCESS) {
+ SLint32 recordingPreset = ma_to_recording_preset__opensl(pConfig->opensl.recordingPreset);
+ resultSL = (*pRecorderConfig)->SetConfiguration(pRecorderConfig, SL_ANDROID_KEY_RECORDING_PRESET, &recordingPreset, sizeof(SLint32));
+ if (resultSL != SL_RESULT_SUCCESS) {
+ /* Failed to set the configuration. Just keep going. */
+ }
+ }
+ }
- if (pDeviceID != NULL) {
- ma_context_get_device_info_callback_data__dsound data;
+ resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->Realize((SLObjectItf)pDevice->opensl.pAudioRecorderObj, SL_BOOLEAN_FALSE);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ ma_device_uninit__opensl(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to realize audio recorder.");
+ return ma_result_from_OpenSL(resultSL);
+ }
- /* ID. */
- MA_COPY_MEMORY(pDeviceInfo->id.dsound, pDeviceID->dsound, 16);
+ resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioRecorderObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_RECORD, &pDevice->opensl.pAudioRecorder);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ ma_device_uninit__opensl(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_RECORD interface.");
+ return ma_result_from_OpenSL(resultSL);
+ }
- /* Name / Description. This is retrieved by enumerating over each device until we find that one that matches the input ID. */
- data.pDeviceID = pDeviceID;
- data.pDeviceInfo = pDeviceInfo;
- data.found = MA_FALSE;
- if (deviceType == ma_device_type_playback) {
- ((ma_DirectSoundEnumerateAProc)pContext->dsound.DirectSoundEnumerateA)(ma_context_get_device_info_callback__dsound, &data);
- } else {
- ((ma_DirectSoundCaptureEnumerateAProc)pContext->dsound.DirectSoundCaptureEnumerateA)(ma_context_get_device_info_callback__dsound, &data);
+ resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioRecorderObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &pDevice->opensl.pBufferQueueCapture);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ ma_device_uninit__opensl(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_ANDROIDSIMPLEBUFFERQUEUE interface.");
+ return ma_result_from_OpenSL(resultSL);
}
- if (!data.found) {
- return MA_NO_DEVICE;
+ resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->RegisterCallback((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture, ma_buffer_queue_callback_capture__opensl_android, pDevice);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ ma_device_uninit__opensl(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to register buffer queue callback.");
+ return ma_result_from_OpenSL(resultSL);
}
- } else {
- /* I don't think there's a way to get the name of the default device with DirectSound. In this case we just need to use defaults. */
- /* ID */
- MA_ZERO_MEMORY(pDeviceInfo->id.dsound, 16);
+ /* The internal format is determined by the "pcm" object. */
+ ma_deconstruct_SLDataFormat_PCM__opensl(&pcm, &pDescriptorCapture->format, &pDescriptorCapture->channels, &pDescriptorCapture->sampleRate, pDescriptorCapture->channelMap, ma_countof(pDescriptorCapture->channelMap));
- /* Name / Description */
- if (deviceType == ma_device_type_playback) {
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
- } else {
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
- }
+ /* Buffer. */
+ pDescriptorCapture->periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptorCapture, pDescriptorCapture->sampleRate, pConfig->performanceProfile);
+ pDevice->opensl.currentBufferIndexCapture = 0;
- pDeviceInfo->isDefault = MA_TRUE;
+ bufferSizeInBytes = pDescriptorCapture->periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels) * pDescriptorCapture->periodCount;
+ pDevice->opensl.pBufferCapture = (ma_uint8*)ma_calloc(bufferSizeInBytes, &pDevice->pContext->allocationCallbacks);
+ if (pDevice->opensl.pBufferCapture == NULL) {
+ ma_device_uninit__opensl(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to allocate memory for data buffer.");
+ return MA_OUT_OF_MEMORY;
+ }
+ MA_ZERO_MEMORY(pDevice->opensl.pBufferCapture, bufferSizeInBytes);
}
- /* Retrieving detailed information is slightly different depending on the device type. */
- if (deviceType == ma_device_type_playback) {
- /* Playback. */
- ma_IDirectSound* pDirectSound;
- MA_DSCAPS caps;
- WORD channels;
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ ma_SLDataFormat_PCM pcm;
+ SLDataSource source;
+ SLDataLocator_OutputMix outmixLocator;
+ SLDataSink sink;
+ SLAndroidConfigurationItf pPlayerConfig;
- result = ma_context_create_IDirectSound__dsound(pContext, ma_share_mode_shared, pDeviceID, &pDirectSound);
- if (result != MA_SUCCESS) {
- return result;
+ ma_SLDataFormat_PCM_init__opensl(pDescriptorPlayback->format, pDescriptorPlayback->channels, pDescriptorPlayback->sampleRate, pDescriptorPlayback->channelMap, &pcm);
+
+ resultSL = (*g_maEngineSL)->CreateOutputMix(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pOutputMixObj, 0, NULL, NULL);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ ma_device_uninit__opensl(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to create output mix.");
+ return ma_result_from_OpenSL(resultSL);
}
- MA_ZERO_OBJECT(&caps);
- caps.dwSize = sizeof(caps);
- hr = ma_IDirectSound_GetCaps(pDirectSound, &caps);
- if (FAILED(hr)) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_GetCaps() failed for playback device.", ma_result_from_HRESULT(hr));
+ resultSL = MA_OPENSL_OBJ(pDevice->opensl.pOutputMixObj)->Realize((SLObjectItf)pDevice->opensl.pOutputMixObj, SL_BOOLEAN_FALSE);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ ma_device_uninit__opensl(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to realize output mix object.");
+ return ma_result_from_OpenSL(resultSL);
}
+ resultSL = MA_OPENSL_OBJ(pDevice->opensl.pOutputMixObj)->GetInterface((SLObjectItf)pDevice->opensl.pOutputMixObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_OUTPUTMIX, &pDevice->opensl.pOutputMix);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ ma_device_uninit__opensl(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_OUTPUTMIX interface.");
+ return ma_result_from_OpenSL(resultSL);
+ }
- /* Channels. Only a single channel count is reported for DirectSound. */
- if ((caps.dwFlags & MA_DSCAPS_PRIMARYSTEREO) != 0) {
- /* It supports at least stereo, but could support more. */
- DWORD speakerConfig;
+ /* Set the output device. */
+ if (pDescriptorPlayback->pDeviceID != NULL) {
+ SLuint32 deviceID_OpenSL = pDescriptorPlayback->pDeviceID->opensl;
+ MA_OPENSL_OUTPUTMIX(pDevice->opensl.pOutputMix)->ReRoute((SLOutputMixItf)pDevice->opensl.pOutputMix, 1, &deviceID_OpenSL);
+ }
- channels = 2;
+ queue.numBuffers = pDescriptorPlayback->periodCount;
- /* Look at the speaker configuration to get a better idea on the channel count. */
- hr = ma_IDirectSound_GetSpeakerConfig(pDirectSound, &speakerConfig);
- if (SUCCEEDED(hr)) {
- ma_get_channels_from_speaker_config__dsound(speakerConfig, &channels, NULL);
- }
- } else {
- /* It does not support stereo, which means we are stuck with mono. */
- channels = 1;
+ source.pLocator = &queue;
+ source.pFormat = (SLDataFormat_PCM*)&pcm;
+
+ outmixLocator.locatorType = SL_DATALOCATOR_OUTPUTMIX;
+ outmixLocator.outputMix = (SLObjectItf)pDevice->opensl.pOutputMixObj;
+
+ sink.pLocator = &outmixLocator;
+ sink.pFormat = NULL;
+
+ resultSL = (*g_maEngineSL)->CreateAudioPlayer(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioPlayerObj, &source, &sink, ma_countof(itfIDs), itfIDs, itfIDsRequired);
+ if (resultSL == SL_RESULT_CONTENT_UNSUPPORTED || resultSL == SL_RESULT_PARAMETER_INVALID) {
+ /* Unsupported format. Fall back to something safer and try again. If this fails, just abort. */
+ pcm.formatType = SL_DATAFORMAT_PCM;
+ pcm.numChannels = 2;
+ ((SLDataFormat_PCM*)&pcm)->samplesPerSec = SL_SAMPLINGRATE_16;
+ pcm.bitsPerSample = 16;
+ pcm.containerSize = pcm.bitsPerSample; /* Always tightly packed for now. */
+ pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
+ resultSL = (*g_maEngineSL)->CreateAudioPlayer(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioPlayerObj, &source, &sink, ma_countof(itfIDs), itfIDs, itfIDsRequired);
}
+ if (resultSL != SL_RESULT_SUCCESS) {
+ ma_device_uninit__opensl(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to create audio player.");
+ return ma_result_from_OpenSL(resultSL);
+ }
- /*
- In DirectSound, our native formats are centered around sample rates. All formats are supported, and we're only reporting a single channel
- count. However, DirectSound can report a range of supported sample rates. We're only going to include standard rates known by miniaudio
- in order to keep the size of this within reason.
- */
- if ((caps.dwFlags & MA_DSCAPS_CONTINUOUSRATE) != 0) {
- /* Multiple sample rates are supported. We'll report in order of our preferred sample rates. */
- size_t iStandardSampleRate;
- for (iStandardSampleRate = 0; iStandardSampleRate < ma_countof(g_maStandardSampleRatePriorities); iStandardSampleRate += 1) {
- ma_uint32 sampleRate = g_maStandardSampleRatePriorities[iStandardSampleRate];
- if (sampleRate >= caps.dwMinSecondarySampleRate && sampleRate <= caps.dwMaxSecondarySampleRate) {
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = ma_format_unknown;
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = channels;
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = sampleRate;
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = 0;
- pDeviceInfo->nativeDataFormatCount += 1;
+
+ /* Set the stream type before realizing the player. */
+ if (pConfig->opensl.streamType != ma_opensl_stream_type_default) {
+ resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioPlayerObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDCONFIGURATION, &pPlayerConfig);
+ if (resultSL == SL_RESULT_SUCCESS) {
+ SLint32 streamType = ma_to_stream_type__opensl(pConfig->opensl.streamType);
+ resultSL = (*pPlayerConfig)->SetConfiguration(pPlayerConfig, SL_ANDROID_KEY_STREAM_TYPE, &streamType, sizeof(SLint32));
+ if (resultSL != SL_RESULT_SUCCESS) {
+ /* Failed to set the configuration. Just keep going. */
}
}
- } else {
- /* Only a single sample rate is supported. */
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = ma_format_unknown;
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = channels;
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = caps.dwMaxSecondarySampleRate;
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = 0;
- pDeviceInfo->nativeDataFormatCount += 1;
}
- ma_IDirectSound_Release(pDirectSound);
- } else {
- /*
- Capture. This is a little different to playback due to the say the supported formats are reported. Technically capture
- devices can support a number of different formats, but for simplicity and consistency with ma_device_init() I'm just
- reporting the best format.
- */
- ma_IDirectSoundCapture* pDirectSoundCapture;
- WORD channels;
- WORD bitsPerSample;
- DWORD sampleRate;
-
- result = ma_context_create_IDirectSoundCapture__dsound(pContext, ma_share_mode_shared, pDeviceID, &pDirectSoundCapture);
- if (result != MA_SUCCESS) {
- return result;
+ resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->Realize((SLObjectItf)pDevice->opensl.pAudioPlayerObj, SL_BOOLEAN_FALSE);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ ma_device_uninit__opensl(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to realize audio player.");
+ return ma_result_from_OpenSL(resultSL);
}
- result = ma_context_get_format_info_for_IDirectSoundCapture__dsound(pContext, pDirectSoundCapture, &channels, &bitsPerSample, &sampleRate);
- if (result != MA_SUCCESS) {
- ma_IDirectSoundCapture_Release(pDirectSoundCapture);
- return result;
+ resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioPlayerObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_PLAY, &pDevice->opensl.pAudioPlayer);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ ma_device_uninit__opensl(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_PLAY interface.");
+ return ma_result_from_OpenSL(resultSL);
}
- ma_IDirectSoundCapture_Release(pDirectSoundCapture);
+ resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioPlayerObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &pDevice->opensl.pBufferQueuePlayback);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ ma_device_uninit__opensl(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_ANDROIDSIMPLEBUFFERQUEUE interface.");
+ return ma_result_from_OpenSL(resultSL);
+ }
- /* The format is always an integer format and is based on the bits per sample. */
- if (bitsPerSample == 8) {
- pDeviceInfo->formats[0] = ma_format_u8;
- } else if (bitsPerSample == 16) {
- pDeviceInfo->formats[0] = ma_format_s16;
- } else if (bitsPerSample == 24) {
- pDeviceInfo->formats[0] = ma_format_s24;
- } else if (bitsPerSample == 32) {
- pDeviceInfo->formats[0] = ma_format_s32;
- } else {
- return MA_FORMAT_NOT_SUPPORTED;
+ resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->RegisterCallback((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback, ma_buffer_queue_callback_playback__opensl_android, pDevice);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ ma_device_uninit__opensl(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to register buffer queue callback.");
+ return ma_result_from_OpenSL(resultSL);
}
- pDeviceInfo->nativeDataFormats[0].channels = channels;
- pDeviceInfo->nativeDataFormats[0].sampleRate = sampleRate;
- pDeviceInfo->nativeDataFormats[0].flags = 0;
- pDeviceInfo->nativeDataFormatCount = 1;
+ /* The internal format is determined by the "pcm" object. */
+ ma_deconstruct_SLDataFormat_PCM__opensl(&pcm, &pDescriptorPlayback->format, &pDescriptorPlayback->channels, &pDescriptorPlayback->sampleRate, pDescriptorPlayback->channelMap, ma_countof(pDescriptorPlayback->channelMap));
+
+ /* Buffer. */
+ pDescriptorPlayback->periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptorPlayback, pDescriptorPlayback->sampleRate, pConfig->performanceProfile);
+ pDevice->opensl.currentBufferIndexPlayback = 0;
+
+ bufferSizeInBytes = pDescriptorPlayback->periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorPlayback->format, pDescriptorPlayback->channels) * pDescriptorPlayback->periodCount;
+ pDevice->opensl.pBufferPlayback = (ma_uint8*)ma_calloc(bufferSizeInBytes, &pDevice->pContext->allocationCallbacks);
+ if (pDevice->opensl.pBufferPlayback == NULL) {
+ ma_device_uninit__opensl(pDevice);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to allocate memory for data buffer.");
+ return MA_OUT_OF_MEMORY;
+ }
+ MA_ZERO_MEMORY(pDevice->opensl.pBufferPlayback, bufferSizeInBytes);
}
return MA_SUCCESS;
+#else
+ return MA_NO_BACKEND; /* Non-Android implementations are not supported. */
+#endif
}
-
-
-static ma_result ma_device_uninit__dsound(ma_device* pDevice)
+static ma_result ma_device_start__opensl(ma_device* pDevice)
{
+ SLresult resultSL;
+ size_t periodSizeInBytes;
+ ma_uint32 iPeriod;
+
MA_ASSERT(pDevice != NULL);
- if (pDevice->dsound.pCaptureBuffer != NULL) {
- ma_IDirectSoundCaptureBuffer_Release((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer);
- }
- if (pDevice->dsound.pCapture != NULL) {
- ma_IDirectSoundCapture_Release((ma_IDirectSoundCapture*)pDevice->dsound.pCapture);
+ MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to start the device. */
+ if (g_maOpenSLInitCounter == 0) {
+ return MA_INVALID_OPERATION;
}
- if (pDevice->dsound.pPlaybackBuffer != NULL) {
- ma_IDirectSoundBuffer_Release((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer);
- }
- if (pDevice->dsound.pPlaybackPrimaryBuffer != NULL) {
- ma_IDirectSoundBuffer_Release((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackPrimaryBuffer);
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ resultSL = MA_OPENSL_RECORD(pDevice->opensl.pAudioRecorder)->SetRecordState((SLRecordItf)pDevice->opensl.pAudioRecorder, SL_RECORDSTATE_RECORDING);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to start internal capture device.");
+ return ma_result_from_OpenSL(resultSL);
+ }
+
+ periodSizeInBytes = pDevice->capture.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
+ for (iPeriod = 0; iPeriod < pDevice->capture.internalPeriods; ++iPeriod) {
+ resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture, pDevice->opensl.pBufferCapture + (periodSizeInBytes * iPeriod), periodSizeInBytes);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ MA_OPENSL_RECORD(pDevice->opensl.pAudioRecorder)->SetRecordState((SLRecordItf)pDevice->opensl.pAudioRecorder, SL_RECORDSTATE_STOPPED);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to enqueue buffer for capture device.");
+ return ma_result_from_OpenSL(resultSL);
+ }
+ }
}
- if (pDevice->dsound.pPlayback != NULL) {
- ma_IDirectSound_Release((ma_IDirectSound*)pDevice->dsound.pPlayback);
+
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ resultSL = MA_OPENSL_PLAY(pDevice->opensl.pAudioPlayer)->SetPlayState((SLPlayItf)pDevice->opensl.pAudioPlayer, SL_PLAYSTATE_PLAYING);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to start internal playback device.");
+ return ma_result_from_OpenSL(resultSL);
+ }
+
+ /* In playback mode (no duplex) we need to load some initial buffers. In duplex mode we need to enqueu silent buffers. */
+ if (pDevice->type == ma_device_type_duplex) {
+ MA_ZERO_MEMORY(pDevice->opensl.pBufferPlayback, pDevice->playback.internalPeriodSizeInFrames * pDevice->playback.internalPeriods * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels));
+ } else {
+ ma_device__read_frames_from_client(pDevice, pDevice->playback.internalPeriodSizeInFrames * pDevice->playback.internalPeriods, pDevice->opensl.pBufferPlayback);
+ }
+
+ periodSizeInBytes = pDevice->playback.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
+ for (iPeriod = 0; iPeriod < pDevice->playback.internalPeriods; ++iPeriod) {
+ resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback, pDevice->opensl.pBufferPlayback + (periodSizeInBytes * iPeriod), periodSizeInBytes);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ MA_OPENSL_PLAY(pDevice->opensl.pAudioPlayer)->SetPlayState((SLPlayItf)pDevice->opensl.pAudioPlayer, SL_PLAYSTATE_STOPPED);
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to enqueue buffer for playback device.");
+ return ma_result_from_OpenSL(resultSL);
+ }
+ }
}
return MA_SUCCESS;
}
-static ma_result ma_config_to_WAVEFORMATEXTENSIBLE(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, const ma_channel* pChannelMap, WAVEFORMATEXTENSIBLE* pWF)
+static ma_result ma_device_drain__opensl(ma_device* pDevice, ma_device_type deviceType)
{
- GUID subformat;
+ SLAndroidSimpleBufferQueueItf pBufferQueue;
- if (format == ma_format_unknown) {
- format = MA_DEFAULT_FORMAT;
+ MA_ASSERT(deviceType == ma_device_type_capture || deviceType == ma_device_type_playback);
+
+ if (pDevice->type == ma_device_type_capture) {
+ pBufferQueue = (SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture;
+ pDevice->opensl.isDrainingCapture = MA_TRUE;
+ } else {
+ pBufferQueue = (SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback;
+ pDevice->opensl.isDrainingPlayback = MA_TRUE;
}
- if (channels == 0) {
- channels = MA_DEFAULT_CHANNELS;
+ for (;;) {
+ SLAndroidSimpleBufferQueueState state;
+
+ MA_OPENSL_BUFFERQUEUE(pBufferQueue)->GetState(pBufferQueue, &state);
+ if (state.count == 0) {
+ break;
+ }
+
+ ma_sleep(10);
}
- if (sampleRate == 0) {
- sampleRate = MA_DEFAULT_SAMPLE_RATE;
+ if (pDevice->type == ma_device_type_capture) {
+ pDevice->opensl.isDrainingCapture = MA_FALSE;
+ } else {
+ pDevice->opensl.isDrainingPlayback = MA_FALSE;
}
- switch (format)
- {
- case ma_format_u8:
- case ma_format_s16:
- case ma_format_s24:
- /*case ma_format_s24_32:*/
- case ma_format_s32:
- {
- subformat = MA_GUID_KSDATAFORMAT_SUBTYPE_PCM;
- } break;
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_stop__opensl(ma_device* pDevice)
+{
+ SLresult resultSL;
+
+ MA_ASSERT(pDevice != NULL);
+
+ MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it before stopping/uninitializing the device. */
+ if (g_maOpenSLInitCounter == 0) {
+ return MA_INVALID_OPERATION;
+ }
+
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ ma_device_drain__opensl(pDevice, ma_device_type_capture);
+
+ resultSL = MA_OPENSL_RECORD(pDevice->opensl.pAudioRecorder)->SetRecordState((SLRecordItf)pDevice->opensl.pAudioRecorder, SL_RECORDSTATE_STOPPED);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to stop internal capture device.");
+ return ma_result_from_OpenSL(resultSL);
+ }
+
+ MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->Clear((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture);
+ }
- case ma_format_f32:
- {
- subformat = MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
- } break;
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ ma_device_drain__opensl(pDevice, ma_device_type_playback);
- default:
- return MA_FORMAT_NOT_SUPPORTED;
+ resultSL = MA_OPENSL_PLAY(pDevice->opensl.pAudioPlayer)->SetPlayState((SLPlayItf)pDevice->opensl.pAudioPlayer, SL_PLAYSTATE_STOPPED);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to stop internal playback device.");
+ return ma_result_from_OpenSL(resultSL);
+ }
+
+ MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->Clear((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback);
}
- MA_ZERO_OBJECT(pWF);
- pWF->Format.cbSize = sizeof(*pWF);
- pWF->Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
- pWF->Format.nChannels = (WORD)channels;
- pWF->Format.nSamplesPerSec = (DWORD)sampleRate;
- pWF->Format.wBitsPerSample = (WORD)(ma_get_bytes_per_sample(format)*8);
- pWF->Format.nBlockAlign = (WORD)(pWF->Format.nChannels * pWF->Format.wBitsPerSample / 8);
- pWF->Format.nAvgBytesPerSec = pWF->Format.nBlockAlign * pWF->Format.nSamplesPerSec;
- pWF->Samples.wValidBitsPerSample = pWF->Format.wBitsPerSample;
- pWF->dwChannelMask = ma_channel_map_to_channel_mask__win32(pChannelMap, channels);
- pWF->SubFormat = subformat;
+ /* Make sure the client is aware that the device has stopped. There may be an OpenSL|ES callback for this, but I haven't found it. */
+ ma_device__on_notification_stopped(pDevice);
return MA_SUCCESS;
}
-static ma_uint32 ma_calculate_period_size_in_frames_from_descriptor__dsound(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile)
+
+static ma_result ma_context_uninit__opensl(ma_context* pContext)
{
- /* DirectSound has a minimum period size of 20ms. */
- ma_uint32 minPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(20, nativeSampleRate);
- ma_uint32 periodSizeInFrames;
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(pContext->backend == ma_backend_opensl);
+ (void)pContext;
- periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, nativeSampleRate, performanceProfile);
- if (periodSizeInFrames < minPeriodSizeInFrames) {
- periodSizeInFrames = minPeriodSizeInFrames;
+ /* Uninit global data. */
+ ma_spinlock_lock(&g_maOpenSLSpinlock);
+ {
+ MA_ASSERT(g_maOpenSLInitCounter > 0); /* If you've triggered this, it means you have ma_context_init/uninit mismatch. Each successful call to ma_context_init() must be matched up with a call to ma_context_uninit(). */
+
+ g_maOpenSLInitCounter -= 1;
+ if (g_maOpenSLInitCounter == 0) {
+ (*g_maEngineObjectSL)->Destroy(g_maEngineObjectSL);
+ }
}
+ ma_spinlock_unlock(&g_maOpenSLSpinlock);
- return periodSizeInFrames;
+ return MA_SUCCESS;
}
-static ma_result ma_device_init__dsound(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+static ma_result ma_dlsym_SLInterfaceID__opensl(ma_context* pContext, const char* pName, ma_handle* pHandle)
{
- ma_result result;
- HRESULT hr;
+ /* We need to return an error if the symbol cannot be found. This is important because there have been reports that some symbols do not exist. */
+ ma_handle* p = (ma_handle*)ma_dlsym(pContext, pContext->opensl.libOpenSLES, pName);
+ if (p == NULL) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_INFO, "[OpenSL] Cannot find symbol %s", pName);
+ return MA_NO_BACKEND;
+ }
- MA_ASSERT(pDevice != NULL);
+ *pHandle = *p;
+ return MA_SUCCESS;
+}
- MA_ZERO_OBJECT(&pDevice->dsound);
+static ma_result ma_context_init_engine_nolock__opensl(ma_context* pContext)
+{
+ g_maOpenSLInitCounter += 1;
+ if (g_maOpenSLInitCounter == 1) {
+ SLresult resultSL;
- if (pConfig->deviceType == ma_device_type_loopback) {
- return MA_DEVICE_TYPE_NOT_SUPPORTED;
- }
+ resultSL = ((ma_slCreateEngine_proc)pContext->opensl.slCreateEngine)(&g_maEngineObjectSL, 0, NULL, 0, NULL, NULL);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ g_maOpenSLInitCounter -= 1;
+ return ma_result_from_OpenSL(resultSL);
+ }
- /*
- Unfortunately DirectSound uses different APIs and data structures for playback and catpure devices. We need to initialize
- the capture device first because we'll want to match it's buffer size and period count on the playback side if we're using
- full-duplex mode.
- */
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
- WAVEFORMATEXTENSIBLE wf;
- MA_DSCBUFFERDESC descDS;
- ma_uint32 periodSizeInFrames;
- ma_uint32 periodCount;
- char rawdata[1024]; /* <-- Ugly hack to avoid a malloc() due to a crappy DirectSound API. */
- WAVEFORMATEXTENSIBLE* pActualFormat;
+ (*g_maEngineObjectSL)->Realize(g_maEngineObjectSL, SL_BOOLEAN_FALSE);
- result = ma_config_to_WAVEFORMATEXTENSIBLE(pDescriptorCapture->format, pDescriptorCapture->channels, pDescriptorCapture->sampleRate, pDescriptorCapture->channelMap, &wf);
- if (result != MA_SUCCESS) {
- return result;
+ resultSL = (*g_maEngineObjectSL)->GetInterface(g_maEngineObjectSL, (SLInterfaceID)pContext->opensl.SL_IID_ENGINE, &g_maEngineSL);
+ if (resultSL != SL_RESULT_SUCCESS) {
+ (*g_maEngineObjectSL)->Destroy(g_maEngineObjectSL);
+ g_maOpenSLInitCounter -= 1;
+ return ma_result_from_OpenSL(resultSL);
}
+ }
- result = ma_context_create_IDirectSoundCapture__dsound(pDevice->pContext, pDescriptorCapture->shareMode, pDescriptorCapture->pDeviceID, (ma_IDirectSoundCapture**)&pDevice->dsound.pCapture);
- if (result != MA_SUCCESS) {
- ma_device_uninit__dsound(pDevice);
- return result;
- }
+ return MA_SUCCESS;
+}
- result = ma_context_get_format_info_for_IDirectSoundCapture__dsound(pDevice->pContext, (ma_IDirectSoundCapture*)pDevice->dsound.pCapture, &wf.Format.nChannels, &wf.Format.wBitsPerSample, &wf.Format.nSamplesPerSec);
- if (result != MA_SUCCESS) {
- ma_device_uninit__dsound(pDevice);
- return result;
- }
+static ma_result ma_context_init__opensl(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+{
+ ma_result result;
- wf.Format.nBlockAlign = (WORD)(wf.Format.nChannels * wf.Format.wBitsPerSample / 8);
- wf.Format.nAvgBytesPerSec = wf.Format.nBlockAlign * wf.Format.nSamplesPerSec;
- wf.Samples.wValidBitsPerSample = wf.Format.wBitsPerSample;
- wf.SubFormat = MA_GUID_KSDATAFORMAT_SUBTYPE_PCM;
+#if !defined(MA_NO_RUNTIME_LINKING)
+ size_t i;
+ const char* libOpenSLESNames[] = {
+ "libOpenSLES.so"
+ };
+#endif
- /* The size of the buffer must be a clean multiple of the period count. */
- periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__dsound(pDescriptorCapture, wf.Format.nSamplesPerSec, pConfig->performanceProfile);
- periodCount = (pDescriptorCapture->periodCount > 0) ? pDescriptorCapture->periodCount : MA_DEFAULT_PERIODS;
+ MA_ASSERT(pContext != NULL);
- MA_ZERO_OBJECT(&descDS);
- descDS.dwSize = sizeof(descDS);
- descDS.dwFlags = 0;
- descDS.dwBufferBytes = periodSizeInFrames * periodCount * wf.Format.nBlockAlign;
- descDS.lpwfxFormat = (WAVEFORMATEX*)&wf;
- hr = ma_IDirectSoundCapture_CreateCaptureBuffer((ma_IDirectSoundCapture*)pDevice->dsound.pCapture, &descDS, (ma_IDirectSoundCaptureBuffer**)&pDevice->dsound.pCaptureBuffer, NULL);
- if (FAILED(hr)) {
- ma_device_uninit__dsound(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCapture_CreateCaptureBuffer() failed for capture device.", ma_result_from_HRESULT(hr));
- }
+ (void)pConfig;
- /* Get the _actual_ properties of the buffer. */
- pActualFormat = (WAVEFORMATEXTENSIBLE*)rawdata;
- hr = ma_IDirectSoundCaptureBuffer_GetFormat((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, (WAVEFORMATEX*)pActualFormat, sizeof(rawdata), NULL);
- if (FAILED(hr)) {
- ma_device_uninit__dsound(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to retrieve the actual format of the capture device's buffer.", ma_result_from_HRESULT(hr));
+#if !defined(MA_NO_RUNTIME_LINKING)
+ /*
+ Dynamically link against libOpenSLES.so. I have now had multiple reports that SL_IID_ANDROIDSIMPLEBUFFERQUEUE cannot be found. One
+ report was happening at compile time and another at runtime. To try working around this, I'm going to link to libOpenSLES at runtime
+ and extract the symbols rather than reference them directly. This should, hopefully, fix these issues as the compiler won't see any
+ references to the symbols and will hopefully skip the checks.
+ */
+ for (i = 0; i < ma_countof(libOpenSLESNames); i += 1) {
+ pContext->opensl.libOpenSLES = ma_dlopen(pContext, libOpenSLESNames[i]);
+ if (pContext->opensl.libOpenSLES != NULL) {
+ break;
}
+ }
- /* We can now start setting the output data formats. */
- pDescriptorCapture->format = ma_format_from_WAVEFORMATEX((WAVEFORMATEX*)pActualFormat);
- pDescriptorCapture->channels = pActualFormat->Format.nChannels;
- pDescriptorCapture->sampleRate = pActualFormat->Format.nSamplesPerSec;
+ if (pContext->opensl.libOpenSLES == NULL) {
+ ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_INFO, "[OpenSL] Could not find libOpenSLES.so");
+ return MA_NO_BACKEND;
+ }
- /* Get the native channel map based on the channel mask. */
- if (pActualFormat->Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
- ma_channel_mask_to_channel_map__win32(pActualFormat->dwChannelMask, pDescriptorCapture->channels, pDescriptorCapture->channelMap);
- } else {
- ma_channel_mask_to_channel_map__win32(wf.dwChannelMask, pDescriptorCapture->channels, pDescriptorCapture->channelMap);
- }
+ result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_ENGINE", &pContext->opensl.SL_IID_ENGINE);
+ if (result != MA_SUCCESS) {
+ ma_dlclose(pContext, pContext->opensl.libOpenSLES);
+ return result;
+ }
- /*
- After getting the actual format the size of the buffer in frames may have actually changed. However, we want this to be as close to what the
- user has asked for as possible, so let's go ahead and release the old capture buffer and create a new one in this case.
- */
- if (periodSizeInFrames != (descDS.dwBufferBytes / ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels) / periodCount)) {
- descDS.dwBufferBytes = periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels) * periodCount;
- ma_IDirectSoundCaptureBuffer_Release((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer);
+ result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_AUDIOIODEVICECAPABILITIES", &pContext->opensl.SL_IID_AUDIOIODEVICECAPABILITIES);
+ if (result != MA_SUCCESS) {
+ ma_dlclose(pContext, pContext->opensl.libOpenSLES);
+ return result;
+ }
- hr = ma_IDirectSoundCapture_CreateCaptureBuffer((ma_IDirectSoundCapture*)pDevice->dsound.pCapture, &descDS, (ma_IDirectSoundCaptureBuffer**)&pDevice->dsound.pCaptureBuffer, NULL);
- if (FAILED(hr)) {
- ma_device_uninit__dsound(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Second attempt at IDirectSoundCapture_CreateCaptureBuffer() failed for capture device.", ma_result_from_HRESULT(hr));
- }
- }
+ result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_ANDROIDSIMPLEBUFFERQUEUE", &pContext->opensl.SL_IID_ANDROIDSIMPLEBUFFERQUEUE);
+ if (result != MA_SUCCESS) {
+ ma_dlclose(pContext, pContext->opensl.libOpenSLES);
+ return result;
+ }
- /* DirectSound should give us a buffer exactly the size we asked for. */
- pDescriptorCapture->periodSizeInFrames = periodSizeInFrames;
- pDescriptorCapture->periodCount = periodCount;
+ result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_RECORD", &pContext->opensl.SL_IID_RECORD);
+ if (result != MA_SUCCESS) {
+ ma_dlclose(pContext, pContext->opensl.libOpenSLES);
+ return result;
}
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- WAVEFORMATEXTENSIBLE wf;
- MA_DSBUFFERDESC descDSPrimary;
- MA_DSCAPS caps;
- char rawdata[1024]; /* <-- Ugly hack to avoid a malloc() due to a crappy DirectSound API. */
- WAVEFORMATEXTENSIBLE* pActualFormat;
- ma_uint32 periodSizeInFrames;
- ma_uint32 periodCount;
- MA_DSBUFFERDESC descDS;
+ result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_PLAY", &pContext->opensl.SL_IID_PLAY);
+ if (result != MA_SUCCESS) {
+ ma_dlclose(pContext, pContext->opensl.libOpenSLES);
+ return result;
+ }
- result = ma_config_to_WAVEFORMATEXTENSIBLE(pDescriptorPlayback->format, pDescriptorPlayback->channels, pDescriptorPlayback->sampleRate, pDescriptorPlayback->channelMap, &wf);
- if (result != MA_SUCCESS) {
- return result;
- }
+ result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_OUTPUTMIX", &pContext->opensl.SL_IID_OUTPUTMIX);
+ if (result != MA_SUCCESS) {
+ ma_dlclose(pContext, pContext->opensl.libOpenSLES);
+ return result;
+ }
- result = ma_context_create_IDirectSound__dsound(pDevice->pContext, pDescriptorPlayback->shareMode, pDescriptorPlayback->pDeviceID, (ma_IDirectSound**)&pDevice->dsound.pPlayback);
- if (result != MA_SUCCESS) {
- ma_device_uninit__dsound(pDevice);
- return result;
- }
+ result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_ANDROIDCONFIGURATION", &pContext->opensl.SL_IID_ANDROIDCONFIGURATION);
+ if (result != MA_SUCCESS) {
+ ma_dlclose(pContext, pContext->opensl.libOpenSLES);
+ return result;
+ }
- MA_ZERO_OBJECT(&descDSPrimary);
- descDSPrimary.dwSize = sizeof(MA_DSBUFFERDESC);
- descDSPrimary.dwFlags = MA_DSBCAPS_PRIMARYBUFFER | MA_DSBCAPS_CTRLVOLUME;
- hr = ma_IDirectSound_CreateSoundBuffer((ma_IDirectSound*)pDevice->dsound.pPlayback, &descDSPrimary, (ma_IDirectSoundBuffer**)&pDevice->dsound.pPlaybackPrimaryBuffer, NULL);
- if (FAILED(hr)) {
- ma_device_uninit__dsound(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_CreateSoundBuffer() failed for playback device's primary buffer.", ma_result_from_HRESULT(hr));
- }
+ pContext->opensl.slCreateEngine = (ma_proc)ma_dlsym(pContext, pContext->opensl.libOpenSLES, "slCreateEngine");
+ if (pContext->opensl.slCreateEngine == NULL) {
+ ma_dlclose(pContext, pContext->opensl.libOpenSLES);
+ ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_INFO, "[OpenSL] Cannot find symbol slCreateEngine.");
+ return MA_NO_BACKEND;
+ }
+#else
+ pContext->opensl.SL_IID_ENGINE = (ma_handle)SL_IID_ENGINE;
+ pContext->opensl.SL_IID_AUDIOIODEVICECAPABILITIES = (ma_handle)SL_IID_AUDIOIODEVICECAPABILITIES;
+ pContext->opensl.SL_IID_ANDROIDSIMPLEBUFFERQUEUE = (ma_handle)SL_IID_ANDROIDSIMPLEBUFFERQUEUE;
+ pContext->opensl.SL_IID_RECORD = (ma_handle)SL_IID_RECORD;
+ pContext->opensl.SL_IID_PLAY = (ma_handle)SL_IID_PLAY;
+ pContext->opensl.SL_IID_OUTPUTMIX = (ma_handle)SL_IID_OUTPUTMIX;
+ pContext->opensl.SL_IID_ANDROIDCONFIGURATION = (ma_handle)SL_IID_ANDROIDCONFIGURATION;
+ pContext->opensl.slCreateEngine = (ma_proc)slCreateEngine;
+#endif
- /* We may want to make some adjustments to the format if we are using defaults. */
- MA_ZERO_OBJECT(&caps);
- caps.dwSize = sizeof(caps);
- hr = ma_IDirectSound_GetCaps((ma_IDirectSound*)pDevice->dsound.pPlayback, &caps);
- if (FAILED(hr)) {
- ma_device_uninit__dsound(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_GetCaps() failed for playback device.", ma_result_from_HRESULT(hr));
- }
+ /* Initialize global data first if applicable. */
+ ma_spinlock_lock(&g_maOpenSLSpinlock);
+ {
+ result = ma_context_init_engine_nolock__opensl(pContext);
+ }
+ ma_spinlock_unlock(&g_maOpenSLSpinlock);
- if (pDescriptorPlayback->channels == 0) {
- if ((caps.dwFlags & MA_DSCAPS_PRIMARYSTEREO) != 0) {
- DWORD speakerConfig;
+ if (result != MA_SUCCESS) {
+ ma_dlclose(pContext, pContext->opensl.libOpenSLES);
+ ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_INFO, "[OpenSL] Failed to initialize OpenSL engine.");
+ return result;
+ }
- /* It supports at least stereo, but could support more. */
- wf.Format.nChannels = 2;
+ pCallbacks->onContextInit = ma_context_init__opensl;
+ pCallbacks->onContextUninit = ma_context_uninit__opensl;
+ pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__opensl;
+ pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__opensl;
+ pCallbacks->onDeviceInit = ma_device_init__opensl;
+ pCallbacks->onDeviceUninit = ma_device_uninit__opensl;
+ pCallbacks->onDeviceStart = ma_device_start__opensl;
+ pCallbacks->onDeviceStop = ma_device_stop__opensl;
+ pCallbacks->onDeviceRead = NULL; /* Not needed because OpenSL|ES is asynchronous. */
+ pCallbacks->onDeviceWrite = NULL; /* Not needed because OpenSL|ES is asynchronous. */
+ pCallbacks->onDeviceDataLoop = NULL; /* Not needed because OpenSL|ES is asynchronous. */
- /* Look at the speaker configuration to get a better idea on the channel count. */
- if (SUCCEEDED(ma_IDirectSound_GetSpeakerConfig((ma_IDirectSound*)pDevice->dsound.pPlayback, &speakerConfig))) {
- ma_get_channels_from_speaker_config__dsound(speakerConfig, &wf.Format.nChannels, &wf.dwChannelMask);
- }
- } else {
- /* It does not support stereo, which means we are stuck with mono. */
- wf.Format.nChannels = 1;
- }
- }
+ return MA_SUCCESS;
+}
+#endif /* OpenSL|ES */
- if (pDescriptorPlayback->sampleRate == 0) {
- /* We base the sample rate on the values returned by GetCaps(). */
- if ((caps.dwFlags & MA_DSCAPS_CONTINUOUSRATE) != 0) {
- wf.Format.nSamplesPerSec = ma_get_best_sample_rate_within_range(caps.dwMinSecondarySampleRate, caps.dwMaxSecondarySampleRate);
- } else {
- wf.Format.nSamplesPerSec = caps.dwMaxSecondarySampleRate;
- }
- }
- wf.Format.nBlockAlign = (WORD)(wf.Format.nChannels * wf.Format.wBitsPerSample / 8);
- wf.Format.nAvgBytesPerSec = wf.Format.nBlockAlign * wf.Format.nSamplesPerSec;
+/******************************************************************************
- /*
- From MSDN:
+Web Audio Backend
- The method succeeds even if the hardware does not support the requested format; DirectSound sets the buffer to the closest
- supported format. To determine whether this has happened, an application can call the GetFormat method for the primary buffer
- and compare the result with the format that was requested with the SetFormat method.
- */
- hr = ma_IDirectSoundBuffer_SetFormat((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackPrimaryBuffer, (WAVEFORMATEX*)&wf);
- if (FAILED(hr)) {
- ma_device_uninit__dsound(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to set format of playback device's primary buffer.", ma_result_from_HRESULT(hr));
- }
+******************************************************************************/
+#ifdef MA_HAS_WEBAUDIO
+#include
- /* Get the _actual_ properties of the buffer. */
- pActualFormat = (WAVEFORMATEXTENSIBLE*)rawdata;
- hr = ma_IDirectSoundBuffer_GetFormat((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackPrimaryBuffer, (WAVEFORMATEX*)pActualFormat, sizeof(rawdata), NULL);
- if (FAILED(hr)) {
- ma_device_uninit__dsound(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to retrieve the actual format of the playback device's primary buffer.", ma_result_from_HRESULT(hr));
- }
+static ma_bool32 ma_is_capture_supported__webaudio()
+{
+ return EM_ASM_INT({
+ return (navigator.mediaDevices !== undefined && navigator.mediaDevices.getUserMedia !== undefined);
+ }, 0) != 0; /* Must pass in a dummy argument for C99 compatibility. */
+}
- /* We now have enough information to start setting some output properties. */
- pDescriptorPlayback->format = ma_format_from_WAVEFORMATEX((WAVEFORMATEX*)pActualFormat);
- pDescriptorPlayback->channels = pActualFormat->Format.nChannels;
- pDescriptorPlayback->sampleRate = pActualFormat->Format.nSamplesPerSec;
+#ifdef __cplusplus
+extern "C" {
+#endif
+void EMSCRIPTEN_KEEPALIVE ma_device_process_pcm_frames_capture__webaudio(ma_device* pDevice, int frameCount, float* pFrames)
+{
+ ma_device_handle_backend_data_callback(pDevice, NULL, pFrames, (ma_uint32)frameCount);
+}
- /* Get the internal channel map based on the channel mask. */
- if (pActualFormat->Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
- ma_channel_mask_to_channel_map__win32(pActualFormat->dwChannelMask, pDescriptorPlayback->channels, pDescriptorPlayback->channelMap);
- } else {
- ma_channel_mask_to_channel_map__win32(wf.dwChannelMask, pDescriptorPlayback->channels, pDescriptorPlayback->channelMap);
- }
+void EMSCRIPTEN_KEEPALIVE ma_device_process_pcm_frames_playback__webaudio(ma_device* pDevice, int frameCount, float* pFrames)
+{
+ ma_device_handle_backend_data_callback(pDevice, pFrames, NULL, (ma_uint32)frameCount);
+}
+#ifdef __cplusplus
+}
+#endif
- /* The size of the buffer must be a clean multiple of the period count. */
- periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__dsound(pDescriptorPlayback, pDescriptorPlayback->sampleRate, pConfig->performanceProfile);
- periodCount = (pDescriptorPlayback->periodCount > 0) ? pDescriptorPlayback->periodCount : MA_DEFAULT_PERIODS;
+static ma_result ma_context_enumerate_devices__webaudio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+{
+ ma_bool32 cbResult = MA_TRUE;
- /*
- Meaning of dwFlags (from MSDN):
+ MA_ASSERT(pContext != NULL);
+ MA_ASSERT(callback != NULL);
- DSBCAPS_CTRLPOSITIONNOTIFY
- The buffer has position notification capability.
+ /* Only supporting default devices for now. */
- DSBCAPS_GLOBALFOCUS
- With this flag set, an application using DirectSound can continue to play its buffers if the user switches focus to
- another application, even if the new application uses DirectSound.
+ /* Playback. */
+ if (cbResult) {
+ ma_device_info deviceInfo;
+ MA_ZERO_OBJECT(&deviceInfo);
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
+ deviceInfo.isDefault = MA_TRUE; /* Only supporting default devices. */
+ cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
+ }
- DSBCAPS_GETCURRENTPOSITION2
- In the first version of DirectSound, the play cursor was significantly ahead of the actual playing sound on emulated
- sound cards; it was directly behind the write cursor. Now, if the DSBCAPS_GETCURRENTPOSITION2 flag is specified, the
- application can get a more accurate play cursor.
- */
- MA_ZERO_OBJECT(&descDS);
- descDS.dwSize = sizeof(descDS);
- descDS.dwFlags = MA_DSBCAPS_CTRLPOSITIONNOTIFY | MA_DSBCAPS_GLOBALFOCUS | MA_DSBCAPS_GETCURRENTPOSITION2;
- descDS.dwBufferBytes = periodSizeInFrames * periodCount * ma_get_bytes_per_frame(pDescriptorPlayback->format, pDescriptorPlayback->channels);
- descDS.lpwfxFormat = (WAVEFORMATEX*)&wf;
- hr = ma_IDirectSound_CreateSoundBuffer((ma_IDirectSound*)pDevice->dsound.pPlayback, &descDS, (ma_IDirectSoundBuffer**)&pDevice->dsound.pPlaybackBuffer, NULL);
- if (FAILED(hr)) {
- ma_device_uninit__dsound(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_CreateSoundBuffer() failed for playback device's secondary buffer.", ma_result_from_HRESULT(hr));
+ /* Capture. */
+ if (cbResult) {
+ if (ma_is_capture_supported__webaudio()) {
+ ma_device_info deviceInfo;
+ MA_ZERO_OBJECT(&deviceInfo);
+ ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
+ deviceInfo.isDefault = MA_TRUE; /* Only supporting default devices. */
+ cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
}
-
- /* DirectSound should give us a buffer exactly the size we asked for. */
- pDescriptorPlayback->periodSizeInFrames = periodSizeInFrames;
- pDescriptorPlayback->periodCount = periodCount;
}
return MA_SUCCESS;
}
-
-static ma_result ma_device_data_loop__dsound(ma_device* pDevice)
+static ma_result ma_context_get_device_info__webaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
{
- ma_result result = MA_SUCCESS;
- ma_uint32 bpfDeviceCapture = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
- ma_uint32 bpfDevicePlayback = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
- HRESULT hr;
- DWORD lockOffsetInBytesCapture;
- DWORD lockSizeInBytesCapture;
- DWORD mappedSizeInBytesCapture;
- DWORD mappedDeviceFramesProcessedCapture;
- void* pMappedDeviceBufferCapture;
- DWORD lockOffsetInBytesPlayback;
- DWORD lockSizeInBytesPlayback;
- DWORD mappedSizeInBytesPlayback;
- void* pMappedDeviceBufferPlayback;
- DWORD prevReadCursorInBytesCapture = 0;
- DWORD prevPlayCursorInBytesPlayback = 0;
- ma_bool32 physicalPlayCursorLoopFlagPlayback = 0;
- DWORD virtualWriteCursorInBytesPlayback = 0;
- ma_bool32 virtualWriteCursorLoopFlagPlayback = 0;
- ma_bool32 isPlaybackDeviceStarted = MA_FALSE;
- ma_uint32 framesWrittenToPlaybackDevice = 0; /* For knowing whether or not the playback device needs to be started. */
- ma_uint32 waitTimeInMilliseconds = 1;
-
- MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(pContext != NULL);
- /* The first thing to do is start the capture device. The playback device is only started after the first period is written. */
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- if (FAILED(ma_IDirectSoundCaptureBuffer_Start((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, MA_DSCBSTART_LOOPING))) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCaptureBuffer_Start() failed.", MA_FAILED_TO_START_BACKEND_DEVICE);
- }
+ if (deviceType == ma_device_type_capture && !ma_is_capture_supported__webaudio()) {
+ return MA_NO_DEVICE;
}
- while (ma_device_get_state(pDevice) == MA_STATE_STARTED) {
- switch (pDevice->type)
- {
- case ma_device_type_duplex:
- {
- DWORD physicalCaptureCursorInBytes;
- DWORD physicalReadCursorInBytes;
- hr = ma_IDirectSoundCaptureBuffer_GetCurrentPosition((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, &physicalCaptureCursorInBytes, &physicalReadCursorInBytes);
- if (FAILED(hr)) {
- return ma_result_from_HRESULT(hr);
- }
+ MA_ZERO_MEMORY(pDeviceInfo->id.webaudio, sizeof(pDeviceInfo->id.webaudio));
- /* If nothing is available we just sleep for a bit and return from this iteration. */
- if (physicalReadCursorInBytes == prevReadCursorInBytesCapture) {
- ma_sleep(waitTimeInMilliseconds);
- continue; /* Nothing is available in the capture buffer. */
- }
+ /* Only supporting default devices for now. */
+ (void)pDeviceID;
+ if (deviceType == ma_device_type_playback) {
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
+ } else {
+ ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
+ }
- /*
- The current position has moved. We need to map all of the captured samples and write them to the playback device, making sure
- we don't return until every frame has been copied over.
- */
- if (prevReadCursorInBytesCapture < physicalReadCursorInBytes) {
- /* The capture position has not looped. This is the simple case. */
- lockOffsetInBytesCapture = prevReadCursorInBytesCapture;
- lockSizeInBytesCapture = (physicalReadCursorInBytes - prevReadCursorInBytesCapture);
- } else {
- /*
- The capture position has looped. This is the more complex case. Map to the end of the buffer. If this does not return anything,
- do it again from the start.
- */
- if (prevReadCursorInBytesCapture < pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture) {
- /* Lock up to the end of the buffer. */
- lockOffsetInBytesCapture = prevReadCursorInBytesCapture;
- lockSizeInBytesCapture = (pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture) - prevReadCursorInBytesCapture;
- } else {
- /* Lock starting from the start of the buffer. */
- lockOffsetInBytesCapture = 0;
- lockSizeInBytesCapture = physicalReadCursorInBytes;
- }
- }
+ /* Only supporting default devices. */
+ pDeviceInfo->isDefault = MA_TRUE;
- if (lockSizeInBytesCapture == 0) {
- ma_sleep(waitTimeInMilliseconds);
- continue; /* Nothing is available in the capture buffer. */
- }
+ /* Web Audio can support any number of channels and sample rates. It only supports f32 formats, however. */
+ pDeviceInfo->nativeDataFormats[0].flags = 0;
+ pDeviceInfo->nativeDataFormats[0].format = ma_format_unknown;
+ pDeviceInfo->nativeDataFormats[0].channels = 0; /* All channels are supported. */
+ pDeviceInfo->nativeDataFormats[0].sampleRate = EM_ASM_INT({
+ try {
+ var temp = new (window.AudioContext || window.webkitAudioContext)();
+ var sampleRate = temp.sampleRate;
+ temp.close();
+ return sampleRate;
+ } catch(e) {
+ return 0;
+ }
+ }, 0); /* Must pass in a dummy argument for C99 compatibility. */
+
+ if (pDeviceInfo->nativeDataFormats[0].sampleRate == 0) {
+ return MA_NO_DEVICE;
+ }
- hr = ma_IDirectSoundCaptureBuffer_Lock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, lockOffsetInBytesCapture, lockSizeInBytesCapture, &pMappedDeviceBufferCapture, &mappedSizeInBytesCapture, NULL, NULL, 0);
- if (FAILED(hr)) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from capture device in preparation for writing to the device.", ma_result_from_HRESULT(hr));
- }
+ pDeviceInfo->nativeDataFormatCount = 1;
+ return MA_SUCCESS;
+}
- /* At this point we have some input data that we need to output. We do not return until every mapped frame of the input data is written to the playback device. */
- mappedDeviceFramesProcessedCapture = 0;
- for (;;) { /* Keep writing to the playback device. */
- ma_uint8 inputFramesInClientFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
- ma_uint32 inputFramesInClientFormatCap = sizeof(inputFramesInClientFormat) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels);
- ma_uint8 outputFramesInClientFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
- ma_uint32 outputFramesInClientFormatCap = sizeof(outputFramesInClientFormat) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
- ma_uint32 outputFramesInClientFormatCount;
- ma_uint32 outputFramesInClientFormatConsumed = 0;
- ma_uint64 clientCapturedFramesToProcess = ma_min(inputFramesInClientFormatCap, outputFramesInClientFormatCap);
- ma_uint64 deviceCapturedFramesToProcess = (mappedSizeInBytesCapture / bpfDeviceCapture) - mappedDeviceFramesProcessedCapture;
- void* pRunningMappedDeviceBufferCapture = ma_offset_ptr(pMappedDeviceBufferCapture, mappedDeviceFramesProcessedCapture * bpfDeviceCapture);
+static void ma_device_uninit_by_index__webaudio(ma_device* pDevice, ma_device_type deviceType, int deviceIndex)
+{
+ MA_ASSERT(pDevice != NULL);
- result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningMappedDeviceBufferCapture, &deviceCapturedFramesToProcess, inputFramesInClientFormat, &clientCapturedFramesToProcess);
- if (result != MA_SUCCESS) {
- break;
- }
+ EM_ASM({
+ var device = miniaudio.get_device_by_index($0);
- outputFramesInClientFormatCount = (ma_uint32)clientCapturedFramesToProcess;
- mappedDeviceFramesProcessedCapture += (ma_uint32)deviceCapturedFramesToProcess;
+ /* Make sure all nodes are disconnected and marked for collection. */
+ if (device.scriptNode !== undefined) {
+ device.scriptNode.onaudioprocess = function(e) {}; /* We want to reset the callback to ensure it doesn't get called after AudioContext.close() has returned. Shouldn't happen since we're disconnecting, but just to be safe... */
+ device.scriptNode.disconnect();
+ device.scriptNode = undefined;
+ }
+ if (device.streamNode !== undefined) {
+ device.streamNode.disconnect();
+ device.streamNode = undefined;
+ }
- ma_device__on_data(pDevice, outputFramesInClientFormat, inputFramesInClientFormat, (ma_uint32)clientCapturedFramesToProcess);
+ /*
+ Stop the device. I think there is a chance the callback could get fired after calling this, hence why we want
+ to clear the callback before closing.
+ */
+ device.webaudio.close();
+ device.webaudio = undefined;
- /* At this point we have input and output data in client format. All we need to do now is convert it to the output device format. This may take a few passes. */
- for (;;) {
- ma_uint32 framesWrittenThisIteration;
- DWORD physicalPlayCursorInBytes;
- DWORD physicalWriteCursorInBytes;
- DWORD availableBytesPlayback;
- DWORD silentPaddingInBytes = 0; /* <-- Must be initialized to 0. */
+ /* Can't forget to free the intermediary buffer. This is the buffer that's shared between JavaScript and C. */
+ if (device.intermediaryBuffer !== undefined) {
+ Module._free(device.intermediaryBuffer);
+ device.intermediaryBuffer = undefined;
+ device.intermediaryBufferView = undefined;
+ device.intermediaryBufferSizeInBytes = undefined;
+ }
- /* We need the physical play and write cursors. */
- if (FAILED(ma_IDirectSoundBuffer_GetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &physicalPlayCursorInBytes, &physicalWriteCursorInBytes))) {
- break;
- }
+ /* Make sure the device is untracked so the slot can be reused later. */
+ miniaudio.untrack_device_by_index($0);
+ }, deviceIndex, deviceType);
+}
- if (physicalPlayCursorInBytes < prevPlayCursorInBytesPlayback) {
- physicalPlayCursorLoopFlagPlayback = !physicalPlayCursorLoopFlagPlayback;
- }
- prevPlayCursorInBytesPlayback = physicalPlayCursorInBytes;
+static ma_result ma_device_uninit__webaudio(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
- /* If there's any bytes available for writing we can do that now. The space between the virtual cursor position and play cursor. */
- if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) {
- /* Same loop iteration. The available bytes wraps all the way around from the virtual write cursor to the physical play cursor. */
- if (physicalPlayCursorInBytes <= virtualWriteCursorInBytesPlayback) {
- availableBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback;
- availableBytesPlayback += physicalPlayCursorInBytes; /* Wrap around. */
- } else {
- /* This is an error. */
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[DirectSound] (Duplex/Playback) WARNING: Play cursor has moved in front of the write cursor (same loop iterations). physicalPlayCursorInBytes=%ld, virtualWriteCursorInBytes=%ld.\n", physicalPlayCursorInBytes, virtualWriteCursorInBytesPlayback);
- availableBytesPlayback = 0;
- }
- } else {
- /* Different loop iterations. The available bytes only goes from the virtual write cursor to the physical play cursor. */
- if (physicalPlayCursorInBytes >= virtualWriteCursorInBytesPlayback) {
- availableBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback;
- } else {
- /* This is an error. */
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[DirectSound] (Duplex/Playback) WARNING: Write cursor has moved behind the play cursor (different loop iterations). physicalPlayCursorInBytes=%ld, virtualWriteCursorInBytes=%ld.\n", physicalPlayCursorInBytes, virtualWriteCursorInBytesPlayback);
- availableBytesPlayback = 0;
- }
- }
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ ma_device_uninit_by_index__webaudio(pDevice, ma_device_type_capture, pDevice->webaudio.indexCapture);
+ }
- /* If there's no room available for writing we need to wait for more. */
- if (availableBytesPlayback == 0) {
- /* If we haven't started the device yet, this will never get beyond 0. In this case we need to get the device started. */
- if (!isPlaybackDeviceStarted) {
- hr = ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING);
- if (FAILED(hr)) {
- ma_IDirectSoundCaptureBuffer_Stop((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed.", ma_result_from_HRESULT(hr));
- }
- isPlaybackDeviceStarted = MA_TRUE;
- } else {
- ma_sleep(waitTimeInMilliseconds);
- continue;
- }
- }
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ ma_device_uninit_by_index__webaudio(pDevice, ma_device_type_playback, pDevice->webaudio.indexPlayback);
+ }
+ return MA_SUCCESS;
+}
- /* Getting here means there room available somewhere. We limit this to either the end of the buffer or the physical play cursor, whichever is closest. */
- lockOffsetInBytesPlayback = virtualWriteCursorInBytesPlayback;
- if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) {
- /* Same loop iteration. Go up to the end of the buffer. */
- lockSizeInBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback;
- } else {
- /* Different loop iterations. Go up to the physical play cursor. */
- lockSizeInBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback;
- }
+static ma_uint32 ma_calculate_period_size_in_frames_from_descriptor__webaudio(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile)
+{
+ /*
+ There have been reports of the default buffer size being too small on some browsers. There have been reports of the default buffer
+ size being too small on some browsers. If we're using default buffer size, we'll make sure the period size is a big biffer than our
+ standard defaults.
+ */
+ ma_uint32 periodSizeInFrames;
- hr = ma_IDirectSoundBuffer_Lock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, lockOffsetInBytesPlayback, lockSizeInBytesPlayback, &pMappedDeviceBufferPlayback, &mappedSizeInBytesPlayback, NULL, NULL, 0);
- if (FAILED(hr)) {
- result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from playback device in preparation for writing to the device.", ma_result_from_HRESULT(hr));
- break;
- }
+ if (pDescriptor->periodSizeInFrames == 0) {
+ if (pDescriptor->periodSizeInMilliseconds == 0) {
+ if (performanceProfile == ma_performance_profile_low_latency) {
+ periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(33, nativeSampleRate); /* 1 frame @ 30 FPS */
+ } else {
+ periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(333, nativeSampleRate);
+ }
+ } else {
+ periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pDescriptor->periodSizeInMilliseconds, nativeSampleRate);
+ }
+ } else {
+ periodSizeInFrames = pDescriptor->periodSizeInFrames;
+ }
- /*
- Experiment: If the playback buffer is being starved, pad it with some silence to get it back in sync. This will cause a glitch, but it may prevent
- endless glitching due to it constantly running out of data.
- */
- if (isPlaybackDeviceStarted) {
- DWORD bytesQueuedForPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - availableBytesPlayback;
- if (bytesQueuedForPlayback < (pDevice->playback.internalPeriodSizeInFrames*bpfDevicePlayback)) {
- silentPaddingInBytes = (pDevice->playback.internalPeriodSizeInFrames*2*bpfDevicePlayback) - bytesQueuedForPlayback;
- if (silentPaddingInBytes > lockSizeInBytesPlayback) {
- silentPaddingInBytes = lockSizeInBytesPlayback;
- }
+ /* The size of the buffer must be a power of 2 and between 256 and 16384. */
+ if (periodSizeInFrames < 256) {
+ periodSizeInFrames = 256;
+ } else if (periodSizeInFrames > 16384) {
+ periodSizeInFrames = 16384;
+ } else {
+ periodSizeInFrames = ma_next_power_of_2(periodSizeInFrames);
+ }
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[DirectSound] (Duplex/Playback) Playback buffer starved. availableBytesPlayback=%ld, silentPaddingInBytes=%ld\n", availableBytesPlayback, silentPaddingInBytes);
- }
- }
+ return periodSizeInFrames;
+}
- /* At this point we have a buffer for output. */
- if (silentPaddingInBytes > 0) {
- MA_ZERO_MEMORY(pMappedDeviceBufferPlayback, silentPaddingInBytes);
- framesWrittenThisIteration = silentPaddingInBytes/bpfDevicePlayback;
- } else {
- ma_uint64 convertedFrameCountIn = (outputFramesInClientFormatCount - outputFramesInClientFormatConsumed);
- ma_uint64 convertedFrameCountOut = mappedSizeInBytesPlayback/bpfDevicePlayback;
- void* pConvertedFramesIn = ma_offset_ptr(outputFramesInClientFormat, outputFramesInClientFormatConsumed * bpfDevicePlayback);
- void* pConvertedFramesOut = pMappedDeviceBufferPlayback;
+static ma_result ma_device_init_by_type__webaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptor, ma_device_type deviceType)
+{
+ int deviceIndex;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ ma_uint32 periodSizeInFrames;
- result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, pConvertedFramesIn, &convertedFrameCountIn, pConvertedFramesOut, &convertedFrameCountOut);
- if (result != MA_SUCCESS) {
- break;
- }
+ MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(pConfig != NULL);
+ MA_ASSERT(deviceType != ma_device_type_duplex);
- outputFramesInClientFormatConsumed += (ma_uint32)convertedFrameCountOut;
- framesWrittenThisIteration = (ma_uint32)convertedFrameCountOut;
- }
+ if (deviceType == ma_device_type_capture && !ma_is_capture_supported__webaudio()) {
+ return MA_NO_DEVICE;
+ }
+ /* We're going to calculate some stuff in C just to simplify the JS code. */
+ channels = (pDescriptor->channels > 0) ? pDescriptor->channels : MA_DEFAULT_CHANNELS;
+ sampleRate = (pDescriptor->sampleRate > 0) ? pDescriptor->sampleRate : MA_DEFAULT_SAMPLE_RATE;
+ periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__webaudio(pDescriptor, sampleRate, pConfig->performanceProfile);
- hr = ma_IDirectSoundBuffer_Unlock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, pMappedDeviceBufferPlayback, framesWrittenThisIteration*bpfDevicePlayback, NULL, 0);
- if (FAILED(hr)) {
- result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from playback device after writing to the device.", ma_result_from_HRESULT(hr));
- break;
- }
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "periodSizeInFrames = %d\n", (int)periodSizeInFrames);
- virtualWriteCursorInBytesPlayback += framesWrittenThisIteration*bpfDevicePlayback;
- if ((virtualWriteCursorInBytesPlayback/bpfDevicePlayback) == pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods) {
- virtualWriteCursorInBytesPlayback = 0;
- virtualWriteCursorLoopFlagPlayback = !virtualWriteCursorLoopFlagPlayback;
- }
+ /* We create the device on the JavaScript side and reference it using an index. We use this to make it possible to reference the device between JavaScript and C. */
+ deviceIndex = EM_ASM_INT({
+ var channels = $0;
+ var sampleRate = $1;
+ var bufferSize = $2; /* In PCM frames. */
+ var isCapture = $3;
+ var pDevice = $4;
- /*
- We may need to start the device. We want two full periods to be written before starting the playback device. Having an extra period adds
- a bit of a buffer to prevent the playback buffer from getting starved.
- */
- framesWrittenToPlaybackDevice += framesWrittenThisIteration;
- if (!isPlaybackDeviceStarted && framesWrittenToPlaybackDevice >= (pDevice->playback.internalPeriodSizeInFrames*2)) {
- hr = ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING);
- if (FAILED(hr)) {
- ma_IDirectSoundCaptureBuffer_Stop((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed.", ma_result_from_HRESULT(hr));
- }
- isPlaybackDeviceStarted = MA_TRUE;
- }
+ if (typeof(window.miniaudio) === 'undefined') {
+ return -1; /* Context not initialized. */
+ }
- if (framesWrittenThisIteration < mappedSizeInBytesPlayback/bpfDevicePlayback) {
- break; /* We're finished with the output data.*/
- }
- }
+ var device = {};
- if (clientCapturedFramesToProcess == 0) {
- break; /* We just consumed every input sample. */
- }
- }
+ /* The AudioContext must be created in a suspended state. */
+ device.webaudio = new (window.AudioContext || window.webkitAudioContext)({sampleRate:sampleRate});
+ device.webaudio.suspend();
+ device.state = 1; /* ma_device_state_stopped */
+ /*
+ We need an intermediary buffer which we use for JavaScript and C interop. This buffer stores interleaved f32 PCM data. Because it's passed between
+ JavaScript and C it needs to be allocated and freed using Module._malloc() and Module._free().
+ */
+ device.intermediaryBufferSizeInBytes = channels * bufferSize * 4;
+ device.intermediaryBuffer = Module._malloc(device.intermediaryBufferSizeInBytes);
+ device.intermediaryBufferView = new Float32Array(Module.HEAPF32.buffer, device.intermediaryBuffer, device.intermediaryBufferSizeInBytes);
- /* At this point we're done with the mapped portion of the capture buffer. */
- hr = ma_IDirectSoundCaptureBuffer_Unlock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, pMappedDeviceBufferCapture, mappedSizeInBytesCapture, NULL, 0);
- if (FAILED(hr)) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from capture device after reading from the device.", ma_result_from_HRESULT(hr));
- }
- prevReadCursorInBytesCapture = (lockOffsetInBytesCapture + mappedSizeInBytesCapture);
- } break;
+ /*
+ Both playback and capture devices use a ScriptProcessorNode for performing per-sample operations.
+ ScriptProcessorNode is actually deprecated so this is likely to be temporary. The way this works for playback is very simple. You just set a callback
+ that's periodically fired, just like a normal audio callback function. But apparently this design is "flawed" and is now deprecated in favour of
+ something called AudioWorklets which _forces_ you to load a _separate_ .js file at run time... nice... Hopefully ScriptProcessorNode will continue to
+ work for years to come, but this may need to change to use AudioSourceBufferNode instead, which I think is what Emscripten uses for it's built-in SDL
+ implementation. I'll be avoiding that insane AudioWorklet API like the plague...
+ For capture it is a bit unintuitive. We use the ScriptProccessorNode _only_ to get the raw PCM data. It is connected to an AudioContext just like the
+ playback case, however we just output silence to the AudioContext instead of passing any real data. It would make more sense to me to use the
+ MediaRecorder API, but unfortunately you need to specify a MIME time (Opus, Vorbis, etc.) for the binary blob that's returned to the client, but I've
+ been unable to figure out how to get this as raw PCM. The closest I can think is to use the MIME type for WAV files and just parse it, but I don't know
+ how well this would work. Although ScriptProccessorNode is deprecated, in practice it seems to have pretty good browser support so I'm leaving it like
+ this for now. If anyone knows how I could get raw PCM data using the MediaRecorder API please let me know!
+ */
+ device.scriptNode = device.webaudio.createScriptProcessor(bufferSize, (isCapture) ? channels : 0, (isCapture) ? 0 : channels);
- case ma_device_type_capture:
- {
- DWORD physicalCaptureCursorInBytes;
- DWORD physicalReadCursorInBytes;
- hr = ma_IDirectSoundCaptureBuffer_GetCurrentPosition((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, &physicalCaptureCursorInBytes, &physicalReadCursorInBytes);
- if (FAILED(hr)) {
- return MA_ERROR;
+ if (isCapture) {
+ device.scriptNode.onaudioprocess = function(e) {
+ if (device.intermediaryBuffer === undefined) {
+ return; /* This means the device has been uninitialized. */
}
- /* If the previous capture position is the same as the current position we need to wait a bit longer. */
- if (prevReadCursorInBytesCapture == physicalReadCursorInBytes) {
- ma_sleep(waitTimeInMilliseconds);
- continue;
+ if (device.intermediaryBufferView.length == 0) {
+ /* Recreate intermediaryBufferView when losing reference to the underlying buffer, probably due to emscripten resizing heap. */
+ device.intermediaryBufferView = new Float32Array(Module.HEAPF32.buffer, device.intermediaryBuffer, device.intermediaryBufferSizeInBytes);
}
- /* Getting here means we have capture data available. */
- if (prevReadCursorInBytesCapture < physicalReadCursorInBytes) {
- /* The capture position has not looped. This is the simple case. */
- lockOffsetInBytesCapture = prevReadCursorInBytesCapture;
- lockSizeInBytesCapture = (physicalReadCursorInBytes - prevReadCursorInBytesCapture);
- } else {
- /*
- The capture position has looped. This is the more complex case. Map to the end of the buffer. If this does not return anything,
- do it again from the start.
- */
- if (prevReadCursorInBytesCapture < pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture) {
- /* Lock up to the end of the buffer. */
- lockOffsetInBytesCapture = prevReadCursorInBytesCapture;
- lockSizeInBytesCapture = (pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture) - prevReadCursorInBytesCapture;
- } else {
- /* Lock starting from the start of the buffer. */
- lockOffsetInBytesCapture = 0;
- lockSizeInBytesCapture = physicalReadCursorInBytes;
- }
+ /* Make sure silence it output to the AudioContext destination. Not doing this will cause sound to come out of the speakers! */
+ for (var iChannel = 0; iChannel < e.outputBuffer.numberOfChannels; ++iChannel) {
+ e.outputBuffer.getChannelData(iChannel).fill(0.0);
}
- if (lockSizeInBytesCapture < pDevice->capture.internalPeriodSizeInFrames) {
- ma_sleep(waitTimeInMilliseconds);
- continue; /* Nothing is available in the capture buffer. */
+ /* There are some situations where we may want to send silence to the client. */
+ var sendSilence = false;
+ if (device.streamNode === undefined) {
+ sendSilence = true;
}
- hr = ma_IDirectSoundCaptureBuffer_Lock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, lockOffsetInBytesCapture, lockSizeInBytesCapture, &pMappedDeviceBufferCapture, &mappedSizeInBytesCapture, NULL, NULL, 0);
- if (FAILED(hr)) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from capture device in preparation for writing to the device.", ma_result_from_HRESULT(hr));
+ /* Sanity check. This will never happen, right? */
+ if (e.inputBuffer.numberOfChannels != channels) {
+ console.log("Capture: Channel count mismatch. " + e.inputBufer.numberOfChannels + " != " + channels + ". Sending silence.");
+ sendSilence = true;
}
- #ifdef MA_DEBUG_OUTPUT
- if (lockSizeInBytesCapture != mappedSizeInBytesCapture) {
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[DirectSound] (Capture) lockSizeInBytesCapture=%ld != mappedSizeInBytesCapture=%ld\n", lockSizeInBytesCapture, mappedSizeInBytesCapture);
- }
- #endif
+ /* This looped design guards against the situation where e.inputBuffer is a different size to the original buffer size. Should never happen in practice. */
+ var totalFramesProcessed = 0;
+ while (totalFramesProcessed < e.inputBuffer.length) {
+ var framesRemaining = e.inputBuffer.length - totalFramesProcessed;
+ var framesToProcess = framesRemaining;
+ if (framesToProcess > (device.intermediaryBufferSizeInBytes/channels/4)) {
+ framesToProcess = (device.intermediaryBufferSizeInBytes/channels/4);
+ }
- ma_device__send_frames_to_client(pDevice, mappedSizeInBytesCapture/bpfDeviceCapture, pMappedDeviceBufferCapture);
+ /* We need to do the reverse of the playback case. We need to interleave the input data and copy it into the intermediary buffer. Then we send it to the client. */
+ if (sendSilence) {
+ device.intermediaryBufferView.fill(0.0);
+ } else {
+ for (var iFrame = 0; iFrame < framesToProcess; ++iFrame) {
+ for (var iChannel = 0; iChannel < e.inputBuffer.numberOfChannels; ++iChannel) {
+ device.intermediaryBufferView[iFrame*channels + iChannel] = e.inputBuffer.getChannelData(iChannel)[totalFramesProcessed + iFrame];
+ }
+ }
+ }
- hr = ma_IDirectSoundCaptureBuffer_Unlock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, pMappedDeviceBufferCapture, mappedSizeInBytesCapture, NULL, 0);
- if (FAILED(hr)) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from capture device after reading from the device.", ma_result_from_HRESULT(hr));
- }
- prevReadCursorInBytesCapture = lockOffsetInBytesCapture + mappedSizeInBytesCapture;
+ /* Send data to the client from our intermediary buffer. */
+ ccall("ma_device_process_pcm_frames_capture__webaudio", "undefined", ["number", "number", "number"], [pDevice, framesToProcess, device.intermediaryBuffer]);
- if (prevReadCursorInBytesCapture == (pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture)) {
- prevReadCursorInBytesCapture = 0;
+ totalFramesProcessed += framesToProcess;
}
- } break;
-
+ };
+ navigator.mediaDevices.getUserMedia({audio:true, video:false})
+ .then(function(stream) {
+ device.streamNode = device.webaudio.createMediaStreamSource(stream);
+ device.streamNode.connect(device.scriptNode);
+ device.scriptNode.connect(device.webaudio.destination);
+ })
+ .catch(function(error) {
+ /* I think this should output silence... */
+ device.scriptNode.connect(device.webaudio.destination);
+ });
+ } else {
+ device.scriptNode.onaudioprocess = function(e) {
+ if (device.intermediaryBuffer === undefined) {
+ return; /* This means the device has been uninitialized. */
+ }
- case ma_device_type_playback:
- {
- DWORD availableBytesPlayback;
- DWORD physicalPlayCursorInBytes;
- DWORD physicalWriteCursorInBytes;
- hr = ma_IDirectSoundBuffer_GetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &physicalPlayCursorInBytes, &physicalWriteCursorInBytes);
- if (FAILED(hr)) {
- break;
+ if(device.intermediaryBufferView.length == 0) {
+ /* Recreate intermediaryBufferView when losing reference to the underlying buffer, probably due to emscripten resizing heap. */
+ device.intermediaryBufferView = new Float32Array(Module.HEAPF32.buffer, device.intermediaryBuffer, device.intermediaryBufferSizeInBytes);
}
- if (physicalPlayCursorInBytes < prevPlayCursorInBytesPlayback) {
- physicalPlayCursorLoopFlagPlayback = !physicalPlayCursorLoopFlagPlayback;
+ var outputSilence = false;
+
+ /* Sanity check. This will never happen, right? */
+ if (e.outputBuffer.numberOfChannels != channels) {
+ console.log("Playback: Channel count mismatch. " + e.outputBufer.numberOfChannels + " != " + channels + ". Outputting silence.");
+ outputSilence = true;
+ return;
}
- prevPlayCursorInBytesPlayback = physicalPlayCursorInBytes;
- /* If there's any bytes available for writing we can do that now. The space between the virtual cursor position and play cursor. */
- if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) {
- /* Same loop iteration. The available bytes wraps all the way around from the virtual write cursor to the physical play cursor. */
- if (physicalPlayCursorInBytes <= virtualWriteCursorInBytesPlayback) {
- availableBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback;
- availableBytesPlayback += physicalPlayCursorInBytes; /* Wrap around. */
- } else {
- /* This is an error. */
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[DirectSound] (Playback) WARNING: Play cursor has moved in front of the write cursor (same loop iterations). physicalPlayCursorInBytes=%ld, virtualWriteCursorInBytes=%ld.\n", physicalPlayCursorInBytes, virtualWriteCursorInBytesPlayback);
- availableBytesPlayback = 0;
- }
- } else {
- /* Different loop iterations. The available bytes only goes from the virtual write cursor to the physical play cursor. */
- if (physicalPlayCursorInBytes >= virtualWriteCursorInBytesPlayback) {
- availableBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback;
- } else {
- /* This is an error. */
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[DirectSound] (Playback) WARNING: Write cursor has moved behind the play cursor (different loop iterations). physicalPlayCursorInBytes=%ld, virtualWriteCursorInBytes=%ld.\n", physicalPlayCursorInBytes, virtualWriteCursorInBytesPlayback);
- availableBytesPlayback = 0;
+ /* This looped design guards against the situation where e.outputBuffer is a different size to the original buffer size. Should never happen in practice. */
+ var totalFramesProcessed = 0;
+ while (totalFramesProcessed < e.outputBuffer.length) {
+ var framesRemaining = e.outputBuffer.length - totalFramesProcessed;
+ var framesToProcess = framesRemaining;
+ if (framesToProcess > (device.intermediaryBufferSizeInBytes/channels/4)) {
+ framesToProcess = (device.intermediaryBufferSizeInBytes/channels/4);
}
- }
- /* If there's no room available for writing we need to wait for more. */
- if (availableBytesPlayback < pDevice->playback.internalPeriodSizeInFrames) {
- /* If we haven't started the device yet, this will never get beyond 0. In this case we need to get the device started. */
- if (availableBytesPlayback == 0 && !isPlaybackDeviceStarted) {
- hr = ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING);
- if (FAILED(hr)) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed.", ma_result_from_HRESULT(hr));
+ /* Read data from the client into our intermediary buffer. */
+ ccall("ma_device_process_pcm_frames_playback__webaudio", "undefined", ["number", "number", "number"], [pDevice, framesToProcess, device.intermediaryBuffer]);
+
+ /* At this point we'll have data in our intermediary buffer which we now need to deinterleave and copy over to the output buffers. */
+ if (outputSilence) {
+ for (var iChannel = 0; iChannel < e.outputBuffer.numberOfChannels; ++iChannel) {
+ e.outputBuffer.getChannelData(iChannel).fill(0.0);
}
- isPlaybackDeviceStarted = MA_TRUE;
} else {
- ma_sleep(waitTimeInMilliseconds);
- continue;
+ for (var iChannel = 0; iChannel < e.outputBuffer.numberOfChannels; ++iChannel) {
+ var outputBuffer = e.outputBuffer.getChannelData(iChannel);
+ var intermediaryBuffer = device.intermediaryBufferView;
+ for (var iFrame = 0; iFrame < framesToProcess; ++iFrame) {
+ outputBuffer[totalFramesProcessed + iFrame] = intermediaryBuffer[iFrame*channels + iChannel];
+ }
+ }
}
- }
- /* Getting here means there room available somewhere. We limit this to either the end of the buffer or the physical play cursor, whichever is closest. */
- lockOffsetInBytesPlayback = virtualWriteCursorInBytesPlayback;
- if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) {
- /* Same loop iteration. Go up to the end of the buffer. */
- lockSizeInBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback;
- } else {
- /* Different loop iterations. Go up to the physical play cursor. */
- lockSizeInBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback;
+ totalFramesProcessed += framesToProcess;
}
+ };
- hr = ma_IDirectSoundBuffer_Lock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, lockOffsetInBytesPlayback, lockSizeInBytesPlayback, &pMappedDeviceBufferPlayback, &mappedSizeInBytesPlayback, NULL, NULL, 0);
- if (FAILED(hr)) {
- result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from playback device in preparation for writing to the device.", ma_result_from_HRESULT(hr));
- break;
- }
+ device.scriptNode.connect(device.webaudio.destination);
+ }
- /* At this point we have a buffer for output. */
- ma_device__read_frames_from_client(pDevice, (mappedSizeInBytesPlayback/bpfDevicePlayback), pMappedDeviceBufferPlayback);
+ return miniaudio.track_device(device);
+ }, channels, sampleRate, periodSizeInFrames, deviceType == ma_device_type_capture, pDevice);
- hr = ma_IDirectSoundBuffer_Unlock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, pMappedDeviceBufferPlayback, mappedSizeInBytesPlayback, NULL, 0);
- if (FAILED(hr)) {
- result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from playback device after writing to the device.", ma_result_from_HRESULT(hr));
- break;
- }
+ if (deviceIndex < 0) {
+ return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
+ }
- virtualWriteCursorInBytesPlayback += mappedSizeInBytesPlayback;
- if (virtualWriteCursorInBytesPlayback == pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) {
- virtualWriteCursorInBytesPlayback = 0;
- virtualWriteCursorLoopFlagPlayback = !virtualWriteCursorLoopFlagPlayback;
- }
+ if (deviceType == ma_device_type_capture) {
+ pDevice->webaudio.indexCapture = deviceIndex;
+ } else {
+ pDevice->webaudio.indexPlayback = deviceIndex;
+ }
- /*
- We may need to start the device. We want two full periods to be written before starting the playback device. Having an extra period adds
- a bit of a buffer to prevent the playback buffer from getting starved.
- */
- framesWrittenToPlaybackDevice += mappedSizeInBytesPlayback/bpfDevicePlayback;
- if (!isPlaybackDeviceStarted && framesWrittenToPlaybackDevice >= pDevice->playback.internalPeriodSizeInFrames) {
- hr = ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING);
- if (FAILED(hr)) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed.", ma_result_from_HRESULT(hr));
- }
- isPlaybackDeviceStarted = MA_TRUE;
- }
- } break;
+ pDescriptor->format = ma_format_f32;
+ pDescriptor->channels = channels;
+ ma_channel_map_init_standard(ma_standard_channel_map_webaudio, pDescriptor->channelMap, ma_countof(pDescriptor->channelMap), pDescriptor->channels);
+ pDescriptor->sampleRate = EM_ASM_INT({ return miniaudio.get_device_by_index($0).webaudio.sampleRate; }, deviceIndex);
+ pDescriptor->periodSizeInFrames = periodSizeInFrames;
+ pDescriptor->periodCount = 1;
+ return MA_SUCCESS;
+}
- default: return MA_INVALID_ARGS; /* Invalid device type. */
+static ma_result ma_device_init__webaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+{
+ ma_result result;
+
+ if (pConfig->deviceType == ma_device_type_loopback) {
+ return MA_DEVICE_TYPE_NOT_SUPPORTED;
+ }
+
+ /* No exclusive mode with Web Audio. */
+ if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) ||
+ ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) {
+ return MA_SHARE_MODE_NOT_SUPPORTED;
+ }
+
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
+ result = ma_device_init_by_type__webaudio(pDevice, pConfig, pDescriptorCapture, ma_device_type_capture);
+ if (result != MA_SUCCESS) {
+ return result;
}
+ }
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ result = ma_device_init_by_type__webaudio(pDevice, pConfig, pDescriptorPlayback, ma_device_type_playback);
if (result != MA_SUCCESS) {
+ if (pConfig->deviceType == ma_device_type_duplex) {
+ ma_device_uninit_by_index__webaudio(pDevice, ma_device_type_capture, pDevice->webaudio.indexCapture);
+ }
return result;
}
}
- /* Getting here means the device is being stopped. */
+ return MA_SUCCESS;
+}
+
+static ma_result ma_device_start__webaudio(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
+
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- hr = ma_IDirectSoundCaptureBuffer_Stop((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer);
- if (FAILED(hr)) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCaptureBuffer_Stop() failed.", ma_result_from_HRESULT(hr));
- }
+ EM_ASM({
+ var device = miniaudio.get_device_by_index($0);
+ device.webaudio.resume();
+ device.state = 2; /* ma_device_state_started */
+ }, pDevice->webaudio.indexCapture);
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- /* The playback device should be drained before stopping. All we do is wait until the available bytes is equal to the size of the buffer. */
- if (isPlaybackDeviceStarted) {
- for (;;) {
- DWORD availableBytesPlayback = 0;
- DWORD physicalPlayCursorInBytes;
- DWORD physicalWriteCursorInBytes;
- hr = ma_IDirectSoundBuffer_GetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &physicalPlayCursorInBytes, &physicalWriteCursorInBytes);
- if (FAILED(hr)) {
- break;
- }
+ EM_ASM({
+ var device = miniaudio.get_device_by_index($0);
+ device.webaudio.resume();
+ device.state = 2; /* ma_device_state_started */
+ }, pDevice->webaudio.indexPlayback);
+ }
- if (physicalPlayCursorInBytes < prevPlayCursorInBytesPlayback) {
- physicalPlayCursorLoopFlagPlayback = !physicalPlayCursorLoopFlagPlayback;
- }
- prevPlayCursorInBytesPlayback = physicalPlayCursorInBytes;
+ return MA_SUCCESS;
+}
- if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) {
- /* Same loop iteration. The available bytes wraps all the way around from the virtual write cursor to the physical play cursor. */
- if (physicalPlayCursorInBytes <= virtualWriteCursorInBytesPlayback) {
- availableBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback;
- availableBytesPlayback += physicalPlayCursorInBytes; /* Wrap around. */
- } else {
- break;
- }
- } else {
- /* Different loop iterations. The available bytes only goes from the virtual write cursor to the physical play cursor. */
- if (physicalPlayCursorInBytes >= virtualWriteCursorInBytesPlayback) {
- availableBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback;
- } else {
- break;
- }
- }
+static ma_result ma_device_stop__webaudio(ma_device* pDevice)
+{
+ MA_ASSERT(pDevice != NULL);
- if (availableBytesPlayback >= (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback)) {
- break;
- }
+ /*
+ From the WebAudio API documentation for AudioContext.suspend():
- ma_sleep(waitTimeInMilliseconds);
- }
- }
+ Suspends the progression of AudioContext's currentTime, allows any current context processing blocks that are already processed to be played to the
+ destination, and then allows the system to release its claim on audio hardware.
- hr = ma_IDirectSoundBuffer_Stop((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer);
- if (FAILED(hr)) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Stop() failed.", ma_result_from_HRESULT(hr));
- }
+ I read this to mean that "any current context processing blocks" are processed by suspend() - i.e. They they are drained. We therefore shouldn't need to
+ do any kind of explicit draining.
+ */
- ma_IDirectSoundBuffer_SetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0);
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ EM_ASM({
+ var device = miniaudio.get_device_by_index($0);
+ device.webaudio.suspend();
+ device.state = 1; /* ma_device_state_stopped */
+ }, pDevice->webaudio.indexCapture);
+ }
+
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ EM_ASM({
+ var device = miniaudio.get_device_by_index($0);
+ device.webaudio.suspend();
+ device.state = 1; /* ma_device_state_stopped */
+ }, pDevice->webaudio.indexPlayback);
}
+ ma_device__on_notification_stopped(pDevice);
+
return MA_SUCCESS;
}
-static ma_result ma_context_uninit__dsound(ma_context* pContext)
+static ma_result ma_context_uninit__webaudio(ma_context* pContext)
{
MA_ASSERT(pContext != NULL);
- MA_ASSERT(pContext->backend == ma_backend_dsound);
+ MA_ASSERT(pContext->backend == ma_backend_webaudio);
- ma_dlclose(pContext, pContext->dsound.hDSoundDLL);
+ /* Nothing needs to be done here. */
+ (void)pContext;
return MA_SUCCESS;
}
-static ma_result ma_context_init__dsound(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+static ma_result ma_context_init__webaudio(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
{
+ int resultFromJS;
+
MA_ASSERT(pContext != NULL);
- (void)pConfig;
+ (void)pConfig; /* Unused. */
- pContext->dsound.hDSoundDLL = ma_dlopen(pContext, "dsound.dll");
- if (pContext->dsound.hDSoundDLL == NULL) {
- return MA_API_NOT_FOUND;
- }
+ /* Here is where our global JavaScript object is initialized. */
+ resultFromJS = EM_ASM_INT({
+ if ((window.AudioContext || window.webkitAudioContext) === undefined) {
+ return 0; /* Web Audio not supported. */
+ }
- pContext->dsound.DirectSoundCreate = ma_dlsym(pContext, pContext->dsound.hDSoundDLL, "DirectSoundCreate");
- pContext->dsound.DirectSoundEnumerateA = ma_dlsym(pContext, pContext->dsound.hDSoundDLL, "DirectSoundEnumerateA");
- pContext->dsound.DirectSoundCaptureCreate = ma_dlsym(pContext, pContext->dsound.hDSoundDLL, "DirectSoundCaptureCreate");
- pContext->dsound.DirectSoundCaptureEnumerateA = ma_dlsym(pContext, pContext->dsound.hDSoundDLL, "DirectSoundCaptureEnumerateA");
+ if (typeof(window.miniaudio) === 'undefined') {
+ window.miniaudio = {};
+ miniaudio.devices = []; /* Device cache for mapping devices to indexes for JavaScript/C interop. */
- pCallbacks->onContextInit = ma_context_init__dsound;
- pCallbacks->onContextUninit = ma_context_uninit__dsound;
- pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__dsound;
- pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__dsound;
- pCallbacks->onDeviceInit = ma_device_init__dsound;
- pCallbacks->onDeviceUninit = ma_device_uninit__dsound;
- pCallbacks->onDeviceStart = NULL; /* Not used. Started in onDeviceDataLoop. */
- pCallbacks->onDeviceStop = NULL; /* Not used. Stopped in onDeviceDataLoop. */
- pCallbacks->onDeviceRead = NULL; /* Not used. Data is read directly in onDeviceDataLoop. */
- pCallbacks->onDeviceWrite = NULL; /* Not used. Data is written directly in onDeviceDataLoop. */
- pCallbacks->onDeviceDataLoop = ma_device_data_loop__dsound;
+ miniaudio.track_device = function(device) {
+ /* Try inserting into a free slot first. */
+ for (var iDevice = 0; iDevice < miniaudio.devices.length; ++iDevice) {
+ if (miniaudio.devices[iDevice] == null) {
+ miniaudio.devices[iDevice] = device;
+ return iDevice;
+ }
+ }
- return MA_SUCCESS;
-}
-#endif
+ /* Getting here means there is no empty slots in the array so we just push to the end. */
+ miniaudio.devices.push(device);
+ return miniaudio.devices.length - 1;
+ };
+ miniaudio.untrack_device_by_index = function(deviceIndex) {
+ /* We just set the device's slot to null. The slot will get reused in the next call to ma_track_device. */
+ miniaudio.devices[deviceIndex] = null;
+ /* Trim the array if possible. */
+ while (miniaudio.devices.length > 0) {
+ if (miniaudio.devices[miniaudio.devices.length-1] == null) {
+ miniaudio.devices.pop();
+ } else {
+ break;
+ }
+ }
+ };
-/******************************************************************************
+ miniaudio.untrack_device = function(device) {
+ for (var iDevice = 0; iDevice < miniaudio.devices.length; ++iDevice) {
+ if (miniaudio.devices[iDevice] == device) {
+ return miniaudio.untrack_device_by_index(iDevice);
+ }
+ }
+ };
-WinMM Backend
+ miniaudio.get_device_by_index = function(deviceIndex) {
+ return miniaudio.devices[deviceIndex];
+ };
-******************************************************************************/
-#ifdef MA_HAS_WINMM
+ miniaudio.unlock_event_types = (function(){
+ return ['touchstart', 'touchend', 'click'];
+ })();
-/*
-Some older compilers don't have WAVEOUTCAPS2A and WAVEINCAPS2A, so we'll need to write this ourselves. These structures
-are exactly the same as the older ones but they have a few GUIDs for manufacturer/product/name identification. I'm keeping
-the names the same as the Win32 library for consistency, but namespaced to avoid naming conflicts with the Win32 version.
-*/
-typedef struct
-{
- WORD wMid;
- WORD wPid;
- MMVERSION vDriverVersion;
- CHAR szPname[MAXPNAMELEN];
- DWORD dwFormats;
- WORD wChannels;
- WORD wReserved1;
- DWORD dwSupport;
- GUID ManufacturerGuid;
- GUID ProductGuid;
- GUID NameGuid;
-} MA_WAVEOUTCAPS2A;
-typedef struct
-{
- WORD wMid;
- WORD wPid;
- MMVERSION vDriverVersion;
- CHAR szPname[MAXPNAMELEN];
- DWORD dwFormats;
- WORD wChannels;
- WORD wReserved1;
- GUID ManufacturerGuid;
- GUID ProductGuid;
- GUID NameGuid;
-} MA_WAVEINCAPS2A;
+ miniaudio.unlock = function() {
+ for(var i = 0; i < miniaudio.devices.length; ++i) {
+ var device = miniaudio.devices[i];
+ if (device != null && device.webaudio != null && device.state === 2 /* ma_device_state_started */) {
+ device.webaudio.resume();
+ }
+ }
+ miniaudio.unlock_event_types.map(function(event_type) {
+ document.removeEventListener(event_type, miniaudio.unlock, true);
+ });
+ };
-typedef UINT (WINAPI * MA_PFN_waveOutGetNumDevs)(void);
-typedef MMRESULT (WINAPI * MA_PFN_waveOutGetDevCapsA)(ma_uintptr uDeviceID, LPWAVEOUTCAPSA pwoc, UINT cbwoc);
-typedef MMRESULT (WINAPI * MA_PFN_waveOutOpen)(LPHWAVEOUT phwo, UINT uDeviceID, LPCWAVEFORMATEX pwfx, DWORD_PTR dwCallback, DWORD_PTR dwInstance, DWORD fdwOpen);
-typedef MMRESULT (WINAPI * MA_PFN_waveOutClose)(HWAVEOUT hwo);
-typedef MMRESULT (WINAPI * MA_PFN_waveOutPrepareHeader)(HWAVEOUT hwo, LPWAVEHDR pwh, UINT cbwh);
-typedef MMRESULT (WINAPI * MA_PFN_waveOutUnprepareHeader)(HWAVEOUT hwo, LPWAVEHDR pwh, UINT cbwh);
-typedef MMRESULT (WINAPI * MA_PFN_waveOutWrite)(HWAVEOUT hwo, LPWAVEHDR pwh, UINT cbwh);
-typedef MMRESULT (WINAPI * MA_PFN_waveOutReset)(HWAVEOUT hwo);
-typedef UINT (WINAPI * MA_PFN_waveInGetNumDevs)(void);
-typedef MMRESULT (WINAPI * MA_PFN_waveInGetDevCapsA)(ma_uintptr uDeviceID, LPWAVEINCAPSA pwic, UINT cbwic);
-typedef MMRESULT (WINAPI * MA_PFN_waveInOpen)(LPHWAVEIN phwi, UINT uDeviceID, LPCWAVEFORMATEX pwfx, DWORD_PTR dwCallback, DWORD_PTR dwInstance, DWORD fdwOpen);
-typedef MMRESULT (WINAPI * MA_PFN_waveInClose)(HWAVEIN hwi);
-typedef MMRESULT (WINAPI * MA_PFN_waveInPrepareHeader)(HWAVEIN hwi, LPWAVEHDR pwh, UINT cbwh);
-typedef MMRESULT (WINAPI * MA_PFN_waveInUnprepareHeader)(HWAVEIN hwi, LPWAVEHDR pwh, UINT cbwh);
-typedef MMRESULT (WINAPI * MA_PFN_waveInAddBuffer)(HWAVEIN hwi, LPWAVEHDR pwh, UINT cbwh);
-typedef MMRESULT (WINAPI * MA_PFN_waveInStart)(HWAVEIN hwi);
-typedef MMRESULT (WINAPI * MA_PFN_waveInReset)(HWAVEIN hwi);
+ miniaudio.unlock_event_types.map(function(event_type) {
+ document.addEventListener(event_type, miniaudio.unlock, true);
+ });
+ }
-static ma_result ma_result_from_MMRESULT(MMRESULT resultMM)
-{
- switch (resultMM) {
- case MMSYSERR_NOERROR: return MA_SUCCESS;
- case MMSYSERR_BADDEVICEID: return MA_INVALID_ARGS;
- case MMSYSERR_INVALHANDLE: return MA_INVALID_ARGS;
- case MMSYSERR_NOMEM: return MA_OUT_OF_MEMORY;
- case MMSYSERR_INVALFLAG: return MA_INVALID_ARGS;
- case MMSYSERR_INVALPARAM: return MA_INVALID_ARGS;
- case MMSYSERR_HANDLEBUSY: return MA_BUSY;
- case MMSYSERR_ERROR: return MA_ERROR;
- default: return MA_ERROR;
+ return 1;
+ }, 0); /* Must pass in a dummy argument for C99 compatibility. */
+
+ if (resultFromJS != 1) {
+ return MA_FAILED_TO_INIT_BACKEND;
}
+
+ pCallbacks->onContextInit = ma_context_init__webaudio;
+ pCallbacks->onContextUninit = ma_context_uninit__webaudio;
+ pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__webaudio;
+ pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__webaudio;
+ pCallbacks->onDeviceInit = ma_device_init__webaudio;
+ pCallbacks->onDeviceUninit = ma_device_uninit__webaudio;
+ pCallbacks->onDeviceStart = ma_device_start__webaudio;
+ pCallbacks->onDeviceStop = ma_device_stop__webaudio;
+ pCallbacks->onDeviceRead = NULL; /* Not needed because WebAudio is asynchronous. */
+ pCallbacks->onDeviceWrite = NULL; /* Not needed because WebAudio is asynchronous. */
+ pCallbacks->onDeviceDataLoop = NULL; /* Not needed because WebAudio is asynchronous. */
+
+ return MA_SUCCESS;
}
+#endif /* Web Audio */
-static char* ma_find_last_character(char* str, char ch)
-{
- char* last;
- if (str == NULL) {
- return NULL;
- }
- last = NULL;
- while (*str != '\0') {
- if (*str == ch) {
- last = str;
+static ma_bool32 ma__is_channel_map_valid(const ma_channel* pChannelMap, ma_uint32 channels)
+{
+ /* A blank channel map should be allowed, in which case it should use an appropriate default which will depend on context. */
+ if (pChannelMap != NULL && pChannelMap[0] != MA_CHANNEL_NONE) {
+ ma_uint32 iChannel;
+
+ if (channels == 0 || channels > MA_MAX_CHANNELS) {
+ return MA_FALSE; /* Channel count out of range. */
}
- str += 1;
+ /* A channel cannot be present in the channel map more than once. */
+ for (iChannel = 0; iChannel < channels; ++iChannel) {
+ ma_uint32 jChannel;
+ for (jChannel = iChannel + 1; jChannel < channels; ++jChannel) {
+ if (pChannelMap[iChannel] == pChannelMap[jChannel]) {
+ return MA_FALSE;
+ }
+ }
+ }
}
- return last;
-}
-
-static ma_uint32 ma_get_period_size_in_bytes(ma_uint32 periodSizeInFrames, ma_format format, ma_uint32 channels)
-{
- return periodSizeInFrames * ma_get_bytes_per_frame(format, channels);
+ return MA_TRUE;
}
-/*
-Our own "WAVECAPS" structure that contains generic information shared between WAVEOUTCAPS2 and WAVEINCAPS2 so
-we can do things generically and typesafely. Names are being kept the same for consistency.
-*/
-typedef struct
-{
- CHAR szPname[MAXPNAMELEN];
- DWORD dwFormats;
- WORD wChannels;
- GUID NameGuid;
-} MA_WAVECAPSA;
-
-static ma_result ma_get_best_info_from_formats_flags__winmm(DWORD dwFormats, WORD channels, WORD* pBitsPerSample, DWORD* pSampleRate)
+static ma_result ma_device__post_init_setup(ma_device* pDevice, ma_device_type deviceType)
{
- WORD bitsPerSample = 0;
- DWORD sampleRate = 0;
+ ma_result result;
- if (pBitsPerSample) {
- *pBitsPerSample = 0;
- }
- if (pSampleRate) {
- *pSampleRate = 0;
- }
+ MA_ASSERT(pDevice != NULL);
- if (channels == 1) {
- bitsPerSample = 16;
- if ((dwFormats & WAVE_FORMAT_48M16) != 0) {
- sampleRate = 48000;
- } else if ((dwFormats & WAVE_FORMAT_44M16) != 0) {
- sampleRate = 44100;
- } else if ((dwFormats & WAVE_FORMAT_2M16) != 0) {
- sampleRate = 22050;
- } else if ((dwFormats & WAVE_FORMAT_1M16) != 0) {
- sampleRate = 11025;
- } else if ((dwFormats & WAVE_FORMAT_96M16) != 0) {
- sampleRate = 96000;
- } else {
- bitsPerSample = 8;
- if ((dwFormats & WAVE_FORMAT_48M08) != 0) {
- sampleRate = 48000;
- } else if ((dwFormats & WAVE_FORMAT_44M08) != 0) {
- sampleRate = 44100;
- } else if ((dwFormats & WAVE_FORMAT_2M08) != 0) {
- sampleRate = 22050;
- } else if ((dwFormats & WAVE_FORMAT_1M08) != 0) {
- sampleRate = 11025;
- } else if ((dwFormats & WAVE_FORMAT_96M08) != 0) {
- sampleRate = 96000;
+ if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex || deviceType == ma_device_type_loopback) {
+ if (pDevice->capture.format == ma_format_unknown) {
+ pDevice->capture.format = pDevice->capture.internalFormat;
+ }
+ if (pDevice->capture.channels == 0) {
+ pDevice->capture.channels = pDevice->capture.internalChannels;
+ }
+ if (pDevice->capture.channelMap[0] == MA_CHANNEL_NONE) {
+ MA_ASSERT(pDevice->capture.channels <= MA_MAX_CHANNELS);
+ if (pDevice->capture.internalChannels == pDevice->capture.channels) {
+ ma_channel_map_copy(pDevice->capture.channelMap, pDevice->capture.internalChannelMap, pDevice->capture.channels);
} else {
- return MA_FORMAT_NOT_SUPPORTED;
+ if (pDevice->capture.channelMixMode == ma_channel_mix_mode_simple) {
+ ma_channel_map_init_blank(pDevice->capture.channelMap, pDevice->capture.channels);
+ } else {
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pDevice->capture.channelMap, ma_countof(pDevice->capture.channelMap), pDevice->capture.channels);
+ }
}
}
- } else {
- bitsPerSample = 16;
- if ((dwFormats & WAVE_FORMAT_48S16) != 0) {
- sampleRate = 48000;
- } else if ((dwFormats & WAVE_FORMAT_44S16) != 0) {
- sampleRate = 44100;
- } else if ((dwFormats & WAVE_FORMAT_2S16) != 0) {
- sampleRate = 22050;
- } else if ((dwFormats & WAVE_FORMAT_1S16) != 0) {
- sampleRate = 11025;
- } else if ((dwFormats & WAVE_FORMAT_96S16) != 0) {
- sampleRate = 96000;
- } else {
- bitsPerSample = 8;
- if ((dwFormats & WAVE_FORMAT_48S08) != 0) {
- sampleRate = 48000;
- } else if ((dwFormats & WAVE_FORMAT_44S08) != 0) {
- sampleRate = 44100;
- } else if ((dwFormats & WAVE_FORMAT_2S08) != 0) {
- sampleRate = 22050;
- } else if ((dwFormats & WAVE_FORMAT_1S08) != 0) {
- sampleRate = 11025;
- } else if ((dwFormats & WAVE_FORMAT_96S08) != 0) {
- sampleRate = 96000;
+ }
+
+ if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) {
+ if (pDevice->playback.format == ma_format_unknown) {
+ pDevice->playback.format = pDevice->playback.internalFormat;
+ }
+ if (pDevice->playback.channels == 0) {
+ pDevice->playback.channels = pDevice->playback.internalChannels;
+ }
+ if (pDevice->playback.channelMap[0] == MA_CHANNEL_NONE) {
+ MA_ASSERT(pDevice->playback.channels <= MA_MAX_CHANNELS);
+ if (pDevice->playback.internalChannels == pDevice->playback.channels) {
+ ma_channel_map_copy(pDevice->playback.channelMap, pDevice->playback.internalChannelMap, pDevice->playback.channels);
} else {
- return MA_FORMAT_NOT_SUPPORTED;
+ if (pDevice->playback.channelMixMode == ma_channel_mix_mode_simple) {
+ ma_channel_map_init_blank(pDevice->playback.channelMap, pDevice->playback.channels);
+ } else {
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pDevice->playback.channelMap, ma_countof(pDevice->playback.channelMap), pDevice->playback.channels);
+ }
}
}
}
- if (pBitsPerSample) {
- *pBitsPerSample = bitsPerSample;
- }
- if (pSampleRate) {
- *pSampleRate = sampleRate;
+ if (pDevice->sampleRate == 0) {
+ if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex || deviceType == ma_device_type_loopback) {
+ pDevice->sampleRate = pDevice->capture.internalSampleRate;
+ } else {
+ pDevice->sampleRate = pDevice->playback.internalSampleRate;
+ }
}
- return MA_SUCCESS;
-}
-
-static ma_result ma_formats_flags_to_WAVEFORMATEX__winmm(DWORD dwFormats, WORD channels, WAVEFORMATEX* pWF)
-{
- ma_result result;
-
- MA_ASSERT(pWF != NULL);
-
- MA_ZERO_OBJECT(pWF);
- pWF->cbSize = sizeof(*pWF);
- pWF->wFormatTag = WAVE_FORMAT_PCM;
- pWF->nChannels = (WORD)channels;
- if (pWF->nChannels > 2) {
- pWF->nChannels = 2;
+ /* Data converters. */
+ if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex || deviceType == ma_device_type_loopback) {
+ /* Converting from internal device format to client format. */
+ ma_data_converter_config converterConfig = ma_data_converter_config_init_default();
+ converterConfig.formatIn = pDevice->capture.internalFormat;
+ converterConfig.channelsIn = pDevice->capture.internalChannels;
+ converterConfig.sampleRateIn = pDevice->capture.internalSampleRate;
+ converterConfig.pChannelMapIn = pDevice->capture.internalChannelMap;
+ converterConfig.formatOut = pDevice->capture.format;
+ converterConfig.channelsOut = pDevice->capture.channels;
+ converterConfig.sampleRateOut = pDevice->sampleRate;
+ converterConfig.pChannelMapOut = pDevice->capture.channelMap;
+ converterConfig.channelMixMode = pDevice->capture.channelMixMode;
+ converterConfig.allowDynamicSampleRate = MA_FALSE;
+ converterConfig.resampling.algorithm = pDevice->resampling.algorithm;
+ converterConfig.resampling.linear.lpfOrder = pDevice->resampling.linear.lpfOrder;
+ converterConfig.resampling.pBackendVTable = pDevice->resampling.pBackendVTable;
+ converterConfig.resampling.pBackendUserData = pDevice->resampling.pBackendUserData;
+
+ /* Make sure the old converter is uninitialized first. */
+ if (ma_device_get_state(pDevice) != ma_device_state_uninitialized) {
+ ma_data_converter_uninit(&pDevice->capture.converter, &pDevice->pContext->allocationCallbacks);
+ }
+
+ result = ma_data_converter_init(&converterConfig, &pDevice->pContext->allocationCallbacks, &pDevice->capture.converter);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
}
- result = ma_get_best_info_from_formats_flags__winmm(dwFormats, channels, &pWF->wBitsPerSample, &pWF->nSamplesPerSec);
- if (result != MA_SUCCESS) {
- return result;
+ if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) {
+ /* Converting from client format to device format. */
+ ma_data_converter_config converterConfig = ma_data_converter_config_init_default();
+ converterConfig.formatIn = pDevice->playback.format;
+ converterConfig.channelsIn = pDevice->playback.channels;
+ converterConfig.sampleRateIn = pDevice->sampleRate;
+ converterConfig.pChannelMapIn = pDevice->playback.channelMap;
+ converterConfig.formatOut = pDevice->playback.internalFormat;
+ converterConfig.channelsOut = pDevice->playback.internalChannels;
+ converterConfig.sampleRateOut = pDevice->playback.internalSampleRate;
+ converterConfig.pChannelMapOut = pDevice->playback.internalChannelMap;
+ converterConfig.channelMixMode = pDevice->playback.channelMixMode;
+ converterConfig.allowDynamicSampleRate = MA_FALSE;
+ converterConfig.resampling.algorithm = pDevice->resampling.algorithm;
+ converterConfig.resampling.linear.lpfOrder = pDevice->resampling.linear.lpfOrder;
+ converterConfig.resampling.pBackendVTable = pDevice->resampling.pBackendVTable;
+ converterConfig.resampling.pBackendUserData = pDevice->resampling.pBackendUserData;
+
+ /* Make sure the old converter is uninitialized first. */
+ if (ma_device_get_state(pDevice) != ma_device_state_uninitialized) {
+ ma_data_converter_uninit(&pDevice->playback.converter, &pDevice->pContext->allocationCallbacks);
+ }
+
+ result = ma_data_converter_init(&converterConfig, &pDevice->pContext->allocationCallbacks, &pDevice->playback.converter);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
}
- pWF->nBlockAlign = (WORD)(pWF->nChannels * pWF->wBitsPerSample / 8);
- pWF->nAvgBytesPerSec = pWF->nBlockAlign * pWF->nSamplesPerSec;
-
- return MA_SUCCESS;
-}
-
-static ma_result ma_context_get_device_info_from_WAVECAPS(ma_context* pContext, MA_WAVECAPSA* pCaps, ma_device_info* pDeviceInfo)
-{
- WORD bitsPerSample;
- DWORD sampleRate;
- ma_result result;
-
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pCaps != NULL);
- MA_ASSERT(pDeviceInfo != NULL);
-
- /*
- Name / Description
-
- Unfortunately the name specified in WAVE(OUT/IN)CAPS2 is limited to 31 characters. This results in an unprofessional looking
- situation where the names of the devices are truncated. To help work around this, we need to look at the name GUID and try
- looking in the registry for the full name. If we can't find it there, we need to just fall back to the default name.
- */
-
- /* Set the default to begin with. */
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), pCaps->szPname, (size_t)-1);
/*
- Now try the registry. There's a few things to consider here:
- - The name GUID can be null, in which we case we just need to stick to the original 31 characters.
- - If the name GUID is not present in the registry we'll also need to stick to the original 31 characters.
- - I like consistency, so I want the returned device names to be consistent with those returned by WASAPI and DirectSound. The
- problem, however is that WASAPI and DirectSound use " ()" format (such as "Speakers (High Definition Audio)"),
- but WinMM does not specificy the component name. From my admittedly limited testing, I've notice the component name seems to
- usually fit within the 31 characters of the fixed sized buffer, so what I'm going to do is parse that string for the component
- name, and then concatenate the name from the registry.
+ In playback mode, if the data converter does not support retrieval of the required number of
+ input frames given a number of output frames, we need to fall back to a heap-allocated cache.
*/
- if (!ma_is_guid_null(&pCaps->NameGuid)) {
- wchar_t guidStrW[256];
- if (((MA_PFN_StringFromGUID2)pContext->win32.StringFromGUID2)(&pCaps->NameGuid, guidStrW, ma_countof(guidStrW)) > 0) {
- char guidStr[256];
- char keyStr[1024];
- HKEY hKey;
-
- WideCharToMultiByte(CP_UTF8, 0, guidStrW, -1, guidStr, sizeof(guidStr), 0, FALSE);
+ if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) {
+ ma_uint64 unused;
- ma_strcpy_s(keyStr, sizeof(keyStr), "SYSTEM\\CurrentControlSet\\Control\\MediaCategories\\");
- ma_strcat_s(keyStr, sizeof(keyStr), guidStr);
+ pDevice->playback.inputCacheConsumed = 0;
+ pDevice->playback.inputCacheRemaining = 0;
- if (((MA_PFN_RegOpenKeyExA)pContext->win32.RegOpenKeyExA)(HKEY_LOCAL_MACHINE, keyStr, 0, KEY_READ, &hKey) == ERROR_SUCCESS) {
- BYTE nameFromReg[512];
- DWORD nameFromRegSize = sizeof(nameFromReg);
- result = ((MA_PFN_RegQueryValueExA)pContext->win32.RegQueryValueExA)(hKey, "Name", 0, NULL, (LPBYTE)nameFromReg, (LPDWORD)&nameFromRegSize);
- ((MA_PFN_RegCloseKey)pContext->win32.RegCloseKey)(hKey);
+ if (deviceType == ma_device_type_duplex || ma_data_converter_get_required_input_frame_count(&pDevice->playback.converter, 1, &unused) != MA_SUCCESS) {
+ /* We need a heap allocated cache. We want to size this based on the period size. */
+ void* pNewInputCache;
+ ma_uint64 newInputCacheCap;
+ ma_uint64 newInputCacheSizeInBytes;
- if (result == ERROR_SUCCESS) {
- /* We have the value from the registry, so now we need to construct the name string. */
- char name[1024];
- if (ma_strcpy_s(name, sizeof(name), pDeviceInfo->name) == 0) {
- char* nameBeg = ma_find_last_character(name, '(');
- if (nameBeg != NULL) {
- size_t leadingLen = (nameBeg - name);
- ma_strncpy_s(nameBeg + 1, sizeof(name) - leadingLen, (const char*)nameFromReg, (size_t)-1);
+ newInputCacheCap = ma_calculate_frame_count_after_resampling(pDevice->playback.internalSampleRate, pDevice->sampleRate, pDevice->playback.internalPeriodSizeInFrames);
- /* The closing ")", if it can fit. */
- if (leadingLen + nameFromRegSize < sizeof(name)-1) {
- ma_strcat_s(name, sizeof(name), ")");
- }
+ newInputCacheSizeInBytes = newInputCacheCap * ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
+ if (newInputCacheSizeInBytes > MA_SIZE_MAX) {
+ ma_free(pDevice->playback.pInputCache, &pDevice->pContext->allocationCallbacks);
+ pDevice->playback.pInputCache = NULL;
+ pDevice->playback.inputCacheCap = 0;
+ return MA_OUT_OF_MEMORY; /* Allocation too big. Should never hit this, but makes the cast below safer for 32-bit builds. */
+ }
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), name, (size_t)-1);
- }
- }
- }
+ pNewInputCache = ma_realloc(pDevice->playback.pInputCache, (size_t)newInputCacheSizeInBytes, &pDevice->pContext->allocationCallbacks);
+ if (pNewInputCache == NULL) {
+ ma_free(pDevice->playback.pInputCache, &pDevice->pContext->allocationCallbacks);
+ pDevice->playback.pInputCache = NULL;
+ pDevice->playback.inputCacheCap = 0;
+ return MA_OUT_OF_MEMORY;
}
+
+ pDevice->playback.pInputCache = pNewInputCache;
+ pDevice->playback.inputCacheCap = newInputCacheCap;
+ } else {
+ /* Heap allocation not required. Make sure we clear out the old cache just in case this function was called in response to a route change. */
+ ma_free(pDevice->playback.pInputCache, &pDevice->pContext->allocationCallbacks);
+ pDevice->playback.pInputCache = NULL;
+ pDevice->playback.inputCacheCap = 0;
}
}
+ return MA_SUCCESS;
+}
- result = ma_get_best_info_from_formats_flags__winmm(pCaps->dwFormats, pCaps->wChannels, &bitsPerSample, &sampleRate);
- if (result != MA_SUCCESS) {
- return result;
+MA_API ma_result ma_device_post_init(ma_device* pDevice, ma_device_type deviceType, const ma_device_descriptor* pDescriptorPlayback, const ma_device_descriptor* pDescriptorCapture)
+{
+ ma_result result;
+
+ if (pDevice == NULL) {
+ return MA_INVALID_ARGS;
}
- if (bitsPerSample == 8) {
- pDeviceInfo->nativeDataFormats[0].format = ma_format_u8;
- } else if (bitsPerSample == 16) {
- pDeviceInfo->nativeDataFormats[0].format = ma_format_s16;
- } else if (bitsPerSample == 24) {
- pDeviceInfo->nativeDataFormats[0].format = ma_format_s24;
- } else if (bitsPerSample == 32) {
- pDeviceInfo->nativeDataFormats[0].format = ma_format_s32;
- } else {
- return MA_FORMAT_NOT_SUPPORTED;
+ /* Capture. */
+ if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex || deviceType == ma_device_type_loopback) {
+ if (ma_device_descriptor_is_valid(pDescriptorCapture) == MA_FALSE) {
+ return MA_INVALID_ARGS;
+ }
+
+ pDevice->capture.internalFormat = pDescriptorCapture->format;
+ pDevice->capture.internalChannels = pDescriptorCapture->channels;
+ pDevice->capture.internalSampleRate = pDescriptorCapture->sampleRate;
+ MA_COPY_MEMORY(pDevice->capture.internalChannelMap, pDescriptorCapture->channelMap, sizeof(pDescriptorCapture->channelMap));
+ pDevice->capture.internalPeriodSizeInFrames = pDescriptorCapture->periodSizeInFrames;
+ pDevice->capture.internalPeriods = pDescriptorCapture->periodCount;
+
+ if (pDevice->capture.internalPeriodSizeInFrames == 0) {
+ pDevice->capture.internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pDescriptorCapture->periodSizeInMilliseconds, pDescriptorCapture->sampleRate);
+ }
}
- pDeviceInfo->nativeDataFormats[0].channels = pCaps->wChannels;
- pDeviceInfo->nativeDataFormats[0].sampleRate = sampleRate;
- pDeviceInfo->nativeDataFormats[0].flags = 0;
- pDeviceInfo->nativeDataFormatCount = 1;
- return MA_SUCCESS;
-}
+ /* Playback. */
+ if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) {
+ if (ma_device_descriptor_is_valid(pDescriptorPlayback) == MA_FALSE) {
+ return MA_INVALID_ARGS;
+ }
-static ma_result ma_context_get_device_info_from_WAVEOUTCAPS2(ma_context* pContext, MA_WAVEOUTCAPS2A* pCaps, ma_device_info* pDeviceInfo)
-{
- MA_WAVECAPSA caps;
+ pDevice->playback.internalFormat = pDescriptorPlayback->format;
+ pDevice->playback.internalChannels = pDescriptorPlayback->channels;
+ pDevice->playback.internalSampleRate = pDescriptorPlayback->sampleRate;
+ MA_COPY_MEMORY(pDevice->playback.internalChannelMap, pDescriptorPlayback->channelMap, sizeof(pDescriptorPlayback->channelMap));
+ pDevice->playback.internalPeriodSizeInFrames = pDescriptorPlayback->periodSizeInFrames;
+ pDevice->playback.internalPeriods = pDescriptorPlayback->periodCount;
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pCaps != NULL);
- MA_ASSERT(pDeviceInfo != NULL);
+ if (pDevice->playback.internalPeriodSizeInFrames == 0) {
+ pDevice->playback.internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pDescriptorPlayback->periodSizeInMilliseconds, pDescriptorPlayback->sampleRate);
+ }
+ }
- MA_COPY_MEMORY(caps.szPname, pCaps->szPname, sizeof(caps.szPname));
- caps.dwFormats = pCaps->dwFormats;
- caps.wChannels = pCaps->wChannels;
- caps.NameGuid = pCaps->NameGuid;
- return ma_context_get_device_info_from_WAVECAPS(pContext, &caps, pDeviceInfo);
-}
+ /*
+ The name of the device can be retrieved from device info. This may be temporary and replaced with a `ma_device_get_info(pDevice, deviceType)` instead.
+ For loopback devices, we need to retrieve the name of the playback device.
+ */
+ {
+ ma_device_info deviceInfo;
-static ma_result ma_context_get_device_info_from_WAVEINCAPS2(ma_context* pContext, MA_WAVEINCAPS2A* pCaps, ma_device_info* pDeviceInfo)
-{
- MA_WAVECAPSA caps;
+ if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex || deviceType == ma_device_type_loopback) {
+ result = ma_device_get_info(pDevice, (deviceType == ma_device_type_loopback) ? ma_device_type_playback : ma_device_type_capture, &deviceInfo);
+ if (result == MA_SUCCESS) {
+ ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), deviceInfo.name, (size_t)-1);
+ } else {
+ /* We failed to retrieve the device info. Fall back to a default name. */
+ if (pDescriptorCapture->pDeviceID == NULL) {
+ ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
+ } else {
+ ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), "Capture Device", (size_t)-1);
+ }
+ }
+ }
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pCaps != NULL);
- MA_ASSERT(pDeviceInfo != NULL);
+ if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) {
+ result = ma_device_get_info(pDevice, ma_device_type_playback, &deviceInfo);
+ if (result == MA_SUCCESS) {
+ ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), deviceInfo.name, (size_t)-1);
+ } else {
+ /* We failed to retrieve the device info. Fall back to a default name. */
+ if (pDescriptorPlayback->pDeviceID == NULL) {
+ ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
+ } else {
+ ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), "Playback Device", (size_t)-1);
+ }
+ }
+ }
+ }
- MA_COPY_MEMORY(caps.szPname, pCaps->szPname, sizeof(caps.szPname));
- caps.dwFormats = pCaps->dwFormats;
- caps.wChannels = pCaps->wChannels;
- caps.NameGuid = pCaps->NameGuid;
- return ma_context_get_device_info_from_WAVECAPS(pContext, &caps, pDeviceInfo);
+ /* Update data conversion. */
+ return ma_device__post_init_setup(pDevice, deviceType); /* TODO: Should probably rename ma_device__post_init_setup() to something better. */
}
-static ma_result ma_context_enumerate_devices__winmm(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+static ma_thread_result MA_THREADCALL ma_worker_thread(void* pData)
{
- UINT playbackDeviceCount;
- UINT captureDeviceCount;
- UINT iPlaybackDevice;
- UINT iCaptureDevice;
+ ma_device* pDevice = (ma_device*)pData;
+ MA_ASSERT(pDevice != NULL);
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(callback != NULL);
+#ifdef MA_WIN32
+ ma_CoInitializeEx(pDevice->pContext, NULL, MA_COINIT_VALUE);
+#endif
- /* Playback. */
- playbackDeviceCount = ((MA_PFN_waveOutGetNumDevs)pContext->winmm.waveOutGetNumDevs)();
- for (iPlaybackDevice = 0; iPlaybackDevice < playbackDeviceCount; ++iPlaybackDevice) {
- MMRESULT result;
- MA_WAVEOUTCAPS2A caps;
+ /*
+ When the device is being initialized it's initial state is set to ma_device_state_uninitialized. Before returning from
+ ma_device_init(), the state needs to be set to something valid. In miniaudio the device's default state immediately
+ after initialization is stopped, so therefore we need to mark the device as such. miniaudio will wait on the worker
+ thread to signal an event to know when the worker thread is ready for action.
+ */
+ ma_device__set_state(pDevice, ma_device_state_stopped);
+ ma_event_signal(&pDevice->stopEvent);
- MA_ZERO_OBJECT(&caps);
+ for (;;) { /* <-- This loop just keeps the thread alive. The main audio loop is inside. */
+ ma_result startResult;
+ ma_result stopResult; /* <-- This will store the result from onDeviceStop(). If it returns an error, we don't fire the stopped notification callback. */
- result = ((MA_PFN_waveOutGetDevCapsA)pContext->winmm.waveOutGetDevCapsA)(iPlaybackDevice, (WAVEOUTCAPSA*)&caps, sizeof(caps));
- if (result == MMSYSERR_NOERROR) {
- ma_device_info deviceInfo;
+ /* We wait on an event to know when something has requested that the device be started and the main loop entered. */
+ ma_event_wait(&pDevice->wakeupEvent);
- MA_ZERO_OBJECT(&deviceInfo);
- deviceInfo.id.winmm = iPlaybackDevice;
+ /* Default result code. */
+ pDevice->workResult = MA_SUCCESS;
- /* The first enumerated device is the default device. */
- if (iPlaybackDevice == 0) {
- deviceInfo.isDefault = MA_TRUE;
- }
+ /* If the reason for the wake up is that we are terminating, just break from the loop. */
+ if (ma_device_get_state(pDevice) == ma_device_state_uninitialized) {
+ break;
+ }
- if (ma_context_get_device_info_from_WAVEOUTCAPS2(pContext, &caps, &deviceInfo) == MA_SUCCESS) {
- ma_bool32 cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
- if (cbResult == MA_FALSE) {
- return MA_SUCCESS; /* Enumeration was stopped. */
- }
- }
+ /*
+ Getting to this point means the device is wanting to get started. The function that has requested that the device
+ be started will be waiting on an event (pDevice->startEvent) which means we need to make sure we signal the event
+ in both the success and error case. It's important that the state of the device is set _before_ signaling the event.
+ */
+ MA_ASSERT(ma_device_get_state(pDevice) == ma_device_state_starting);
+
+ /* If the device has a start callback, start it now. */
+ if (pDevice->pContext->callbacks.onDeviceStart != NULL) {
+ startResult = pDevice->pContext->callbacks.onDeviceStart(pDevice);
+ } else {
+ startResult = MA_SUCCESS;
}
- }
- /* Capture. */
- captureDeviceCount = ((MA_PFN_waveInGetNumDevs)pContext->winmm.waveInGetNumDevs)();
- for (iCaptureDevice = 0; iCaptureDevice < captureDeviceCount; ++iCaptureDevice) {
- MMRESULT result;
- MA_WAVEINCAPS2A caps;
+ /*
+ If starting was not successful we'll need to loop back to the start and wait for something
+ to happen (pDevice->wakeupEvent).
+ */
+ if (startResult != MA_SUCCESS) {
+ pDevice->workResult = startResult;
+ ma_event_signal(&pDevice->startEvent); /* <-- Always signal the start event so ma_device_start() can return as it'll be waiting on it. */
+ continue;
+ }
- MA_ZERO_OBJECT(&caps);
+ /* Make sure the state is set appropriately. */
+ ma_device__set_state(pDevice, ma_device_state_started); /* <-- Set this before signaling the event so that the state is always guaranteed to be good after ma_device_start() has returned. */
+ ma_event_signal(&pDevice->startEvent);
- result = ((MA_PFN_waveInGetDevCapsA)pContext->winmm.waveInGetDevCapsA)(iCaptureDevice, (WAVEINCAPSA*)&caps, sizeof(caps));
- if (result == MMSYSERR_NOERROR) {
- ma_device_info deviceInfo;
+ ma_device__on_notification_started(pDevice);
- MA_ZERO_OBJECT(&deviceInfo);
- deviceInfo.id.winmm = iCaptureDevice;
+ if (pDevice->pContext->callbacks.onDeviceDataLoop != NULL) {
+ pDevice->pContext->callbacks.onDeviceDataLoop(pDevice);
+ } else {
+ /* The backend is not using a custom main loop implementation, so now fall back to the blocking read-write implementation. */
+ ma_device_audio_thread__default_read_write(pDevice);
+ }
- /* The first enumerated device is the default device. */
- if (iCaptureDevice == 0) {
- deviceInfo.isDefault = MA_TRUE;
- }
+ /* Getting here means we have broken from the main loop which happens the application has requested that device be stopped. */
+ if (pDevice->pContext->callbacks.onDeviceStop != NULL) {
+ stopResult = pDevice->pContext->callbacks.onDeviceStop(pDevice);
+ } else {
+ stopResult = MA_SUCCESS; /* No stop callback with the backend. Just assume successful. */
+ }
- if (ma_context_get_device_info_from_WAVEINCAPS2(pContext, &caps, &deviceInfo) == MA_SUCCESS) {
- ma_bool32 cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
- if (cbResult == MA_FALSE) {
- return MA_SUCCESS; /* Enumeration was stopped. */
- }
- }
+ /*
+ After the device has stopped, make sure an event is posted. Don't post a stopped event if
+ stopping failed. This can happen on some backends when the underlying stream has been
+ stopped due to the device being physically unplugged or disabled via an OS setting.
+ */
+ if (stopResult == MA_SUCCESS) {
+ ma_device__on_notification_stopped(pDevice);
}
+
+ /* A function somewhere is waiting for the device to have stopped for real so we need to signal an event to allow it to continue. */
+ ma_device__set_state(pDevice, ma_device_state_stopped);
+ ma_event_signal(&pDevice->stopEvent);
}
- return MA_SUCCESS;
+#ifdef MA_WIN32
+ ma_CoUninitialize(pDevice->pContext);
+#endif
+
+ return (ma_thread_result)0;
}
-static ma_result ma_context_get_device_info__winmm(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+
+/* Helper for determining whether or not the given device is initialized. */
+static ma_bool32 ma_device__is_initialized(ma_device* pDevice)
{
- UINT winMMDeviceID;
+ if (pDevice == NULL) {
+ return MA_FALSE;
+ }
- MA_ASSERT(pContext != NULL);
+ return ma_device_get_state(pDevice) != ma_device_state_uninitialized;
+}
- winMMDeviceID = 0;
- if (pDeviceID != NULL) {
- winMMDeviceID = (UINT)pDeviceID->winmm;
- }
- pDeviceInfo->id.winmm = winMMDeviceID;
+#ifdef MA_WIN32
+static ma_result ma_context_uninit_backend_apis__win32(ma_context* pContext)
+{
+ /* For some reason UWP complains when CoUninitialize() is called. I'm just not going to call it on UWP. */
+#ifdef MA_WIN32_DESKTOP
+ ma_CoUninitialize(pContext);
+ ma_dlclose(pContext, pContext->win32.hUser32DLL);
+ ma_dlclose(pContext, pContext->win32.hOle32DLL);
+ ma_dlclose(pContext, pContext->win32.hAdvapi32DLL);
+#else
+ (void)pContext;
+#endif
- /* The first ID is the default device. */
- if (winMMDeviceID == 0) {
- pDeviceInfo->isDefault = MA_TRUE;
+ return MA_SUCCESS;
+}
+
+static ma_result ma_context_init_backend_apis__win32(ma_context* pContext)
+{
+#ifdef MA_WIN32_DESKTOP
+ /* Ole32.dll */
+ pContext->win32.hOle32DLL = ma_dlopen(pContext, "ole32.dll");
+ if (pContext->win32.hOle32DLL == NULL) {
+ return MA_FAILED_TO_INIT_BACKEND;
}
- if (deviceType == ma_device_type_playback) {
- MMRESULT result;
- MA_WAVEOUTCAPS2A caps;
+ pContext->win32.CoInitializeEx = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "CoInitializeEx");
+ pContext->win32.CoUninitialize = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "CoUninitialize");
+ pContext->win32.CoCreateInstance = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "CoCreateInstance");
+ pContext->win32.CoTaskMemFree = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "CoTaskMemFree");
+ pContext->win32.PropVariantClear = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "PropVariantClear");
+ pContext->win32.StringFromGUID2 = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "StringFromGUID2");
- MA_ZERO_OBJECT(&caps);
- result = ((MA_PFN_waveOutGetDevCapsA)pContext->winmm.waveOutGetDevCapsA)(winMMDeviceID, (WAVEOUTCAPSA*)&caps, sizeof(caps));
- if (result == MMSYSERR_NOERROR) {
- return ma_context_get_device_info_from_WAVEOUTCAPS2(pContext, &caps, pDeviceInfo);
- }
- } else {
- MMRESULT result;
- MA_WAVEINCAPS2A caps;
+ /* User32.dll */
+ pContext->win32.hUser32DLL = ma_dlopen(pContext, "user32.dll");
+ if (pContext->win32.hUser32DLL == NULL) {
+ return MA_FAILED_TO_INIT_BACKEND;
+ }
- MA_ZERO_OBJECT(&caps);
+ pContext->win32.GetForegroundWindow = (ma_proc)ma_dlsym(pContext, pContext->win32.hUser32DLL, "GetForegroundWindow");
+ pContext->win32.GetDesktopWindow = (ma_proc)ma_dlsym(pContext, pContext->win32.hUser32DLL, "GetDesktopWindow");
- result = ((MA_PFN_waveInGetDevCapsA)pContext->winmm.waveInGetDevCapsA)(winMMDeviceID, (WAVEINCAPSA*)&caps, sizeof(caps));
- if (result == MMSYSERR_NOERROR) {
- return ma_context_get_device_info_from_WAVEINCAPS2(pContext, &caps, pDeviceInfo);
- }
+
+ /* Advapi32.dll */
+ pContext->win32.hAdvapi32DLL = ma_dlopen(pContext, "advapi32.dll");
+ if (pContext->win32.hAdvapi32DLL == NULL) {
+ return MA_FAILED_TO_INIT_BACKEND;
}
- return MA_NO_DEVICE;
+ pContext->win32.RegOpenKeyExA = (ma_proc)ma_dlsym(pContext, pContext->win32.hAdvapi32DLL, "RegOpenKeyExA");
+ pContext->win32.RegCloseKey = (ma_proc)ma_dlsym(pContext, pContext->win32.hAdvapi32DLL, "RegCloseKey");
+ pContext->win32.RegQueryValueExA = (ma_proc)ma_dlsym(pContext, pContext->win32.hAdvapi32DLL, "RegQueryValueExA");
+#endif
+
+ ma_CoInitializeEx(pContext, NULL, MA_COINIT_VALUE);
+ return MA_SUCCESS;
}
+#else
+static ma_result ma_context_uninit_backend_apis__nix(ma_context* pContext)
+{
+#if defined(MA_USE_RUNTIME_LINKING_FOR_PTHREAD) && !defined(MA_NO_RUNTIME_LINKING)
+ ma_dlclose(pContext, pContext->posix.pthreadSO);
+#else
+ (void)pContext;
+#endif
+ return MA_SUCCESS;
+}
-static ma_result ma_device_uninit__winmm(ma_device* pDevice)
+static ma_result ma_context_init_backend_apis__nix(ma_context* pContext)
{
- MA_ASSERT(pDevice != NULL);
+ /* pthread */
+#if defined(MA_USE_RUNTIME_LINKING_FOR_PTHREAD) && !defined(MA_NO_RUNTIME_LINKING)
+ const char* libpthreadFileNames[] = {
+ "libpthread.so",
+ "libpthread.so.0",
+ "libpthread.dylib"
+ };
+ size_t i;
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- ((MA_PFN_waveInClose)pDevice->pContext->winmm.waveInClose)((HWAVEIN)pDevice->winmm.hDeviceCapture);
- CloseHandle((HANDLE)pDevice->winmm.hEventCapture);
+ for (i = 0; i < sizeof(libpthreadFileNames) / sizeof(libpthreadFileNames[0]); ++i) {
+ pContext->posix.pthreadSO = ma_dlopen(pContext, libpthreadFileNames[i]);
+ if (pContext->posix.pthreadSO != NULL) {
+ break;
+ }
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- ((MA_PFN_waveOutReset)pDevice->pContext->winmm.waveOutReset)((HWAVEOUT)pDevice->winmm.hDevicePlayback);
- ((MA_PFN_waveOutClose)pDevice->pContext->winmm.waveOutClose)((HWAVEOUT)pDevice->winmm.hDevicePlayback);
- CloseHandle((HANDLE)pDevice->winmm.hEventPlayback);
+ if (pContext->posix.pthreadSO == NULL) {
+ return MA_FAILED_TO_INIT_BACKEND;
}
- ma__free_from_callbacks(pDevice->winmm._pHeapData, &pDevice->pContext->allocationCallbacks);
-
- MA_ZERO_OBJECT(&pDevice->winmm); /* Safety. */
+ pContext->posix.pthread_create = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_create");
+ pContext->posix.pthread_join = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_join");
+ pContext->posix.pthread_mutex_init = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_mutex_init");
+ pContext->posix.pthread_mutex_destroy = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_mutex_destroy");
+ pContext->posix.pthread_mutex_lock = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_mutex_lock");
+ pContext->posix.pthread_mutex_unlock = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_mutex_unlock");
+ pContext->posix.pthread_cond_init = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_cond_init");
+ pContext->posix.pthread_cond_destroy = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_cond_destroy");
+ pContext->posix.pthread_cond_wait = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_cond_wait");
+ pContext->posix.pthread_cond_signal = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_cond_signal");
+ pContext->posix.pthread_attr_init = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_attr_init");
+ pContext->posix.pthread_attr_destroy = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_attr_destroy");
+ pContext->posix.pthread_attr_setschedpolicy = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_attr_setschedpolicy");
+ pContext->posix.pthread_attr_getschedparam = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_attr_getschedparam");
+ pContext->posix.pthread_attr_setschedparam = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_attr_setschedparam");
+#else
+ pContext->posix.pthread_create = (ma_proc)pthread_create;
+ pContext->posix.pthread_join = (ma_proc)pthread_join;
+ pContext->posix.pthread_mutex_init = (ma_proc)pthread_mutex_init;
+ pContext->posix.pthread_mutex_destroy = (ma_proc)pthread_mutex_destroy;
+ pContext->posix.pthread_mutex_lock = (ma_proc)pthread_mutex_lock;
+ pContext->posix.pthread_mutex_unlock = (ma_proc)pthread_mutex_unlock;
+ pContext->posix.pthread_cond_init = (ma_proc)pthread_cond_init;
+ pContext->posix.pthread_cond_destroy = (ma_proc)pthread_cond_destroy;
+ pContext->posix.pthread_cond_wait = (ma_proc)pthread_cond_wait;
+ pContext->posix.pthread_cond_signal = (ma_proc)pthread_cond_signal;
+ pContext->posix.pthread_attr_init = (ma_proc)pthread_attr_init;
+ pContext->posix.pthread_attr_destroy = (ma_proc)pthread_attr_destroy;
+#if !defined(__EMSCRIPTEN__)
+ pContext->posix.pthread_attr_setschedpolicy = (ma_proc)pthread_attr_setschedpolicy;
+ pContext->posix.pthread_attr_getschedparam = (ma_proc)pthread_attr_getschedparam;
+ pContext->posix.pthread_attr_setschedparam = (ma_proc)pthread_attr_setschedparam;
+#endif
+#endif
return MA_SUCCESS;
}
+#endif
-static ma_uint32 ma_calculate_period_size_in_frames_from_descriptor__winmm(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile)
+static ma_result ma_context_init_backend_apis(ma_context* pContext)
{
- /* WinMM has a minimum period size of 40ms. */
- ma_uint32 minPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(40, nativeSampleRate);
- ma_uint32 periodSizeInFrames;
-
- periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, nativeSampleRate, performanceProfile);
- if (periodSizeInFrames < minPeriodSizeInFrames) {
- periodSizeInFrames = minPeriodSizeInFrames;
- }
+ ma_result result;
+#ifdef MA_WIN32
+ result = ma_context_init_backend_apis__win32(pContext);
+#else
+ result = ma_context_init_backend_apis__nix(pContext);
+#endif
- return periodSizeInFrames;
+ return result;
}
-static ma_result ma_device_init__winmm(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+static ma_result ma_context_uninit_backend_apis(ma_context* pContext)
{
- const char* errorMsg = "";
- ma_result errorCode = MA_ERROR;
- ma_result result = MA_SUCCESS;
- ma_uint32 heapSize;
- UINT winMMDeviceIDPlayback = 0;
- UINT winMMDeviceIDCapture = 0;
-
- MA_ASSERT(pDevice != NULL);
+ ma_result result;
+#ifdef MA_WIN32
+ result = ma_context_uninit_backend_apis__win32(pContext);
+#else
+ result = ma_context_uninit_backend_apis__nix(pContext);
+#endif
- MA_ZERO_OBJECT(&pDevice->winmm);
+ return result;
+}
- if (pConfig->deviceType == ma_device_type_loopback) {
- return MA_DEVICE_TYPE_NOT_SUPPORTED;
- }
- /* No exlusive mode with WinMM. */
- if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) ||
- ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) {
- return MA_SHARE_MODE_NOT_SUPPORTED;
- }
+static ma_bool32 ma_context_is_backend_asynchronous(ma_context* pContext)
+{
+ MA_ASSERT(pContext != NULL);
- if (pDescriptorPlayback->pDeviceID != NULL) {
- winMMDeviceIDPlayback = (UINT)pDescriptorPlayback->pDeviceID->winmm;
- }
- if (pDescriptorCapture->pDeviceID != NULL) {
- winMMDeviceIDCapture = (UINT)pDescriptorCapture->pDeviceID->winmm;
+ if (pContext->callbacks.onDeviceRead == NULL && pContext->callbacks.onDeviceWrite == NULL) {
+ if (pContext->callbacks.onDeviceDataLoop == NULL) {
+ return MA_TRUE;
+ } else {
+ return MA_FALSE;
+ }
+ } else {
+ return MA_FALSE;
}
+}
- /* The capture device needs to be initialized first. */
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
- WAVEINCAPSA caps;
- WAVEFORMATEX wf;
- MMRESULT resultMM;
-
- /* We use an event to know when a new fragment needs to be enqueued. */
- pDevice->winmm.hEventCapture = (ma_handle)CreateEventW(NULL, TRUE, TRUE, NULL);
- if (pDevice->winmm.hEventCapture == NULL) {
- errorMsg = "[WinMM] Failed to create event for fragment enqueing for the capture device.", errorCode = ma_result_from_GetLastError(GetLastError());
- goto on_error;
- }
- /* The format should be based on the device's actual format. */
- if (((MA_PFN_waveInGetDevCapsA)pDevice->pContext->winmm.waveInGetDevCapsA)(winMMDeviceIDCapture, &caps, sizeof(caps)) != MMSYSERR_NOERROR) {
- errorMsg = "[WinMM] Failed to retrieve internal device caps.", errorCode = MA_FORMAT_NOT_SUPPORTED;
- goto on_error;
- }
+/* The default capacity doesn't need to be too big. */
+#ifndef MA_DEFAULT_DEVICE_JOB_QUEUE_CAPACITY
+#define MA_DEFAULT_DEVICE_JOB_QUEUE_CAPACITY 32
+#endif
- result = ma_formats_flags_to_WAVEFORMATEX__winmm(caps.dwFormats, caps.wChannels, &wf);
- if (result != MA_SUCCESS) {
- errorMsg = "[WinMM] Could not find appropriate format for internal device.", errorCode = result;
- goto on_error;
- }
+MA_API ma_device_job_thread_config ma_device_job_thread_config_init(void)
+{
+ ma_device_job_thread_config config;
- resultMM = ((MA_PFN_waveInOpen)pDevice->pContext->winmm.waveInOpen)((LPHWAVEIN)&pDevice->winmm.hDeviceCapture, winMMDeviceIDCapture, &wf, (DWORD_PTR)pDevice->winmm.hEventCapture, (DWORD_PTR)pDevice, CALLBACK_EVENT | WAVE_ALLOWSYNC);
- if (resultMM != MMSYSERR_NOERROR) {
- errorMsg = "[WinMM] Failed to open capture device.", errorCode = MA_FAILED_TO_OPEN_BACKEND_DEVICE;
- goto on_error;
- }
+ MA_ZERO_OBJECT(&config);
+ config.noThread = MA_FALSE;
+ config.jobQueueCapacity = MA_DEFAULT_DEVICE_JOB_QUEUE_CAPACITY;
+ config.jobQueueFlags = 0;
- pDescriptorCapture->format = ma_format_from_WAVEFORMATEX(&wf);
- pDescriptorCapture->channels = wf.nChannels;
- pDescriptorCapture->sampleRate = wf.nSamplesPerSec;
- ma_get_standard_channel_map(ma_standard_channel_map_microsoft, pDescriptorCapture->channels, pDescriptorCapture->channelMap);
- pDescriptorCapture->periodCount = pDescriptorCapture->periodCount;
- pDescriptorCapture->periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__winmm(pDescriptorCapture, pDescriptorCapture->sampleRate, pConfig->performanceProfile);
- }
+ return config;
+}
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- WAVEOUTCAPSA caps;
- WAVEFORMATEX wf;
- MMRESULT resultMM;
- /* We use an event to know when a new fragment needs to be enqueued. */
- pDevice->winmm.hEventPlayback = (ma_handle)CreateEvent(NULL, TRUE, TRUE, NULL);
- if (pDevice->winmm.hEventPlayback == NULL) {
- errorMsg = "[WinMM] Failed to create event for fragment enqueing for the playback device.", errorCode = ma_result_from_GetLastError(GetLastError());
- goto on_error;
- }
+static ma_thread_result MA_THREADCALL ma_device_job_thread_entry(void* pUserData)
+{
+ ma_device_job_thread* pJobThread = (ma_device_job_thread*)pUserData;
+ MA_ASSERT(pJobThread != NULL);
- /* The format should be based on the device's actual format. */
- if (((MA_PFN_waveOutGetDevCapsA)pDevice->pContext->winmm.waveOutGetDevCapsA)(winMMDeviceIDPlayback, &caps, sizeof(caps)) != MMSYSERR_NOERROR) {
- errorMsg = "[WinMM] Failed to retrieve internal device caps.", errorCode = MA_FORMAT_NOT_SUPPORTED;
- goto on_error;
- }
+ for (;;) {
+ ma_result result;
+ ma_job job;
- result = ma_formats_flags_to_WAVEFORMATEX__winmm(caps.dwFormats, caps.wChannels, &wf);
+ result = ma_device_job_thread_next(pJobThread, &job);
if (result != MA_SUCCESS) {
- errorMsg = "[WinMM] Could not find appropriate format for internal device.", errorCode = result;
- goto on_error;
+ break;
}
- resultMM = ((MA_PFN_waveOutOpen)pDevice->pContext->winmm.waveOutOpen)((LPHWAVEOUT)&pDevice->winmm.hDevicePlayback, winMMDeviceIDPlayback, &wf, (DWORD_PTR)pDevice->winmm.hEventPlayback, (DWORD_PTR)pDevice, CALLBACK_EVENT | WAVE_ALLOWSYNC);
- if (resultMM != MMSYSERR_NOERROR) {
- errorMsg = "[WinMM] Failed to open playback device.", errorCode = MA_FAILED_TO_OPEN_BACKEND_DEVICE;
- goto on_error;
+ if (job.toc.breakup.code == MA_JOB_TYPE_QUIT) {
+ break;
}
- pDescriptorPlayback->format = ma_format_from_WAVEFORMATEX(&wf);
- pDescriptorPlayback->channels = wf.nChannels;
- pDescriptorPlayback->sampleRate = wf.nSamplesPerSec;
- ma_get_standard_channel_map(ma_standard_channel_map_microsoft, pDescriptorPlayback->channels, pDescriptorPlayback->channelMap);
- pDescriptorPlayback->periodCount = pDescriptorPlayback->periodCount;
- pDescriptorPlayback->periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__winmm(pDescriptorPlayback, pDescriptorPlayback->sampleRate, pConfig->performanceProfile);
+ ma_job_process(&job);
}
- /*
- The heap allocated data is allocated like so:
+ return (ma_thread_result)0;
+}
- [Capture WAVEHDRs][Playback WAVEHDRs][Capture Intermediary Buffer][Playback Intermediary Buffer]
- */
- heapSize = 0;
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
- heapSize += sizeof(WAVEHDR)*pDescriptorCapture->periodCount + (pDescriptorCapture->periodSizeInFrames * pDescriptorCapture->periodCount * ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels));
- }
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- heapSize += sizeof(WAVEHDR)*pDescriptorPlayback->periodCount + (pDescriptorPlayback->periodSizeInFrames * pDescriptorPlayback->periodCount * ma_get_bytes_per_frame(pDescriptorPlayback->format, pDescriptorPlayback->channels));
- }
+MA_API ma_result ma_device_job_thread_init(const ma_device_job_thread_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_device_job_thread* pJobThread)
+{
+ ma_result result;
+ ma_job_queue_config jobQueueConfig;
- pDevice->winmm._pHeapData = (ma_uint8*)ma__calloc_from_callbacks(heapSize, &pDevice->pContext->allocationCallbacks);
- if (pDevice->winmm._pHeapData == NULL) {
- errorMsg = "[WinMM] Failed to allocate memory for the intermediary buffer.", errorCode = MA_OUT_OF_MEMORY;
- goto on_error;
+ if (pJobThread == NULL) {
+ return MA_INVALID_ARGS;
}
- MA_ZERO_MEMORY(pDevice->winmm._pHeapData, heapSize);
+ MA_ZERO_OBJECT(pJobThread);
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
- ma_uint32 iPeriod;
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
- if (pConfig->deviceType == ma_device_type_capture) {
- pDevice->winmm.pWAVEHDRCapture = pDevice->winmm._pHeapData;
- pDevice->winmm.pIntermediaryBufferCapture = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*(pDescriptorCapture->periodCount));
- } else {
- pDevice->winmm.pWAVEHDRCapture = pDevice->winmm._pHeapData;
- pDevice->winmm.pIntermediaryBufferCapture = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*(pDescriptorCapture->periodCount + pDescriptorPlayback->periodCount));
- }
- /* Prepare headers. */
- for (iPeriod = 0; iPeriod < pDescriptorCapture->periodCount; ++iPeriod) {
- ma_uint32 periodSizeInBytes = ma_get_period_size_in_bytes(pDescriptorCapture->periodSizeInFrames, pDescriptorCapture->format, pDescriptorCapture->channels);
+ /* Initialize the job queue before the thread to ensure it's in a valid state. */
+ jobQueueConfig = ma_job_queue_config_init(pConfig->jobQueueFlags, pConfig->jobQueueCapacity);
- ((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].lpData = (LPSTR)(pDevice->winmm.pIntermediaryBufferCapture + (periodSizeInBytes*iPeriod));
- ((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwBufferLength = periodSizeInBytes;
- ((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwFlags = 0L;
- ((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwLoops = 0L;
- ((MA_PFN_waveInPrepareHeader)pDevice->pContext->winmm.waveInPrepareHeader)((HWAVEIN)pDevice->winmm.hDeviceCapture, &((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod], sizeof(WAVEHDR));
+ result = ma_job_queue_init(&jobQueueConfig, pAllocationCallbacks, &pJobThread->jobQueue);
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to initialize job queue. */
+ }
- /*
- The user data of the WAVEHDR structure is a single flag the controls whether or not it is ready for writing. Consider it to be named "isLocked". A value of 0 means
- it's unlocked and available for writing. A value of 1 means it's locked.
- */
- ((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwUser = 0;
+
+ /* The thread needs to be initialized after the job queue to ensure the thread doesn't try to access it prematurely. */
+ if (pConfig->noThread == MA_FALSE) {
+ result = ma_thread_create(&pJobThread->thread, ma_thread_priority_normal, 0, ma_device_job_thread_entry, pJobThread, pAllocationCallbacks);
+ if (result != MA_SUCCESS) {
+ ma_job_queue_uninit(&pJobThread->jobQueue, pAllocationCallbacks);
+ return result; /* Failed to create the job thread. */
}
+
+ pJobThread->_hasThread = MA_TRUE;
+ } else {
+ pJobThread->_hasThread = MA_FALSE;
}
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- ma_uint32 iPeriod;
- if (pConfig->deviceType == ma_device_type_playback) {
- pDevice->winmm.pWAVEHDRPlayback = pDevice->winmm._pHeapData;
- pDevice->winmm.pIntermediaryBufferPlayback = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*pDescriptorPlayback->periodCount);
- } else {
- pDevice->winmm.pWAVEHDRPlayback = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*(pDescriptorCapture->periodCount));
- pDevice->winmm.pIntermediaryBufferPlayback = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*(pDescriptorCapture->periodCount + pDescriptorPlayback->periodCount)) + (pDescriptorCapture->periodSizeInFrames*pDescriptorCapture->periodCount*ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels));
- }
+ return MA_SUCCESS;
+}
- /* Prepare headers. */
- for (iPeriod = 0; iPeriod < pDescriptorPlayback->periodCount; ++iPeriod) {
- ma_uint32 periodSizeInBytes = ma_get_period_size_in_bytes(pDescriptorPlayback->periodSizeInFrames, pDescriptorPlayback->format, pDescriptorPlayback->channels);
+MA_API void ma_device_job_thread_uninit(ma_device_job_thread* pJobThread, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pJobThread == NULL) {
+ return;
+ }
- ((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].lpData = (LPSTR)(pDevice->winmm.pIntermediaryBufferPlayback + (periodSizeInBytes*iPeriod));
- ((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwBufferLength = periodSizeInBytes;
- ((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwFlags = 0L;
- ((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwLoops = 0L;
- ((MA_PFN_waveOutPrepareHeader)pDevice->pContext->winmm.waveOutPrepareHeader)((HWAVEOUT)pDevice->winmm.hDevicePlayback, &((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod], sizeof(WAVEHDR));
+ /* The first thing to do is post a quit message to the job queue. If we're using a thread we'll need to wait for it. */
+ {
+ ma_job job = ma_job_init(MA_JOB_TYPE_QUIT);
+ ma_device_job_thread_post(pJobThread, &job);
+ }
- /*
- The user data of the WAVEHDR structure is a single flag the controls whether or not it is ready for writing. Consider it to be named "isLocked". A value of 0 means
- it's unlocked and available for writing. A value of 1 means it's locked.
- */
- ((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwUser = 0;
- }
+ /* Wait for the thread to terminate naturally. */
+ if (pJobThread->_hasThread) {
+ ma_thread_wait(&pJobThread->thread);
}
- return MA_SUCCESS;
+ /* At this point the thread should be terminated so we can safely uninitialize the job queue. */
+ ma_job_queue_uninit(&pJobThread->jobQueue, pAllocationCallbacks);
+}
-on_error:
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- if (pDevice->winmm.pWAVEHDRCapture != NULL) {
- ma_uint32 iPeriod;
- for (iPeriod = 0; iPeriod < pDescriptorCapture->periodCount; ++iPeriod) {
- ((MA_PFN_waveInUnprepareHeader)pDevice->pContext->winmm.waveInUnprepareHeader)((HWAVEIN)pDevice->winmm.hDeviceCapture, &((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod], sizeof(WAVEHDR));
- }
- }
+MA_API ma_result ma_device_job_thread_post(ma_device_job_thread* pJobThread, const ma_job* pJob)
+{
+ if (pJobThread == NULL || pJob == NULL) {
+ return MA_INVALID_ARGS;
+ }
- ((MA_PFN_waveInClose)pDevice->pContext->winmm.waveInClose)((HWAVEIN)pDevice->winmm.hDeviceCapture);
+ return ma_job_queue_post(&pJobThread->jobQueue, pJob);
+}
+
+MA_API ma_result ma_device_job_thread_next(ma_device_job_thread* pJobThread, ma_job* pJob)
+{
+ if (pJob == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- if (pDevice->winmm.pWAVEHDRCapture != NULL) {
- ma_uint32 iPeriod;
- for (iPeriod = 0; iPeriod < pDescriptorPlayback->periodCount; ++iPeriod) {
- ((MA_PFN_waveOutUnprepareHeader)pDevice->pContext->winmm.waveOutUnprepareHeader)((HWAVEOUT)pDevice->winmm.hDevicePlayback, &((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod], sizeof(WAVEHDR));
- }
- }
+ MA_ZERO_OBJECT(pJob);
- ((MA_PFN_waveOutClose)pDevice->pContext->winmm.waveOutClose)((HWAVEOUT)pDevice->winmm.hDevicePlayback);
+ if (pJobThread == NULL) {
+ return MA_INVALID_ARGS;
}
- ma__free_from_callbacks(pDevice->winmm._pHeapData, &pDevice->pContext->allocationCallbacks);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, errorMsg, errorCode);
+ return ma_job_queue_next(&pJobThread->jobQueue, pJob);
}
-static ma_result ma_device_start__winmm(ma_device* pDevice)
+
+
+MA_API ma_context_config ma_context_config_init(void)
{
- MA_ASSERT(pDevice != NULL);
+ ma_context_config config;
+ MA_ZERO_OBJECT(&config);
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- MMRESULT resultMM;
- WAVEHDR* pWAVEHDR;
- ma_uint32 iPeriod;
+ return config;
+}
- pWAVEHDR = (WAVEHDR*)pDevice->winmm.pWAVEHDRCapture;
+MA_API ma_result ma_context_init(const ma_backend backends[], ma_uint32 backendCount, const ma_context_config* pConfig, ma_context* pContext)
+{
+ ma_result result;
+ ma_context_config defaultConfig;
+ ma_backend defaultBackends[ma_backend_null+1];
+ ma_uint32 iBackend;
+ ma_backend* pBackendsToIterate;
+ ma_uint32 backendsToIterateCount;
- /* Make sure the event is reset to a non-signaled state to ensure we don't prematurely return from WaitForSingleObject(). */
- ResetEvent((HANDLE)pDevice->winmm.hEventCapture);
+ if (pContext == NULL) {
+ return MA_INVALID_ARGS;
+ }
- /* To start the device we attach all of the buffers and then start it. As the buffers are filled with data we will get notifications. */
- for (iPeriod = 0; iPeriod < pDevice->capture.internalPeriods; ++iPeriod) {
- resultMM = ((MA_PFN_waveInAddBuffer)pDevice->pContext->winmm.waveInAddBuffer)((HWAVEIN)pDevice->winmm.hDeviceCapture, &((LPWAVEHDR)pDevice->winmm.pWAVEHDRCapture)[iPeriod], sizeof(WAVEHDR));
- if (resultMM != MMSYSERR_NOERROR) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WinMM] Failed to attach input buffers to capture device in preparation for capture.", ma_result_from_MMRESULT(resultMM));
- }
+ MA_ZERO_OBJECT(pContext);
- /* Make sure all of the buffers start out locked. We don't want to access them until the backend tells us we can. */
- pWAVEHDR[iPeriod].dwUser = 1; /* 1 = locked. */
- }
+ /* Always make sure the config is set first to ensure properties are available as soon as possible. */
+ if (pConfig == NULL) {
+ defaultConfig = ma_context_config_init();
+ pConfig = &defaultConfig;
+ }
- /* Capture devices need to be explicitly started, unlike playback devices. */
- resultMM = ((MA_PFN_waveInStart)pDevice->pContext->winmm.waveInStart)((HWAVEIN)pDevice->winmm.hDeviceCapture);
- if (resultMM != MMSYSERR_NOERROR) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WinMM] Failed to start backend device.", ma_result_from_MMRESULT(resultMM));
+ /* Allocation callbacks need to come first because they'll be passed around to other areas. */
+ result = ma_allocation_callbacks_init_copy(&pContext->allocationCallbacks, &pConfig->allocationCallbacks);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ /* Get a lot set up first so we can start logging ASAP. */
+ if (pConfig->pLog != NULL) {
+ pContext->pLog = pConfig->pLog;
+ } else {
+ result = ma_log_init(&pContext->allocationCallbacks, &pContext->log);
+ if (result == MA_SUCCESS) {
+ pContext->pLog = &pContext->log;
+ } else {
+ pContext->pLog = NULL; /* Logging is not available. */
}
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- /* Don't need to do anything for playback. It'll be started automatically in ma_device_start__winmm(). */
+ pContext->threadPriority = pConfig->threadPriority;
+ pContext->threadStackSize = pConfig->threadStackSize;
+ pContext->pUserData = pConfig->pUserData;
+
+ /* Backend APIs need to be initialized first. This is where external libraries will be loaded and linked. */
+ result = ma_context_init_backend_apis(pContext);
+ if (result != MA_SUCCESS) {
+ return result;
}
- return MA_SUCCESS;
-}
+ for (iBackend = 0; iBackend <= ma_backend_null; ++iBackend) {
+ defaultBackends[iBackend] = (ma_backend)iBackend;
+ }
-static ma_result ma_device_stop__winmm(ma_device* pDevice)
-{
- MMRESULT resultMM;
+ pBackendsToIterate = (ma_backend*)backends;
+ backendsToIterateCount = backendCount;
+ if (pBackendsToIterate == NULL) {
+ pBackendsToIterate = (ma_backend*)defaultBackends;
+ backendsToIterateCount = ma_countof(defaultBackends);
+ }
- MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(pBackendsToIterate != NULL);
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- if (pDevice->winmm.hDeviceCapture == NULL) {
- return MA_INVALID_ARGS;
- }
+ for (iBackend = 0; iBackend < backendsToIterateCount; iBackend += 1) {
+ ma_backend backend = pBackendsToIterate[iBackend];
- resultMM = ((MA_PFN_waveInReset)pDevice->pContext->winmm.waveInReset)((HWAVEIN)pDevice->winmm.hDeviceCapture);
- if (resultMM != MMSYSERR_NOERROR) {
- ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WinMM] WARNING: Failed to reset capture device.", ma_result_from_MMRESULT(resultMM));
- }
- }
+ /* Make sure all callbacks are reset so we don't accidentally drag in any from previously failed initialization attempts. */
+ MA_ZERO_OBJECT(&pContext->callbacks);
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- ma_uint32 iPeriod;
- WAVEHDR* pWAVEHDR;
+ /* These backends are using the new callback system. */
+ switch (backend) {
+ #ifdef MA_HAS_WASAPI
+ case ma_backend_wasapi:
+ {
+ pContext->callbacks.onContextInit = ma_context_init__wasapi;
+ } break;
+ #endif
+ #ifdef MA_HAS_DSOUND
+ case ma_backend_dsound:
+ {
+ pContext->callbacks.onContextInit = ma_context_init__dsound;
+ } break;
+ #endif
+ #ifdef MA_HAS_WINMM
+ case ma_backend_winmm:
+ {
+ pContext->callbacks.onContextInit = ma_context_init__winmm;
+ } break;
+ #endif
+ #ifdef MA_HAS_COREAUDIO
+ case ma_backend_coreaudio:
+ {
+ pContext->callbacks.onContextInit = ma_context_init__coreaudio;
+ } break;
+ #endif
+ #ifdef MA_HAS_SNDIO
+ case ma_backend_sndio:
+ {
+ pContext->callbacks.onContextInit = ma_context_init__sndio;
+ } break;
+ #endif
+ #ifdef MA_HAS_AUDIO4
+ case ma_backend_audio4:
+ {
+ pContext->callbacks.onContextInit = ma_context_init__audio4;
+ } break;
+ #endif
+ #ifdef MA_HAS_OSS
+ case ma_backend_oss:
+ {
+ pContext->callbacks.onContextInit = ma_context_init__oss;
+ } break;
+ #endif
+ #ifdef MA_HAS_PULSEAUDIO
+ case ma_backend_pulseaudio:
+ {
+ pContext->callbacks.onContextInit = ma_context_init__pulse;
+ } break;
+ #endif
+ #ifdef MA_HAS_ALSA
+ case ma_backend_alsa:
+ {
+ pContext->callbacks.onContextInit = ma_context_init__alsa;
+ } break;
+ #endif
+ #ifdef MA_HAS_JACK
+ case ma_backend_jack:
+ {
+ pContext->callbacks.onContextInit = ma_context_init__jack;
+ } break;
+ #endif
+ #ifdef MA_HAS_AAUDIO
+ case ma_backend_aaudio:
+ {
+ pContext->callbacks.onContextInit = ma_context_init__aaudio;
+ } break;
+ #endif
+ #ifdef MA_HAS_OPENSL
+ case ma_backend_opensl:
+ {
+ pContext->callbacks.onContextInit = ma_context_init__opensl;
+ } break;
+ #endif
+ #ifdef MA_HAS_WEBAUDIO
+ case ma_backend_webaudio:
+ {
+ pContext->callbacks.onContextInit = ma_context_init__webaudio;
+ } break;
+ #endif
+ #ifdef MA_HAS_CUSTOM
+ case ma_backend_custom:
+ {
+ /* Slightly different logic for custom backends. Custom backends can optionally set all of their callbacks in the config. */
+ pContext->callbacks = pConfig->custom;
+ } break;
+ #endif
+ #ifdef MA_HAS_NULL
+ case ma_backend_null:
+ {
+ pContext->callbacks.onContextInit = ma_context_init__null;
+ } break;
+ #endif
- if (pDevice->winmm.hDevicePlayback == NULL) {
- return MA_INVALID_ARGS;
+ default: break;
}
- /* We need to drain the device. To do this we just loop over each header and if it's locked just wait for the event. */
- pWAVEHDR = (WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback;
- for (iPeriod = 0; iPeriod < pDevice->playback.internalPeriods; iPeriod += 1) {
- if (pWAVEHDR[iPeriod].dwUser == 1) { /* 1 = locked. */
- if (WaitForSingleObject((HANDLE)pDevice->winmm.hEventPlayback, INFINITE) != WAIT_OBJECT_0) {
- break; /* An error occurred so just abandon ship and stop the device without draining. */
- }
+ if (pContext->callbacks.onContextInit != NULL) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "Attempting to initialize %s backend...\n", ma_get_backend_name(backend));
+ result = pContext->callbacks.onContextInit(pContext, pConfig, &pContext->callbacks);
+ } else {
+ result = MA_NO_BACKEND;
+ }
- pWAVEHDR[iPeriod].dwUser = 0;
+ /* If this iteration was successful, return. */
+ if (result == MA_SUCCESS) {
+ result = ma_mutex_init(&pContext->deviceEnumLock);
+ if (result != MA_SUCCESS) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_WARNING, "Failed to initialize mutex for device enumeration. ma_context_get_devices() is not thread safe.\n");
}
- }
- resultMM = ((MA_PFN_waveOutReset)pDevice->pContext->winmm.waveOutReset)((HWAVEOUT)pDevice->winmm.hDevicePlayback);
- if (resultMM != MMSYSERR_NOERROR) {
- ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WinMM] WARNING: Failed to reset playback device.", ma_result_from_MMRESULT(resultMM));
+ result = ma_mutex_init(&pContext->deviceInfoLock);
+ if (result != MA_SUCCESS) {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_WARNING, "Failed to initialize mutex for device info retrieval. ma_context_get_device_info() is not thread safe.\n");
+ }
+
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "System Architecture:\n");
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, " Endian: %s\n", ma_is_little_endian() ? "LE" : "BE");
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, " SSE2: %s\n", ma_has_sse2() ? "YES" : "NO");
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, " AVX2: %s\n", ma_has_avx2() ? "YES" : "NO");
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, " NEON: %s\n", ma_has_neon() ? "YES" : "NO");
+
+ pContext->backend = backend;
+ return result;
+ } else {
+ ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "Failed to initialize %s backend.\n", ma_get_backend_name(backend));
}
}
- return MA_SUCCESS;
+ /* If we get here it means an error occurred. */
+ MA_ZERO_OBJECT(pContext); /* Safety. */
+ return MA_NO_BACKEND;
}
-static ma_result ma_device_write__winmm(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten)
+MA_API ma_result ma_context_uninit(ma_context* pContext)
{
- ma_result result = MA_SUCCESS;
- MMRESULT resultMM;
- ma_uint32 totalFramesWritten;
- WAVEHDR* pWAVEHDR;
-
- MA_ASSERT(pDevice != NULL);
- MA_ASSERT(pPCMFrames != NULL);
-
- if (pFramesWritten != NULL) {
- *pFramesWritten = 0;
+ if (pContext == NULL) {
+ return MA_INVALID_ARGS;
}
- pWAVEHDR = (WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback;
-
- /* Keep processing as much data as possible. */
- totalFramesWritten = 0;
- while (totalFramesWritten < frameCount) {
- /* If the current header has some space available we need to write part of it. */
- if (pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwUser == 0) { /* 0 = unlocked. */
- /*
- This header has room in it. We copy as much of it as we can. If we end up fully consuming the buffer we need to
- write it out and move on to the next iteration.
- */
- ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
- ma_uint32 framesRemainingInHeader = (pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwBufferLength/bpf) - pDevice->winmm.headerFramesConsumedPlayback;
+ if (pContext->callbacks.onContextUninit != NULL) {
+ pContext->callbacks.onContextUninit(pContext);
+ }
- ma_uint32 framesToCopy = ma_min(framesRemainingInHeader, (frameCount - totalFramesWritten));
- const void* pSrc = ma_offset_ptr(pPCMFrames, totalFramesWritten*bpf);
- void* pDst = ma_offset_ptr(pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].lpData, pDevice->winmm.headerFramesConsumedPlayback*bpf);
- MA_COPY_MEMORY(pDst, pSrc, framesToCopy*bpf);
+ ma_mutex_uninit(&pContext->deviceEnumLock);
+ ma_mutex_uninit(&pContext->deviceInfoLock);
+ ma_free(pContext->pDeviceInfos, &pContext->allocationCallbacks);
+ ma_context_uninit_backend_apis(pContext);
- pDevice->winmm.headerFramesConsumedPlayback += framesToCopy;
- totalFramesWritten += framesToCopy;
+ if (pContext->pLog == &pContext->log) {
+ ma_log_uninit(&pContext->log);
+ }
- /* If we've consumed the buffer entirely we need to write it out to the device. */
- if (pDevice->winmm.headerFramesConsumedPlayback == (pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwBufferLength/bpf)) {
- pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwUser = 1; /* 1 = locked. */
- pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwFlags &= ~WHDR_DONE; /* <-- Need to make sure the WHDR_DONE flag is unset. */
+ return MA_SUCCESS;
+}
- /* Make sure the event is reset to a non-signaled state to ensure we don't prematurely return from WaitForSingleObject(). */
- ResetEvent((HANDLE)pDevice->winmm.hEventPlayback);
+MA_API size_t ma_context_sizeof()
+{
+ return sizeof(ma_context);
+}
- /* The device will be started here. */
- resultMM = ((MA_PFN_waveOutWrite)pDevice->pContext->winmm.waveOutWrite)((HWAVEOUT)pDevice->winmm.hDevicePlayback, &pWAVEHDR[pDevice->winmm.iNextHeaderPlayback], sizeof(WAVEHDR));
- if (resultMM != MMSYSERR_NOERROR) {
- result = ma_result_from_MMRESULT(resultMM);
- ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WinMM] waveOutWrite() failed.", result);
- break;
- }
- /* Make sure we move to the next header. */
- pDevice->winmm.iNextHeaderPlayback = (pDevice->winmm.iNextHeaderPlayback + 1) % pDevice->playback.internalPeriods;
- pDevice->winmm.headerFramesConsumedPlayback = 0;
- }
+MA_API ma_log* ma_context_get_log(ma_context* pContext)
+{
+ if (pContext == NULL) {
+ return NULL;
+ }
- /* If at this point we have consumed the entire input buffer we can return. */
- MA_ASSERT(totalFramesWritten <= frameCount);
- if (totalFramesWritten == frameCount) {
- break;
- }
+ return pContext->pLog;
+}
- /* Getting here means there's more to process. */
- continue;
- }
- /* Getting here means there isn't enough room in the buffer and we need to wait for one to become available. */
- if (WaitForSingleObject((HANDLE)pDevice->winmm.hEventPlayback, INFINITE) != WAIT_OBJECT_0) {
- result = MA_ERROR;
- break;
- }
+MA_API ma_result ma_context_enumerate_devices(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+{
+ ma_result result;
- /* Something happened. If the next buffer has been marked as done we need to reset a bit of state. */
- if ((pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwFlags & WHDR_DONE) != 0) {
- pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwUser = 0; /* 0 = unlocked (make it available for writing). */
- pDevice->winmm.headerFramesConsumedPlayback = 0;
- }
+ if (pContext == NULL || callback == NULL) {
+ return MA_INVALID_ARGS;
+ }
- /* If the device has been stopped we need to break. */
- if (ma_device_get_state(pDevice) != MA_STATE_STARTED) {
- break;
- }
+ if (pContext->callbacks.onContextEnumerateDevices == NULL) {
+ return MA_INVALID_OPERATION;
}
- if (pFramesWritten != NULL) {
- *pFramesWritten = totalFramesWritten;
+ ma_mutex_lock(&pContext->deviceEnumLock);
+ {
+ result = pContext->callbacks.onContextEnumerateDevices(pContext, callback, pUserData);
}
+ ma_mutex_unlock(&pContext->deviceEnumLock);
return result;
}
-static ma_result ma_device_read__winmm(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead)
-{
- ma_result result = MA_SUCCESS;
- MMRESULT resultMM;
- ma_uint32 totalFramesRead;
- WAVEHDR* pWAVEHDR;
- MA_ASSERT(pDevice != NULL);
- MA_ASSERT(pPCMFrames != NULL);
+static ma_bool32 ma_context_get_devices__enum_callback(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pInfo, void* pUserData)
+{
+ /*
+ We need to insert the device info into our main internal buffer. Where it goes depends on the device type. If it's a capture device
+ it's just appended to the end. If it's a playback device it's inserted just before the first capture device.
+ */
- if (pFramesRead != NULL) {
- *pFramesRead = 0;
- }
+ /*
+ First make sure we have room. Since the number of devices we add to the list is usually relatively small I've decided to use a
+ simple fixed size increment for buffer expansion.
+ */
+ const ma_uint32 bufferExpansionCount = 2;
+ const ma_uint32 totalDeviceInfoCount = pContext->playbackDeviceInfoCount + pContext->captureDeviceInfoCount;
- pWAVEHDR = (WAVEHDR*)pDevice->winmm.pWAVEHDRCapture;
+ if (totalDeviceInfoCount >= pContext->deviceInfoCapacity) {
+ ma_uint32 newCapacity = pContext->deviceInfoCapacity + bufferExpansionCount;
+ ma_device_info* pNewInfos = (ma_device_info*)ma_realloc(pContext->pDeviceInfos, sizeof(*pContext->pDeviceInfos)*newCapacity, &pContext->allocationCallbacks);
+ if (pNewInfos == NULL) {
+ return MA_FALSE; /* Out of memory. */
+ }
- /* Keep processing as much data as possible. */
- totalFramesRead = 0;
- while (totalFramesRead < frameCount) {
- /* If the current header has some space available we need to write part of it. */
- if (pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwUser == 0) { /* 0 = unlocked. */
- /* The buffer is available for reading. If we fully consume it we need to add it back to the buffer. */
- ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
- ma_uint32 framesRemainingInHeader = (pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwBufferLength/bpf) - pDevice->winmm.headerFramesConsumedCapture;
+ pContext->pDeviceInfos = pNewInfos;
+ pContext->deviceInfoCapacity = newCapacity;
+ }
- ma_uint32 framesToCopy = ma_min(framesRemainingInHeader, (frameCount - totalFramesRead));
- const void* pSrc = ma_offset_ptr(pWAVEHDR[pDevice->winmm.iNextHeaderCapture].lpData, pDevice->winmm.headerFramesConsumedCapture*bpf);
- void* pDst = ma_offset_ptr(pPCMFrames, totalFramesRead*bpf);
- MA_COPY_MEMORY(pDst, pSrc, framesToCopy*bpf);
+ if (deviceType == ma_device_type_playback) {
+ /* Playback. Insert just before the first capture device. */
- pDevice->winmm.headerFramesConsumedCapture += framesToCopy;
- totalFramesRead += framesToCopy;
+ /* The first thing to do is move all of the capture devices down a slot. */
+ ma_uint32 iFirstCaptureDevice = pContext->playbackDeviceInfoCount;
+ size_t iCaptureDevice;
+ for (iCaptureDevice = totalDeviceInfoCount; iCaptureDevice > iFirstCaptureDevice; --iCaptureDevice) {
+ pContext->pDeviceInfos[iCaptureDevice] = pContext->pDeviceInfos[iCaptureDevice-1];
+ }
- /* If we've consumed the buffer entirely we need to add it back to the device. */
- if (pDevice->winmm.headerFramesConsumedCapture == (pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwBufferLength/bpf)) {
- pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwUser = 1; /* 1 = locked. */
- pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwFlags &= ~WHDR_DONE; /* <-- Need to make sure the WHDR_DONE flag is unset. */
+ /* Now just insert where the first capture device was before moving it down a slot. */
+ pContext->pDeviceInfos[iFirstCaptureDevice] = *pInfo;
+ pContext->playbackDeviceInfoCount += 1;
+ } else {
+ /* Capture. Insert at the end. */
+ pContext->pDeviceInfos[totalDeviceInfoCount] = *pInfo;
+ pContext->captureDeviceInfoCount += 1;
+ }
- /* Make sure the event is reset to a non-signaled state to ensure we don't prematurely return from WaitForSingleObject(). */
- ResetEvent((HANDLE)pDevice->winmm.hEventCapture);
+ (void)pUserData;
+ return MA_TRUE;
+}
- /* The device will be started here. */
- resultMM = ((MA_PFN_waveInAddBuffer)pDevice->pContext->winmm.waveInAddBuffer)((HWAVEIN)pDevice->winmm.hDeviceCapture, &((LPWAVEHDR)pDevice->winmm.pWAVEHDRCapture)[pDevice->winmm.iNextHeaderCapture], sizeof(WAVEHDR));
- if (resultMM != MMSYSERR_NOERROR) {
- result = ma_result_from_MMRESULT(resultMM);
- ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[WinMM] waveInAddBuffer() failed.", result);
- break;
- }
+MA_API ma_result ma_context_get_devices(ma_context* pContext, ma_device_info** ppPlaybackDeviceInfos, ma_uint32* pPlaybackDeviceCount, ma_device_info** ppCaptureDeviceInfos, ma_uint32* pCaptureDeviceCount)
+{
+ ma_result result;
- /* Make sure we move to the next header. */
- pDevice->winmm.iNextHeaderCapture = (pDevice->winmm.iNextHeaderCapture + 1) % pDevice->capture.internalPeriods;
- pDevice->winmm.headerFramesConsumedCapture = 0;
- }
+ /* Safety. */
+ if (ppPlaybackDeviceInfos != NULL) *ppPlaybackDeviceInfos = NULL;
+ if (pPlaybackDeviceCount != NULL) *pPlaybackDeviceCount = 0;
+ if (ppCaptureDeviceInfos != NULL) *ppCaptureDeviceInfos = NULL;
+ if (pCaptureDeviceCount != NULL) *pCaptureDeviceCount = 0;
- /* If at this point we have filled the entire input buffer we can return. */
- MA_ASSERT(totalFramesRead <= frameCount);
- if (totalFramesRead == frameCount) {
- break;
- }
+ if (pContext == NULL) {
+ return MA_INVALID_ARGS;
+ }
- /* Getting here means there's more to process. */
- continue;
- }
+ if (pContext->callbacks.onContextEnumerateDevices == NULL) {
+ return MA_INVALID_OPERATION;
+ }
- /* Getting here means there isn't enough any data left to send to the client which means we need to wait for more. */
- if (WaitForSingleObject((HANDLE)pDevice->winmm.hEventCapture, INFINITE) != WAIT_OBJECT_0) {
- result = MA_ERROR;
- break;
- }
+ /* Note that we don't use ma_context_enumerate_devices() here because we want to do locking at a higher level. */
+ ma_mutex_lock(&pContext->deviceEnumLock);
+ {
+ /* Reset everything first. */
+ pContext->playbackDeviceInfoCount = 0;
+ pContext->captureDeviceInfoCount = 0;
- /* Something happened. If the next buffer has been marked as done we need to reset a bit of state. */
- if ((pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwFlags & WHDR_DONE) != 0) {
- pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwUser = 0; /* 0 = unlocked (make it available for reading). */
- pDevice->winmm.headerFramesConsumedCapture = 0;
- }
+ /* Now enumerate over available devices. */
+ result = pContext->callbacks.onContextEnumerateDevices(pContext, ma_context_get_devices__enum_callback, NULL);
+ if (result == MA_SUCCESS) {
+ /* Playback devices. */
+ if (ppPlaybackDeviceInfos != NULL) {
+ *ppPlaybackDeviceInfos = pContext->pDeviceInfos;
+ }
+ if (pPlaybackDeviceCount != NULL) {
+ *pPlaybackDeviceCount = pContext->playbackDeviceInfoCount;
+ }
- /* If the device has been stopped we need to break. */
- if (ma_device_get_state(pDevice) != MA_STATE_STARTED) {
- break;
+ /* Capture devices. */
+ if (ppCaptureDeviceInfos != NULL) {
+ *ppCaptureDeviceInfos = pContext->pDeviceInfos + pContext->playbackDeviceInfoCount; /* Capture devices come after playback devices. */
+ }
+ if (pCaptureDeviceCount != NULL) {
+ *pCaptureDeviceCount = pContext->captureDeviceInfoCount;
+ }
}
}
-
- if (pFramesRead != NULL) {
- *pFramesRead = totalFramesRead;
- }
+ ma_mutex_unlock(&pContext->deviceEnumLock);
return result;
}
-static ma_result ma_context_uninit__winmm(ma_context* pContext)
+MA_API ma_result ma_context_get_device_info(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
{
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pContext->backend == ma_backend_winmm);
+ ma_result result;
+ ma_device_info deviceInfo;
- ma_dlclose(pContext, pContext->winmm.hWinMM);
- return MA_SUCCESS;
-}
+ /* NOTE: Do not clear pDeviceInfo on entry. The reason is the pDeviceID may actually point to pDeviceInfo->id which will break things. */
+ if (pContext == NULL || pDeviceInfo == NULL) {
+ return MA_INVALID_ARGS;
+ }
-static ma_result ma_context_init__winmm(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
-{
- MA_ASSERT(pContext != NULL);
+ MA_ZERO_OBJECT(&deviceInfo);
- (void)pConfig;
+ /* Help the backend out by copying over the device ID if we have one. */
+ if (pDeviceID != NULL) {
+ MA_COPY_MEMORY(&deviceInfo.id, pDeviceID, sizeof(*pDeviceID));
+ }
- pContext->winmm.hWinMM = ma_dlopen(pContext, "winmm.dll");
- if (pContext->winmm.hWinMM == NULL) {
- return MA_NO_BACKEND;
+ if (pContext->callbacks.onContextGetDeviceInfo == NULL) {
+ return MA_INVALID_OPERATION;
}
- pContext->winmm.waveOutGetNumDevs = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutGetNumDevs");
- pContext->winmm.waveOutGetDevCapsA = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutGetDevCapsA");
- pContext->winmm.waveOutOpen = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutOpen");
- pContext->winmm.waveOutClose = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutClose");
- pContext->winmm.waveOutPrepareHeader = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutPrepareHeader");
- pContext->winmm.waveOutUnprepareHeader = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutUnprepareHeader");
- pContext->winmm.waveOutWrite = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutWrite");
- pContext->winmm.waveOutReset = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutReset");
- pContext->winmm.waveInGetNumDevs = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInGetNumDevs");
- pContext->winmm.waveInGetDevCapsA = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInGetDevCapsA");
- pContext->winmm.waveInOpen = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInOpen");
- pContext->winmm.waveInClose = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInClose");
- pContext->winmm.waveInPrepareHeader = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInPrepareHeader");
- pContext->winmm.waveInUnprepareHeader = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInUnprepareHeader");
- pContext->winmm.waveInAddBuffer = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInAddBuffer");
- pContext->winmm.waveInStart = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInStart");
- pContext->winmm.waveInReset = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInReset");
+ ma_mutex_lock(&pContext->deviceInfoLock);
+ {
+ result = pContext->callbacks.onContextGetDeviceInfo(pContext, deviceType, pDeviceID, &deviceInfo);
+ }
+ ma_mutex_unlock(&pContext->deviceInfoLock);
- pCallbacks->onContextInit = ma_context_init__winmm;
- pCallbacks->onContextUninit = ma_context_uninit__winmm;
- pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__winmm;
- pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__winmm;
- pCallbacks->onDeviceInit = ma_device_init__winmm;
- pCallbacks->onDeviceUninit = ma_device_uninit__winmm;
- pCallbacks->onDeviceStart = ma_device_start__winmm;
- pCallbacks->onDeviceStop = ma_device_stop__winmm;
- pCallbacks->onDeviceRead = ma_device_read__winmm;
- pCallbacks->onDeviceWrite = ma_device_write__winmm;
- pCallbacks->onDeviceDataLoop = NULL; /* This is a blocking read-write API, so this can be NULL since miniaudio will manage the audio thread for us. */
+ *pDeviceInfo = deviceInfo;
+ return result;
+}
- return MA_SUCCESS;
+MA_API ma_bool32 ma_context_is_loopback_supported(ma_context* pContext)
+{
+ if (pContext == NULL) {
+ return MA_FALSE;
+ }
+
+ return ma_is_loopback_supported(pContext->backend);
}
-#endif
+MA_API ma_device_config ma_device_config_init(ma_device_type deviceType)
+{
+ ma_device_config config;
+ MA_ZERO_OBJECT(&config);
+ config.deviceType = deviceType;
+ config.resampling = ma_resampler_config_init(ma_format_unknown, 0, 0, 0, ma_resample_algorithm_linear); /* Format/channels/rate don't matter here. */
+ return config;
+}
-/******************************************************************************
+MA_API ma_result ma_device_init(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice)
+{
+ ma_result result;
+ ma_device_descriptor descriptorPlayback;
+ ma_device_descriptor descriptorCapture;
-ALSA Backend
+ /* The context can be null, in which case we self-manage it. */
+ if (pContext == NULL) {
+ return ma_device_init_ex(NULL, 0, NULL, pConfig, pDevice);
+ }
-******************************************************************************/
-#ifdef MA_HAS_ALSA
+ if (pDevice == NULL) {
+ return MA_INVALID_ARGS;
+ }
-#include /* poll(), struct pollfd */
-#include /* eventfd() */
+ MA_ZERO_OBJECT(pDevice);
-#ifdef MA_NO_RUNTIME_LINKING
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
-/* asoundlib.h marks some functions with "inline" which isn't always supported. Need to emulate it. */
-#if !defined(__cplusplus)
- #if defined(__STRICT_ANSI__)
- #if !defined(inline)
- #define inline __inline__ __attribute__((always_inline))
- #define MA_INLINE_DEFINED
- #endif
- #endif
-#endif
-#include
-#if defined(MA_INLINE_DEFINED)
- #undef inline
- #undef MA_INLINE_DEFINED
-#endif
+ /* Check that we have our callbacks defined. */
+ if (pContext->callbacks.onDeviceInit == NULL) {
+ return MA_INVALID_OPERATION;
+ }
-typedef snd_pcm_uframes_t ma_snd_pcm_uframes_t;
-typedef snd_pcm_sframes_t ma_snd_pcm_sframes_t;
-typedef snd_pcm_stream_t ma_snd_pcm_stream_t;
-typedef snd_pcm_format_t ma_snd_pcm_format_t;
-typedef snd_pcm_access_t ma_snd_pcm_access_t;
-typedef snd_pcm_t ma_snd_pcm_t;
-typedef snd_pcm_hw_params_t ma_snd_pcm_hw_params_t;
-typedef snd_pcm_sw_params_t ma_snd_pcm_sw_params_t;
-typedef snd_pcm_format_mask_t ma_snd_pcm_format_mask_t;
-typedef snd_pcm_info_t ma_snd_pcm_info_t;
-typedef snd_pcm_channel_area_t ma_snd_pcm_channel_area_t;
-typedef snd_pcm_chmap_t ma_snd_pcm_chmap_t;
-typedef snd_pcm_state_t ma_snd_pcm_state_t;
+ /* Basic config validation. */
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
+ if (pConfig->capture.channels > MA_MAX_CHANNELS) {
+ return MA_INVALID_ARGS;
+ }
-/* snd_pcm_stream_t */
-#define MA_SND_PCM_STREAM_PLAYBACK SND_PCM_STREAM_PLAYBACK
-#define MA_SND_PCM_STREAM_CAPTURE SND_PCM_STREAM_CAPTURE
+ if (!ma__is_channel_map_valid(pConfig->capture.pChannelMap, pConfig->capture.channels)) {
+ return MA_INVALID_ARGS;
+ }
+ }
-/* snd_pcm_format_t */
-#define MA_SND_PCM_FORMAT_UNKNOWN SND_PCM_FORMAT_UNKNOWN
-#define MA_SND_PCM_FORMAT_U8 SND_PCM_FORMAT_U8
-#define MA_SND_PCM_FORMAT_S16_LE SND_PCM_FORMAT_S16_LE
-#define MA_SND_PCM_FORMAT_S16_BE SND_PCM_FORMAT_S16_BE
-#define MA_SND_PCM_FORMAT_S24_LE SND_PCM_FORMAT_S24_LE
-#define MA_SND_PCM_FORMAT_S24_BE SND_PCM_FORMAT_S24_BE
-#define MA_SND_PCM_FORMAT_S32_LE SND_PCM_FORMAT_S32_LE
-#define MA_SND_PCM_FORMAT_S32_BE SND_PCM_FORMAT_S32_BE
-#define MA_SND_PCM_FORMAT_FLOAT_LE SND_PCM_FORMAT_FLOAT_LE
-#define MA_SND_PCM_FORMAT_FLOAT_BE SND_PCM_FORMAT_FLOAT_BE
-#define MA_SND_PCM_FORMAT_FLOAT64_LE SND_PCM_FORMAT_FLOAT64_LE
-#define MA_SND_PCM_FORMAT_FLOAT64_BE SND_PCM_FORMAT_FLOAT64_BE
-#define MA_SND_PCM_FORMAT_MU_LAW SND_PCM_FORMAT_MU_LAW
-#define MA_SND_PCM_FORMAT_A_LAW SND_PCM_FORMAT_A_LAW
-#define MA_SND_PCM_FORMAT_S24_3LE SND_PCM_FORMAT_S24_3LE
-#define MA_SND_PCM_FORMAT_S24_3BE SND_PCM_FORMAT_S24_3BE
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex || pConfig->deviceType == ma_device_type_loopback) {
+ if (pConfig->playback.channels > MA_MAX_CHANNELS) {
+ return MA_INVALID_ARGS;
+ }
-/* ma_snd_pcm_access_t */
-#define MA_SND_PCM_ACCESS_MMAP_INTERLEAVED SND_PCM_ACCESS_MMAP_INTERLEAVED
-#define MA_SND_PCM_ACCESS_MMAP_NONINTERLEAVED SND_PCM_ACCESS_MMAP_NONINTERLEAVED
-#define MA_SND_PCM_ACCESS_MMAP_COMPLEX SND_PCM_ACCESS_MMAP_COMPLEX
-#define MA_SND_PCM_ACCESS_RW_INTERLEAVED SND_PCM_ACCESS_RW_INTERLEAVED
-#define MA_SND_PCM_ACCESS_RW_NONINTERLEAVED SND_PCM_ACCESS_RW_NONINTERLEAVED
+ if (!ma__is_channel_map_valid(pConfig->playback.pChannelMap, pConfig->playback.channels)) {
+ return MA_INVALID_ARGS;
+ }
+ }
-/* Channel positions. */
-#define MA_SND_CHMAP_UNKNOWN SND_CHMAP_UNKNOWN
-#define MA_SND_CHMAP_NA SND_CHMAP_NA
-#define MA_SND_CHMAP_MONO SND_CHMAP_MONO
-#define MA_SND_CHMAP_FL SND_CHMAP_FL
-#define MA_SND_CHMAP_FR SND_CHMAP_FR
-#define MA_SND_CHMAP_RL SND_CHMAP_RL
-#define MA_SND_CHMAP_RR SND_CHMAP_RR
-#define MA_SND_CHMAP_FC SND_CHMAP_FC
-#define MA_SND_CHMAP_LFE SND_CHMAP_LFE
-#define MA_SND_CHMAP_SL SND_CHMAP_SL
-#define MA_SND_CHMAP_SR SND_CHMAP_SR
-#define MA_SND_CHMAP_RC SND_CHMAP_RC
-#define MA_SND_CHMAP_FLC SND_CHMAP_FLC
-#define MA_SND_CHMAP_FRC SND_CHMAP_FRC
-#define MA_SND_CHMAP_RLC SND_CHMAP_RLC
-#define MA_SND_CHMAP_RRC SND_CHMAP_RRC
-#define MA_SND_CHMAP_FLW SND_CHMAP_FLW
-#define MA_SND_CHMAP_FRW SND_CHMAP_FRW
-#define MA_SND_CHMAP_FLH SND_CHMAP_FLH
-#define MA_SND_CHMAP_FCH SND_CHMAP_FCH
-#define MA_SND_CHMAP_FRH SND_CHMAP_FRH
-#define MA_SND_CHMAP_TC SND_CHMAP_TC
-#define MA_SND_CHMAP_TFL SND_CHMAP_TFL
-#define MA_SND_CHMAP_TFR SND_CHMAP_TFR
-#define MA_SND_CHMAP_TFC SND_CHMAP_TFC
-#define MA_SND_CHMAP_TRL SND_CHMAP_TRL
-#define MA_SND_CHMAP_TRR SND_CHMAP_TRR
-#define MA_SND_CHMAP_TRC SND_CHMAP_TRC
-#define MA_SND_CHMAP_TFLC SND_CHMAP_TFLC
-#define MA_SND_CHMAP_TFRC SND_CHMAP_TFRC
-#define MA_SND_CHMAP_TSL SND_CHMAP_TSL
-#define MA_SND_CHMAP_TSR SND_CHMAP_TSR
-#define MA_SND_CHMAP_LLFE SND_CHMAP_LLFE
-#define MA_SND_CHMAP_RLFE SND_CHMAP_RLFE
-#define MA_SND_CHMAP_BC SND_CHMAP_BC
-#define MA_SND_CHMAP_BLC SND_CHMAP_BLC
-#define MA_SND_CHMAP_BRC SND_CHMAP_BRC
+ pDevice->pContext = pContext;
-/* Open mode flags. */
-#define MA_SND_PCM_NO_AUTO_RESAMPLE SND_PCM_NO_AUTO_RESAMPLE
-#define MA_SND_PCM_NO_AUTO_CHANNELS SND_PCM_NO_AUTO_CHANNELS
-#define MA_SND_PCM_NO_AUTO_FORMAT SND_PCM_NO_AUTO_FORMAT
-#else
-#include /* For EPIPE, etc. */
-typedef unsigned long ma_snd_pcm_uframes_t;
-typedef long ma_snd_pcm_sframes_t;
-typedef int ma_snd_pcm_stream_t;
-typedef int ma_snd_pcm_format_t;
-typedef int ma_snd_pcm_access_t;
-typedef int ma_snd_pcm_state_t;
-typedef struct ma_snd_pcm_t ma_snd_pcm_t;
-typedef struct ma_snd_pcm_hw_params_t ma_snd_pcm_hw_params_t;
-typedef struct ma_snd_pcm_sw_params_t ma_snd_pcm_sw_params_t;
-typedef struct ma_snd_pcm_format_mask_t ma_snd_pcm_format_mask_t;
-typedef struct ma_snd_pcm_info_t ma_snd_pcm_info_t;
-typedef struct
-{
- void* addr;
- unsigned int first;
- unsigned int step;
-} ma_snd_pcm_channel_area_t;
-typedef struct
-{
- unsigned int channels;
- unsigned int pos[1];
-} ma_snd_pcm_chmap_t;
+ /* Set the user data and log callback ASAP to ensure it is available for the entire initialization process. */
+ pDevice->pUserData = pConfig->pUserData;
+ pDevice->onData = pConfig->dataCallback;
+ pDevice->onNotification = pConfig->notificationCallback;
+ pDevice->onStop = pConfig->stopCallback;
-/* snd_pcm_state_t */
-#define MA_SND_PCM_STATE_OPEN 0
-#define MA_SND_PCM_STATE_SETUP 1
-#define MA_SND_PCM_STATE_PREPARED 2
-#define MA_SND_PCM_STATE_RUNNING 3
-#define MA_SND_PCM_STATE_XRUN 4
-#define MA_SND_PCM_STATE_DRAINING 5
-#define MA_SND_PCM_STATE_PAUSED 6
-#define MA_SND_PCM_STATE_SUSPENDED 7
-#define MA_SND_PCM_STATE_DISCONNECTED 8
+ if (pConfig->playback.pDeviceID != NULL) {
+ MA_COPY_MEMORY(&pDevice->playback.id, pConfig->playback.pDeviceID, sizeof(pDevice->playback.id));
+ pDevice->playback.pID = &pDevice->playback.id;
+ } else {
+ pDevice->playback.pID = NULL;
+ }
-/* snd_pcm_stream_t */
-#define MA_SND_PCM_STREAM_PLAYBACK 0
-#define MA_SND_PCM_STREAM_CAPTURE 1
+ if (pConfig->capture.pDeviceID != NULL) {
+ MA_COPY_MEMORY(&pDevice->capture.id, pConfig->capture.pDeviceID, sizeof(pDevice->capture.id));
+ pDevice->capture.pID = &pDevice->capture.id;
+ } else {
+ pDevice->capture.pID = NULL;
+ }
-/* snd_pcm_format_t */
-#define MA_SND_PCM_FORMAT_UNKNOWN -1
-#define MA_SND_PCM_FORMAT_U8 1
-#define MA_SND_PCM_FORMAT_S16_LE 2
-#define MA_SND_PCM_FORMAT_S16_BE 3
-#define MA_SND_PCM_FORMAT_S24_LE 6
-#define MA_SND_PCM_FORMAT_S24_BE 7
-#define MA_SND_PCM_FORMAT_S32_LE 10
-#define MA_SND_PCM_FORMAT_S32_BE 11
-#define MA_SND_PCM_FORMAT_FLOAT_LE 14
-#define MA_SND_PCM_FORMAT_FLOAT_BE 15
-#define MA_SND_PCM_FORMAT_FLOAT64_LE 16
-#define MA_SND_PCM_FORMAT_FLOAT64_BE 17
-#define MA_SND_PCM_FORMAT_MU_LAW 20
-#define MA_SND_PCM_FORMAT_A_LAW 21
-#define MA_SND_PCM_FORMAT_S24_3LE 32
-#define MA_SND_PCM_FORMAT_S24_3BE 33
+ pDevice->noPreSilencedOutputBuffer = pConfig->noPreSilencedOutputBuffer;
+ pDevice->noClip = pConfig->noClip;
+ pDevice->noDisableDenormals = pConfig->noDisableDenormals;
+ pDevice->noFixedSizedCallback = pConfig->noFixedSizedCallback;
+ pDevice->masterVolumeFactor = 1;
-/* snd_pcm_access_t */
-#define MA_SND_PCM_ACCESS_MMAP_INTERLEAVED 0
-#define MA_SND_PCM_ACCESS_MMAP_NONINTERLEAVED 1
-#define MA_SND_PCM_ACCESS_MMAP_COMPLEX 2
-#define MA_SND_PCM_ACCESS_RW_INTERLEAVED 3
-#define MA_SND_PCM_ACCESS_RW_NONINTERLEAVED 4
+ pDevice->type = pConfig->deviceType;
+ pDevice->sampleRate = pConfig->sampleRate;
+ pDevice->resampling.algorithm = pConfig->resampling.algorithm;
+ pDevice->resampling.linear.lpfOrder = pConfig->resampling.linear.lpfOrder;
+ pDevice->resampling.pBackendVTable = pConfig->resampling.pBackendVTable;
+ pDevice->resampling.pBackendUserData = pConfig->resampling.pBackendUserData;
-/* Channel positions. */
-#define MA_SND_CHMAP_UNKNOWN 0
-#define MA_SND_CHMAP_NA 1
-#define MA_SND_CHMAP_MONO 2
-#define MA_SND_CHMAP_FL 3
-#define MA_SND_CHMAP_FR 4
-#define MA_SND_CHMAP_RL 5
-#define MA_SND_CHMAP_RR 6
-#define MA_SND_CHMAP_FC 7
-#define MA_SND_CHMAP_LFE 8
-#define MA_SND_CHMAP_SL 9
-#define MA_SND_CHMAP_SR 10
-#define MA_SND_CHMAP_RC 11
-#define MA_SND_CHMAP_FLC 12
-#define MA_SND_CHMAP_FRC 13
-#define MA_SND_CHMAP_RLC 14
-#define MA_SND_CHMAP_RRC 15
-#define MA_SND_CHMAP_FLW 16
-#define MA_SND_CHMAP_FRW 17
-#define MA_SND_CHMAP_FLH 18
-#define MA_SND_CHMAP_FCH 19
-#define MA_SND_CHMAP_FRH 20
-#define MA_SND_CHMAP_TC 21
-#define MA_SND_CHMAP_TFL 22
-#define MA_SND_CHMAP_TFR 23
-#define MA_SND_CHMAP_TFC 24
-#define MA_SND_CHMAP_TRL 25
-#define MA_SND_CHMAP_TRR 26
-#define MA_SND_CHMAP_TRC 27
-#define MA_SND_CHMAP_TFLC 28
-#define MA_SND_CHMAP_TFRC 29
-#define MA_SND_CHMAP_TSL 30
-#define MA_SND_CHMAP_TSR 31
-#define MA_SND_CHMAP_LLFE 32
-#define MA_SND_CHMAP_RLFE 33
-#define MA_SND_CHMAP_BC 34
-#define MA_SND_CHMAP_BLC 35
-#define MA_SND_CHMAP_BRC 36
+ pDevice->capture.shareMode = pConfig->capture.shareMode;
+ pDevice->capture.format = pConfig->capture.format;
+ pDevice->capture.channels = pConfig->capture.channels;
+ ma_channel_map_copy_or_default(pDevice->capture.channelMap, ma_countof(pDevice->capture.channelMap), pConfig->capture.pChannelMap, pConfig->capture.channels);
+ pDevice->capture.channelMixMode = pConfig->capture.channelMixMode;
-/* Open mode flags. */
-#define MA_SND_PCM_NO_AUTO_RESAMPLE 0x00010000
-#define MA_SND_PCM_NO_AUTO_CHANNELS 0x00020000
-#define MA_SND_PCM_NO_AUTO_FORMAT 0x00040000
-#endif
+ pDevice->playback.shareMode = pConfig->playback.shareMode;
+ pDevice->playback.format = pConfig->playback.format;
+ pDevice->playback.channels = pConfig->playback.channels;
+ ma_channel_map_copy_or_default(pDevice->playback.channelMap, ma_countof(pDevice->playback.channelMap), pConfig->playback.pChannelMap, pConfig->playback.channels);
+ pDevice->playback.channelMixMode = pConfig->playback.channelMixMode;
-typedef int (* ma_snd_pcm_open_proc) (ma_snd_pcm_t **pcm, const char *name, ma_snd_pcm_stream_t stream, int mode);
-typedef int (* ma_snd_pcm_close_proc) (ma_snd_pcm_t *pcm);
-typedef size_t (* ma_snd_pcm_hw_params_sizeof_proc) (void);
-typedef int (* ma_snd_pcm_hw_params_any_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params);
-typedef int (* ma_snd_pcm_hw_params_set_format_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t val);
-typedef int (* ma_snd_pcm_hw_params_set_format_first_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t *format);
-typedef void (* ma_snd_pcm_hw_params_get_format_mask_proc) (ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_mask_t *mask);
-typedef int (* ma_snd_pcm_hw_params_set_channels_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val);
-typedef int (* ma_snd_pcm_hw_params_set_channels_near_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *val);
-typedef int (* ma_snd_pcm_hw_params_set_channels_minmax_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *minimum, unsigned int *maximum);
-typedef int (* ma_snd_pcm_hw_params_set_rate_resample_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val);
-typedef int (* ma_snd_pcm_hw_params_set_rate_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val, int dir);
-typedef int (* ma_snd_pcm_hw_params_set_rate_near_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *val, int *dir);
-typedef int (* ma_snd_pcm_hw_params_set_buffer_size_near_proc)(ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_uframes_t *val);
-typedef int (* ma_snd_pcm_hw_params_set_periods_near_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *val, int *dir);
-typedef int (* ma_snd_pcm_hw_params_set_access_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_access_t _access);
-typedef int (* ma_snd_pcm_hw_params_get_format_proc) (const ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t *format);
-typedef int (* ma_snd_pcm_hw_params_get_channels_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val);
-typedef int (* ma_snd_pcm_hw_params_get_channels_min_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val);
-typedef int (* ma_snd_pcm_hw_params_get_channels_max_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val);
-typedef int (* ma_snd_pcm_hw_params_get_rate_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *rate, int *dir);
-typedef int (* ma_snd_pcm_hw_params_get_rate_min_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *rate, int *dir);
-typedef int (* ma_snd_pcm_hw_params_get_rate_max_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *rate, int *dir);
-typedef int (* ma_snd_pcm_hw_params_get_buffer_size_proc) (const ma_snd_pcm_hw_params_t *params, ma_snd_pcm_uframes_t *val);
-typedef int (* ma_snd_pcm_hw_params_get_periods_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val, int *dir);
-typedef int (* ma_snd_pcm_hw_params_get_access_proc) (const ma_snd_pcm_hw_params_t *params, ma_snd_pcm_access_t *_access);
-typedef int (* ma_snd_pcm_hw_params_test_format_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t val);
-typedef int (* ma_snd_pcm_hw_params_test_channels_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val);
-typedef int (* ma_snd_pcm_hw_params_test_rate_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val, int dir);
-typedef int (* ma_snd_pcm_hw_params_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params);
-typedef size_t (* ma_snd_pcm_sw_params_sizeof_proc) (void);
-typedef int (* ma_snd_pcm_sw_params_current_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params);
-typedef int (* ma_snd_pcm_sw_params_get_boundary_proc) (const ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t* val);
-typedef int (* ma_snd_pcm_sw_params_set_avail_min_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t val);
-typedef int (* ma_snd_pcm_sw_params_set_start_threshold_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t val);
-typedef int (* ma_snd_pcm_sw_params_set_stop_threshold_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t val);
-typedef int (* ma_snd_pcm_sw_params_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params);
-typedef size_t (* ma_snd_pcm_format_mask_sizeof_proc) (void);
-typedef int (* ma_snd_pcm_format_mask_test_proc) (const ma_snd_pcm_format_mask_t *mask, ma_snd_pcm_format_t val);
-typedef ma_snd_pcm_chmap_t * (* ma_snd_pcm_get_chmap_proc) (ma_snd_pcm_t *pcm);
-typedef ma_snd_pcm_state_t (* ma_snd_pcm_state_proc) (ma_snd_pcm_t *pcm);
-typedef int (* ma_snd_pcm_prepare_proc) (ma_snd_pcm_t *pcm);
-typedef int (* ma_snd_pcm_start_proc) (ma_snd_pcm_t *pcm);
-typedef int (* ma_snd_pcm_drop_proc) (ma_snd_pcm_t *pcm);
-typedef int (* ma_snd_pcm_drain_proc) (ma_snd_pcm_t *pcm);
-typedef int (* ma_snd_pcm_reset_proc) (ma_snd_pcm_t *pcm);
-typedef int (* ma_snd_device_name_hint_proc) (int card, const char *iface, void ***hints);
-typedef char * (* ma_snd_device_name_get_hint_proc) (const void *hint, const char *id);
-typedef int (* ma_snd_card_get_index_proc) (const char *name);
-typedef int (* ma_snd_device_name_free_hint_proc) (void **hints);
-typedef int (* ma_snd_pcm_mmap_begin_proc) (ma_snd_pcm_t *pcm, const ma_snd_pcm_channel_area_t **areas, ma_snd_pcm_uframes_t *offset, ma_snd_pcm_uframes_t *frames);
-typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_mmap_commit_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_uframes_t offset, ma_snd_pcm_uframes_t frames);
-typedef int (* ma_snd_pcm_recover_proc) (ma_snd_pcm_t *pcm, int err, int silent);
-typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_readi_proc) (ma_snd_pcm_t *pcm, void *buffer, ma_snd_pcm_uframes_t size);
-typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_writei_proc) (ma_snd_pcm_t *pcm, const void *buffer, ma_snd_pcm_uframes_t size);
-typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_avail_proc) (ma_snd_pcm_t *pcm);
-typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_avail_update_proc) (ma_snd_pcm_t *pcm);
-typedef int (* ma_snd_pcm_wait_proc) (ma_snd_pcm_t *pcm, int timeout);
-typedef int (* ma_snd_pcm_nonblock_proc) (ma_snd_pcm_t *pcm, int nonblock);
-typedef int (* ma_snd_pcm_info_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_info_t* info);
-typedef size_t (* ma_snd_pcm_info_sizeof_proc) (void);
-typedef const char* (* ma_snd_pcm_info_get_name_proc) (const ma_snd_pcm_info_t* info);
-typedef int (* ma_snd_pcm_poll_descriptors_proc) (ma_snd_pcm_t *pcm, struct pollfd *pfds, unsigned int space);
-typedef int (* ma_snd_pcm_poll_descriptors_count_proc) (ma_snd_pcm_t *pcm);
-typedef int (* ma_snd_pcm_poll_descriptors_revents_proc) (ma_snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents);
-typedef int (* ma_snd_config_update_free_global_proc) (void);
-/* This array specifies each of the common devices that can be used for both playback and capture. */
-static const char* g_maCommonDeviceNamesALSA[] = {
- "default",
- "null",
- "pulse",
- "jack"
-};
+ result = ma_mutex_init(&pDevice->startStopLock);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
-/* This array allows us to blacklist specific playback devices. */
-static const char* g_maBlacklistedPlaybackDeviceNamesALSA[] = {
- ""
-};
+ /*
+ When the device is started, the worker thread is the one that does the actual startup of the backend device. We
+ use a semaphore to wait for the background thread to finish the work. The same applies for stopping the device.
-/* This array allows us to blacklist specific capture devices. */
-static const char* g_maBlacklistedCaptureDeviceNamesALSA[] = {
- ""
-};
+ Each of these semaphores is released internally by the worker thread when the work is completed. The start
+ semaphore is also used to wake up the worker thread.
+ */
+ result = ma_event_init(&pDevice->wakeupEvent);
+ if (result != MA_SUCCESS) {
+ ma_mutex_uninit(&pDevice->startStopLock);
+ return result;
+ }
+
+ result = ma_event_init(&pDevice->startEvent);
+ if (result != MA_SUCCESS) {
+ ma_event_uninit(&pDevice->wakeupEvent);
+ ma_mutex_uninit(&pDevice->startStopLock);
+ return result;
+ }
+ result = ma_event_init(&pDevice->stopEvent);
+ if (result != MA_SUCCESS) {
+ ma_event_uninit(&pDevice->startEvent);
+ ma_event_uninit(&pDevice->wakeupEvent);
+ ma_mutex_uninit(&pDevice->startStopLock);
+ return result;
+ }
-static ma_snd_pcm_format_t ma_convert_ma_format_to_alsa_format(ma_format format)
-{
- ma_snd_pcm_format_t ALSAFormats[] = {
- MA_SND_PCM_FORMAT_UNKNOWN, /* ma_format_unknown */
- MA_SND_PCM_FORMAT_U8, /* ma_format_u8 */
- MA_SND_PCM_FORMAT_S16_LE, /* ma_format_s16 */
- MA_SND_PCM_FORMAT_S24_3LE, /* ma_format_s24 */
- MA_SND_PCM_FORMAT_S32_LE, /* ma_format_s32 */
- MA_SND_PCM_FORMAT_FLOAT_LE /* ma_format_f32 */
- };
- if (ma_is_big_endian()) {
- ALSAFormats[0] = MA_SND_PCM_FORMAT_UNKNOWN;
- ALSAFormats[1] = MA_SND_PCM_FORMAT_U8;
- ALSAFormats[2] = MA_SND_PCM_FORMAT_S16_BE;
- ALSAFormats[3] = MA_SND_PCM_FORMAT_S24_3BE;
- ALSAFormats[4] = MA_SND_PCM_FORMAT_S32_BE;
- ALSAFormats[5] = MA_SND_PCM_FORMAT_FLOAT_BE;
+ MA_ZERO_OBJECT(&descriptorPlayback);
+ descriptorPlayback.pDeviceID = pConfig->playback.pDeviceID;
+ descriptorPlayback.shareMode = pConfig->playback.shareMode;
+ descriptorPlayback.format = pConfig->playback.format;
+ descriptorPlayback.channels = pConfig->playback.channels;
+ descriptorPlayback.sampleRate = pConfig->sampleRate;
+ ma_channel_map_copy_or_default(descriptorPlayback.channelMap, ma_countof(descriptorPlayback.channelMap), pConfig->playback.pChannelMap, pConfig->playback.channels);
+ descriptorPlayback.periodSizeInFrames = pConfig->periodSizeInFrames;
+ descriptorPlayback.periodSizeInMilliseconds = pConfig->periodSizeInMilliseconds;
+ descriptorPlayback.periodCount = pConfig->periods;
+
+ if (descriptorPlayback.periodCount == 0) {
+ descriptorPlayback.periodCount = MA_DEFAULT_PERIODS;
}
- return ALSAFormats[format];
-}
-static ma_format ma_format_from_alsa(ma_snd_pcm_format_t formatALSA)
-{
- if (ma_is_little_endian()) {
- switch (formatALSA) {
- case MA_SND_PCM_FORMAT_S16_LE: return ma_format_s16;
- case MA_SND_PCM_FORMAT_S24_3LE: return ma_format_s24;
- case MA_SND_PCM_FORMAT_S32_LE: return ma_format_s32;
- case MA_SND_PCM_FORMAT_FLOAT_LE: return ma_format_f32;
- default: break;
- }
- } else {
- switch (formatALSA) {
- case MA_SND_PCM_FORMAT_S16_BE: return ma_format_s16;
- case MA_SND_PCM_FORMAT_S24_3BE: return ma_format_s24;
- case MA_SND_PCM_FORMAT_S32_BE: return ma_format_s32;
- case MA_SND_PCM_FORMAT_FLOAT_BE: return ma_format_f32;
- default: break;
- }
- }
+ MA_ZERO_OBJECT(&descriptorCapture);
+ descriptorCapture.pDeviceID = pConfig->capture.pDeviceID;
+ descriptorCapture.shareMode = pConfig->capture.shareMode;
+ descriptorCapture.format = pConfig->capture.format;
+ descriptorCapture.channels = pConfig->capture.channels;
+ descriptorCapture.sampleRate = pConfig->sampleRate;
+ ma_channel_map_copy_or_default(descriptorCapture.channelMap, ma_countof(descriptorCapture.channelMap), pConfig->capture.pChannelMap, pConfig->capture.channels);
+ descriptorCapture.periodSizeInFrames = pConfig->periodSizeInFrames;
+ descriptorCapture.periodSizeInMilliseconds = pConfig->periodSizeInMilliseconds;
+ descriptorCapture.periodCount = pConfig->periods;
- /* Endian agnostic. */
- switch (formatALSA) {
- case MA_SND_PCM_FORMAT_U8: return ma_format_u8;
- default: return ma_format_unknown;
+ if (descriptorCapture.periodCount == 0) {
+ descriptorCapture.periodCount = MA_DEFAULT_PERIODS;
}
-}
-static ma_channel ma_convert_alsa_channel_position_to_ma_channel(unsigned int alsaChannelPos)
-{
- switch (alsaChannelPos)
- {
- case MA_SND_CHMAP_MONO: return MA_CHANNEL_MONO;
- case MA_SND_CHMAP_FL: return MA_CHANNEL_FRONT_LEFT;
- case MA_SND_CHMAP_FR: return MA_CHANNEL_FRONT_RIGHT;
- case MA_SND_CHMAP_RL: return MA_CHANNEL_BACK_LEFT;
- case MA_SND_CHMAP_RR: return MA_CHANNEL_BACK_RIGHT;
- case MA_SND_CHMAP_FC: return MA_CHANNEL_FRONT_CENTER;
- case MA_SND_CHMAP_LFE: return MA_CHANNEL_LFE;
- case MA_SND_CHMAP_SL: return MA_CHANNEL_SIDE_LEFT;
- case MA_SND_CHMAP_SR: return MA_CHANNEL_SIDE_RIGHT;
- case MA_SND_CHMAP_RC: return MA_CHANNEL_BACK_CENTER;
- case MA_SND_CHMAP_FLC: return MA_CHANNEL_FRONT_LEFT_CENTER;
- case MA_SND_CHMAP_FRC: return MA_CHANNEL_FRONT_RIGHT_CENTER;
- case MA_SND_CHMAP_RLC: return 0;
- case MA_SND_CHMAP_RRC: return 0;
- case MA_SND_CHMAP_FLW: return 0;
- case MA_SND_CHMAP_FRW: return 0;
- case MA_SND_CHMAP_FLH: return 0;
- case MA_SND_CHMAP_FCH: return 0;
- case MA_SND_CHMAP_FRH: return 0;
- case MA_SND_CHMAP_TC: return MA_CHANNEL_TOP_CENTER;
- case MA_SND_CHMAP_TFL: return MA_CHANNEL_TOP_FRONT_LEFT;
- case MA_SND_CHMAP_TFR: return MA_CHANNEL_TOP_FRONT_RIGHT;
- case MA_SND_CHMAP_TFC: return MA_CHANNEL_TOP_FRONT_CENTER;
- case MA_SND_CHMAP_TRL: return MA_CHANNEL_TOP_BACK_LEFT;
- case MA_SND_CHMAP_TRR: return MA_CHANNEL_TOP_BACK_RIGHT;
- case MA_SND_CHMAP_TRC: return MA_CHANNEL_TOP_BACK_CENTER;
- default: break;
+
+ result = pContext->callbacks.onDeviceInit(pDevice, pConfig, &descriptorPlayback, &descriptorCapture);
+ if (result != MA_SUCCESS) {
+ ma_event_uninit(&pDevice->startEvent);
+ ma_event_uninit(&pDevice->wakeupEvent);
+ ma_mutex_uninit(&pDevice->startStopLock);
+ return result;
}
- return 0;
-}
+#if 0
+ /*
+ On output the descriptors will contain the *actual* data format of the device. We need this to know how to convert the data between
+ the requested format and the internal format.
+ */
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex || pConfig->deviceType == ma_device_type_loopback) {
+ if (!ma_device_descriptor_is_valid(&descriptorCapture)) {
+ ma_device_uninit(pDevice);
+ return MA_INVALID_ARGS;
+ }
-static ma_bool32 ma_is_common_device_name__alsa(const char* name)
-{
- size_t iName;
- for (iName = 0; iName < ma_countof(g_maCommonDeviceNamesALSA); ++iName) {
- if (ma_strcmp(name, g_maCommonDeviceNamesALSA[iName]) == 0) {
- return MA_TRUE;
+ pDevice->capture.internalFormat = descriptorCapture.format;
+ pDevice->capture.internalChannels = descriptorCapture.channels;
+ pDevice->capture.internalSampleRate = descriptorCapture.sampleRate;
+ ma_channel_map_copy(pDevice->capture.internalChannelMap, descriptorCapture.channelMap, descriptorCapture.channels);
+ pDevice->capture.internalPeriodSizeInFrames = descriptorCapture.periodSizeInFrames;
+ pDevice->capture.internalPeriods = descriptorCapture.periodCount;
+
+ if (pDevice->capture.internalPeriodSizeInFrames == 0) {
+ pDevice->capture.internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(descriptorCapture.periodSizeInMilliseconds, descriptorCapture.sampleRate);
}
}
- return MA_FALSE;
-}
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ if (!ma_device_descriptor_is_valid(&descriptorPlayback)) {
+ ma_device_uninit(pDevice);
+ return MA_INVALID_ARGS;
+ }
+ pDevice->playback.internalFormat = descriptorPlayback.format;
+ pDevice->playback.internalChannels = descriptorPlayback.channels;
+ pDevice->playback.internalSampleRate = descriptorPlayback.sampleRate;
+ ma_channel_map_copy(pDevice->playback.internalChannelMap, descriptorPlayback.channelMap, descriptorPlayback.channels);
+ pDevice->playback.internalPeriodSizeInFrames = descriptorPlayback.periodSizeInFrames;
+ pDevice->playback.internalPeriods = descriptorPlayback.periodCount;
-static ma_bool32 ma_is_playback_device_blacklisted__alsa(const char* name)
-{
- size_t iName;
- for (iName = 0; iName < ma_countof(g_maBlacklistedPlaybackDeviceNamesALSA); ++iName) {
- if (ma_strcmp(name, g_maBlacklistedPlaybackDeviceNamesALSA[iName]) == 0) {
- return MA_TRUE;
+ if (pDevice->playback.internalPeriodSizeInFrames == 0) {
+ pDevice->playback.internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(descriptorPlayback.periodSizeInMilliseconds, descriptorPlayback.sampleRate);
}
}
- return MA_FALSE;
-}
-static ma_bool32 ma_is_capture_device_blacklisted__alsa(const char* name)
-{
- size_t iName;
- for (iName = 0; iName < ma_countof(g_maBlacklistedCaptureDeviceNamesALSA); ++iName) {
- if (ma_strcmp(name, g_maBlacklistedCaptureDeviceNamesALSA[iName]) == 0) {
- return MA_TRUE;
+ /*
+ The name of the device can be retrieved from device info. This may be temporary and replaced with a `ma_device_get_info(pDevice, deviceType)` instead.
+ For loopback devices, we need to retrieve the name of the playback device.
+ */
+ {
+ ma_device_info deviceInfo;
+
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex || pConfig->deviceType == ma_device_type_loopback) {
+ result = ma_device_get_info(pDevice, (pConfig->deviceType == ma_device_type_loopback) ? ma_device_type_playback : ma_device_type_capture, &deviceInfo);
+ if (result == MA_SUCCESS) {
+ ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), deviceInfo.name, (size_t)-1);
+ } else {
+ /* We failed to retrieve the device info. Fall back to a default name. */
+ if (descriptorCapture.pDeviceID == NULL) {
+ ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
+ } else {
+ ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), "Capture Device", (size_t)-1);
+ }
+ }
+ }
+
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ result = ma_device_get_info(pDevice, ma_device_type_playback, &deviceInfo);
+ if (result == MA_SUCCESS) {
+ ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), deviceInfo.name, (size_t)-1);
+ } else {
+ /* We failed to retrieve the device info. Fall back to a default name. */
+ if (descriptorPlayback.pDeviceID == NULL) {
+ ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
+ } else {
+ ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), "Playback Device", (size_t)-1);
+ }
+ }
}
}
- return MA_FALSE;
-}
-static ma_bool32 ma_is_device_blacklisted__alsa(ma_device_type deviceType, const char* name)
-{
- if (deviceType == ma_device_type_playback) {
- return ma_is_playback_device_blacklisted__alsa(name);
- } else {
- return ma_is_capture_device_blacklisted__alsa(name);
+ ma_device__post_init_setup(pDevice, pConfig->deviceType);
+#endif
+
+ result = ma_device_post_init(pDevice, pConfig->deviceType, &descriptorPlayback, &descriptorCapture);
+ if (result != MA_SUCCESS) {
+ ma_device_uninit(pDevice);
+ return result;
}
-}
-static const char* ma_find_char(const char* str, char c, int* index)
-{
- int i = 0;
- for (;;) {
- if (str[i] == '\0') {
- if (index) *index = -1;
- return NULL;
- }
- if (str[i] == c) {
- if (index) *index = i;
- return str + i;
+ /*
+ If we're using fixed sized callbacks we'll need to make use of an intermediary buffer. Needs to
+ be done after post_init_setup() because we'll need access to the sample rate.
+ */
+ if (pConfig->noFixedSizedCallback == MA_FALSE) {
+ /* We're using a fixed sized data callback so we'll need an intermediary buffer. */
+ ma_uint32 intermediaryBufferCap = pConfig->periodSizeInFrames;
+ if (intermediaryBufferCap == 0) {
+ intermediaryBufferCap = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->periodSizeInMilliseconds, pDevice->sampleRate);
}
- i += 1;
- }
+ if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex || pConfig->deviceType == ma_device_type_loopback) {
+ ma_uint32 intermediaryBufferSizeInBytes;
- /* Should never get here, but treat it as though the character was not found to make me feel better inside. */
- if (index) *index = -1;
- return NULL;
-}
+ pDevice->capture.intermediaryBufferLen = 0;
+ pDevice->capture.intermediaryBufferCap = intermediaryBufferCap;
+ if (pDevice->capture.intermediaryBufferCap == 0) {
+ pDevice->capture.intermediaryBufferCap = pDevice->capture.internalPeriodSizeInFrames;
+ }
-static ma_bool32 ma_is_device_name_in_hw_format__alsa(const char* hwid)
-{
- /* This function is just checking whether or not hwid is in "hw:%d,%d" format. */
+ intermediaryBufferSizeInBytes = pDevice->capture.intermediaryBufferCap * ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels);
- int commaPos;
- const char* dev;
- int i;
+ pDevice->capture.pIntermediaryBuffer = ma_malloc((size_t)intermediaryBufferSizeInBytes, &pContext->allocationCallbacks);
+ if (pDevice->capture.pIntermediaryBuffer == NULL) {
+ ma_device_uninit(pDevice);
+ return MA_OUT_OF_MEMORY;
+ }
- if (hwid == NULL) {
- return MA_FALSE;
- }
+ /* Silence the buffer for safety. */
+ ma_silence_pcm_frames(pDevice->capture.pIntermediaryBuffer, pDevice->capture.intermediaryBufferCap, pDevice->capture.format, pDevice->capture.channels);
+ pDevice->capture.intermediaryBufferLen = pDevice->capture.intermediaryBufferCap;
+ }
- if (hwid[0] != 'h' || hwid[1] != 'w' || hwid[2] != ':') {
- return MA_FALSE;
- }
+ if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+ ma_uint64 intermediaryBufferSizeInBytes;
+
+ pDevice->playback.intermediaryBufferLen = 0;
+ if (pConfig->deviceType == ma_device_type_duplex) {
+ pDevice->playback.intermediaryBufferCap = pDevice->capture.intermediaryBufferCap; /* In duplex mode, make sure the intermediary buffer is always the same size as the capture side. */
+ } else {
+ pDevice->playback.intermediaryBufferCap = intermediaryBufferCap;
+ if (pDevice->playback.intermediaryBufferCap == 0) {
+ pDevice->playback.intermediaryBufferCap = pDevice->playback.internalPeriodSizeInFrames;
+ }
+ }
- hwid += 3;
+ intermediaryBufferSizeInBytes = pDevice->playback.intermediaryBufferCap * ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
+
+ pDevice->playback.pIntermediaryBuffer = ma_malloc((size_t)intermediaryBufferSizeInBytes, &pContext->allocationCallbacks);
+ if (pDevice->playback.pIntermediaryBuffer == NULL) {
+ ma_device_uninit(pDevice);
+ return MA_OUT_OF_MEMORY;
+ }
- dev = ma_find_char(hwid, ',', &commaPos);
- if (dev == NULL) {
- return MA_FALSE;
+ /* Silence the buffer for safety. */
+ ma_silence_pcm_frames(pDevice->playback.pIntermediaryBuffer, pDevice->playback.intermediaryBufferCap, pDevice->playback.format, pDevice->playback.channels);
+ pDevice->playback.intermediaryBufferLen = 0;
+ }
} else {
- dev += 1; /* Skip past the ",". */
+ /* Not using a fixed sized data callback so no need for an intermediary buffer. */
}
- /* Check if the part between the ":" and the "," contains only numbers. If not, return false. */
- for (i = 0; i < commaPos; ++i) {
- if (hwid[i] < '0' || hwid[i] > '9') {
- return MA_FALSE;
+
+ /* Some backends don't require the worker thread. */
+ if (!ma_context_is_backend_asynchronous(pContext)) {
+ /* The worker thread. */
+ result = ma_thread_create(&pDevice->thread, pContext->threadPriority, pContext->threadStackSize, ma_worker_thread, pDevice, &pContext->allocationCallbacks);
+ if (result != MA_SUCCESS) {
+ ma_device_uninit(pDevice);
+ return result;
}
- }
- /* Check if everything after the "," is numeric. If not, return false. */
- i = 0;
- while (dev[i] != '\0') {
- if (dev[i] < '0' || dev[i] > '9') {
- return MA_FALSE;
+ /* Wait for the worker thread to put the device into it's stopped state for real. */
+ ma_event_wait(&pDevice->stopEvent);
+ MA_ASSERT(ma_device_get_state(pDevice) == ma_device_state_stopped);
+ } else {
+ /*
+ If the backend is asynchronous and the device is duplex, we'll need an intermediary ring buffer. Note that this needs to be done
+ after ma_device__post_init_setup().
+ */
+ if (ma_context_is_backend_asynchronous(pContext)) {
+ if (pConfig->deviceType == ma_device_type_duplex) {
+ result = ma_duplex_rb_init(pDevice->capture.format, pDevice->capture.channels, pDevice->sampleRate, pDevice->capture.internalSampleRate, pDevice->capture.internalPeriodSizeInFrames, &pDevice->pContext->allocationCallbacks, &pDevice->duplexRB);
+ if (result != MA_SUCCESS) {
+ ma_device_uninit(pDevice);
+ return result;
+ }
+ }
}
- i += 1;
+
+ ma_device__set_state(pDevice, ma_device_state_stopped);
}
- return MA_TRUE;
+ /* Log device information. */
+ {
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[%s]\n", ma_get_backend_name(pDevice->pContext->backend));
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+ char name[MA_MAX_DEVICE_NAME_LENGTH + 1];
+ ma_device_get_name(pDevice, ma_device_type_capture, name, sizeof(name), NULL);
+
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " %s (%s)\n", name, "Capture");
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Format: %s -> %s\n", ma_get_format_name(pDevice->capture.internalFormat), ma_get_format_name(pDevice->capture.format));
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Channels: %d -> %d\n", pDevice->capture.internalChannels, pDevice->capture.channels);
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Sample Rate: %d -> %d\n", pDevice->capture.internalSampleRate, pDevice->sampleRate);
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Buffer Size: %d*%d (%d)\n", pDevice->capture.internalPeriodSizeInFrames, pDevice->capture.internalPeriods, (pDevice->capture.internalPeriodSizeInFrames * pDevice->capture.internalPeriods));
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Conversion:\n");
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Pre Format Conversion: %s\n", pDevice->capture.converter.hasPreFormatConversion ? "YES" : "NO");
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Post Format Conversion: %s\n", pDevice->capture.converter.hasPostFormatConversion ? "YES" : "NO");
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Channel Routing: %s\n", pDevice->capture.converter.hasChannelConverter ? "YES" : "NO");
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Resampling: %s\n", pDevice->capture.converter.hasResampler ? "YES" : "NO");
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Passthrough: %s\n", pDevice->capture.converter.isPassthrough ? "YES" : "NO");
+ }
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ char name[MA_MAX_DEVICE_NAME_LENGTH + 1];
+ ma_device_get_name(pDevice, ma_device_type_playback, name, sizeof(name), NULL);
+
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " %s (%s)\n", name, "Playback");
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Format: %s -> %s\n", ma_get_format_name(pDevice->playback.format), ma_get_format_name(pDevice->playback.internalFormat));
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Channels: %d -> %d\n", pDevice->playback.channels, pDevice->playback.internalChannels);
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Sample Rate: %d -> %d\n", pDevice->sampleRate, pDevice->playback.internalSampleRate);
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Buffer Size: %d*%d (%d)\n", pDevice->playback.internalPeriodSizeInFrames, pDevice->playback.internalPeriods, (pDevice->playback.internalPeriodSizeInFrames * pDevice->playback.internalPeriods));
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Conversion:\n");
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Pre Format Conversion: %s\n", pDevice->playback.converter.hasPreFormatConversion ? "YES" : "NO");
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Post Format Conversion: %s\n", pDevice->playback.converter.hasPostFormatConversion ? "YES" : "NO");
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Channel Routing: %s\n", pDevice->playback.converter.hasChannelConverter ? "YES" : "NO");
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Resampling: %s\n", pDevice->playback.converter.hasResampler ? "YES" : "NO");
+ ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Passthrough: %s\n", pDevice->playback.converter.isPassthrough ? "YES" : "NO");
+ }
+ }
+
+ MA_ASSERT(ma_device_get_state(pDevice) == ma_device_state_stopped);
+ return MA_SUCCESS;
}
-static int ma_convert_device_name_to_hw_format__alsa(ma_context* pContext, char* dst, size_t dstSize, const char* src) /* Returns 0 on success, non-0 on error. */
+MA_API ma_result ma_device_init_ex(const ma_backend backends[], ma_uint32 backendCount, const ma_context_config* pContextConfig, const ma_device_config* pConfig, ma_device* pDevice)
{
- /* src should look something like this: "hw:CARD=I82801AAICH,DEV=0" */
-
- int colonPos;
- int commaPos;
- char card[256];
- const char* dev;
- int cardIndex;
+ ma_result result;
+ ma_context* pContext;
+ ma_backend defaultBackends[ma_backend_null+1];
+ ma_uint32 iBackend;
+ ma_backend* pBackendsToIterate;
+ ma_uint32 backendsToIterateCount;
+ ma_allocation_callbacks allocationCallbacks;
- if (dst == NULL) {
- return -1;
- }
- if (dstSize < 7) {
- return -1; /* Absolute minimum size of the output buffer is 7 bytes. */
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- *dst = '\0'; /* Safety. */
- if (src == NULL) {
- return -1;
+ if (pContextConfig != NULL) {
+ result = ma_allocation_callbacks_init_copy(&allocationCallbacks, &pContextConfig->allocationCallbacks);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ } else {
+ allocationCallbacks = ma_allocation_callbacks_init_default();
}
- /* If the input name is already in "hw:%d,%d" format, just return that verbatim. */
- if (ma_is_device_name_in_hw_format__alsa(src)) {
- return ma_strcpy_s(dst, dstSize, src);
- }
- src = ma_find_char(src, ':', &colonPos);
- if (src == NULL) {
- return -1; /* Couldn't find a colon */
+ pContext = (ma_context*)ma_malloc(sizeof(*pContext), &allocationCallbacks);
+ if (pContext == NULL) {
+ return MA_OUT_OF_MEMORY;
}
- dev = ma_find_char(src, ',', &commaPos);
- if (dev == NULL) {
- dev = "0";
- ma_strncpy_s(card, sizeof(card), src+6, (size_t)-1); /* +6 = ":CARD=" */
- } else {
- dev = dev + 5; /* +5 = ",DEV=" */
- ma_strncpy_s(card, sizeof(card), src+6, commaPos-6); /* +6 = ":CARD=" */
+ for (iBackend = 0; iBackend <= ma_backend_null; ++iBackend) {
+ defaultBackends[iBackend] = (ma_backend)iBackend;
}
- cardIndex = ((ma_snd_card_get_index_proc)pContext->alsa.snd_card_get_index)(card);
- if (cardIndex < 0) {
- return -2; /* Failed to retrieve the card index. */
+ pBackendsToIterate = (ma_backend*)backends;
+ backendsToIterateCount = backendCount;
+ if (pBackendsToIterate == NULL) {
+ pBackendsToIterate = (ma_backend*)defaultBackends;
+ backendsToIterateCount = ma_countof(defaultBackends);
}
+ result = MA_NO_BACKEND;
- /* Construction. */
- dst[0] = 'h'; dst[1] = 'w'; dst[2] = ':';
- if (ma_itoa_s(cardIndex, dst+3, dstSize-3, 10) != 0) {
- return -3;
- }
- if (ma_strcat_s(dst, dstSize, ",") != 0) {
- return -3;
+ for (iBackend = 0; iBackend < backendsToIterateCount; ++iBackend) {
+ result = ma_context_init(&pBackendsToIterate[iBackend], 1, pContextConfig, pContext);
+ if (result == MA_SUCCESS) {
+ result = ma_device_init(pContext, pConfig, pDevice);
+ if (result == MA_SUCCESS) {
+ break; /* Success. */
+ } else {
+ ma_context_uninit(pContext); /* Failure. */
+ }
+ }
}
- if (ma_strcat_s(dst, dstSize, dev) != 0) {
- return -3;
+
+ if (result != MA_SUCCESS) {
+ ma_free(pContext, &allocationCallbacks);
+ return result;
}
- return 0;
+ pDevice->isOwnerOfContext = MA_TRUE;
+ return result;
}
-static ma_bool32 ma_does_id_exist_in_list__alsa(ma_device_id* pUniqueIDs, ma_uint32 count, const char* pHWID)
+MA_API void ma_device_uninit(ma_device* pDevice)
{
- ma_uint32 i;
-
- MA_ASSERT(pHWID != NULL);
+ if (!ma_device__is_initialized(pDevice)) {
+ return;
+ }
- for (i = 0; i < count; ++i) {
- if (ma_strcmp(pUniqueIDs[i].alsa, pHWID) == 0) {
- return MA_TRUE;
- }
+ /* Make sure the device is stopped first. The backends will probably handle this naturally, but I like to do it explicitly for my own sanity. */
+ if (ma_device_is_started(pDevice)) {
+ ma_device_stop(pDevice);
}
- return MA_FALSE;
-}
+ /* Putting the device into an uninitialized state will make the worker thread return. */
+ ma_device__set_state(pDevice, ma_device_state_uninitialized);
+ /* Wake up the worker thread and wait for it to properly terminate. */
+ if (!ma_context_is_backend_asynchronous(pDevice->pContext)) {
+ ma_event_signal(&pDevice->wakeupEvent);
+ ma_thread_wait(&pDevice->thread);
+ }
-static ma_result ma_context_open_pcm__alsa(ma_context* pContext, ma_share_mode shareMode, ma_device_type deviceType, const ma_device_id* pDeviceID, int openMode, ma_snd_pcm_t** ppPCM)
-{
- ma_snd_pcm_t* pPCM;
- ma_snd_pcm_stream_t stream;
+ if (pDevice->pContext->callbacks.onDeviceUninit != NULL) {
+ pDevice->pContext->callbacks.onDeviceUninit(pDevice);
+ }
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(ppPCM != NULL);
- *ppPCM = NULL;
- pPCM = NULL;
+ ma_event_uninit(&pDevice->stopEvent);
+ ma_event_uninit(&pDevice->startEvent);
+ ma_event_uninit(&pDevice->wakeupEvent);
+ ma_mutex_uninit(&pDevice->startStopLock);
- stream = (deviceType == ma_device_type_playback) ? MA_SND_PCM_STREAM_PLAYBACK : MA_SND_PCM_STREAM_CAPTURE;
+ if (ma_context_is_backend_asynchronous(pDevice->pContext)) {
+ if (pDevice->type == ma_device_type_duplex) {
+ ma_duplex_rb_uninit(&pDevice->duplexRB);
+ }
+ }
- if (pDeviceID == NULL) {
- ma_bool32 isDeviceOpen;
- size_t i;
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) {
+ ma_data_converter_uninit(&pDevice->capture.converter, &pDevice->pContext->allocationCallbacks);
+ }
+ if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+ ma_data_converter_uninit(&pDevice->playback.converter, &pDevice->pContext->allocationCallbacks);
+ }
- /*
- We're opening the default device. I don't know if trying anything other than "default" is necessary, but it makes
- me feel better to try as hard as we can get to get _something_ working.
- */
- const char* defaultDeviceNames[] = {
- "default",
- NULL,
- NULL,
- NULL,
- NULL,
- NULL,
- NULL
- };
+ if (pDevice->playback.pInputCache != NULL) {
+ ma_free(pDevice->playback.pInputCache, &pDevice->pContext->allocationCallbacks);
+ }
- if (shareMode == ma_share_mode_exclusive) {
- defaultDeviceNames[1] = "hw";
- defaultDeviceNames[2] = "hw:0";
- defaultDeviceNames[3] = "hw:0,0";
- } else {
- if (deviceType == ma_device_type_playback) {
- defaultDeviceNames[1] = "dmix";
- defaultDeviceNames[2] = "dmix:0";
- defaultDeviceNames[3] = "dmix:0,0";
- } else {
- defaultDeviceNames[1] = "dsnoop";
- defaultDeviceNames[2] = "dsnoop:0";
- defaultDeviceNames[3] = "dsnoop:0,0";
- }
- defaultDeviceNames[4] = "hw";
- defaultDeviceNames[5] = "hw:0";
- defaultDeviceNames[6] = "hw:0,0";
- }
+ if (pDevice->capture.pIntermediaryBuffer != NULL) {
+ ma_free(pDevice->capture.pIntermediaryBuffer, &pDevice->pContext->allocationCallbacks);
+ }
+ if (pDevice->playback.pIntermediaryBuffer != NULL) {
+ ma_free(pDevice->playback.pIntermediaryBuffer, &pDevice->pContext->allocationCallbacks);
+ }
- isDeviceOpen = MA_FALSE;
- for (i = 0; i < ma_countof(defaultDeviceNames); ++i) {
- if (defaultDeviceNames[i] != NULL && defaultDeviceNames[i][0] != '\0') {
- if (((ma_snd_pcm_open_proc)pContext->alsa.snd_pcm_open)(&pPCM, defaultDeviceNames[i], stream, openMode) == 0) {
- isDeviceOpen = MA_TRUE;
- break;
- }
- }
- }
+ if (pDevice->isOwnerOfContext) {
+ ma_allocation_callbacks allocationCallbacks = pDevice->pContext->allocationCallbacks;
- if (!isDeviceOpen) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_open() failed when trying to open an appropriate default device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE);
- }
- } else {
- /*
- We're trying to open a specific device. There's a few things to consider here:
+ ma_context_uninit(pDevice->pContext);
+ ma_free(pDevice->pContext, &allocationCallbacks);
+ }
- miniaudio recongnizes a special format of device id that excludes the "hw", "dmix", etc. prefix. It looks like this: ":0,0", ":0,1", etc. When
- an ID of this format is specified, it indicates to miniaudio that it can try different combinations of plugins ("hw", "dmix", etc.) until it
- finds an appropriate one that works. This comes in very handy when trying to open a device in shared mode ("dmix"), vs exclusive mode ("hw").
- */
+ MA_ZERO_OBJECT(pDevice);
+}
- /* May end up needing to make small adjustments to the ID, so make a copy. */
- ma_device_id deviceID = *pDeviceID;
- int resultALSA = -ENODEV;
+MA_API ma_context* ma_device_get_context(ma_device* pDevice)
+{
+ if (pDevice == NULL) {
+ return NULL;
+ }
- if (deviceID.alsa[0] != ':') {
- /* The ID is not in ":0,0" format. Use the ID exactly as-is. */
- resultALSA = ((ma_snd_pcm_open_proc)pContext->alsa.snd_pcm_open)(&pPCM, deviceID.alsa, stream, openMode);
- } else {
- char hwid[256];
+ return pDevice->pContext;
+}
- /* The ID is in ":0,0" format. Try different plugins depending on the shared mode. */
- if (deviceID.alsa[1] == '\0') {
- deviceID.alsa[0] = '\0'; /* An ID of ":" should be converted to "". */
- }
+MA_API ma_log* ma_device_get_log(ma_device* pDevice)
+{
+ return ma_context_get_log(ma_device_get_context(pDevice));
+}
- if (shareMode == ma_share_mode_shared) {
- if (deviceType == ma_device_type_playback) {
- ma_strcpy_s(hwid, sizeof(hwid), "dmix");
- } else {
- ma_strcpy_s(hwid, sizeof(hwid), "dsnoop");
- }
+MA_API ma_result ma_device_get_info(ma_device* pDevice, ma_device_type type, ma_device_info* pDeviceInfo)
+{
+ if (pDeviceInfo == NULL) {
+ return MA_INVALID_ARGS;
+ }
- if (ma_strcat_s(hwid, sizeof(hwid), deviceID.alsa) == 0) {
- resultALSA = ((ma_snd_pcm_open_proc)pContext->alsa.snd_pcm_open)(&pPCM, hwid, stream, openMode);
- }
- }
+ MA_ZERO_OBJECT(pDeviceInfo);
- /* If at this point we still don't have an open device it means we're either preferencing exclusive mode or opening with "dmix"/"dsnoop" failed. */
- if (resultALSA != 0) {
- ma_strcpy_s(hwid, sizeof(hwid), "hw");
- if (ma_strcat_s(hwid, sizeof(hwid), deviceID.alsa) == 0) {
- resultALSA = ((ma_snd_pcm_open_proc)pContext->alsa.snd_pcm_open)(&pPCM, hwid, stream, openMode);
- }
- }
- }
+ if (pDevice == NULL) {
+ return MA_INVALID_ARGS;
+ }
- if (resultALSA < 0) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_open() failed.", ma_result_from_errno(-resultALSA));
- }
+ /* If the onDeviceGetInfo() callback is set, use that. Otherwise we'll fall back to ma_context_get_device_info(). */
+ if (pDevice->pContext->callbacks.onDeviceGetInfo != NULL) {
+ return pDevice->pContext->callbacks.onDeviceGetInfo(pDevice, type, pDeviceInfo);
}
- *ppPCM = pPCM;
- return MA_SUCCESS;
+ /* Getting here means onDeviceGetInfo is not implemented so we need to fall back to an alternative. */
+ if (type == ma_device_type_playback) {
+ return ma_context_get_device_info(pDevice->pContext, type, pDevice->playback.pID, pDeviceInfo);
+ } else {
+ return ma_context_get_device_info(pDevice->pContext, type, pDevice->capture.pID, pDeviceInfo);
+ }
}
-
-static ma_result ma_context_enumerate_devices__alsa(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+MA_API ma_result ma_device_get_name(ma_device* pDevice, ma_device_type type, char* pName, size_t nameCap, size_t* pLengthNotIncludingNullTerminator)
{
- int resultALSA;
- ma_bool32 cbResult = MA_TRUE;
- char** ppDeviceHints;
- ma_device_id* pUniqueIDs = NULL;
- ma_uint32 uniqueIDCount = 0;
- char** ppNextDeviceHint;
+ ma_result result;
+ ma_device_info deviceInfo;
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(callback != NULL);
+ if (pLengthNotIncludingNullTerminator != NULL) {
+ *pLengthNotIncludingNullTerminator = 0;
+ }
- ma_mutex_lock(&pContext->alsa.internalDeviceEnumLock);
+ if (pName != NULL && nameCap > 0) {
+ pName[0] = '\0';
+ }
- resultALSA = ((ma_snd_device_name_hint_proc)pContext->alsa.snd_device_name_hint)(-1, "pcm", (void***)&ppDeviceHints);
- if (resultALSA < 0) {
- ma_mutex_unlock(&pContext->alsa.internalDeviceEnumLock);
- return ma_result_from_errno(-resultALSA);
+ result = ma_device_get_info(pDevice, type, &deviceInfo);
+ if (result != MA_SUCCESS) {
+ return result;
}
- ppNextDeviceHint = ppDeviceHints;
- while (*ppNextDeviceHint != NULL) {
- char* NAME = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "NAME");
- char* DESC = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "DESC");
- char* IOID = ((ma_snd_device_name_get_hint_proc)pContext->alsa.snd_device_name_get_hint)(*ppNextDeviceHint, "IOID");
- ma_device_type deviceType = ma_device_type_playback;
- ma_bool32 stopEnumeration = MA_FALSE;
- char hwid[sizeof(pUniqueIDs->alsa)];
- ma_device_info deviceInfo;
+ if (pName != NULL) {
+ ma_strncpy_s(pName, nameCap, deviceInfo.name, (size_t)-1);
- if ((IOID == NULL || ma_strcmp(IOID, "Output") == 0)) {
- deviceType = ma_device_type_playback;
+ /*
+ For safety, make sure the length is based on the truncated output string rather than the
+ source. Otherwise the caller might assume the output buffer contains more content than it
+ actually does.
+ */
+ if (pLengthNotIncludingNullTerminator != NULL) {
+ *pLengthNotIncludingNullTerminator = strlen(pName);
}
- if ((IOID != NULL && ma_strcmp(IOID, "Input" ) == 0)) {
- deviceType = ma_device_type_capture;
+ } else {
+ /* Name not specified. Just report the length of the source string. */
+ if (pLengthNotIncludingNullTerminator != NULL) {
+ *pLengthNotIncludingNullTerminator = strlen(deviceInfo.name);
}
+ }
- if (NAME != NULL) {
- if (pContext->alsa.useVerboseDeviceEnumeration) {
- /* Verbose mode. Use the name exactly as-is. */
- ma_strncpy_s(hwid, sizeof(hwid), NAME, (size_t)-1);
- } else {
- /* Simplified mode. Use ":%d,%d" format. */
- if (ma_convert_device_name_to_hw_format__alsa(pContext, hwid, sizeof(hwid), NAME) == 0) {
- /*
- At this point, hwid looks like "hw:0,0". In simplified enumeration mode, we actually want to strip off the
- plugin name so it looks like ":0,0". The reason for this is that this special format is detected at device
- initialization time and is used as an indicator to try and use the most appropriate plugin depending on the
- device type and sharing mode.
- */
- char* dst = hwid;
- char* src = hwid+2;
- while ((*dst++ = *src++));
- } else {
- /* Conversion to "hw:%d,%d" failed. Just use the name as-is. */
- ma_strncpy_s(hwid, sizeof(hwid), NAME, (size_t)-1);
- }
-
- if (ma_does_id_exist_in_list__alsa(pUniqueIDs, uniqueIDCount, hwid)) {
- goto next_device; /* The device has already been enumerated. Move on to the next one. */
- } else {
- /* The device has not yet been enumerated. Make sure it's added to our list so that it's not enumerated again. */
- size_t oldCapacity = sizeof(*pUniqueIDs) * uniqueIDCount;
- size_t newCapacity = sizeof(*pUniqueIDs) * (uniqueIDCount + 1);
- ma_device_id* pNewUniqueIDs = (ma_device_id*)ma__realloc_from_callbacks(pUniqueIDs, newCapacity, oldCapacity, &pContext->allocationCallbacks);
- if (pNewUniqueIDs == NULL) {
- goto next_device; /* Failed to allocate memory. */
- }
+ return MA_SUCCESS;
+}
- pUniqueIDs = pNewUniqueIDs;
- MA_COPY_MEMORY(pUniqueIDs[uniqueIDCount].alsa, hwid, sizeof(hwid));
- uniqueIDCount += 1;
- }
- }
- } else {
- MA_ZERO_MEMORY(hwid, sizeof(hwid));
- }
+MA_API ma_result ma_device_start(ma_device* pDevice)
+{
+ ma_result result;
- MA_ZERO_OBJECT(&deviceInfo);
- ma_strncpy_s(deviceInfo.id.alsa, sizeof(deviceInfo.id.alsa), hwid, (size_t)-1);
+ if (pDevice == NULL) {
+ return MA_INVALID_ARGS;
+ }
- /*
- There's no good way to determine whether or not a device is the default on Linux. We're just going to do something simple and
- just use the name of "default" as the indicator.
- */
- if (ma_strcmp(deviceInfo.id.alsa, "default") == 0) {
- deviceInfo.isDefault = MA_TRUE;
- }
+ if (ma_device_get_state(pDevice) == ma_device_state_uninitialized) {
+ return MA_INVALID_OPERATION; /* Not initialized. */
+ }
+ if (ma_device_get_state(pDevice) == ma_device_state_started) {
+ return MA_SUCCESS; /* Already started. */
+ }
- /*
- DESC is the friendly name. We treat this slightly differently depending on whether or not we are using verbose
- device enumeration. In verbose mode we want to take the entire description so that the end-user can distinguish
- between the subdevices of each card/dev pair. In simplified mode, however, we only want the first part of the
- description.
+ ma_mutex_lock(&pDevice->startStopLock);
+ {
+ /* Starting and stopping are wrapped in a mutex which means we can assert that the device is in a stopped or paused state. */
+ MA_ASSERT(ma_device_get_state(pDevice) == ma_device_state_stopped);
- The value in DESC seems to be split into two lines, with the first line being the name of the device and the
- second line being a description of the device. I don't like having the description be across two lines because
- it makes formatting ugly and annoying. I'm therefore deciding to put it all on a single line with the second line
- being put into parentheses. In simplified mode I'm just stripping the second line entirely.
- */
- if (DESC != NULL) {
- int lfPos;
- const char* line2 = ma_find_char(DESC, '\n', &lfPos);
- if (line2 != NULL) {
- line2 += 1; /* Skip past the new-line character. */
+ ma_device__set_state(pDevice, ma_device_state_starting);
- if (pContext->alsa.useVerboseDeviceEnumeration) {
- /* Verbose mode. Put the second line in brackets. */
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), DESC, lfPos);
- ma_strcat_s (deviceInfo.name, sizeof(deviceInfo.name), " (");
- ma_strcat_s (deviceInfo.name, sizeof(deviceInfo.name), line2);
- ma_strcat_s (deviceInfo.name, sizeof(deviceInfo.name), ")");
- } else {
- /* Simplified mode. Strip the second line entirely. */
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), DESC, lfPos);
- }
+ /* Asynchronous backends need to be handled differently. */
+ if (ma_context_is_backend_asynchronous(pDevice->pContext)) {
+ if (pDevice->pContext->callbacks.onDeviceStart != NULL) {
+ result = pDevice->pContext->callbacks.onDeviceStart(pDevice);
} else {
- /* There's no second line. Just copy the whole description. */
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), DESC, (size_t)-1);
+ result = MA_INVALID_OPERATION;
}
- }
-
- if (!ma_is_device_blacklisted__alsa(deviceType, NAME)) {
- cbResult = callback(pContext, deviceType, &deviceInfo, pUserData);
- }
- /*
- Some devices are both playback and capture, but they are only enumerated by ALSA once. We need to fire the callback
- again for the other device type in this case. We do this for known devices.
- */
- if (cbResult) {
- if (ma_is_common_device_name__alsa(NAME)) {
- if (deviceType == ma_device_type_playback) {
- if (!ma_is_capture_device_blacklisted__alsa(NAME)) {
- cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
- }
- } else {
- if (!ma_is_playback_device_blacklisted__alsa(NAME)) {
- cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
- }
- }
+ if (result == MA_SUCCESS) {
+ ma_device__set_state(pDevice, ma_device_state_started);
+ ma_device__on_notification_started(pDevice);
}
+ } else {
+ /*
+ Synchronous backends are started by signaling an event that's being waited on in the worker thread. We first wake up the
+ thread and then wait for the start event.
+ */
+ ma_event_signal(&pDevice->wakeupEvent);
+
+ /*
+ Wait for the worker thread to finish starting the device. Note that the worker thread will be the one who puts the device
+ into the started state. Don't call ma_device__set_state() here.
+ */
+ ma_event_wait(&pDevice->startEvent);
+ result = pDevice->workResult;
}
- if (cbResult == MA_FALSE) {
- stopEnumeration = MA_TRUE;
+ /* We changed the state from stopped to started, so if we failed, make sure we put the state back to stopped. */
+ if (result != MA_SUCCESS) {
+ ma_device__set_state(pDevice, ma_device_state_stopped);
}
+ }
+ ma_mutex_unlock(&pDevice->startStopLock);
- next_device:
- free(NAME);
- free(DESC);
- free(IOID);
- ppNextDeviceHint += 1;
+ return result;
+}
- /* We need to stop enumeration if the callback returned false. */
- if (stopEnumeration) {
- break;
- }
+MA_API ma_result ma_device_stop(ma_device* pDevice)
+{
+ ma_result result;
+
+ if (pDevice == NULL) {
+ return MA_INVALID_ARGS;
}
- ma__free_from_callbacks(pUniqueIDs, &pContext->allocationCallbacks);
- ((ma_snd_device_name_free_hint_proc)pContext->alsa.snd_device_name_free_hint)((void**)ppDeviceHints);
+ if (ma_device_get_state(pDevice) == ma_device_state_uninitialized) {
+ return MA_INVALID_OPERATION; /* Not initialized. */
+ }
- ma_mutex_unlock(&pContext->alsa.internalDeviceEnumLock);
+ if (ma_device_get_state(pDevice) == ma_device_state_stopped) {
+ return MA_SUCCESS; /* Already stopped. */
+ }
- return MA_SUCCESS;
-}
+ ma_mutex_lock(&pDevice->startStopLock);
+ {
+ /* Starting and stopping are wrapped in a mutex which means we can assert that the device is in a started or paused state. */
+ MA_ASSERT(ma_device_get_state(pDevice) == ma_device_state_started);
+ ma_device__set_state(pDevice, ma_device_state_stopping);
-typedef struct
-{
- ma_device_type deviceType;
- const ma_device_id* pDeviceID;
- ma_share_mode shareMode;
- ma_device_info* pDeviceInfo;
- ma_bool32 foundDevice;
-} ma_context_get_device_info_enum_callback_data__alsa;
+ /* Asynchronous backends need to be handled differently. */
+ if (ma_context_is_backend_asynchronous(pDevice->pContext)) {
+ /* Asynchronous backends must have a stop operation. */
+ if (pDevice->pContext->callbacks.onDeviceStop != NULL) {
+ result = pDevice->pContext->callbacks.onDeviceStop(pDevice);
+ } else {
+ result = MA_INVALID_OPERATION;
+ }
-static ma_bool32 ma_context_get_device_info_enum_callback__alsa(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pDeviceInfo, void* pUserData)
-{
- ma_context_get_device_info_enum_callback_data__alsa* pData = (ma_context_get_device_info_enum_callback_data__alsa*)pUserData;
- MA_ASSERT(pData != NULL);
+ ma_device__set_state(pDevice, ma_device_state_stopped);
+ } else {
+ /*
+ Synchronous backends. The stop callback is always called from the worker thread. Do not call the stop callback here. If
+ the backend is implementing it's own audio thread loop we'll need to wake it up if required. Note that we need to make
+ sure the state of the device is *not* playing right now, which it shouldn't be since we set it above. This is super
+ important though, so I'm asserting it here as well for extra safety in case we accidentally change something later.
+ */
+ MA_ASSERT(ma_device_get_state(pDevice) != ma_device_state_started);
- (void)pContext;
+ if (pDevice->pContext->callbacks.onDeviceDataLoopWakeup != NULL) {
+ pDevice->pContext->callbacks.onDeviceDataLoopWakeup(pDevice);
+ }
- if (pData->pDeviceID == NULL && ma_strcmp(pDeviceInfo->id.alsa, "default") == 0) {
- ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), pDeviceInfo->name, (size_t)-1);
- pData->foundDevice = MA_TRUE;
- } else {
- if (pData->deviceType == deviceType && (pData->pDeviceID != NULL && ma_strcmp(pData->pDeviceID->alsa, pDeviceInfo->id.alsa) == 0)) {
- ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), pDeviceInfo->name, (size_t)-1);
- pData->foundDevice = MA_TRUE;
+ /*
+ We need to wait for the worker thread to become available for work before returning. Note that the worker thread will be
+ the one who puts the device into the stopped state. Don't call ma_device__set_state() here.
+ */
+ ma_event_wait(&pDevice->stopEvent);
+ result = MA_SUCCESS;
}
}
+ ma_mutex_unlock(&pDevice->startStopLock);
- /* Keep enumerating until we have found the device. */
- return !pData->foundDevice;
+ return result;
}
-static void ma_context_test_rate_and_add_native_data_format__alsa(ma_context* pContext, ma_snd_pcm_t* pPCM, ma_snd_pcm_hw_params_t* pHWParams, ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 flags, ma_device_info* pDeviceInfo)
+MA_API ma_bool32 ma_device_is_started(const ma_device* pDevice)
{
- MA_ASSERT(pPCM != NULL);
- MA_ASSERT(pHWParams != NULL);
- MA_ASSERT(pDeviceInfo != NULL);
+ return ma_device_get_state(pDevice) == ma_device_state_started;
+}
- if (pDeviceInfo->nativeDataFormatCount < ma_countof(pDeviceInfo->nativeDataFormats) && ((ma_snd_pcm_hw_params_test_rate_proc)pContext->alsa.snd_pcm_hw_params_test_rate)(pPCM, pHWParams, sampleRate, 0) == 0) {
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = format;
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = channels;
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = sampleRate;
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = flags;
- pDeviceInfo->nativeDataFormatCount += 1;
+MA_API ma_device_state ma_device_get_state(const ma_device* pDevice)
+{
+ if (pDevice == NULL) {
+ return ma_device_state_uninitialized;
}
+
+ return (ma_device_state)c89atomic_load_i32((ma_int32*)&pDevice->state); /* Naughty cast to get rid of a const warning. */
}
-static void ma_context_iterate_rates_and_add_native_data_format__alsa(ma_context* pContext, ma_snd_pcm_t* pPCM, ma_snd_pcm_hw_params_t* pHWParams, ma_format format, ma_uint32 channels, ma_uint32 flags, ma_device_info* pDeviceInfo)
+MA_API ma_result ma_device_set_master_volume(ma_device* pDevice, float volume)
{
- ma_uint32 iSampleRate;
- unsigned int minSampleRate;
- unsigned int maxSampleRate;
- int sampleRateDir; /* Not used. Just passed into snd_pcm_hw_params_get_rate_min/max(). */
+ if (pDevice == NULL) {
+ return MA_INVALID_ARGS;
+ }
- /* There could be a range. */
- ((ma_snd_pcm_hw_params_get_rate_min_proc)pContext->alsa.snd_pcm_hw_params_get_rate_min)(pHWParams, &minSampleRate, &sampleRateDir);
- ((ma_snd_pcm_hw_params_get_rate_max_proc)pContext->alsa.snd_pcm_hw_params_get_rate_max)(pHWParams, &maxSampleRate, &sampleRateDir);
+ if (volume < 0.0f) {
+ return MA_INVALID_ARGS;
+ }
- /* Make sure our sample rates are clamped to sane values. Stupid devices like "pulse" will reports rates like "1" which is ridiculus. */
- minSampleRate = ma_clamp(minSampleRate, (unsigned int)ma_standard_sample_rate_min, (unsigned int)ma_standard_sample_rate_max);
- maxSampleRate = ma_clamp(maxSampleRate, (unsigned int)ma_standard_sample_rate_min, (unsigned int)ma_standard_sample_rate_max);
+ c89atomic_exchange_f32(&pDevice->masterVolumeFactor, volume);
- for (iSampleRate = 0; iSampleRate < ma_countof(g_maStandardSampleRatePriorities); iSampleRate += 1) {
- ma_uint32 standardSampleRate = g_maStandardSampleRatePriorities[iSampleRate];
+ return MA_SUCCESS;
+}
- if (standardSampleRate >= minSampleRate && standardSampleRate <= maxSampleRate) {
- ma_context_test_rate_and_add_native_data_format__alsa(pContext, pPCM, pHWParams, format, channels, standardSampleRate, flags, pDeviceInfo);
- }
+MA_API ma_result ma_device_get_master_volume(ma_device* pDevice, float* pVolume)
+{
+ if (pVolume == NULL) {
+ return MA_INVALID_ARGS;
}
- /* Now make sure our min and max rates are included just in case they aren't in the range of our standard rates. */
- if (!ma_is_standard_sample_rate(minSampleRate)) {
- ma_context_test_rate_and_add_native_data_format__alsa(pContext, pPCM, pHWParams, format, channels, minSampleRate, flags, pDeviceInfo);
+ if (pDevice == NULL) {
+ *pVolume = 0;
+ return MA_INVALID_ARGS;
}
- if (!ma_is_standard_sample_rate(maxSampleRate) && maxSampleRate != minSampleRate) {
- ma_context_test_rate_and_add_native_data_format__alsa(pContext, pPCM, pHWParams, format, channels, maxSampleRate, flags, pDeviceInfo);
+ *pVolume = c89atomic_load_f32(&pDevice->masterVolumeFactor);
+
+ return MA_SUCCESS;
+}
+
+MA_API ma_result ma_device_set_master_volume_db(ma_device* pDevice, float gainDB)
+{
+ if (gainDB > 0) {
+ return MA_INVALID_ARGS;
}
+
+ return ma_device_set_master_volume(pDevice, ma_volume_db_to_linear(gainDB));
}
-static ma_result ma_context_get_device_info__alsa(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+MA_API ma_result ma_device_get_master_volume_db(ma_device* pDevice, float* pGainDB)
{
- ma_context_get_device_info_enum_callback_data__alsa data;
+ float factor;
ma_result result;
- int resultALSA;
- ma_snd_pcm_t* pPCM;
- ma_snd_pcm_hw_params_t* pHWParams;
- ma_uint32 iFormat;
- ma_uint32 iChannel;
- MA_ASSERT(pContext != NULL);
+ if (pGainDB == NULL) {
+ return MA_INVALID_ARGS;
+ }
- /* We just enumerate to find basic information about the device. */
- data.deviceType = deviceType;
- data.pDeviceID = pDeviceID;
- data.pDeviceInfo = pDeviceInfo;
- data.foundDevice = MA_FALSE;
- result = ma_context_enumerate_devices__alsa(pContext, ma_context_get_device_info_enum_callback__alsa, &data);
+ result = ma_device_get_master_volume(pDevice, &factor);
if (result != MA_SUCCESS) {
+ *pGainDB = 0;
return result;
}
- if (!data.foundDevice) {
- return MA_NO_DEVICE;
- }
+ *pGainDB = ma_volume_linear_to_db(factor);
- if (ma_strcmp(pDeviceInfo->id.alsa, "default") == 0) {
- pDeviceInfo->isDefault = MA_TRUE;
+ return MA_SUCCESS;
+}
+
+
+MA_API ma_result ma_device_handle_backend_data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
+{
+ if (pDevice == NULL) {
+ return MA_INVALID_ARGS;
}
- /* For detailed info we need to open the device. */
- result = ma_context_open_pcm__alsa(pContext, ma_share_mode_shared, deviceType, pDeviceID, 0, &pPCM);
- if (result != MA_SUCCESS) {
- return result;
+ if (pOutput == NULL && pInput == NULL) {
+ return MA_INVALID_ARGS;
}
- /* We need to initialize a HW parameters object in order to know what formats are supported. */
- pHWParams = (ma_snd_pcm_hw_params_t*)ma__calloc_from_callbacks(((ma_snd_pcm_hw_params_sizeof_proc)pContext->alsa.snd_pcm_hw_params_sizeof)(), &pContext->allocationCallbacks);
- if (pHWParams == NULL) {
- ((ma_snd_pcm_close_proc)pContext->alsa.snd_pcm_close)(pPCM);
- return MA_OUT_OF_MEMORY;
+ if (pDevice->type == ma_device_type_duplex) {
+ if (pInput != NULL) {
+ ma_device__handle_duplex_callback_capture(pDevice, frameCount, pInput, &pDevice->duplexRB.rb);
+ }
+
+ if (pOutput != NULL) {
+ ma_device__handle_duplex_callback_playback(pDevice, frameCount, pOutput, &pDevice->duplexRB.rb);
+ }
+ } else {
+ if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_loopback) {
+ if (pInput == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ ma_device__send_frames_to_client(pDevice, frameCount, pInput);
+ }
+
+ if (pDevice->type == ma_device_type_playback) {
+ if (pOutput == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ ma_device__read_frames_from_client(pDevice, frameCount, pOutput);
+ }
}
- resultALSA = ((ma_snd_pcm_hw_params_any_proc)pContext->alsa.snd_pcm_hw_params_any)(pPCM, pHWParams);
- if (resultALSA < 0) {
- ma__free_from_callbacks(pHWParams, &pContext->allocationCallbacks);
- ((ma_snd_pcm_close_proc)pContext->alsa.snd_pcm_close)(pPCM);
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to initialize hardware parameters. snd_pcm_hw_params_any() failed.", ma_result_from_errno(-resultALSA));
+ return MA_SUCCESS;
+}
+
+MA_API ma_uint32 ma_calculate_buffer_size_in_frames_from_descriptor(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile)
+{
+ if (pDescriptor == NULL) {
+ return 0;
}
/*
- Some ALSA devices can support many permutations of formats, channels and rates. We only support
- a fixed number of permutations which means we need to employ some strategies to ensure the best
- combinations are returned. An example is the "pulse" device which can do it's own data conversion
- in software and as a result can support any combination of format, channels and rate.
-
- We want to ensure the the first data formats are the best. We have a list of favored sample
- formats and sample rates, so these will be the basis of our iteration.
+ We must have a non-0 native sample rate, but some backends don't allow retrieval of this at the
+ time when the size of the buffer needs to be determined. In this case we need to just take a best
+ guess and move on. We'll try using the sample rate in pDescriptor first. If that's not set we'll
+ just fall back to MA_DEFAULT_SAMPLE_RATE.
*/
+ if (nativeSampleRate == 0) {
+ nativeSampleRate = pDescriptor->sampleRate;
+ }
+ if (nativeSampleRate == 0) {
+ nativeSampleRate = MA_DEFAULT_SAMPLE_RATE;
+ }
- /* Formats. We just iterate over our standard formats and test them, making sure we reset the configuration space each iteration. */
- for (iFormat = 0; iFormat < ma_countof(g_maFormatPriorities); iFormat += 1) {
- ma_format format = g_maFormatPriorities[iFormat];
+ MA_ASSERT(nativeSampleRate != 0);
- /*
- For each format we need to make sure we reset the configuration space so we don't return
- channel counts and rates that aren't compatible with a format.
- */
- ((ma_snd_pcm_hw_params_any_proc)pContext->alsa.snd_pcm_hw_params_any)(pPCM, pHWParams);
+ if (pDescriptor->periodSizeInFrames == 0) {
+ if (pDescriptor->periodSizeInMilliseconds == 0) {
+ if (performanceProfile == ma_performance_profile_low_latency) {
+ return ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY, nativeSampleRate);
+ } else {
+ return ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE, nativeSampleRate);
+ }
+ } else {
+ return ma_calculate_buffer_size_in_frames_from_milliseconds(pDescriptor->periodSizeInMilliseconds, nativeSampleRate);
+ }
+ } else {
+ return pDescriptor->periodSizeInFrames;
+ }
+}
+#endif /* MA_NO_DEVICE_IO */
- /* Test the format first. If this fails it means the format is not supported and we can skip it. */
- if (((ma_snd_pcm_hw_params_test_format_proc)pContext->alsa.snd_pcm_hw_params_test_format)(pPCM, pHWParams, ma_convert_ma_format_to_alsa_format(format)) == 0) {
- /* The format is supported. */
- unsigned int minChannels;
- unsigned int maxChannels;
- /*
- The configuration space needs to be restricted to this format so we can get an accurate
- picture of which sample rates and channel counts are support with this format.
- */
- ((ma_snd_pcm_hw_params_set_format_proc)pContext->alsa.snd_pcm_hw_params_set_format)(pPCM, pHWParams, ma_convert_ma_format_to_alsa_format(format));
+MA_API ma_uint32 ma_calculate_buffer_size_in_milliseconds_from_frames(ma_uint32 bufferSizeInFrames, ma_uint32 sampleRate)
+{
+ /* Prevent a division by zero. */
+ if (sampleRate == 0) {
+ return 0;
+ }
- /* Now we need to check for supported channels. */
- ((ma_snd_pcm_hw_params_get_channels_min_proc)pContext->alsa.snd_pcm_hw_params_get_channels_min)(pHWParams, &minChannels);
- ((ma_snd_pcm_hw_params_get_channels_max_proc)pContext->alsa.snd_pcm_hw_params_get_channels_max)(pHWParams, &maxChannels);
+ return bufferSizeInFrames*1000 / sampleRate;
+}
- if (minChannels > MA_MAX_CHANNELS) {
- continue; /* Too many channels. */
- }
- if (maxChannels < MA_MIN_CHANNELS) {
- continue; /* Not enough channels. */
- }
+MA_API ma_uint32 ma_calculate_buffer_size_in_frames_from_milliseconds(ma_uint32 bufferSizeInMilliseconds, ma_uint32 sampleRate)
+{
+ /* Prevent a division by zero. */
+ if (sampleRate == 0) {
+ return 0;
+ }
- /*
- Make sure the channel count is clamped. This is mainly intended for the max channels
- because some devices can report an unbound maximum.
- */
- minChannels = ma_clamp(minChannels, MA_MIN_CHANNELS, MA_MAX_CHANNELS);
- maxChannels = ma_clamp(maxChannels, MA_MIN_CHANNELS, MA_MAX_CHANNELS);
+ return bufferSizeInMilliseconds*sampleRate / 1000;
+}
- if (minChannels == MA_MIN_CHANNELS && maxChannels == MA_MAX_CHANNELS) {
- /* The device supports all channels. Don't iterate over every single one. Instead just set the channels to 0 which means all channels are supported. */
- ma_context_iterate_rates_and_add_native_data_format__alsa(pContext, pPCM, pHWParams, format, 0, 0, pDeviceInfo); /* Intentionally setting the channel count to 0 as that means all channels are supported. */
- } else {
- /* The device only supports a specific set of channels. We need to iterate over all of them. */
- for (iChannel = minChannels; iChannel <= maxChannels; iChannel += 1) {
- /* Test the channel before applying it to the configuration space. */
- unsigned int channels = iChannel;
+MA_API void ma_copy_pcm_frames(void* dst, const void* src, ma_uint64 frameCount, ma_format format, ma_uint32 channels)
+{
+ if (dst == src) {
+ return; /* No-op. */
+ }
+
+ ma_copy_memory_64(dst, src, frameCount * ma_get_bytes_per_frame(format, channels));
+}
+
+MA_API void ma_silence_pcm_frames(void* p, ma_uint64 frameCount, ma_format format, ma_uint32 channels)
+{
+ if (format == ma_format_u8) {
+ ma_uint64 sampleCount = frameCount * channels;
+ ma_uint64 iSample;
+ for (iSample = 0; iSample < sampleCount; iSample += 1) {
+ ((ma_uint8*)p)[iSample] = 128;
+ }
+ } else {
+ ma_zero_memory_64(p, frameCount * ma_get_bytes_per_frame(format, channels));
+ }
+}
+
+MA_API void* ma_offset_pcm_frames_ptr(void* p, ma_uint64 offsetInFrames, ma_format format, ma_uint32 channels)
+{
+ return ma_offset_ptr(p, offsetInFrames * ma_get_bytes_per_frame(format, channels));
+}
+
+MA_API const void* ma_offset_pcm_frames_const_ptr(const void* p, ma_uint64 offsetInFrames, ma_format format, ma_uint32 channels)
+{
+ return ma_offset_ptr(p, offsetInFrames * ma_get_bytes_per_frame(format, channels));
+}
- /* Make sure our channel range is reset before testing again or else we'll always fail the test. */
- ((ma_snd_pcm_hw_params_any_proc)pContext->alsa.snd_pcm_hw_params_any)(pPCM, pHWParams);
- ((ma_snd_pcm_hw_params_set_format_proc)pContext->alsa.snd_pcm_hw_params_set_format)(pPCM, pHWParams, ma_convert_ma_format_to_alsa_format(format));
- if (((ma_snd_pcm_hw_params_test_channels_proc)pContext->alsa.snd_pcm_hw_params_test_channels)(pPCM, pHWParams, channels) == 0) {
- /* The channel count is supported. */
+MA_API void ma_clip_samples_u8(ma_uint8* pDst, const ma_int16* pSrc, ma_uint64 count)
+{
+ ma_uint64 iSample;
- /* The configuration space now needs to be restricted to the channel count before extracting the sample rate. */
- ((ma_snd_pcm_hw_params_set_channels_proc)pContext->alsa.snd_pcm_hw_params_set_channels)(pPCM, pHWParams, channels);
+ MA_ASSERT(pDst != NULL);
+ MA_ASSERT(pSrc != NULL);
- /* Only after the configuration space has been restricted to the specific channel count should we iterate over our sample rates. */
- ma_context_iterate_rates_and_add_native_data_format__alsa(pContext, pPCM, pHWParams, format, channels, 0, pDeviceInfo);
- } else {
- /* The channel count is not supported. Skip. */
- }
- }
- }
- } else {
- /* The format is not supported. Skip. */
- }
+ for (iSample = 0; iSample < count; iSample += 1) {
+ pDst[iSample] = ma_clip_u8(pSrc[iSample]);
}
+}
- ma__free_from_callbacks(pHWParams, &pContext->allocationCallbacks);
+MA_API void ma_clip_samples_s16(ma_int16* pDst, const ma_int32* pSrc, ma_uint64 count)
+{
+ ma_uint64 iSample;
- ((ma_snd_pcm_close_proc)pContext->alsa.snd_pcm_close)(pPCM);
- return MA_SUCCESS;
+ MA_ASSERT(pDst != NULL);
+ MA_ASSERT(pSrc != NULL);
+
+ for (iSample = 0; iSample < count; iSample += 1) {
+ pDst[iSample] = ma_clip_s16(pSrc[iSample]);
+ }
}
-static ma_result ma_device_uninit__alsa(ma_device* pDevice)
+MA_API void ma_clip_samples_s24(ma_uint8* pDst, const ma_int64* pSrc, ma_uint64 count)
{
- MA_ASSERT(pDevice != NULL);
+ ma_uint64 iSample;
- if ((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture) {
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture);
- close(pDevice->alsa.wakeupfdCapture);
- ma_free(pDevice->alsa.pPollDescriptorsCapture, &pDevice->pContext->allocationCallbacks);
+ MA_ASSERT(pDst != NULL);
+ MA_ASSERT(pSrc != NULL);
+
+ for (iSample = 0; iSample < count; iSample += 1) {
+ ma_int64 s = ma_clip_s24(pSrc[iSample]);
+ pDst[iSample*3 + 0] = (ma_uint8)((s & 0x000000FF) >> 0);
+ pDst[iSample*3 + 1] = (ma_uint8)((s & 0x0000FF00) >> 8);
+ pDst[iSample*3 + 2] = (ma_uint8)((s & 0x00FF0000) >> 16);
}
+}
- if ((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback) {
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback);
- close(pDevice->alsa.wakeupfdPlayback);
- ma_free(pDevice->alsa.pPollDescriptorsPlayback, &pDevice->pContext->allocationCallbacks);
+MA_API void ma_clip_samples_s32(ma_int32* pDst, const ma_int64* pSrc, ma_uint64 count)
+{
+ ma_uint64 iSample;
+
+ MA_ASSERT(pDst != NULL);
+ MA_ASSERT(pSrc != NULL);
+
+ for (iSample = 0; iSample < count; iSample += 1) {
+ pDst[iSample] = ma_clip_s32(pSrc[iSample]);
}
+}
- return MA_SUCCESS;
+MA_API void ma_clip_samples_f32(float* pDst, const float* pSrc, ma_uint64 count)
+{
+ ma_uint64 iSample;
+
+ MA_ASSERT(pDst != NULL);
+ MA_ASSERT(pSrc != NULL);
+
+ for (iSample = 0; iSample < count; iSample += 1) {
+ pDst[iSample] = ma_clip_f32(pSrc[iSample]);
+ }
}
-static ma_result ma_device_init_by_type__alsa(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptor, ma_device_type deviceType)
+MA_API void ma_clip_pcm_frames(void* pDst, const void* pSrc, ma_uint64 frameCount, ma_format format, ma_uint32 channels)
{
- ma_result result;
- int resultALSA;
- ma_snd_pcm_t* pPCM;
- ma_bool32 isUsingMMap;
- ma_snd_pcm_format_t formatALSA;
- ma_format internalFormat;
- ma_uint32 internalChannels;
- ma_uint32 internalSampleRate;
- ma_channel internalChannelMap[MA_MAX_CHANNELS];
- ma_uint32 internalPeriodSizeInFrames;
- ma_uint32 internalPeriods;
- int openMode;
- ma_snd_pcm_hw_params_t* pHWParams;
- ma_snd_pcm_sw_params_t* pSWParams;
- ma_snd_pcm_uframes_t bufferBoundary;
- int pollDescriptorCount;
- struct pollfd* pPollDescriptors;
- int wakeupfd;
+ ma_uint64 sampleCount;
- MA_ASSERT(pConfig != NULL);
- MA_ASSERT(deviceType != ma_device_type_duplex); /* This function should only be called for playback _or_ capture, never duplex. */
- MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(pDst != NULL);
+ MA_ASSERT(pSrc != NULL);
- formatALSA = ma_convert_ma_format_to_alsa_format(pDescriptor->format);
+ sampleCount = frameCount * channels;
- openMode = 0;
- if (pConfig->alsa.noAutoResample) {
- openMode |= MA_SND_PCM_NO_AUTO_RESAMPLE;
- }
- if (pConfig->alsa.noAutoChannels) {
- openMode |= MA_SND_PCM_NO_AUTO_CHANNELS;
+ switch (format) {
+ case ma_format_u8: ma_clip_samples_u8( (ma_uint8*)pDst, (const ma_int16*)pSrc, sampleCount); break;
+ case ma_format_s16: ma_clip_samples_s16((ma_int16*)pDst, (const ma_int32*)pSrc, sampleCount); break;
+ case ma_format_s24: ma_clip_samples_s24((ma_uint8*)pDst, (const ma_int64*)pSrc, sampleCount); break;
+ case ma_format_s32: ma_clip_samples_s32((ma_int32*)pDst, (const ma_int64*)pSrc, sampleCount); break;
+ case ma_format_f32: ma_clip_samples_f32(( float*)pDst, (const float*)pSrc, sampleCount); break;
+
+ /* Do nothing if we don't know the format. We're including these here to silence a compiler warning about enums not being handled by the switch. */
+ case ma_format_unknown:
+ case ma_format_count:
+ break;
}
- if (pConfig->alsa.noAutoFormat) {
- openMode |= MA_SND_PCM_NO_AUTO_FORMAT;
+}
+
+
+MA_API void ma_copy_and_apply_volume_factor_u8(ma_uint8* pSamplesOut, const ma_uint8* pSamplesIn, ma_uint64 sampleCount, float factor)
+{
+ ma_uint64 iSample;
+
+ if (pSamplesOut == NULL || pSamplesIn == NULL) {
+ return;
}
- result = ma_context_open_pcm__alsa(pDevice->pContext, pDescriptor->shareMode, deviceType, pDescriptor->pDeviceID, openMode, &pPCM);
- if (result != MA_SUCCESS) {
- return result;
+ for (iSample = 0; iSample < sampleCount; iSample += 1) {
+ pSamplesOut[iSample] = (ma_uint8)(pSamplesIn[iSample] * factor);
}
+}
+MA_API void ma_copy_and_apply_volume_factor_s16(ma_int16* pSamplesOut, const ma_int16* pSamplesIn, ma_uint64 sampleCount, float factor)
+{
+ ma_uint64 iSample;
- /* Hardware parameters. */
- pHWParams = (ma_snd_pcm_hw_params_t*)ma__calloc_from_callbacks(((ma_snd_pcm_hw_params_sizeof_proc)pDevice->pContext->alsa.snd_pcm_hw_params_sizeof)(), &pDevice->pContext->allocationCallbacks);
- if (pHWParams == NULL) {
- return MA_OUT_OF_MEMORY;
+ if (pSamplesOut == NULL || pSamplesIn == NULL) {
+ return;
}
- resultALSA = ((ma_snd_pcm_hw_params_any_proc)pDevice->pContext->alsa.snd_pcm_hw_params_any)(pPCM, pHWParams);
- if (resultALSA < 0) {
- ma__free_from_callbacks(pHWParams, &pDevice->pContext->allocationCallbacks);
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to initialize hardware parameters. snd_pcm_hw_params_any() failed.", ma_result_from_errno(-resultALSA));
+ for (iSample = 0; iSample < sampleCount; iSample += 1) {
+ pSamplesOut[iSample] = (ma_int16)(pSamplesIn[iSample] * factor);
}
+}
- /* MMAP Mode. Try using interleaved MMAP access. If this fails, fall back to standard readi/writei. */
- isUsingMMap = MA_FALSE;
-#if 0 /* NOTE: MMAP mode temporarily disabled. */
- if (deviceType != ma_device_type_capture) { /* <-- Disabling MMAP mode for capture devices because I apparently do not have a device that supports it which means I can't test it... Contributions welcome. */
- if (!pConfig->alsa.noMMap && ma_device__is_async(pDevice)) {
- if (((ma_snd_pcm_hw_params_set_access_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_access)(pPCM, pHWParams, MA_SND_PCM_ACCESS_MMAP_INTERLEAVED) == 0) {
- pDevice->alsa.isUsingMMap = MA_TRUE;
- }
- }
- }
-#endif
+MA_API void ma_copy_and_apply_volume_factor_s24(void* pSamplesOut, const void* pSamplesIn, ma_uint64 sampleCount, float factor)
+{
+ ma_uint64 iSample;
+ ma_uint8* pSamplesOut8;
+ ma_uint8* pSamplesIn8;
- if (!isUsingMMap) {
- resultALSA = ((ma_snd_pcm_hw_params_set_access_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_access)(pPCM, pHWParams, MA_SND_PCM_ACCESS_RW_INTERLEAVED);
- if (resultALSA < 0) {
- ma__free_from_callbacks(pHWParams, &pDevice->pContext->allocationCallbacks);
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set access mode to neither SND_PCM_ACCESS_MMAP_INTERLEAVED nor SND_PCM_ACCESS_RW_INTERLEAVED. snd_pcm_hw_params_set_access() failed.", ma_result_from_errno(-resultALSA));
- }
+ if (pSamplesOut == NULL || pSamplesIn == NULL) {
+ return;
}
- /*
- Most important properties first. The documentation for OSS (yes, I know this is ALSA!) recommends format, channels, then sample rate. I can't
- find any documentation for ALSA specifically, so I'm going to copy the recommendation for OSS.
- */
+ pSamplesOut8 = (ma_uint8*)pSamplesOut;
+ pSamplesIn8 = (ma_uint8*)pSamplesIn;
- /* Format. */
- {
- /*
- At this point we should have a list of supported formats, so now we need to find the best one. We first check if the requested format is
- supported, and if so, use that one. If it's not supported, we just run though a list of formats and try to find the best one.
- */
- if (formatALSA == MA_SND_PCM_FORMAT_UNKNOWN || ((ma_snd_pcm_hw_params_test_format_proc)pDevice->pContext->alsa.snd_pcm_hw_params_test_format)(pPCM, pHWParams, formatALSA) != 0) {
- /* We're either requesting the native format or the specified format is not supported. */
- size_t iFormat;
+ for (iSample = 0; iSample < sampleCount; iSample += 1) {
+ ma_int32 sampleS32;
- formatALSA = MA_SND_PCM_FORMAT_UNKNOWN;
- for (iFormat = 0; iFormat < ma_countof(g_maFormatPriorities); ++iFormat) {
- if (((ma_snd_pcm_hw_params_test_format_proc)pDevice->pContext->alsa.snd_pcm_hw_params_test_format)(pPCM, pHWParams, ma_convert_ma_format_to_alsa_format(g_maFormatPriorities[iFormat])) == 0) {
- formatALSA = ma_convert_ma_format_to_alsa_format(g_maFormatPriorities[iFormat]);
- break;
- }
- }
+ sampleS32 = (ma_int32)(((ma_uint32)(pSamplesIn8[iSample*3+0]) << 8) | ((ma_uint32)(pSamplesIn8[iSample*3+1]) << 16) | ((ma_uint32)(pSamplesIn8[iSample*3+2])) << 24);
+ sampleS32 = (ma_int32)(sampleS32 * factor);
- if (formatALSA == MA_SND_PCM_FORMAT_UNKNOWN) {
- ma__free_from_callbacks(pHWParams, &pDevice->pContext->allocationCallbacks);
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Format not supported. The device does not support any miniaudio formats.", MA_FORMAT_NOT_SUPPORTED);
- }
- }
+ pSamplesOut8[iSample*3+0] = (ma_uint8)(((ma_uint32)sampleS32 & 0x0000FF00) >> 8);
+ pSamplesOut8[iSample*3+1] = (ma_uint8)(((ma_uint32)sampleS32 & 0x00FF0000) >> 16);
+ pSamplesOut8[iSample*3+2] = (ma_uint8)(((ma_uint32)sampleS32 & 0xFF000000) >> 24);
+ }
+}
- resultALSA = ((ma_snd_pcm_hw_params_set_format_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_format)(pPCM, pHWParams, formatALSA);
- if (resultALSA < 0) {
- ma__free_from_callbacks(pHWParams, &pDevice->pContext->allocationCallbacks);
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Format not supported. snd_pcm_hw_params_set_format() failed.", ma_result_from_errno(-resultALSA));
- }
+MA_API void ma_copy_and_apply_volume_factor_s32(ma_int32* pSamplesOut, const ma_int32* pSamplesIn, ma_uint64 sampleCount, float factor)
+{
+ ma_uint64 iSample;
- internalFormat = ma_format_from_alsa(formatALSA);
- if (internalFormat == ma_format_unknown) {
- ma__free_from_callbacks(pHWParams, &pDevice->pContext->allocationCallbacks);
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] The chosen format is not supported by miniaudio.", MA_FORMAT_NOT_SUPPORTED);
- }
+ if (pSamplesOut == NULL || pSamplesIn == NULL) {
+ return;
}
- /* Channels. */
- {
- unsigned int channels = pDescriptor->channels;
- if (channels == 0) {
- channels = MA_DEFAULT_CHANNELS;
- }
+ for (iSample = 0; iSample < sampleCount; iSample += 1) {
+ pSamplesOut[iSample] = (ma_int32)(pSamplesIn[iSample] * factor);
+ }
+}
- resultALSA = ((ma_snd_pcm_hw_params_set_channels_near_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_channels_near)(pPCM, pHWParams, &channels);
- if (resultALSA < 0) {
- ma__free_from_callbacks(pHWParams, &pDevice->pContext->allocationCallbacks);
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set channel count. snd_pcm_hw_params_set_channels_near() failed.", ma_result_from_errno(-resultALSA));
- }
+MA_API void ma_copy_and_apply_volume_factor_f32(float* pSamplesOut, const float* pSamplesIn, ma_uint64 sampleCount, float factor)
+{
+ ma_uint64 iSample;
- internalChannels = (ma_uint32)channels;
+ if (pSamplesOut == NULL || pSamplesIn == NULL) {
+ return;
}
- /* Sample Rate */
- {
- unsigned int sampleRate;
+ if (factor == 1) {
+ if (pSamplesOut == pSamplesIn) {
+ /* In place. No-op. */
+ } else {
+ /* Just a copy. */
+ for (iSample = 0; iSample < sampleCount; iSample += 1) {
+ pSamplesOut[iSample] = pSamplesIn[iSample];
+ }
+ }
+ } else {
+ for (iSample = 0; iSample < sampleCount; iSample += 1) {
+ pSamplesOut[iSample] = pSamplesIn[iSample] * factor;
+ }
+ }
+}
- /*
- It appears there's either a bug in ALSA, a bug in some drivers, or I'm doing something silly; but having resampling enabled causes
- problems with some device configurations when used in conjunction with MMAP access mode. To fix this problem we need to disable
- resampling.
+MA_API void ma_apply_volume_factor_u8(ma_uint8* pSamples, ma_uint64 sampleCount, float factor)
+{
+ ma_copy_and_apply_volume_factor_u8(pSamples, pSamples, sampleCount, factor);
+}
- To reproduce this problem, open the "plug:dmix" device, and set the sample rate to 44100. Internally, it looks like dmix uses a
- sample rate of 48000. The hardware parameters will get set correctly with no errors, but it looks like the 44100 -> 48000 resampling
- doesn't work properly - but only with MMAP access mode. You will notice skipping/crackling in the audio, and it'll run at a slightly
- faster rate.
+MA_API void ma_apply_volume_factor_s16(ma_int16* pSamples, ma_uint64 sampleCount, float factor)
+{
+ ma_copy_and_apply_volume_factor_s16(pSamples, pSamples, sampleCount, factor);
+}
- miniaudio has built-in support for sample rate conversion (albeit low quality at the moment), so disabling resampling should be fine
- for us. The only problem is that it won't be taking advantage of any kind of hardware-accelerated resampling and it won't be very
- good quality until I get a chance to improve the quality of miniaudio's software sample rate conversion.
+MA_API void ma_apply_volume_factor_s24(void* pSamples, ma_uint64 sampleCount, float factor)
+{
+ ma_copy_and_apply_volume_factor_s24(pSamples, pSamples, sampleCount, factor);
+}
- I don't currently know if the dmix plugin is the only one with this error. Indeed, this is the only one I've been able to reproduce
- this error with. In the future, we may want to restrict the disabling of resampling to only known bad plugins.
- */
- ((ma_snd_pcm_hw_params_set_rate_resample_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_rate_resample)(pPCM, pHWParams, 0);
+MA_API void ma_apply_volume_factor_s32(ma_int32* pSamples, ma_uint64 sampleCount, float factor)
+{
+ ma_copy_and_apply_volume_factor_s32(pSamples, pSamples, sampleCount, factor);
+}
- sampleRate = pDescriptor->sampleRate;
- if (sampleRate == 0) {
- sampleRate = MA_DEFAULT_SAMPLE_RATE;
- }
+MA_API void ma_apply_volume_factor_f32(float* pSamples, ma_uint64 sampleCount, float factor)
+{
+ ma_copy_and_apply_volume_factor_f32(pSamples, pSamples, sampleCount, factor);
+}
- resultALSA = ((ma_snd_pcm_hw_params_set_rate_near_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_rate_near)(pPCM, pHWParams, &sampleRate, 0);
- if (resultALSA < 0) {
- ma__free_from_callbacks(pHWParams, &pDevice->pContext->allocationCallbacks);
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Sample rate not supported. snd_pcm_hw_params_set_rate_near() failed.", ma_result_from_errno(-resultALSA));
- }
+MA_API void ma_copy_and_apply_volume_factor_pcm_frames_u8(ma_uint8* pFramesOut, const ma_uint8* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor)
+{
+ ma_copy_and_apply_volume_factor_u8(pFramesOut, pFramesIn, frameCount*channels, factor);
+}
- internalSampleRate = (ma_uint32)sampleRate;
- }
+MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s16(ma_int16* pFramesOut, const ma_int16* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor)
+{
+ ma_copy_and_apply_volume_factor_s16(pFramesOut, pFramesIn, frameCount*channels, factor);
+}
- /* Periods. */
- {
- ma_uint32 periods = pDescriptor->periodCount;
+MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s24(void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor)
+{
+ ma_copy_and_apply_volume_factor_s24(pFramesOut, pFramesIn, frameCount*channels, factor);
+}
- resultALSA = ((ma_snd_pcm_hw_params_set_periods_near_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_periods_near)(pPCM, pHWParams, &periods, NULL);
- if (resultALSA < 0) {
- ma__free_from_callbacks(pHWParams, &pDevice->pContext->allocationCallbacks);
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set period count. snd_pcm_hw_params_set_periods_near() failed.", ma_result_from_errno(-resultALSA));
- }
+MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s32(ma_int32* pFramesOut, const ma_int32* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor)
+{
+ ma_copy_and_apply_volume_factor_s32(pFramesOut, pFramesIn, frameCount*channels, factor);
+}
- internalPeriods = periods;
- }
+MA_API void ma_copy_and_apply_volume_factor_pcm_frames_f32(float* pFramesOut, const float* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor)
+{
+ ma_copy_and_apply_volume_factor_f32(pFramesOut, pFramesIn, frameCount*channels, factor);
+}
- /* Buffer Size */
+MA_API void ma_copy_and_apply_volume_factor_pcm_frames(void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount, ma_format format, ma_uint32 channels, float factor)
+{
+ switch (format)
{
- ma_snd_pcm_uframes_t actualBufferSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, internalSampleRate, pConfig->performanceProfile) * internalPeriods;
+ case ma_format_u8: ma_copy_and_apply_volume_factor_pcm_frames_u8 ((ma_uint8*)pFramesOut, (const ma_uint8*)pFramesIn, frameCount, channels, factor); return;
+ case ma_format_s16: ma_copy_and_apply_volume_factor_pcm_frames_s16((ma_int16*)pFramesOut, (const ma_int16*)pFramesIn, frameCount, channels, factor); return;
+ case ma_format_s24: ma_copy_and_apply_volume_factor_pcm_frames_s24( pFramesOut, pFramesIn, frameCount, channels, factor); return;
+ case ma_format_s32: ma_copy_and_apply_volume_factor_pcm_frames_s32((ma_int32*)pFramesOut, (const ma_int32*)pFramesIn, frameCount, channels, factor); return;
+ case ma_format_f32: ma_copy_and_apply_volume_factor_pcm_frames_f32( (float*)pFramesOut, (const float*)pFramesIn, frameCount, channels, factor); return;
+ default: return; /* Do nothing. */
+ }
+}
- resultALSA = ((ma_snd_pcm_hw_params_set_buffer_size_near_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_buffer_size_near)(pPCM, pHWParams, &actualBufferSizeInFrames);
- if (resultALSA < 0) {
- ma__free_from_callbacks(pHWParams, &pDevice->pContext->allocationCallbacks);
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set buffer size for device. snd_pcm_hw_params_set_buffer_size() failed.", ma_result_from_errno(-resultALSA));
- }
+MA_API void ma_apply_volume_factor_pcm_frames_u8(ma_uint8* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor)
+{
+ ma_copy_and_apply_volume_factor_pcm_frames_u8(pFrames, pFrames, frameCount, channels, factor);
+}
- internalPeriodSizeInFrames = actualBufferSizeInFrames / internalPeriods;
- }
+MA_API void ma_apply_volume_factor_pcm_frames_s16(ma_int16* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor)
+{
+ ma_copy_and_apply_volume_factor_pcm_frames_s16(pFrames, pFrames, frameCount, channels, factor);
+}
- /* Apply hardware parameters. */
- resultALSA = ((ma_snd_pcm_hw_params_proc)pDevice->pContext->alsa.snd_pcm_hw_params)(pPCM, pHWParams);
- if (resultALSA < 0) {
- ma__free_from_callbacks(pHWParams, &pDevice->pContext->allocationCallbacks);
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set hardware parameters. snd_pcm_hw_params() failed.", ma_result_from_errno(-resultALSA));
- }
+MA_API void ma_apply_volume_factor_pcm_frames_s24(void* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor)
+{
+ ma_copy_and_apply_volume_factor_pcm_frames_s24(pFrames, pFrames, frameCount, channels, factor);
+}
- ma__free_from_callbacks(pHWParams, &pDevice->pContext->allocationCallbacks);
- pHWParams = NULL;
+MA_API void ma_apply_volume_factor_pcm_frames_s32(ma_int32* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor)
+{
+ ma_copy_and_apply_volume_factor_pcm_frames_s32(pFrames, pFrames, frameCount, channels, factor);
+}
+MA_API void ma_apply_volume_factor_pcm_frames_f32(float* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor)
+{
+ ma_copy_and_apply_volume_factor_pcm_frames_f32(pFrames, pFrames, frameCount, channels, factor);
+}
- /* Software parameters. */
- pSWParams = (ma_snd_pcm_sw_params_t*)ma__calloc_from_callbacks(((ma_snd_pcm_sw_params_sizeof_proc)pDevice->pContext->alsa.snd_pcm_sw_params_sizeof)(), &pDevice->pContext->allocationCallbacks);
- if (pSWParams == NULL) {
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return MA_OUT_OF_MEMORY;
- }
+MA_API void ma_apply_volume_factor_pcm_frames(void* pFramesOut, ma_uint64 frameCount, ma_format format, ma_uint32 channels, float factor)
+{
+ ma_copy_and_apply_volume_factor_pcm_frames(pFramesOut, pFramesOut, frameCount, format, channels, factor);
+}
- resultALSA = ((ma_snd_pcm_sw_params_current_proc)pDevice->pContext->alsa.snd_pcm_sw_params_current)(pPCM, pSWParams);
- if (resultALSA < 0) {
- ma__free_from_callbacks(pSWParams, &pDevice->pContext->allocationCallbacks);
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to initialize software parameters. snd_pcm_sw_params_current() failed.", ma_result_from_errno(-resultALSA));
- }
- resultALSA = ((ma_snd_pcm_sw_params_set_avail_min_proc)pDevice->pContext->alsa.snd_pcm_sw_params_set_avail_min)(pPCM, pSWParams, ma_prev_power_of_2(internalPeriodSizeInFrames));
- if (resultALSA < 0) {
- ma__free_from_callbacks(pSWParams, &pDevice->pContext->allocationCallbacks);
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_sw_params_set_avail_min() failed.", ma_result_from_errno(-resultALSA));
- }
+MA_API void ma_copy_and_apply_volume_factor_per_channel_f32(float* pFramesOut, const float* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float* pChannelGains)
+{
+ ma_uint64 iFrame;
- resultALSA = ((ma_snd_pcm_sw_params_get_boundary_proc)pDevice->pContext->alsa.snd_pcm_sw_params_get_boundary)(pSWParams, &bufferBoundary);
- if (resultALSA < 0) {
- bufferBoundary = internalPeriodSizeInFrames * internalPeriods;
+ if (channels == 2) {
+ /* TODO: Do an optimized implementation for stereo and mono. Can do a SIMD optimized implementation as well. */
}
- if (deviceType == ma_device_type_playback && !isUsingMMap) { /* Only playback devices in writei/readi mode need a start threshold. */
- /*
- Subtle detail here with the start threshold. When in playback-only mode (no full-duplex) we can set the start threshold to
- the size of a period. But for full-duplex we need to set it such that it is at least two periods.
- */
- resultALSA = ((ma_snd_pcm_sw_params_set_start_threshold_proc)pDevice->pContext->alsa.snd_pcm_sw_params_set_start_threshold)(pPCM, pSWParams, internalPeriodSizeInFrames*2);
- if (resultALSA < 0) {
- ma__free_from_callbacks(pSWParams, &pDevice->pContext->allocationCallbacks);
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set start threshold for playback device. snd_pcm_sw_params_set_start_threshold() failed.", ma_result_from_errno(-resultALSA));
- }
-
- resultALSA = ((ma_snd_pcm_sw_params_set_stop_threshold_proc)pDevice->pContext->alsa.snd_pcm_sw_params_set_stop_threshold)(pPCM, pSWParams, bufferBoundary);
- if (resultALSA < 0) { /* Set to boundary to loop instead of stop in the event of an xrun. */
- ma__free_from_callbacks(pSWParams, &pDevice->pContext->allocationCallbacks);
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set stop threshold for playback device. snd_pcm_sw_params_set_stop_threshold() failed.", ma_result_from_errno(-resultALSA));
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ pFramesOut[iFrame * channels + iChannel] = pFramesIn[iFrame * channels + iChannel] * pChannelGains[iChannel];
}
}
+}
- resultALSA = ((ma_snd_pcm_sw_params_proc)pDevice->pContext->alsa.snd_pcm_sw_params)(pPCM, pSWParams);
- if (resultALSA < 0) {
- ma__free_from_callbacks(pSWParams, &pDevice->pContext->allocationCallbacks);
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set software parameters. snd_pcm_sw_params() failed.", ma_result_from_errno(-resultALSA));
- }
- ma__free_from_callbacks(pSWParams, &pDevice->pContext->allocationCallbacks);
- pSWParams = NULL;
+static MA_INLINE ma_int16 ma_apply_volume_unclipped_u8(ma_int16 x, ma_int16 volume)
+{
+ return (ma_int16)(((ma_int32)x * (ma_int32)volume) >> 8);
+}
- /* Grab the internal channel map. For now we're not going to bother trying to change the channel map and instead just do it ourselves. */
- {
- ma_snd_pcm_chmap_t* pChmap = ((ma_snd_pcm_get_chmap_proc)pDevice->pContext->alsa.snd_pcm_get_chmap)(pPCM);
- if (pChmap != NULL) {
- ma_uint32 iChannel;
+static MA_INLINE ma_int32 ma_apply_volume_unclipped_s16(ma_int32 x, ma_int16 volume)
+{
+ return (ma_int32)((x * volume) >> 8);
+}
- /* There are cases where the returned channel map can have a different channel count than was returned by snd_pcm_hw_params_set_channels_near(). */
- if (pChmap->channels >= internalChannels) {
- /* Drop excess channels. */
- for (iChannel = 0; iChannel < internalChannels; ++iChannel) {
- internalChannelMap[iChannel] = ma_convert_alsa_channel_position_to_ma_channel(pChmap->pos[iChannel]);
- }
- } else {
- ma_uint32 i;
+static MA_INLINE ma_int64 ma_apply_volume_unclipped_s24(ma_int64 x, ma_int16 volume)
+{
+ return (ma_int64)((x * volume) >> 8);
+}
- /*
- Excess channels use defaults. Do an initial fill with defaults, overwrite the first pChmap->channels, validate to ensure there are no duplicate
- channels. If validation fails, fall back to defaults.
- */
- ma_bool32 isValid = MA_TRUE;
+static MA_INLINE ma_int64 ma_apply_volume_unclipped_s32(ma_int64 x, ma_int16 volume)
+{
+ return (ma_int64)((x * volume) >> 8);
+}
- /* Fill with defaults. */
- ma_get_standard_channel_map(ma_standard_channel_map_alsa, internalChannels, internalChannelMap);
+static MA_INLINE float ma_apply_volume_unclipped_f32(float x, float volume)
+{
+ return x * volume;
+}
- /* Overwrite first pChmap->channels channels. */
- for (iChannel = 0; iChannel < pChmap->channels; ++iChannel) {
- internalChannelMap[iChannel] = ma_convert_alsa_channel_position_to_ma_channel(pChmap->pos[iChannel]);
- }
- /* Validate. */
- for (i = 0; i < internalChannels && isValid; ++i) {
- ma_uint32 j;
- for (j = i+1; j < internalChannels; ++j) {
- if (internalChannelMap[i] == internalChannelMap[j]) {
- isValid = MA_FALSE;
- break;
- }
- }
- }
+MA_API void ma_copy_and_apply_volume_and_clip_samples_u8(ma_uint8* pDst, const ma_int16* pSrc, ma_uint64 count, float volume)
+{
+ ma_uint64 iSample;
+ ma_int16 volumeFixed;
- /* If our channel map is invalid, fall back to defaults. */
- if (!isValid) {
- ma_get_standard_channel_map(ma_standard_channel_map_alsa, internalChannels, internalChannelMap);
- }
- }
+ MA_ASSERT(pDst != NULL);
+ MA_ASSERT(pSrc != NULL);
- free(pChmap);
- pChmap = NULL;
- } else {
- /* Could not retrieve the channel map. Fall back to a hard-coded assumption. */
- ma_get_standard_channel_map(ma_standard_channel_map_alsa, internalChannels, internalChannelMap);
- }
+ volumeFixed = ma_float_to_fixed_16(volume);
+
+ for (iSample = 0; iSample < count; iSample += 1) {
+ pDst[iSample] = ma_clip_u8(ma_apply_volume_unclipped_u8(pSrc[iSample], volumeFixed));
}
+}
+MA_API void ma_copy_and_apply_volume_and_clip_samples_s16(ma_int16* pDst, const ma_int32* pSrc, ma_uint64 count, float volume)
+{
+ ma_uint64 iSample;
+ ma_int16 volumeFixed;
- /*
- We need to retrieve the poll descriptors so we can use poll() to wait for data to become
- available for reading or writing. There's no well defined maximum for this so we're just going
- to allocate this on the heap.
- */
- pollDescriptorCount = ((ma_snd_pcm_poll_descriptors_count_proc)pDevice->pContext->alsa.snd_pcm_poll_descriptors_count)(pPCM);
- if (pollDescriptorCount <= 0) {
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to retrieve poll descriptors count.", MA_ERROR);
- }
+ MA_ASSERT(pDst != NULL);
+ MA_ASSERT(pSrc != NULL);
- pPollDescriptors = (struct pollfd*)ma_malloc(sizeof(*pPollDescriptors) * (pollDescriptorCount + 1), &pDevice->pContext->allocationCallbacks/*, MA_ALLOCATION_TYPE_GENERAL*/); /* +1 because we want room for the wakeup descriptor. */
- if (pPollDescriptors == NULL) {
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to allocate memory for poll descriptors.", MA_OUT_OF_MEMORY);
- }
+ volumeFixed = ma_float_to_fixed_16(volume);
- /*
- We need an eventfd to wakeup from poll() and avoid a deadlock in situations where the driver
- never returns from writei() and readi(). This has been observed with the "pulse" device.
- */
- wakeupfd = eventfd(0, 0);
- if (wakeupfd < 0) {
- ma_free(pPollDescriptors, &pDevice->pContext->allocationCallbacks);
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to create eventfd for poll wakeup.", ma_result_from_errno(errno));
+ for (iSample = 0; iSample < count; iSample += 1) {
+ pDst[iSample] = ma_clip_s16(ma_apply_volume_unclipped_s16(pSrc[iSample], volumeFixed));
}
+}
- /* We'll place the wakeup fd at the start of the buffer. */
- pPollDescriptors[0].fd = wakeupfd;
- pPollDescriptors[0].events = POLLIN; /* We only care about waiting to read from the wakeup file descriptor. */
- pPollDescriptors[0].revents = 0;
+MA_API void ma_copy_and_apply_volume_and_clip_samples_s24(ma_uint8* pDst, const ma_int64* pSrc, ma_uint64 count, float volume)
+{
+ ma_uint64 iSample;
+ ma_int16 volumeFixed;
- /* We can now extract the PCM poll descriptors which we place after the wakeup descriptor. */
- pollDescriptorCount = ((ma_snd_pcm_poll_descriptors_proc)pDevice->pContext->alsa.snd_pcm_poll_descriptors)(pPCM, pPollDescriptors + 1, pollDescriptorCount); /* +1 because we want to place these descriptors after the wakeup descriptor. */
- if (pollDescriptorCount <= 0) {
- close(wakeupfd);
- ma_free(pPollDescriptors, &pDevice->pContext->allocationCallbacks);
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to retrieve poll descriptors.", MA_ERROR);
+ MA_ASSERT(pDst != NULL);
+ MA_ASSERT(pSrc != NULL);
+
+ volumeFixed = ma_float_to_fixed_16(volume);
+
+ for (iSample = 0; iSample < count; iSample += 1) {
+ ma_int64 s = ma_clip_s24(ma_apply_volume_unclipped_s24(pSrc[iSample], volumeFixed));
+ pDst[iSample*3 + 0] = (ma_uint8)((s & 0x000000FF) >> 0);
+ pDst[iSample*3 + 1] = (ma_uint8)((s & 0x0000FF00) >> 8);
+ pDst[iSample*3 + 2] = (ma_uint8)((s & 0x00FF0000) >> 16);
}
+}
- if (deviceType == ma_device_type_capture) {
- pDevice->alsa.pollDescriptorCountCapture = pollDescriptorCount;
- pDevice->alsa.pPollDescriptorsCapture = pPollDescriptors;
- pDevice->alsa.wakeupfdCapture = wakeupfd;
- } else {
- pDevice->alsa.pollDescriptorCountPlayback = pollDescriptorCount;
- pDevice->alsa.pPollDescriptorsPlayback = pPollDescriptors;
- pDevice->alsa.wakeupfdPlayback = wakeupfd;
+MA_API void ma_copy_and_apply_volume_and_clip_samples_s32(ma_int32* pDst, const ma_int64* pSrc, ma_uint64 count, float volume)
+{
+ ma_uint64 iSample;
+ ma_int16 volumeFixed;
+
+ MA_ASSERT(pDst != NULL);
+ MA_ASSERT(pSrc != NULL);
+
+ volumeFixed = ma_float_to_fixed_16(volume);
+
+ for (iSample = 0; iSample < count; iSample += 1) {
+ pDst[iSample] = ma_clip_s32(ma_apply_volume_unclipped_s32(pSrc[iSample], volumeFixed));
}
+}
+MA_API void ma_copy_and_apply_volume_and_clip_samples_f32(float* pDst, const float* pSrc, ma_uint64 count, float volume)
+{
+ ma_uint64 iSample;
- /* We're done. Prepare the device. */
- resultALSA = ((ma_snd_pcm_prepare_proc)pDevice->pContext->alsa.snd_pcm_prepare)(pPCM);
- if (resultALSA < 0) {
- close(wakeupfd);
- ma_free(pPollDescriptors, &pDevice->pContext->allocationCallbacks);
- ((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to prepare device.", ma_result_from_errno(-resultALSA));
+ MA_ASSERT(pDst != NULL);
+ MA_ASSERT(pSrc != NULL);
+
+ /* For the f32 case we need to make sure this supports in-place processing where the input and output buffers are the same. */
+
+ for (iSample = 0; iSample < count; iSample += 1) {
+ pDst[iSample] = ma_clip_f32(ma_apply_volume_unclipped_f32(pSrc[iSample], volume));
}
+}
+MA_API void ma_copy_and_apply_volume_and_clip_pcm_frames(void* pDst, const void* pSrc, ma_uint64 frameCount, ma_format format, ma_uint32 channels, float volume)
+{
+ MA_ASSERT(pDst != NULL);
+ MA_ASSERT(pSrc != NULL);
- if (deviceType == ma_device_type_capture) {
- pDevice->alsa.pPCMCapture = (ma_ptr)pPCM;
- pDevice->alsa.isUsingMMapCapture = isUsingMMap;
+ if (volume == 1) {
+ ma_clip_pcm_frames(pDst, pSrc, frameCount, format, channels); /* Optimized case for volume = 1. */
+ } else if (volume == 0) {
+ ma_silence_pcm_frames(pDst, frameCount, format, channels); /* Optimized case for volume = 0. */
} else {
- pDevice->alsa.pPCMPlayback = (ma_ptr)pPCM;
- pDevice->alsa.isUsingMMapPlayback = isUsingMMap;
+ ma_uint64 sampleCount = frameCount * channels;
+
+ switch (format) {
+ case ma_format_u8: ma_copy_and_apply_volume_and_clip_samples_u8( (ma_uint8*)pDst, (const ma_int16*)pSrc, sampleCount, volume); break;
+ case ma_format_s16: ma_copy_and_apply_volume_and_clip_samples_s16((ma_int16*)pDst, (const ma_int32*)pSrc, sampleCount, volume); break;
+ case ma_format_s24: ma_copy_and_apply_volume_and_clip_samples_s24((ma_uint8*)pDst, (const ma_int64*)pSrc, sampleCount, volume); break;
+ case ma_format_s32: ma_copy_and_apply_volume_and_clip_samples_s32((ma_int32*)pDst, (const ma_int64*)pSrc, sampleCount, volume); break;
+ case ma_format_f32: ma_copy_and_apply_volume_and_clip_samples_f32(( float*)pDst, (const float*)pSrc, sampleCount, volume); break;
+
+ /* Do nothing if we don't know the format. We're including these here to silence a compiler warning about enums not being handled by the switch. */
+ case ma_format_unknown:
+ case ma_format_count:
+ break;
+ }
}
+}
+
- pDescriptor->format = internalFormat;
- pDescriptor->channels = internalChannels;
- pDescriptor->sampleRate = internalSampleRate;
- ma_channel_map_copy(pDescriptor->channelMap, internalChannelMap, ma_min(internalChannels, MA_MAX_CHANNELS));
- pDescriptor->periodSizeInFrames = internalPeriodSizeInFrames;
- pDescriptor->periodCount = internalPeriods;
- return MA_SUCCESS;
+MA_API float ma_volume_linear_to_db(float factor)
+{
+ return 20*ma_log10f(factor);
}
-static ma_result ma_device_init__alsa(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+MA_API float ma_volume_db_to_linear(float gain)
{
- MA_ASSERT(pDevice != NULL);
+ return ma_powf(10, gain/20.0f);
+}
- MA_ZERO_OBJECT(&pDevice->alsa);
- if (pConfig->deviceType == ma_device_type_loopback) {
- return MA_DEVICE_TYPE_NOT_SUPPORTED;
- }
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
- ma_result result = ma_device_init_by_type__alsa(pDevice, pConfig, pDescriptorCapture, ma_device_type_capture);
- if (result != MA_SUCCESS) {
- return result;
- }
- }
+/**************************************************************************************************************************************************************
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- ma_result result = ma_device_init_by_type__alsa(pDevice, pConfig, pDescriptorPlayback, ma_device_type_playback);
- if (result != MA_SUCCESS) {
- return result;
- }
- }
+Format Conversion
- return MA_SUCCESS;
+**************************************************************************************************************************************************************/
+
+static MA_INLINE ma_int16 ma_pcm_sample_f32_to_s16(float x)
+{
+ return (ma_int16)(x * 32767.0f);
}
-static ma_result ma_device_start__alsa(ma_device* pDevice)
+static MA_INLINE ma_int16 ma_pcm_sample_u8_to_s16_no_scale(ma_uint8 x)
{
- int resultALSA;
+ return (ma_int16)((ma_int16)x - 128);
+}
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- resultALSA = ((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture);
- if (resultALSA < 0) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to start capture device.", ma_result_from_errno(-resultALSA));
- }
- }
+static MA_INLINE ma_int64 ma_pcm_sample_s24_to_s32_no_scale(const ma_uint8* x)
+{
+ return (ma_int64)(((ma_uint64)x[0] << 40) | ((ma_uint64)x[1] << 48) | ((ma_uint64)x[2] << 56)) >> 40; /* Make sure the sign bits are maintained. */
+}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- /* Don't need to do anything for playback because it'll be started automatically when enough data has been written. */
- }
+static MA_INLINE void ma_pcm_sample_s32_to_s24_no_scale(ma_int64 x, ma_uint8* s24)
+{
+ s24[0] = (ma_uint8)((x & 0x000000FF) >> 0);
+ s24[1] = (ma_uint8)((x & 0x0000FF00) >> 8);
+ s24[2] = (ma_uint8)((x & 0x00FF0000) >> 16);
+}
- return MA_SUCCESS;
+
+/* u8 */
+MA_API void ma_pcm_u8_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ (void)ditherMode;
+ ma_copy_memory_64(dst, src, count * sizeof(ma_uint8));
}
-static ma_result ma_device_stop__alsa(ma_device* pDevice)
+
+static MA_INLINE void ma_pcm_u8_to_s16__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Dropping capture device... ");
- ((ma_snd_pcm_drop_proc)pDevice->pContext->alsa.snd_pcm_drop)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture);
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "Done\n");
+ ma_int16* dst_s16 = (ma_int16*)dst;
+ const ma_uint8* src_u8 = (const ma_uint8*)src;
- /* We need to prepare the device again, otherwise we won't be able to restart the device. */
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Preparing capture device... ");
- if (((ma_snd_pcm_prepare_proc)pDevice->pContext->alsa.snd_pcm_prepare)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture) < 0) {
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "Failed\n");
- } else {
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "Done\n");
- }
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ ma_int16 x = src_u8[i];
+ x = (ma_int16)(x - 128);
+ x = (ma_int16)(x << 8);
+ dst_s16[i] = x;
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Dropping playback device... ");
- ((ma_snd_pcm_drop_proc)pDevice->pContext->alsa.snd_pcm_drop)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback);
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "Done\n");
+ (void)ditherMode;
+}
- /* We need to prepare the device again, otherwise we won't be able to restart the device. */
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Preparing playback device... ");
- if (((ma_snd_pcm_prepare_proc)pDevice->pContext->alsa.snd_pcm_prepare)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback) < 0) {
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "Failed\n");
- } else {
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "Done\n");
- }
- }
+static MA_INLINE void ma_pcm_u8_to_s16__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_u8_to_s16__reference(dst, src, count, ditherMode);
+}
- return MA_SUCCESS;
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE void ma_pcm_u8_to_s16__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_u8_to_s16__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE void ma_pcm_u8_to_s16__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_u8_to_s16__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE void ma_pcm_u8_to_s16__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_u8_to_s16__optimized(dst, src, count, ditherMode);
}
+#endif
-static ma_result ma_device_wait__alsa(ma_device* pDevice, ma_snd_pcm_t* pPCM, struct pollfd* pPollDescriptors, int pollDescriptorCount, short requiredEvent)
+MA_API void ma_pcm_u8_to_s16(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- for (;;) {
- unsigned short revents;
- int resultALSA;
- int resultPoll = poll(pPollDescriptors, pollDescriptorCount, -1);
- if (resultPoll < 0) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] poll() failed.", ma_result_from_errno(errno));
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_u8_to_s16__reference(dst, src, count, ditherMode);
+#else
+ # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
+ if (ma_has_avx2()) {
+ ma_pcm_u8_to_s16__avx2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
+ if (ma_has_sse2()) {
+ ma_pcm_u8_to_s16__sse2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
+ if (ma_has_neon()) {
+ ma_pcm_u8_to_s16__neon(dst, src, count, ditherMode);
+ } else
+ #endif
+ {
+ ma_pcm_u8_to_s16__optimized(dst, src, count, ditherMode);
}
+#endif
+}
- /*
- Before checking the ALSA poll descriptor flag we need to check if the wakeup descriptor
- has had it's POLLIN flag set. If so, we need to actually read the data and then exit
- function. The wakeup descriptor will be the first item in the descriptors buffer.
- */
- if ((pPollDescriptors[0].revents & POLLIN) != 0) {
- ma_uint64 t;
- read(pPollDescriptors[0].fd, &t, sizeof(t)); /* <-- Important that we read here so that the next write() does not block. */
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] POLLIN set for wakeupfd\n");
+static MA_INLINE void ma_pcm_u8_to_s24__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_uint8* dst_s24 = (ma_uint8*)dst;
+ const ma_uint8* src_u8 = (const ma_uint8*)src;
- return MA_DEVICE_NOT_STARTED;
- }
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ ma_int16 x = src_u8[i];
+ x = (ma_int16)(x - 128);
- /*
- Getting here means that some data should be able to be read. We need to use ALSA to
- translate the revents flags for us.
- */
- resultALSA = ((ma_snd_pcm_poll_descriptors_revents_proc)pDevice->pContext->alsa.snd_pcm_poll_descriptors_revents)(pPCM, pPollDescriptors + 1, pollDescriptorCount - 1, &revents); /* +1, -1 to ignore the wakeup descriptor. */
- if (resultALSA < 0) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_poll_descriptors_revents() failed.", ma_result_from_errno(-resultALSA));
- }
+ dst_s24[i*3+0] = 0;
+ dst_s24[i*3+1] = 0;
+ dst_s24[i*3+2] = (ma_uint8)((ma_int8)x);
+ }
- if ((revents & POLLERR) != 0) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] POLLERR detected.", ma_result_from_errno(errno));
- }
+ (void)ditherMode;
+}
- if ((revents & requiredEvent) == requiredEvent) {
- break; /* We're done. Data available for reading or writing. */
+static MA_INLINE void ma_pcm_u8_to_s24__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_u8_to_s24__reference(dst, src, count, ditherMode);
+}
+
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE void ma_pcm_u8_to_s24__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_u8_to_s24__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE void ma_pcm_u8_to_s24__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_u8_to_s24__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE void ma_pcm_u8_to_s24__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_u8_to_s24__optimized(dst, src, count, ditherMode);
+}
+#endif
+
+MA_API void ma_pcm_u8_to_s24(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_u8_to_s24__reference(dst, src, count, ditherMode);
+#else
+ # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
+ if (ma_has_avx2()) {
+ ma_pcm_u8_to_s24__avx2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
+ if (ma_has_sse2()) {
+ ma_pcm_u8_to_s24__sse2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
+ if (ma_has_neon()) {
+ ma_pcm_u8_to_s24__neon(dst, src, count, ditherMode);
+ } else
+ #endif
+ {
+ ma_pcm_u8_to_s24__optimized(dst, src, count, ditherMode);
}
+#endif
+}
+
+
+static MA_INLINE void ma_pcm_u8_to_s32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_int32* dst_s32 = (ma_int32*)dst;
+ const ma_uint8* src_u8 = (const ma_uint8*)src;
+
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ ma_int32 x = src_u8[i];
+ x = x - 128;
+ x = x << 24;
+ dst_s32[i] = x;
}
- return MA_SUCCESS;
+ (void)ditherMode;
}
-static ma_result ma_device_wait_read__alsa(ma_device* pDevice)
+static MA_INLINE void ma_pcm_u8_to_s32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- return ma_device_wait__alsa(pDevice, (ma_snd_pcm_t*)pDevice->alsa.pPCMCapture, (struct pollfd*)pDevice->alsa.pPollDescriptorsCapture, pDevice->alsa.pollDescriptorCountCapture + 1, POLLIN); /* +1 to account for the wakeup descriptor. */
+ ma_pcm_u8_to_s32__reference(dst, src, count, ditherMode);
}
-static ma_result ma_device_wait_write__alsa(ma_device* pDevice)
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE void ma_pcm_u8_to_s32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- return ma_device_wait__alsa(pDevice, (ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback, (struct pollfd*)pDevice->alsa.pPollDescriptorsPlayback, pDevice->alsa.pollDescriptorCountPlayback + 1, POLLOUT); /* +1 to account for the wakeup descriptor. */
+ ma_pcm_u8_to_s32__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE void ma_pcm_u8_to_s32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_u8_to_s32__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE void ma_pcm_u8_to_s32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_u8_to_s32__optimized(dst, src, count, ditherMode);
}
+#endif
-static ma_result ma_device_read__alsa(ma_device* pDevice, void* pFramesOut, ma_uint32 frameCount, ma_uint32* pFramesRead)
+MA_API void ma_pcm_u8_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_u8_to_s32__reference(dst, src, count, ditherMode);
+#else
+ # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
+ if (ma_has_avx2()) {
+ ma_pcm_u8_to_s32__avx2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
+ if (ma_has_sse2()) {
+ ma_pcm_u8_to_s32__sse2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
+ if (ma_has_neon()) {
+ ma_pcm_u8_to_s32__neon(dst, src, count, ditherMode);
+ } else
+ #endif
+ {
+ ma_pcm_u8_to_s32__optimized(dst, src, count, ditherMode);
+ }
+#endif
+}
+
+
+static MA_INLINE void ma_pcm_u8_to_f32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- ma_snd_pcm_sframes_t resultALSA;
+ float* dst_f32 = (float*)dst;
+ const ma_uint8* src_u8 = (const ma_uint8*)src;
- MA_ASSERT(pDevice != NULL);
- MA_ASSERT(pFramesOut != NULL);
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ float x = (float)src_u8[i];
+ x = x * 0.00784313725490196078f; /* 0..255 to 0..2 */
+ x = x - 1; /* 0..2 to -1..1 */
- if (pFramesRead != NULL) {
- *pFramesRead = 0;
+ dst_f32[i] = x;
}
- while (ma_device_get_state(pDevice) == MA_STATE_STARTED) {
- ma_result result;
+ (void)ditherMode;
+}
- /* The first thing to do is wait for data to become available for reading. This will return an error code if the device has been stopped. */
- result = ma_device_wait_read__alsa(pDevice);
- if (result != MA_SUCCESS) {
- return result;
- }
+static MA_INLINE void ma_pcm_u8_to_f32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_u8_to_f32__reference(dst, src, count, ditherMode);
+}
- /* Getting here means we should have data available. */
- resultALSA = ((ma_snd_pcm_readi_proc)pDevice->pContext->alsa.snd_pcm_readi)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture, pFramesOut, frameCount);
- if (resultALSA >= 0) {
- break; /* Success. */
- } else {
- if (resultALSA == -EAGAIN) {
- /*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "TRACE: EGAIN (read)\n");*/
- continue; /* Try again. */
- } else if (resultALSA == -EPIPE) {
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "TRACE: EPIPE (read)\n");
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE void ma_pcm_u8_to_f32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_u8_to_f32__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE void ma_pcm_u8_to_f32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_u8_to_f32__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE void ma_pcm_u8_to_f32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_u8_to_f32__optimized(dst, src, count, ditherMode);
+}
+#endif
- /* Overrun. Recover and try again. If this fails we need to return an error. */
- resultALSA = ((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture, resultALSA, MA_TRUE);
- if (resultALSA < 0) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to recover device after overrun.", ma_result_from_errno((int)-resultALSA));
- }
+MA_API void ma_pcm_u8_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_u8_to_f32__reference(dst, src, count, ditherMode);
+#else
+ # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
+ if (ma_has_avx2()) {
+ ma_pcm_u8_to_f32__avx2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
+ if (ma_has_sse2()) {
+ ma_pcm_u8_to_f32__sse2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
+ if (ma_has_neon()) {
+ ma_pcm_u8_to_f32__neon(dst, src, count, ditherMode);
+ } else
+ #endif
+ {
+ ma_pcm_u8_to_f32__optimized(dst, src, count, ditherMode);
+ }
+#endif
+}
- resultALSA = ((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture);
- if (resultALSA < 0) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to start device after underrun.", ma_result_from_errno((int)-resultALSA));
- }
- continue; /* Try reading again. */
- }
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+static MA_INLINE void ma_pcm_interleave_u8__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
+{
+ ma_uint8* dst_u8 = (ma_uint8*)dst;
+ const ma_uint8** src_u8 = (const ma_uint8**)src;
+
+ ma_uint64 iFrame;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ dst_u8[iFrame*channels + iChannel] = src_u8[iChannel][iFrame];
}
}
+}
+#else
+static MA_INLINE void ma_pcm_interleave_u8__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
+{
+ ma_uint8* dst_u8 = (ma_uint8*)dst;
+ const ma_uint8** src_u8 = (const ma_uint8**)src;
- if (pFramesRead != NULL) {
- *pFramesRead = resultALSA;
+ if (channels == 1) {
+ ma_copy_memory_64(dst, src[0], frameCount * sizeof(ma_uint8));
+ } else if (channels == 2) {
+ ma_uint64 iFrame;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ dst_u8[iFrame*2 + 0] = src_u8[0][iFrame];
+ dst_u8[iFrame*2 + 1] = src_u8[1][iFrame];
+ }
+ } else {
+ ma_uint64 iFrame;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ dst_u8[iFrame*channels + iChannel] = src_u8[iChannel][iFrame];
+ }
+ }
}
-
- return MA_SUCCESS;
}
+#endif
-static ma_result ma_device_write__alsa(ma_device* pDevice, const void* pFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten)
+MA_API void ma_pcm_interleave_u8(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
{
- ma_snd_pcm_sframes_t resultALSA;
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_interleave_u8__reference(dst, src, frameCount, channels);
+#else
+ ma_pcm_interleave_u8__optimized(dst, src, frameCount, channels);
+#endif
+}
- MA_ASSERT(pDevice != NULL);
- MA_ASSERT(pFrames != NULL);
- if (pFramesWritten != NULL) {
- *pFramesWritten = 0;
+static MA_INLINE void ma_pcm_deinterleave_u8__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
+{
+ ma_uint8** dst_u8 = (ma_uint8**)dst;
+ const ma_uint8* src_u8 = (const ma_uint8*)src;
+
+ ma_uint64 iFrame;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ dst_u8[iChannel][iFrame] = src_u8[iFrame*channels + iChannel];
+ }
}
+}
- while (ma_device_get_state(pDevice) == MA_STATE_STARTED) {
- ma_result result;
+static MA_INLINE void ma_pcm_deinterleave_u8__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
+{
+ ma_pcm_deinterleave_u8__reference(dst, src, frameCount, channels);
+}
- /* The first thing to do is wait for space to become available for writing. This will return an error code if the device has been stopped. */
- result = ma_device_wait_write__alsa(pDevice);
- if (result != MA_SUCCESS) {
- return result;
- }
+MA_API void ma_pcm_deinterleave_u8(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_deinterleave_u8__reference(dst, src, frameCount, channels);
+#else
+ ma_pcm_deinterleave_u8__optimized(dst, src, frameCount, channels);
+#endif
+}
- resultALSA = ((ma_snd_pcm_writei_proc)pDevice->pContext->alsa.snd_pcm_writei)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback, pFrames, frameCount);
- if (resultALSA >= 0) {
- break; /* Success. */
- } else {
- if (resultALSA == -EAGAIN) {
- /*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "TRACE: EGAIN (write)\n");*/
- continue; /* Try again. */
- } else if (resultALSA == -EPIPE) {
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "TRACE: EPIPE (write)\n");
- /* Underrun. Recover and try again. If this fails we need to return an error. */
- resultALSA = ((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback, resultALSA, MA_TRUE); /* MA_TRUE=silent (don't print anything on error). */
- if (resultALSA < 0) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to recover device after underrun.", ma_result_from_errno((int)-resultALSA));
- }
+/* s16 */
+static MA_INLINE void ma_pcm_s16_to_u8__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_uint8* dst_u8 = (ma_uint8*)dst;
+ const ma_int16* src_s16 = (const ma_int16*)src;
- /*
- In my testing I have had a situation where writei() does not automatically restart the device even though I've set it
- up as such in the software parameters. What will happen is writei() will block indefinitely even though the number of
- frames is well beyond the auto-start threshold. To work around this I've needed to add an explicit start here. Not sure
- if this is me just being stupid and not recovering the device properly, but this definitely feels like something isn't
- quite right here.
- */
- resultALSA = ((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback);
- if (resultALSA < 0) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[ALSA] Failed to start device after underrun.", ma_result_from_errno((int)-resultALSA));
- }
+ if (ditherMode == ma_dither_mode_none) {
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ ma_int16 x = src_s16[i];
+ x = (ma_int16)(x >> 8);
+ x = (ma_int16)(x + 128);
+ dst_u8[i] = (ma_uint8)x;
+ }
+ } else {
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ ma_int16 x = src_s16[i];
- continue; /* Try writing again. */
+ /* Dither. Don't overflow. */
+ ma_int32 dither = ma_dither_s32(ditherMode, -0x80, 0x7F);
+ if ((x + dither) <= 0x7FFF) {
+ x = (ma_int16)(x + dither);
+ } else {
+ x = 0x7FFF;
}
+
+ x = (ma_int16)(x >> 8);
+ x = (ma_int16)(x + 128);
+ dst_u8[i] = (ma_uint8)x;
}
}
+}
- if (pFramesWritten != NULL) {
- *pFramesWritten = resultALSA;
- }
+static MA_INLINE void ma_pcm_s16_to_u8__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s16_to_u8__reference(dst, src, count, ditherMode);
+}
+
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE void ma_pcm_s16_to_u8__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s16_to_u8__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE void ma_pcm_s16_to_u8__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s16_to_u8__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE void ma_pcm_s16_to_u8__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s16_to_u8__optimized(dst, src, count, ditherMode);
+}
+#endif
- return MA_SUCCESS;
+MA_API void ma_pcm_s16_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_s16_to_u8__reference(dst, src, count, ditherMode);
+#else
+ # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
+ if (ma_has_avx2()) {
+ ma_pcm_s16_to_u8__avx2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
+ if (ma_has_sse2()) {
+ ma_pcm_s16_to_u8__sse2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
+ if (ma_has_neon()) {
+ ma_pcm_s16_to_u8__neon(dst, src, count, ditherMode);
+ } else
+ #endif
+ {
+ ma_pcm_s16_to_u8__optimized(dst, src, count, ditherMode);
+ }
+#endif
}
-static ma_result ma_device_data_loop_wakeup__alsa(ma_device* pDevice)
+
+MA_API void ma_pcm_s16_to_s16(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- ma_uint64 t = 1;
+ (void)ditherMode;
+ ma_copy_memory_64(dst, src, count * sizeof(ma_int16));
+}
- MA_ASSERT(pDevice != NULL);
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Waking up... ");
+static MA_INLINE void ma_pcm_s16_to_s24__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_uint8* dst_s24 = (ma_uint8*)dst;
+ const ma_int16* src_s16 = (const ma_int16*)src;
- /* Write to an eventfd to trigger a wakeup from poll() and abort any reading or writing. */
- if (pDevice->alsa.pPollDescriptorsCapture != NULL) {
- write(pDevice->alsa.wakeupfdCapture, &t, sizeof(t));
- }
- if (pDevice->alsa.pPollDescriptorsPlayback != NULL) {
- write(pDevice->alsa.wakeupfdPlayback, &t, sizeof(t));
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ dst_s24[i*3+0] = 0;
+ dst_s24[i*3+1] = (ma_uint8)(src_s16[i] & 0xFF);
+ dst_s24[i*3+2] = (ma_uint8)(src_s16[i] >> 8);
}
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "Done\n");
-
- return MA_SUCCESS;
+ (void)ditherMode;
}
-static ma_result ma_context_uninit__alsa(ma_context* pContext)
+static MA_INLINE void ma_pcm_s16_to_s24__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pContext->backend == ma_backend_alsa);
-
- /* Clean up memory for memory leak checkers. */
- ((ma_snd_config_update_free_global_proc)pContext->alsa.snd_config_update_free_global)();
+ ma_pcm_s16_to_s24__reference(dst, src, count, ditherMode);
+}
-#ifndef MA_NO_RUNTIME_LINKING
- ma_dlclose(pContext, pContext->alsa.asoundSO);
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE void ma_pcm_s16_to_s24__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s16_to_s24__optimized(dst, src, count, ditherMode);
+}
#endif
-
- ma_mutex_uninit(&pContext->alsa.internalDeviceEnumLock);
-
- return MA_SUCCESS;
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE void ma_pcm_s16_to_s24__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s16_to_s24__optimized(dst, src, count, ditherMode);
}
-
-static ma_result ma_context_init__alsa(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+#endif
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE void ma_pcm_s16_to_s24__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
-#ifndef MA_NO_RUNTIME_LINKING
- const char* libasoundNames[] = {
- "libasound.so.2",
- "libasound.so"
- };
- size_t i;
+ ma_pcm_s16_to_s24__optimized(dst, src, count, ditherMode);
+}
+#endif
- for (i = 0; i < ma_countof(libasoundNames); ++i) {
- pContext->alsa.asoundSO = ma_dlopen(pContext, libasoundNames[i]);
- if (pContext->alsa.asoundSO != NULL) {
- break;
+MA_API void ma_pcm_s16_to_s24(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_s16_to_s24__reference(dst, src, count, ditherMode);
+#else
+ # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
+ if (ma_has_avx2()) {
+ ma_pcm_s16_to_s24__avx2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
+ if (ma_has_sse2()) {
+ ma_pcm_s16_to_s24__sse2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
+ if (ma_has_neon()) {
+ ma_pcm_s16_to_s24__neon(dst, src, count, ditherMode);
+ } else
+ #endif
+ {
+ ma_pcm_s16_to_s24__optimized(dst, src, count, ditherMode);
}
- }
+#endif
+}
- if (pContext->alsa.asoundSO == NULL) {
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[ALSA] Failed to open shared object.\n");
- return MA_NO_BACKEND;
- }
- pContext->alsa.snd_pcm_open = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_open");
- pContext->alsa.snd_pcm_close = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_close");
- pContext->alsa.snd_pcm_hw_params_sizeof = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_sizeof");
- pContext->alsa.snd_pcm_hw_params_any = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_any");
- pContext->alsa.snd_pcm_hw_params_set_format = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_format");
- pContext->alsa.snd_pcm_hw_params_set_format_first = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_format_first");
- pContext->alsa.snd_pcm_hw_params_get_format_mask = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_format_mask");
- pContext->alsa.snd_pcm_hw_params_set_channels = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_channels");
- pContext->alsa.snd_pcm_hw_params_set_channels_near = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_channels_near");
- pContext->alsa.snd_pcm_hw_params_set_channels_minmax = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_channels_minmax");
- pContext->alsa.snd_pcm_hw_params_set_rate_resample = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_rate_resample");
- pContext->alsa.snd_pcm_hw_params_set_rate = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_rate");
- pContext->alsa.snd_pcm_hw_params_set_rate_near = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_rate_near");
- pContext->alsa.snd_pcm_hw_params_set_buffer_size_near = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_buffer_size_near");
- pContext->alsa.snd_pcm_hw_params_set_periods_near = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_periods_near");
- pContext->alsa.snd_pcm_hw_params_set_access = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_set_access");
- pContext->alsa.snd_pcm_hw_params_get_format = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_format");
- pContext->alsa.snd_pcm_hw_params_get_channels = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_channels");
- pContext->alsa.snd_pcm_hw_params_get_channels_min = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_channels_min");
- pContext->alsa.snd_pcm_hw_params_get_channels_max = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_channels_max");
- pContext->alsa.snd_pcm_hw_params_get_rate = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_rate");
- pContext->alsa.snd_pcm_hw_params_get_rate_min = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_rate_min");
- pContext->alsa.snd_pcm_hw_params_get_rate_max = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_rate_max");
- pContext->alsa.snd_pcm_hw_params_get_buffer_size = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_buffer_size");
- pContext->alsa.snd_pcm_hw_params_get_periods = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_periods");
- pContext->alsa.snd_pcm_hw_params_get_access = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_get_access");
- pContext->alsa.snd_pcm_hw_params_test_format = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_test_format");
- pContext->alsa.snd_pcm_hw_params_test_channels = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_test_channels");
- pContext->alsa.snd_pcm_hw_params_test_rate = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params_test_rate");
- pContext->alsa.snd_pcm_hw_params = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_hw_params");
- pContext->alsa.snd_pcm_sw_params_sizeof = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_sizeof");
- pContext->alsa.snd_pcm_sw_params_current = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_current");
- pContext->alsa.snd_pcm_sw_params_get_boundary = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_get_boundary");
- pContext->alsa.snd_pcm_sw_params_set_avail_min = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_set_avail_min");
- pContext->alsa.snd_pcm_sw_params_set_start_threshold = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_set_start_threshold");
- pContext->alsa.snd_pcm_sw_params_set_stop_threshold = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params_set_stop_threshold");
- pContext->alsa.snd_pcm_sw_params = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_sw_params");
- pContext->alsa.snd_pcm_format_mask_sizeof = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_format_mask_sizeof");
- pContext->alsa.snd_pcm_format_mask_test = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_format_mask_test");
- pContext->alsa.snd_pcm_get_chmap = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_get_chmap");
- pContext->alsa.snd_pcm_state = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_state");
- pContext->alsa.snd_pcm_prepare = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_prepare");
- pContext->alsa.snd_pcm_start = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_start");
- pContext->alsa.snd_pcm_drop = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_drop");
- pContext->alsa.snd_pcm_drain = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_drain");
- pContext->alsa.snd_pcm_reset = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_reset");
- pContext->alsa.snd_device_name_hint = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_device_name_hint");
- pContext->alsa.snd_device_name_get_hint = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_device_name_get_hint");
- pContext->alsa.snd_card_get_index = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_card_get_index");
- pContext->alsa.snd_device_name_free_hint = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_device_name_free_hint");
- pContext->alsa.snd_pcm_mmap_begin = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_mmap_begin");
- pContext->alsa.snd_pcm_mmap_commit = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_mmap_commit");
- pContext->alsa.snd_pcm_recover = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_recover");
- pContext->alsa.snd_pcm_readi = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_readi");
- pContext->alsa.snd_pcm_writei = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_writei");
- pContext->alsa.snd_pcm_avail = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_avail");
- pContext->alsa.snd_pcm_avail_update = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_avail_update");
- pContext->alsa.snd_pcm_wait = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_wait");
- pContext->alsa.snd_pcm_nonblock = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_nonblock");
- pContext->alsa.snd_pcm_info = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_info");
- pContext->alsa.snd_pcm_info_sizeof = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_info_sizeof");
- pContext->alsa.snd_pcm_info_get_name = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_info_get_name");
- pContext->alsa.snd_pcm_poll_descriptors = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_poll_descriptors");
- pContext->alsa.snd_pcm_poll_descriptors_count = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_poll_descriptors_count");
- pContext->alsa.snd_pcm_poll_descriptors_revents = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_pcm_poll_descriptors_revents");
- pContext->alsa.snd_config_update_free_global = (ma_proc)ma_dlsym(pContext, pContext->alsa.asoundSO, "snd_config_update_free_global");
-#else
- /* The system below is just for type safety. */
- ma_snd_pcm_open_proc _snd_pcm_open = snd_pcm_open;
- ma_snd_pcm_close_proc _snd_pcm_close = snd_pcm_close;
- ma_snd_pcm_hw_params_sizeof_proc _snd_pcm_hw_params_sizeof = snd_pcm_hw_params_sizeof;
- ma_snd_pcm_hw_params_any_proc _snd_pcm_hw_params_any = snd_pcm_hw_params_any;
- ma_snd_pcm_hw_params_set_format_proc _snd_pcm_hw_params_set_format = snd_pcm_hw_params_set_format;
- ma_snd_pcm_hw_params_set_format_first_proc _snd_pcm_hw_params_set_format_first = snd_pcm_hw_params_set_format_first;
- ma_snd_pcm_hw_params_get_format_mask_proc _snd_pcm_hw_params_get_format_mask = snd_pcm_hw_params_get_format_mask;
- ma_snd_pcm_hw_params_set_channels_proc _snd_pcm_hw_params_set_channels = snd_pcm_hw_params_set_channels;
- ma_snd_pcm_hw_params_set_channels_near_proc _snd_pcm_hw_params_set_channels_near = snd_pcm_hw_params_set_channels_near;
- ma_snd_pcm_hw_params_set_rate_resample_proc _snd_pcm_hw_params_set_rate_resample = snd_pcm_hw_params_set_rate_resample;
- ma_snd_pcm_hw_params_set_rate_near _snd_pcm_hw_params_set_rate = snd_pcm_hw_params_set_rate;
- ma_snd_pcm_hw_params_set_rate_near_proc _snd_pcm_hw_params_set_rate_near = snd_pcm_hw_params_set_rate_near;
- ma_snd_pcm_hw_params_set_rate_minmax_proc _snd_pcm_hw_params_set_rate_minmax = snd_pcm_hw_params_set_rate_minmax;
- ma_snd_pcm_hw_params_set_buffer_size_near_proc _snd_pcm_hw_params_set_buffer_size_near = snd_pcm_hw_params_set_buffer_size_near;
- ma_snd_pcm_hw_params_set_periods_near_proc _snd_pcm_hw_params_set_periods_near = snd_pcm_hw_params_set_periods_near;
- ma_snd_pcm_hw_params_set_access_proc _snd_pcm_hw_params_set_access = snd_pcm_hw_params_set_access;
- ma_snd_pcm_hw_params_get_format_proc _snd_pcm_hw_params_get_format = snd_pcm_hw_params_get_format;
- ma_snd_pcm_hw_params_get_channels_proc _snd_pcm_hw_params_get_channels = snd_pcm_hw_params_get_channels;
- ma_snd_pcm_hw_params_get_channels_min_proc _snd_pcm_hw_params_get_channels_min = snd_pcm_hw_params_get_channels_min;
- ma_snd_pcm_hw_params_get_channels_max_proc _snd_pcm_hw_params_get_channels_max = snd_pcm_hw_params_get_channels_max;
- ma_snd_pcm_hw_params_get_rate_proc _snd_pcm_hw_params_get_rate = snd_pcm_hw_params_get_rate;
- ma_snd_pcm_hw_params_get_rate_min_proc _snd_pcm_hw_params_get_rate_min = snd_pcm_hw_params_get_rate_min;
- ma_snd_pcm_hw_params_get_rate_max_proc _snd_pcm_hw_params_get_rate_max = snd_pcm_hw_params_get_rate_max;
- ma_snd_pcm_hw_params_get_buffer_size_proc _snd_pcm_hw_params_get_buffer_size = snd_pcm_hw_params_get_buffer_size;
- ma_snd_pcm_hw_params_get_periods_proc _snd_pcm_hw_params_get_periods = snd_pcm_hw_params_get_periods;
- ma_snd_pcm_hw_params_get_access_proc _snd_pcm_hw_params_get_access = snd_pcm_hw_params_get_access;
- ma_snd_pcm_hw_params_test_format_proc _snd_pcm_hw_params_test_format = snd_pcm_hw_params_test_format;
- ma_snd_pcm_hw_params_test_channels_proc _snd_pcm_hw_params_test_channels = snd_pcm_hw_params_test_channels;
- ma_snd_pcm_hw_params_test_rate_proc _snd_pcm_hw_params_test_rate = snd_pcm_hw_params_test_rate;
- ma_snd_pcm_hw_params_proc _snd_pcm_hw_params = snd_pcm_hw_params;
- ma_snd_pcm_sw_params_sizeof_proc _snd_pcm_sw_params_sizeof = snd_pcm_sw_params_sizeof;
- ma_snd_pcm_sw_params_current_proc _snd_pcm_sw_params_current = snd_pcm_sw_params_current;
- ma_snd_pcm_sw_params_get_boundary_proc _snd_pcm_sw_params_get_boundary = snd_pcm_sw_params_get_boundary;
- ma_snd_pcm_sw_params_set_avail_min_proc _snd_pcm_sw_params_set_avail_min = snd_pcm_sw_params_set_avail_min;
- ma_snd_pcm_sw_params_set_start_threshold_proc _snd_pcm_sw_params_set_start_threshold = snd_pcm_sw_params_set_start_threshold;
- ma_snd_pcm_sw_params_set_stop_threshold_proc _snd_pcm_sw_params_set_stop_threshold = snd_pcm_sw_params_set_stop_threshold;
- ma_snd_pcm_sw_params_proc _snd_pcm_sw_params = snd_pcm_sw_params;
- ma_snd_pcm_format_mask_sizeof_proc _snd_pcm_format_mask_sizeof = snd_pcm_format_mask_sizeof;
- ma_snd_pcm_format_mask_test_proc _snd_pcm_format_mask_test = snd_pcm_format_mask_test;
- ma_snd_pcm_get_chmap_proc _snd_pcm_get_chmap = snd_pcm_get_chmap;
- ma_snd_pcm_state_proc _snd_pcm_state = snd_pcm_state;
- ma_snd_pcm_prepare_proc _snd_pcm_prepare = snd_pcm_prepare;
- ma_snd_pcm_start_proc _snd_pcm_start = snd_pcm_start;
- ma_snd_pcm_drop_proc _snd_pcm_drop = snd_pcm_drop;
- ma_snd_pcm_drain_proc _snd_pcm_drain = snd_pcm_drain;
- ma_snd_pcm_reset_proc _snd_pcm_reset = snd_pcm_reset;
- ma_snd_device_name_hint_proc _snd_device_name_hint = snd_device_name_hint;
- ma_snd_device_name_get_hint_proc _snd_device_name_get_hint = snd_device_name_get_hint;
- ma_snd_card_get_index_proc _snd_card_get_index = snd_card_get_index;
- ma_snd_device_name_free_hint_proc _snd_device_name_free_hint = snd_device_name_free_hint;
- ma_snd_pcm_mmap_begin_proc _snd_pcm_mmap_begin = snd_pcm_mmap_begin;
- ma_snd_pcm_mmap_commit_proc _snd_pcm_mmap_commit = snd_pcm_mmap_commit;
- ma_snd_pcm_recover_proc _snd_pcm_recover = snd_pcm_recover;
- ma_snd_pcm_readi_proc _snd_pcm_readi = snd_pcm_readi;
- ma_snd_pcm_writei_proc _snd_pcm_writei = snd_pcm_writei;
- ma_snd_pcm_avail_proc _snd_pcm_avail = snd_pcm_avail;
- ma_snd_pcm_avail_update_proc _snd_pcm_avail_update = snd_pcm_avail_update;
- ma_snd_pcm_wait_proc _snd_pcm_wait = snd_pcm_wait;
- ma_snd_pcm_nonblock_proc _snd_pcm_nonblock = snd_pcm_nonblock;
- ma_snd_pcm_info_proc _snd_pcm_info = snd_pcm_info;
- ma_snd_pcm_info_sizeof_proc _snd_pcm_info_sizeof = snd_pcm_info_sizeof;
- ma_snd_pcm_info_get_name_proc _snd_pcm_info_get_name = snd_pcm_info_get_name;
- ma_snd_pcm_poll_descriptors _snd_pcm_poll_descriptors = snd_pcm_poll_descriptors;
- ma_snd_pcm_poll_descriptors_count _snd_pcm_poll_descriptors_count = snd_pcm_poll_descriptors_count;
- ma_snd_pcm_poll_descriptors_revents _snd_pcm_poll_descriptors_revents = snd_pcm_poll_descriptors_revents;
- ma_snd_config_update_free_global_proc _snd_config_update_free_global = snd_config_update_free_global;
+static MA_INLINE void ma_pcm_s16_to_s32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_int32* dst_s32 = (ma_int32*)dst;
+ const ma_int16* src_s16 = (const ma_int16*)src;
- pContext->alsa.snd_pcm_open = (ma_proc)_snd_pcm_open;
- pContext->alsa.snd_pcm_close = (ma_proc)_snd_pcm_close;
- pContext->alsa.snd_pcm_hw_params_sizeof = (ma_proc)_snd_pcm_hw_params_sizeof;
- pContext->alsa.snd_pcm_hw_params_any = (ma_proc)_snd_pcm_hw_params_any;
- pContext->alsa.snd_pcm_hw_params_set_format = (ma_proc)_snd_pcm_hw_params_set_format;
- pContext->alsa.snd_pcm_hw_params_set_format_first = (ma_proc)_snd_pcm_hw_params_set_format_first;
- pContext->alsa.snd_pcm_hw_params_get_format_mask = (ma_proc)_snd_pcm_hw_params_get_format_mask;
- pContext->alsa.snd_pcm_hw_params_set_channels = (ma_proc)_snd_pcm_hw_params_set_channels;
- pContext->alsa.snd_pcm_hw_params_set_channels_near = (ma_proc)_snd_pcm_hw_params_set_channels_near;
- pContext->alsa.snd_pcm_hw_params_set_channels_minmax = (ma_proc)_snd_pcm_hw_params_set_channels_minmax;
- pContext->alsa.snd_pcm_hw_params_set_rate_resample = (ma_proc)_snd_pcm_hw_params_set_rate_resample;
- pContext->alsa.snd_pcm_hw_params_set_rate = (ma_proc)_snd_pcm_hw_params_set_rate;
- pContext->alsa.snd_pcm_hw_params_set_rate_near = (ma_proc)_snd_pcm_hw_params_set_rate_near;
- pContext->alsa.snd_pcm_hw_params_set_buffer_size_near = (ma_proc)_snd_pcm_hw_params_set_buffer_size_near;
- pContext->alsa.snd_pcm_hw_params_set_periods_near = (ma_proc)_snd_pcm_hw_params_set_periods_near;
- pContext->alsa.snd_pcm_hw_params_set_access = (ma_proc)_snd_pcm_hw_params_set_access;
- pContext->alsa.snd_pcm_hw_params_get_format = (ma_proc)_snd_pcm_hw_params_get_format;
- pContext->alsa.snd_pcm_hw_params_get_channels = (ma_proc)_snd_pcm_hw_params_get_channels;
- pContext->alsa.snd_pcm_hw_params_get_channels_min = (ma_proc)_snd_pcm_hw_params_get_channels_min;
- pContext->alsa.snd_pcm_hw_params_get_channels_max = (ma_proc)_snd_pcm_hw_params_get_channels_max;
- pContext->alsa.snd_pcm_hw_params_get_rate = (ma_proc)_snd_pcm_hw_params_get_rate;
- pContext->alsa.snd_pcm_hw_params_get_rate_min = (ma_proc)_snd_pcm_hw_params_get_rate_min;
- pContext->alsa.snd_pcm_hw_params_get_rate_max = (ma_proc)_snd_pcm_hw_params_get_rate_max;
- pContext->alsa.snd_pcm_hw_params_get_buffer_size = (ma_proc)_snd_pcm_hw_params_get_buffer_size;
- pContext->alsa.snd_pcm_hw_params_get_periods = (ma_proc)_snd_pcm_hw_params_get_periods;
- pContext->alsa.snd_pcm_hw_params_get_access = (ma_proc)_snd_pcm_hw_params_get_access;
- pContext->alsa.snd_pcm_hw_params_test_format = (ma_proc)_snd_pcm_hw_params_test_format;
- pContext->alsa.snd_pcm_hw_params_test_channels = (ma_proc)_snd_pcm_hw_params_test_channels;
- pContext->alsa.snd_pcm_hw_params_test_rate = (ma_proc)_snd_pcm_hw_params_test_rate;
- pContext->alsa.snd_pcm_hw_params = (ma_proc)_snd_pcm_hw_params;
- pContext->alsa.snd_pcm_sw_params_sizeof = (ma_proc)_snd_pcm_sw_params_sizeof;
- pContext->alsa.snd_pcm_sw_params_current = (ma_proc)_snd_pcm_sw_params_current;
- pContext->alsa.snd_pcm_sw_params_get_boundary = (ma_proc)_snd_pcm_sw_params_get_boundary;
- pContext->alsa.snd_pcm_sw_params_set_avail_min = (ma_proc)_snd_pcm_sw_params_set_avail_min;
- pContext->alsa.snd_pcm_sw_params_set_start_threshold = (ma_proc)_snd_pcm_sw_params_set_start_threshold;
- pContext->alsa.snd_pcm_sw_params_set_stop_threshold = (ma_proc)_snd_pcm_sw_params_set_stop_threshold;
- pContext->alsa.snd_pcm_sw_params = (ma_proc)_snd_pcm_sw_params;
- pContext->alsa.snd_pcm_format_mask_sizeof = (ma_proc)_snd_pcm_format_mask_sizeof;
- pContext->alsa.snd_pcm_format_mask_test = (ma_proc)_snd_pcm_format_mask_test;
- pContext->alsa.snd_pcm_get_chmap = (ma_proc)_snd_pcm_get_chmap;
- pContext->alsa.snd_pcm_state = (ma_proc)_snd_pcm_state;
- pContext->alsa.snd_pcm_prepare = (ma_proc)_snd_pcm_prepare;
- pContext->alsa.snd_pcm_start = (ma_proc)_snd_pcm_start;
- pContext->alsa.snd_pcm_drop = (ma_proc)_snd_pcm_drop;
- pContext->alsa.snd_pcm_drain = (ma_proc)_snd_pcm_drain;
- pContext->alsa.snd_pcm_reset = (ma_proc)_snd_pcm_reset;
- pContext->alsa.snd_device_name_hint = (ma_proc)_snd_device_name_hint;
- pContext->alsa.snd_device_name_get_hint = (ma_proc)_snd_device_name_get_hint;
- pContext->alsa.snd_card_get_index = (ma_proc)_snd_card_get_index;
- pContext->alsa.snd_device_name_free_hint = (ma_proc)_snd_device_name_free_hint;
- pContext->alsa.snd_pcm_mmap_begin = (ma_proc)_snd_pcm_mmap_begin;
- pContext->alsa.snd_pcm_mmap_commit = (ma_proc)_snd_pcm_mmap_commit;
- pContext->alsa.snd_pcm_recover = (ma_proc)_snd_pcm_recover;
- pContext->alsa.snd_pcm_readi = (ma_proc)_snd_pcm_readi;
- pContext->alsa.snd_pcm_writei = (ma_proc)_snd_pcm_writei;
- pContext->alsa.snd_pcm_avail = (ma_proc)_snd_pcm_avail;
- pContext->alsa.snd_pcm_avail_update = (ma_proc)_snd_pcm_avail_update;
- pContext->alsa.snd_pcm_wait = (ma_proc)_snd_pcm_wait;
- pContext->alsa.snd_pcm_nonblock = (ma_proc)_snd_pcm_nonblock;
- pContext->alsa.snd_pcm_info = (ma_proc)_snd_pcm_info;
- pContext->alsa.snd_pcm_info_sizeof = (ma_proc)_snd_pcm_info_sizeof;
- pContext->alsa.snd_pcm_info_get_name = (ma_proc)_snd_pcm_info_get_name;
- pContext->alsa.snd_pcm_poll_descriptors = (ma_proc)_snd_pcm_poll_descriptors;
- pContext->alsa.snd_pcm_poll_descriptors_count = (ma_proc)_snd_pcm_poll_descriptors_count;
- pContext->alsa.snd_pcm_poll_descriptors_revents = (ma_proc)_snd_pcm_poll_descriptors_revents;
- pContext->alsa.snd_config_update_free_global = (ma_proc)_snd_config_update_free_global;
-#endif
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ dst_s32[i] = src_s16[i] << 16;
+ }
- pContext->alsa.useVerboseDeviceEnumeration = pConfig->alsa.useVerboseDeviceEnumeration;
+ (void)ditherMode;
+}
- if (ma_mutex_init(&pContext->alsa.internalDeviceEnumLock) != MA_SUCCESS) {
- ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[ALSA] WARNING: Failed to initialize mutex for internal device enumeration.", MA_ERROR);
- }
+static MA_INLINE void ma_pcm_s16_to_s32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s16_to_s32__reference(dst, src, count, ditherMode);
+}
- pCallbacks->onContextInit = ma_context_init__alsa;
- pCallbacks->onContextUninit = ma_context_uninit__alsa;
- pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__alsa;
- pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__alsa;
- pCallbacks->onDeviceInit = ma_device_init__alsa;
- pCallbacks->onDeviceUninit = ma_device_uninit__alsa;
- pCallbacks->onDeviceStart = ma_device_start__alsa;
- pCallbacks->onDeviceStop = ma_device_stop__alsa;
- pCallbacks->onDeviceRead = ma_device_read__alsa;
- pCallbacks->onDeviceWrite = ma_device_write__alsa;
- pCallbacks->onDeviceDataLoop = NULL;
- pCallbacks->onDeviceDataLoopWakeup = ma_device_data_loop_wakeup__alsa;
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE void ma_pcm_s16_to_s32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s16_to_s32__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE void ma_pcm_s16_to_s32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s16_to_s32__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE void ma_pcm_s16_to_s32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s16_to_s32__optimized(dst, src, count, ditherMode);
+}
+#endif
- return MA_SUCCESS;
+MA_API void ma_pcm_s16_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_s16_to_s32__reference(dst, src, count, ditherMode);
+#else
+ # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
+ if (ma_has_avx2()) {
+ ma_pcm_s16_to_s32__avx2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
+ if (ma_has_sse2()) {
+ ma_pcm_s16_to_s32__sse2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
+ if (ma_has_neon()) {
+ ma_pcm_s16_to_s32__neon(dst, src, count, ditherMode);
+ } else
+ #endif
+ {
+ ma_pcm_s16_to_s32__optimized(dst, src, count, ditherMode);
+ }
+#endif
}
-#endif /* ALSA */
+static MA_INLINE void ma_pcm_s16_to_f32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ float* dst_f32 = (float*)dst;
+ const ma_int16* src_s16 = (const ma_int16*)src;
-/******************************************************************************
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ float x = (float)src_s16[i];
-PulseAudio Backend
+#if 0
+ /* The accurate way. */
+ x = x + 32768.0f; /* -32768..32767 to 0..65535 */
+ x = x * 0.00003051804379339284f; /* 0..65535 to 0..2 */
+ x = x - 1; /* 0..2 to -1..1 */
+#else
+ /* The fast way. */
+ x = x * 0.000030517578125f; /* -32768..32767 to -1..0.999969482421875 */
+#endif
-******************************************************************************/
-#ifdef MA_HAS_PULSEAUDIO
-/*
-The PulseAudio API, along with Apple's Core Audio, is the worst of the maintream audio APIs. This is a brief description of what's going on
-in the PulseAudio backend. I apologize if this gets a bit ranty for your liking - you might want to skip this discussion.
+ dst_f32[i] = x;
+ }
-PulseAudio has something they call the "Simple API", which unfortunately isn't suitable for miniaudio. I've not seen anywhere where it
-allows you to enumerate over devices, nor does it seem to support the ability to stop and start streams. Looking at the documentation, it
-appears as though the stream is constantly running and you prevent sound from being emitted or captured by simply not calling the read or
-write functions. This is not a professional solution as it would be much better to *actually* stop the underlying stream. Perhaps the
-simple API has some smarts to do this automatically, but I'm not sure. Another limitation with the simple API is that it seems inefficient
-when you want to have multiple streams to a single context. For these reasons, miniaudio is not using the simple API.
+ (void)ditherMode;
+}
-Since we're not using the simple API, we're left with the asynchronous API as our only other option. And boy, is this where it starts to
-get fun, and I don't mean that in a good way...
+static MA_INLINE void ma_pcm_s16_to_f32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s16_to_f32__reference(dst, src, count, ditherMode);
+}
-The problems start with the very name of the API - "asynchronous". Yes, this is an asynchronous oriented API which means your commands
-don't immediately take effect. You instead need to issue your commands, and then wait for them to complete. The waiting mechanism is
-enabled through the use of a "main loop". In the asychronous API you cannot get away from the main loop, and the main loop is where almost
-all of PulseAudio's problems stem from.
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE void ma_pcm_s16_to_f32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s16_to_f32__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE void ma_pcm_s16_to_f32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s16_to_f32__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE void ma_pcm_s16_to_f32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s16_to_f32__optimized(dst, src, count, ditherMode);
+}
+#endif
-When you first initialize PulseAudio you need an object referred to as "main loop". You can implement this yourself by defining your own
-vtable, but it's much easier to just use one of the built-in main loop implementations. There's two generic implementations called
-pa_mainloop and pa_threaded_mainloop, and another implementation specific to GLib called pa_glib_mainloop. We're using pa_threaded_mainloop
-because it simplifies management of the worker thread. The idea of the main loop object is pretty self explanatory - you're supposed to use
-it to implement a worker thread which runs in a loop. The main loop is where operations are actually executed.
+MA_API void ma_pcm_s16_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_s16_to_f32__reference(dst, src, count, ditherMode);
+#else
+ # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
+ if (ma_has_avx2()) {
+ ma_pcm_s16_to_f32__avx2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
+ if (ma_has_sse2()) {
+ ma_pcm_s16_to_f32__sse2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
+ if (ma_has_neon()) {
+ ma_pcm_s16_to_f32__neon(dst, src, count, ditherMode);
+ } else
+ #endif
+ {
+ ma_pcm_s16_to_f32__optimized(dst, src, count, ditherMode);
+ }
+#endif
+}
-To initialize the main loop, you just use `pa_threaded_mainloop_new()`. This is the first function you'll call. You can then get a pointer
-to the vtable with `pa_threaded_mainloop_get_api()` (the main loop vtable is called `pa_mainloop_api`). Again, you can bypass the threaded
-main loop object entirely and just implement `pa_mainloop_api` directly, but there's no need for it unless you're doing something extremely
-specialized such as if you want to integrate it into your application's existing main loop infrastructure.
-(EDIT 2021-01-26: miniaudio is no longer using `pa_threaded_mainloop` due to this issue: https://github.com/mackron/miniaudio/issues/262.
-It is now using `pa_mainloop` which turns out to be a simpler solution anyway. The rest of this rant still applies, however.)
+static MA_INLINE void ma_pcm_interleave_s16__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
+{
+ ma_int16* dst_s16 = (ma_int16*)dst;
+ const ma_int16** src_s16 = (const ma_int16**)src;
-Once you have your main loop vtable (the `pa_mainloop_api` object) you can create the PulseAudio context. This is very similar to
-miniaudio's context and they map to each other quite well. You have one context to many streams, which is basically the same as miniaudio's
-one `ma_context` to many `ma_device`s. Here's where it starts to get annoying, however. When you first create the PulseAudio context, which
-is done with `pa_context_new()`, it's not actually connected to anything. When you connect, you call `pa_context_connect()`. However, if
-you remember, PulseAudio is an asynchronous API. That means you cannot just assume the context is connected after `pa_context_context()`
-has returned. You instead need to wait for it to connect. To do this, you need to either wait for a callback to get fired, which you can
-set with `pa_context_set_state_callback()`, or you can continuously poll the context's state. Either way, you need to run this in a loop.
-All objects from here out are created from the context, and, I believe, you can't be creating these objects until the context is connected.
-This waiting loop is therefore unavoidable. In order for the waiting to ever complete, however, the main loop needs to be running. Before
-attempting to connect the context, the main loop needs to be started with `pa_threaded_mainloop_start()`.
+ ma_uint64 iFrame;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ dst_s16[iFrame*channels + iChannel] = src_s16[iChannel][iFrame];
+ }
+ }
+}
+
+static MA_INLINE void ma_pcm_interleave_s16__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
+{
+ ma_pcm_interleave_s16__reference(dst, src, frameCount, channels);
+}
-The reason for this asynchronous design is to support cases where you're connecting to a remote server, say through a local network or an
-internet connection. However, the *VAST* majority of cases don't involve this at all - they just connect to a local "server" running on the
-host machine. The fact that this would be the default rather than making `pa_context_connect()` synchronous tends to boggle the mind.
+MA_API void ma_pcm_interleave_s16(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_interleave_s16__reference(dst, src, frameCount, channels);
+#else
+ ma_pcm_interleave_s16__optimized(dst, src, frameCount, channels);
+#endif
+}
-Once the context has been created and connected you can start creating a stream. A PulseAudio stream is analogous to miniaudio's device.
-The initialization of a stream is fairly standard - you configure some attributes (analogous to miniaudio's device config) and then call
-`pa_stream_new()` to actually create it. Here is where we start to get into "operations". When configuring the stream, you can get
-information about the source (such as sample format, sample rate, etc.), however it's not synchronous. Instead, a `pa_operation` object
-is returned from `pa_context_get_source_info_by_name()` (capture) or `pa_context_get_sink_info_by_name()` (playback). Then, you need to
-run a loop (again!) to wait for the operation to complete which you can determine via a callback or polling, just like we did with the
-context. Then, as an added bonus, you need to decrement the reference counter of the `pa_operation` object to ensure memory is cleaned up.
-All of that just to retrieve basic information about a device!
-Once the basic information about the device has been retrieved, miniaudio can now create the stream with `ma_stream_new()`. Like the
-context, this needs to be connected. But we need to be careful here, because we're now about to introduce one of the most horrific design
-choices in PulseAudio.
+static MA_INLINE void ma_pcm_deinterleave_s16__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
+{
+ ma_int16** dst_s16 = (ma_int16**)dst;
+ const ma_int16* src_s16 = (const ma_int16*)src;
-PulseAudio allows you to specify a callback that is fired when data can be written to or read from a stream. The language is important here
-because PulseAudio takes it literally, specifically the "can be". You would think these callbacks would be appropriate as the place for
-writing and reading data to and from the stream, and that would be right, except when it's not. When you initialize the stream, you can
-set a flag that tells PulseAudio to not start the stream automatically. This is required because miniaudio does not auto-start devices
-straight after initialization - you need to call `ma_device_start()` manually. The problem is that even when this flag is specified,
-PulseAudio will immediately fire it's write or read callback. This is *technically* correct (based on the wording in the documentation)
-because indeed, data *can* be written at this point. The problem is that it's not *practical*. It makes sense that the write/read callback
-would be where a program will want to write or read data to or from the stream, but when it's called before the application has even
-requested that the stream be started, it's just not practical because the program probably isn't ready for any kind of data delivery at
-that point (it may still need to load files or whatnot). Instead, this callback should only be fired when the application requests the
-stream be started which is how it works with literally *every* other callback-based audio API. Since miniaudio forbids firing of the data
-callback until the device has been started (as it should be with *all* callback based APIs), logic needs to be added to ensure miniaudio
-doesn't just blindly fire the application-defined data callback from within the PulseAudio callback before the stream has actually been
-started. The device state is used for this - if the state is anything other than `MA_STATE_STARTING` or `MA_STATE_STARTED`, the main data
-callback is not fired.
+ ma_uint64 iFrame;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ dst_s16[iChannel][iFrame] = src_s16[iFrame*channels + iChannel];
+ }
+ }
+}
-This, unfortunately, is not the end of the problems with the PulseAudio write callback. Any normal callback based audio API will
-continuously fire the callback at regular intervals based on the size of the internal buffer. This will only ever be fired when the device
-is running, and will be fired regardless of whether or not the user actually wrote anything to the device/stream. This not the case in
-PulseAudio. In PulseAudio, the data callback will *only* be called if you wrote something to it previously. That means, if you don't call
-`pa_stream_write()`, the callback will not get fired. On the surface you wouldn't think this would matter because you should be always
-writing data, and if you don't have anything to write, just write silence. That's fine until you want to drain the stream. You see, if
-you're continuously writing data to the stream, the stream will never get drained! That means in order to drain the stream, you need to
-*not* write data to it! But remember, when you don't write data to the stream, the callback won't get fired again! Why is draining
-important? Because that's how we've defined stopping to work in miniaudio. In miniaudio, stopping the device requires it to be drained
-before returning from ma_device_stop(). So we've stopped the device, which requires us to drain, but draining requires us to *not* write
-data to the stream (or else it won't ever complete draining), but not writing to the stream means the callback won't get fired again!
+static MA_INLINE void ma_pcm_deinterleave_s16__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
+{
+ ma_pcm_deinterleave_s16__reference(dst, src, frameCount, channels);
+}
-This becomes a problem when stopping and then restarting the device. When the device is stopped, it's drained, which requires us to *not*
-write anything to the stream. But then, since we didn't write anything to it, the write callback will *never* get called again if we just
-resume the stream naively. This means that starting the stream requires us to write data to the stream from outside the callback. This
-disconnect is something PulseAudio has got seriously wrong - there should only ever be a single source of data delivery, that being the
-callback. (I have tried using `pa_stream_flush()` to trigger the write callback to fire, but this just doesn't work for some reason.)
+MA_API void ma_pcm_deinterleave_s16(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_deinterleave_s16__reference(dst, src, frameCount, channels);
+#else
+ ma_pcm_deinterleave_s16__optimized(dst, src, frameCount, channels);
+#endif
+}
-Once you've created the stream, you need to connect it which involves the whole waiting procedure. This is the same process as the context,
-only this time you'll poll for the state with `pa_stream_get_status()`. The starting and stopping of a streaming is referred to as
-"corking" in PulseAudio. The analogy is corking a barrel. To start the stream, you uncork it, to stop it you cork it. Personally I think
-it's silly - why would you not just call it "starting" and "stopping" like any other normal audio API? Anyway, the act of corking is, you
-guessed it, asynchronous. This means you'll need our waiting loop as usual. Again, why this asynchronous design is the default is
-absolutely beyond me. Would it really be that hard to just make it run synchronously?
-Teardown is pretty simple (what?!). It's just a matter of calling the relevant `_unref()` function on each object in reverse order that
-they were initialized in.
+/* s24 */
+static MA_INLINE void ma_pcm_s24_to_u8__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_uint8* dst_u8 = (ma_uint8*)dst;
+ const ma_uint8* src_s24 = (const ma_uint8*)src;
-That's about it from the PulseAudio side. A bit ranty, I know, but they really need to fix that main loop and callback system. They're
-embarrassingly unpractical. The main loop thing is an easy fix - have synchronous versions of all APIs. If an application wants these to
-run asynchronously, they can execute them in a separate thread themselves. The desire to run these asynchronously is such a niche
-requirement - it makes no sense to make it the default. The stream write callback needs to be change, or an alternative provided, that is
-constantly fired, regardless of whether or not `pa_stream_write()` has been called, and it needs to take a pointer to a buffer as a
-parameter which the program just writes to directly rather than having to call `pa_stream_writable_size()` and `pa_stream_write()`. These
-changes alone will change PulseAudio from one of the worst audio APIs to one of the best.
-*/
+ if (ditherMode == ma_dither_mode_none) {
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ dst_u8[i] = (ma_uint8)((ma_int8)src_s24[i*3 + 2] + 128);
+ }
+ } else {
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ ma_int32 x = (ma_int32)(((ma_uint32)(src_s24[i*3+0]) << 8) | ((ma_uint32)(src_s24[i*3+1]) << 16) | ((ma_uint32)(src_s24[i*3+2])) << 24);
+ /* Dither. Don't overflow. */
+ ma_int32 dither = ma_dither_s32(ditherMode, -0x800000, 0x7FFFFF);
+ if ((ma_int64)x + dither <= 0x7FFFFFFF) {
+ x = x + dither;
+ } else {
+ x = 0x7FFFFFFF;
+ }
-/*
-It is assumed pulseaudio.h is available when linking at compile time. When linking at compile time, we use the declarations in the header
-to check for type safety. We cannot do this when linking at run time because the header might not be available.
-*/
-#ifdef MA_NO_RUNTIME_LINKING
+ x = x >> 24;
+ x = x + 128;
+ dst_u8[i] = (ma_uint8)x;
+ }
+ }
+}
-/* pulseaudio.h marks some functions with "inline" which isn't always supported. Need to emulate it. */
-#if !defined(__cplusplus)
- #if defined(__STRICT_ANSI__)
- #if !defined(inline)
- #define inline __inline__ __attribute__((always_inline))
- #define MA_INLINE_DEFINED
- #endif
- #endif
+static MA_INLINE void ma_pcm_s24_to_u8__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s24_to_u8__reference(dst, src, count, ditherMode);
+}
+
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE void ma_pcm_s24_to_u8__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s24_to_u8__optimized(dst, src, count, ditherMode);
+}
#endif
-#include
-#if defined(MA_INLINE_DEFINED)
- #undef inline
- #undef MA_INLINE_DEFINED
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE void ma_pcm_s24_to_u8__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s24_to_u8__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE void ma_pcm_s24_to_u8__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s24_to_u8__optimized(dst, src, count, ditherMode);
+}
#endif
-#define MA_PA_OK PA_OK
-#define MA_PA_ERR_ACCESS PA_ERR_ACCESS
-#define MA_PA_ERR_INVALID PA_ERR_INVALID
-#define MA_PA_ERR_NOENTITY PA_ERR_NOENTITY
-#define MA_PA_ERR_NOTSUPPORTED PA_ERR_NOTSUPPORTED
-
-#define MA_PA_CHANNELS_MAX PA_CHANNELS_MAX
-#define MA_PA_RATE_MAX PA_RATE_MAX
-
-typedef pa_context_flags_t ma_pa_context_flags_t;
-#define MA_PA_CONTEXT_NOFLAGS PA_CONTEXT_NOFLAGS
-#define MA_PA_CONTEXT_NOAUTOSPAWN PA_CONTEXT_NOAUTOSPAWN
-#define MA_PA_CONTEXT_NOFAIL PA_CONTEXT_NOFAIL
+MA_API void ma_pcm_s24_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_s24_to_u8__reference(dst, src, count, ditherMode);
+#else
+ # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
+ if (ma_has_avx2()) {
+ ma_pcm_s24_to_u8__avx2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
+ if (ma_has_sse2()) {
+ ma_pcm_s24_to_u8__sse2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
+ if (ma_has_neon()) {
+ ma_pcm_s24_to_u8__neon(dst, src, count, ditherMode);
+ } else
+ #endif
+ {
+ ma_pcm_s24_to_u8__optimized(dst, src, count, ditherMode);
+ }
+#endif
+}
-typedef pa_stream_flags_t ma_pa_stream_flags_t;
-#define MA_PA_STREAM_NOFLAGS PA_STREAM_NOFLAGS
-#define MA_PA_STREAM_START_CORKED PA_STREAM_START_CORKED
-#define MA_PA_STREAM_INTERPOLATE_TIMING PA_STREAM_INTERPOLATE_TIMING
-#define MA_PA_STREAM_NOT_MONOTONIC PA_STREAM_NOT_MONOTONIC
-#define MA_PA_STREAM_AUTO_TIMING_UPDATE PA_STREAM_AUTO_TIMING_UPDATE
-#define MA_PA_STREAM_NO_REMAP_CHANNELS PA_STREAM_NO_REMAP_CHANNELS
-#define MA_PA_STREAM_NO_REMIX_CHANNELS PA_STREAM_NO_REMIX_CHANNELS
-#define MA_PA_STREAM_FIX_FORMAT PA_STREAM_FIX_FORMAT
-#define MA_PA_STREAM_FIX_RATE PA_STREAM_FIX_RATE
-#define MA_PA_STREAM_FIX_CHANNELS PA_STREAM_FIX_CHANNELS
-#define MA_PA_STREAM_DONT_MOVE PA_STREAM_DONT_MOVE
-#define MA_PA_STREAM_VARIABLE_RATE PA_STREAM_VARIABLE_RATE
-#define MA_PA_STREAM_PEAK_DETECT PA_STREAM_PEAK_DETECT
-#define MA_PA_STREAM_START_MUTED PA_STREAM_START_MUTED
-#define MA_PA_STREAM_ADJUST_LATENCY PA_STREAM_ADJUST_LATENCY
-#define MA_PA_STREAM_EARLY_REQUESTS PA_STREAM_EARLY_REQUESTS
-#define MA_PA_STREAM_DONT_INHIBIT_AUTO_SUSPEND PA_STREAM_DONT_INHIBIT_AUTO_SUSPEND
-#define MA_PA_STREAM_START_UNMUTED PA_STREAM_START_UNMUTED
-#define MA_PA_STREAM_FAIL_ON_SUSPEND PA_STREAM_FAIL_ON_SUSPEND
-#define MA_PA_STREAM_RELATIVE_VOLUME PA_STREAM_RELATIVE_VOLUME
-#define MA_PA_STREAM_PASSTHROUGH PA_STREAM_PASSTHROUGH
-typedef pa_sink_flags_t ma_pa_sink_flags_t;
-#define MA_PA_SINK_NOFLAGS PA_SINK_NOFLAGS
-#define MA_PA_SINK_HW_VOLUME_CTRL PA_SINK_HW_VOLUME_CTRL
-#define MA_PA_SINK_LATENCY PA_SINK_LATENCY
-#define MA_PA_SINK_HARDWARE PA_SINK_HARDWARE
-#define MA_PA_SINK_NETWORK PA_SINK_NETWORK
-#define MA_PA_SINK_HW_MUTE_CTRL PA_SINK_HW_MUTE_CTRL
-#define MA_PA_SINK_DECIBEL_VOLUME PA_SINK_DECIBEL_VOLUME
-#define MA_PA_SINK_FLAT_VOLUME PA_SINK_FLAT_VOLUME
-#define MA_PA_SINK_DYNAMIC_LATENCY PA_SINK_DYNAMIC_LATENCY
-#define MA_PA_SINK_SET_FORMATS PA_SINK_SET_FORMATS
+static MA_INLINE void ma_pcm_s24_to_s16__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_int16* dst_s16 = (ma_int16*)dst;
+ const ma_uint8* src_s24 = (const ma_uint8*)src;
-typedef pa_source_flags_t ma_pa_source_flags_t;
-#define MA_PA_SOURCE_NOFLAGS PA_SOURCE_NOFLAGS
-#define MA_PA_SOURCE_HW_VOLUME_CTRL PA_SOURCE_HW_VOLUME_CTRL
-#define MA_PA_SOURCE_LATENCY PA_SOURCE_LATENCY
-#define MA_PA_SOURCE_HARDWARE PA_SOURCE_HARDWARE
-#define MA_PA_SOURCE_NETWORK PA_SOURCE_NETWORK
-#define MA_PA_SOURCE_HW_MUTE_CTRL PA_SOURCE_HW_MUTE_CTRL
-#define MA_PA_SOURCE_DECIBEL_VOLUME PA_SOURCE_DECIBEL_VOLUME
-#define MA_PA_SOURCE_DYNAMIC_LATENCY PA_SOURCE_DYNAMIC_LATENCY
-#define MA_PA_SOURCE_FLAT_VOLUME PA_SOURCE_FLAT_VOLUME
+ if (ditherMode == ma_dither_mode_none) {
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ ma_uint16 dst_lo = ((ma_uint16)src_s24[i*3 + 1]);
+ ma_uint16 dst_hi = (ma_uint16)((ma_uint16)src_s24[i*3 + 2] << 8);
+ dst_s16[i] = (ma_int16)(dst_lo | dst_hi);
+ }
+ } else {
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ ma_int32 x = (ma_int32)(((ma_uint32)(src_s24[i*3+0]) << 8) | ((ma_uint32)(src_s24[i*3+1]) << 16) | ((ma_uint32)(src_s24[i*3+2])) << 24);
-typedef pa_context_state_t ma_pa_context_state_t;
-#define MA_PA_CONTEXT_UNCONNECTED PA_CONTEXT_UNCONNECTED
-#define MA_PA_CONTEXT_CONNECTING PA_CONTEXT_CONNECTING
-#define MA_PA_CONTEXT_AUTHORIZING PA_CONTEXT_AUTHORIZING
-#define MA_PA_CONTEXT_SETTING_NAME PA_CONTEXT_SETTING_NAME
-#define MA_PA_CONTEXT_READY PA_CONTEXT_READY
-#define MA_PA_CONTEXT_FAILED PA_CONTEXT_FAILED
-#define MA_PA_CONTEXT_TERMINATED PA_CONTEXT_TERMINATED
+ /* Dither. Don't overflow. */
+ ma_int32 dither = ma_dither_s32(ditherMode, -0x8000, 0x7FFF);
+ if ((ma_int64)x + dither <= 0x7FFFFFFF) {
+ x = x + dither;
+ } else {
+ x = 0x7FFFFFFF;
+ }
-typedef pa_stream_state_t ma_pa_stream_state_t;
-#define MA_PA_STREAM_UNCONNECTED PA_STREAM_UNCONNECTED
-#define MA_PA_STREAM_CREATING PA_STREAM_CREATING
-#define MA_PA_STREAM_READY PA_STREAM_READY
-#define MA_PA_STREAM_FAILED PA_STREAM_FAILED
-#define MA_PA_STREAM_TERMINATED PA_STREAM_TERMINATED
+ x = x >> 16;
+ dst_s16[i] = (ma_int16)x;
+ }
+ }
+}
-typedef pa_operation_state_t ma_pa_operation_state_t;
-#define MA_PA_OPERATION_RUNNING PA_OPERATION_RUNNING
-#define MA_PA_OPERATION_DONE PA_OPERATION_DONE
-#define MA_PA_OPERATION_CANCELLED PA_OPERATION_CANCELLED
+static MA_INLINE void ma_pcm_s24_to_s16__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s24_to_s16__reference(dst, src, count, ditherMode);
+}
-typedef pa_sink_state_t ma_pa_sink_state_t;
-#define MA_PA_SINK_INVALID_STATE PA_SINK_INVALID_STATE
-#define MA_PA_SINK_RUNNING PA_SINK_RUNNING
-#define MA_PA_SINK_IDLE PA_SINK_IDLE
-#define MA_PA_SINK_SUSPENDED PA_SINK_SUSPENDED
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE void ma_pcm_s24_to_s16__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s24_to_s16__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE void ma_pcm_s24_to_s16__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s24_to_s16__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE void ma_pcm_s24_to_s16__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s24_to_s16__optimized(dst, src, count, ditherMode);
+}
+#endif
-typedef pa_source_state_t ma_pa_source_state_t;
-#define MA_PA_SOURCE_INVALID_STATE PA_SOURCE_INVALID_STATE
-#define MA_PA_SOURCE_RUNNING PA_SOURCE_RUNNING
-#define MA_PA_SOURCE_IDLE PA_SOURCE_IDLE
-#define MA_PA_SOURCE_SUSPENDED PA_SOURCE_SUSPENDED
+MA_API void ma_pcm_s24_to_s16(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_s24_to_s16__reference(dst, src, count, ditherMode);
+#else
+ # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
+ if (ma_has_avx2()) {
+ ma_pcm_s24_to_s16__avx2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
+ if (ma_has_sse2()) {
+ ma_pcm_s24_to_s16__sse2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
+ if (ma_has_neon()) {
+ ma_pcm_s24_to_s16__neon(dst, src, count, ditherMode);
+ } else
+ #endif
+ {
+ ma_pcm_s24_to_s16__optimized(dst, src, count, ditherMode);
+ }
+#endif
+}
-typedef pa_seek_mode_t ma_pa_seek_mode_t;
-#define MA_PA_SEEK_RELATIVE PA_SEEK_RELATIVE
-#define MA_PA_SEEK_ABSOLUTE PA_SEEK_ABSOLUTE
-#define MA_PA_SEEK_RELATIVE_ON_READ PA_SEEK_RELATIVE_ON_READ
-#define MA_PA_SEEK_RELATIVE_END PA_SEEK_RELATIVE_END
-typedef pa_channel_position_t ma_pa_channel_position_t;
-#define MA_PA_CHANNEL_POSITION_INVALID PA_CHANNEL_POSITION_INVALID
-#define MA_PA_CHANNEL_POSITION_MONO PA_CHANNEL_POSITION_MONO
-#define MA_PA_CHANNEL_POSITION_FRONT_LEFT PA_CHANNEL_POSITION_FRONT_LEFT
-#define MA_PA_CHANNEL_POSITION_FRONT_RIGHT PA_CHANNEL_POSITION_FRONT_RIGHT
-#define MA_PA_CHANNEL_POSITION_FRONT_CENTER PA_CHANNEL_POSITION_FRONT_CENTER
-#define MA_PA_CHANNEL_POSITION_REAR_CENTER PA_CHANNEL_POSITION_REAR_CENTER
-#define MA_PA_CHANNEL_POSITION_REAR_LEFT PA_CHANNEL_POSITION_REAR_LEFT
-#define MA_PA_CHANNEL_POSITION_REAR_RIGHT PA_CHANNEL_POSITION_REAR_RIGHT
-#define MA_PA_CHANNEL_POSITION_LFE PA_CHANNEL_POSITION_LFE
-#define MA_PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER
-#define MA_PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER
-#define MA_PA_CHANNEL_POSITION_SIDE_LEFT PA_CHANNEL_POSITION_SIDE_LEFT
-#define MA_PA_CHANNEL_POSITION_SIDE_RIGHT PA_CHANNEL_POSITION_SIDE_RIGHT
-#define MA_PA_CHANNEL_POSITION_AUX0 PA_CHANNEL_POSITION_AUX0
-#define MA_PA_CHANNEL_POSITION_AUX1 PA_CHANNEL_POSITION_AUX1
-#define MA_PA_CHANNEL_POSITION_AUX2 PA_CHANNEL_POSITION_AUX2
-#define MA_PA_CHANNEL_POSITION_AUX3 PA_CHANNEL_POSITION_AUX3
-#define MA_PA_CHANNEL_POSITION_AUX4 PA_CHANNEL_POSITION_AUX4
-#define MA_PA_CHANNEL_POSITION_AUX5 PA_CHANNEL_POSITION_AUX5
-#define MA_PA_CHANNEL_POSITION_AUX6 PA_CHANNEL_POSITION_AUX6
-#define MA_PA_CHANNEL_POSITION_AUX7 PA_CHANNEL_POSITION_AUX7
-#define MA_PA_CHANNEL_POSITION_AUX8 PA_CHANNEL_POSITION_AUX8
-#define MA_PA_CHANNEL_POSITION_AUX9 PA_CHANNEL_POSITION_AUX9
-#define MA_PA_CHANNEL_POSITION_AUX10 PA_CHANNEL_POSITION_AUX10
-#define MA_PA_CHANNEL_POSITION_AUX11 PA_CHANNEL_POSITION_AUX11
-#define MA_PA_CHANNEL_POSITION_AUX12 PA_CHANNEL_POSITION_AUX12
-#define MA_PA_CHANNEL_POSITION_AUX13 PA_CHANNEL_POSITION_AUX13
-#define MA_PA_CHANNEL_POSITION_AUX14 PA_CHANNEL_POSITION_AUX14
-#define MA_PA_CHANNEL_POSITION_AUX15 PA_CHANNEL_POSITION_AUX15
-#define MA_PA_CHANNEL_POSITION_AUX16 PA_CHANNEL_POSITION_AUX16
-#define MA_PA_CHANNEL_POSITION_AUX17 PA_CHANNEL_POSITION_AUX17
-#define MA_PA_CHANNEL_POSITION_AUX18 PA_CHANNEL_POSITION_AUX18
-#define MA_PA_CHANNEL_POSITION_AUX19 PA_CHANNEL_POSITION_AUX19
-#define MA_PA_CHANNEL_POSITION_AUX20 PA_CHANNEL_POSITION_AUX20
-#define MA_PA_CHANNEL_POSITION_AUX21 PA_CHANNEL_POSITION_AUX21
-#define MA_PA_CHANNEL_POSITION_AUX22 PA_CHANNEL_POSITION_AUX22
-#define MA_PA_CHANNEL_POSITION_AUX23 PA_CHANNEL_POSITION_AUX23
-#define MA_PA_CHANNEL_POSITION_AUX24 PA_CHANNEL_POSITION_AUX24
-#define MA_PA_CHANNEL_POSITION_AUX25 PA_CHANNEL_POSITION_AUX25
-#define MA_PA_CHANNEL_POSITION_AUX26 PA_CHANNEL_POSITION_AUX26
-#define MA_PA_CHANNEL_POSITION_AUX27 PA_CHANNEL_POSITION_AUX27
-#define MA_PA_CHANNEL_POSITION_AUX28 PA_CHANNEL_POSITION_AUX28
-#define MA_PA_CHANNEL_POSITION_AUX29 PA_CHANNEL_POSITION_AUX29
-#define MA_PA_CHANNEL_POSITION_AUX30 PA_CHANNEL_POSITION_AUX30
-#define MA_PA_CHANNEL_POSITION_AUX31 PA_CHANNEL_POSITION_AUX31
-#define MA_PA_CHANNEL_POSITION_TOP_CENTER PA_CHANNEL_POSITION_TOP_CENTER
-#define MA_PA_CHANNEL_POSITION_TOP_FRONT_LEFT PA_CHANNEL_POSITION_TOP_FRONT_LEFT
-#define MA_PA_CHANNEL_POSITION_TOP_FRONT_RIGHT PA_CHANNEL_POSITION_TOP_FRONT_RIGHT
-#define MA_PA_CHANNEL_POSITION_TOP_FRONT_CENTER PA_CHANNEL_POSITION_TOP_FRONT_CENTER
-#define MA_PA_CHANNEL_POSITION_TOP_REAR_LEFT PA_CHANNEL_POSITION_TOP_REAR_LEFT
-#define MA_PA_CHANNEL_POSITION_TOP_REAR_RIGHT PA_CHANNEL_POSITION_TOP_REAR_RIGHT
-#define MA_PA_CHANNEL_POSITION_TOP_REAR_CENTER PA_CHANNEL_POSITION_TOP_REAR_CENTER
-#define MA_PA_CHANNEL_POSITION_LEFT PA_CHANNEL_POSITION_LEFT
-#define MA_PA_CHANNEL_POSITION_RIGHT PA_CHANNEL_POSITION_RIGHT
-#define MA_PA_CHANNEL_POSITION_CENTER PA_CHANNEL_POSITION_CENTER
-#define MA_PA_CHANNEL_POSITION_SUBWOOFER PA_CHANNEL_POSITION_SUBWOOFER
+MA_API void ma_pcm_s24_to_s24(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ (void)ditherMode;
-typedef pa_channel_map_def_t ma_pa_channel_map_def_t;
-#define MA_PA_CHANNEL_MAP_AIFF PA_CHANNEL_MAP_AIFF
-#define MA_PA_CHANNEL_MAP_ALSA PA_CHANNEL_MAP_ALSA
-#define MA_PA_CHANNEL_MAP_AUX PA_CHANNEL_MAP_AUX
-#define MA_PA_CHANNEL_MAP_WAVEEX PA_CHANNEL_MAP_WAVEEX
-#define MA_PA_CHANNEL_MAP_OSS PA_CHANNEL_MAP_OSS
-#define MA_PA_CHANNEL_MAP_DEFAULT PA_CHANNEL_MAP_DEFAULT
+ ma_copy_memory_64(dst, src, count * 3);
+}
-typedef pa_sample_format_t ma_pa_sample_format_t;
-#define MA_PA_SAMPLE_INVALID PA_SAMPLE_INVALID
-#define MA_PA_SAMPLE_U8 PA_SAMPLE_U8
-#define MA_PA_SAMPLE_ALAW PA_SAMPLE_ALAW
-#define MA_PA_SAMPLE_ULAW PA_SAMPLE_ULAW
-#define MA_PA_SAMPLE_S16LE PA_SAMPLE_S16LE
-#define MA_PA_SAMPLE_S16BE PA_SAMPLE_S16BE
-#define MA_PA_SAMPLE_FLOAT32LE PA_SAMPLE_FLOAT32LE
-#define MA_PA_SAMPLE_FLOAT32BE PA_SAMPLE_FLOAT32BE
-#define MA_PA_SAMPLE_S32LE PA_SAMPLE_S32LE
-#define MA_PA_SAMPLE_S32BE PA_SAMPLE_S32BE
-#define MA_PA_SAMPLE_S24LE PA_SAMPLE_S24LE
-#define MA_PA_SAMPLE_S24BE PA_SAMPLE_S24BE
-#define MA_PA_SAMPLE_S24_32LE PA_SAMPLE_S24_32LE
-#define MA_PA_SAMPLE_S24_32BE PA_SAMPLE_S24_32BE
-typedef pa_mainloop ma_pa_mainloop;
-typedef pa_threaded_mainloop ma_pa_threaded_mainloop;
-typedef pa_mainloop_api ma_pa_mainloop_api;
-typedef pa_context ma_pa_context;
-typedef pa_operation ma_pa_operation;
-typedef pa_stream ma_pa_stream;
-typedef pa_spawn_api ma_pa_spawn_api;
-typedef pa_buffer_attr ma_pa_buffer_attr;
-typedef pa_channel_map ma_pa_channel_map;
-typedef pa_cvolume ma_pa_cvolume;
-typedef pa_sample_spec ma_pa_sample_spec;
-typedef pa_sink_info ma_pa_sink_info;
-typedef pa_source_info ma_pa_source_info;
+static MA_INLINE void ma_pcm_s24_to_s32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_int32* dst_s32 = (ma_int32*)dst;
+ const ma_uint8* src_s24 = (const ma_uint8*)src;
-typedef pa_context_notify_cb_t ma_pa_context_notify_cb_t;
-typedef pa_sink_info_cb_t ma_pa_sink_info_cb_t;
-typedef pa_source_info_cb_t ma_pa_source_info_cb_t;
-typedef pa_stream_success_cb_t ma_pa_stream_success_cb_t;
-typedef pa_stream_request_cb_t ma_pa_stream_request_cb_t;
-typedef pa_stream_notify_cb_t ma_pa_stream_notify_cb_t;
-typedef pa_free_cb_t ma_pa_free_cb_t;
-#else
-#define MA_PA_OK 0
-#define MA_PA_ERR_ACCESS 1
-#define MA_PA_ERR_INVALID 2
-#define MA_PA_ERR_NOENTITY 5
-#define MA_PA_ERR_NOTSUPPORTED 19
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ dst_s32[i] = (ma_int32)(((ma_uint32)(src_s24[i*3+0]) << 8) | ((ma_uint32)(src_s24[i*3+1]) << 16) | ((ma_uint32)(src_s24[i*3+2])) << 24);
+ }
-#define MA_PA_CHANNELS_MAX 32
-#define MA_PA_RATE_MAX 384000
+ (void)ditherMode;
+}
-typedef int ma_pa_context_flags_t;
-#define MA_PA_CONTEXT_NOFLAGS 0x00000000
-#define MA_PA_CONTEXT_NOAUTOSPAWN 0x00000001
-#define MA_PA_CONTEXT_NOFAIL 0x00000002
+static MA_INLINE void ma_pcm_s24_to_s32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s24_to_s32__reference(dst, src, count, ditherMode);
+}
-typedef int ma_pa_stream_flags_t;
-#define MA_PA_STREAM_NOFLAGS 0x00000000
-#define MA_PA_STREAM_START_CORKED 0x00000001
-#define MA_PA_STREAM_INTERPOLATE_TIMING 0x00000002
-#define MA_PA_STREAM_NOT_MONOTONIC 0x00000004
-#define MA_PA_STREAM_AUTO_TIMING_UPDATE 0x00000008
-#define MA_PA_STREAM_NO_REMAP_CHANNELS 0x00000010
-#define MA_PA_STREAM_NO_REMIX_CHANNELS 0x00000020
-#define MA_PA_STREAM_FIX_FORMAT 0x00000040
-#define MA_PA_STREAM_FIX_RATE 0x00000080
-#define MA_PA_STREAM_FIX_CHANNELS 0x00000100
-#define MA_PA_STREAM_DONT_MOVE 0x00000200
-#define MA_PA_STREAM_VARIABLE_RATE 0x00000400
-#define MA_PA_STREAM_PEAK_DETECT 0x00000800
-#define MA_PA_STREAM_START_MUTED 0x00001000
-#define MA_PA_STREAM_ADJUST_LATENCY 0x00002000
-#define MA_PA_STREAM_EARLY_REQUESTS 0x00004000
-#define MA_PA_STREAM_DONT_INHIBIT_AUTO_SUSPEND 0x00008000
-#define MA_PA_STREAM_START_UNMUTED 0x00010000
-#define MA_PA_STREAM_FAIL_ON_SUSPEND 0x00020000
-#define MA_PA_STREAM_RELATIVE_VOLUME 0x00040000
-#define MA_PA_STREAM_PASSTHROUGH 0x00080000
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE void ma_pcm_s24_to_s32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s24_to_s32__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE void ma_pcm_s24_to_s32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s24_to_s32__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE void ma_pcm_s24_to_s32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s24_to_s32__optimized(dst, src, count, ditherMode);
+}
+#endif
-typedef int ma_pa_sink_flags_t;
-#define MA_PA_SINK_NOFLAGS 0x00000000
-#define MA_PA_SINK_HW_VOLUME_CTRL 0x00000001
-#define MA_PA_SINK_LATENCY 0x00000002
-#define MA_PA_SINK_HARDWARE 0x00000004
-#define MA_PA_SINK_NETWORK 0x00000008
-#define MA_PA_SINK_HW_MUTE_CTRL 0x00000010
-#define MA_PA_SINK_DECIBEL_VOLUME 0x00000020
-#define MA_PA_SINK_FLAT_VOLUME 0x00000040
-#define MA_PA_SINK_DYNAMIC_LATENCY 0x00000080
-#define MA_PA_SINK_SET_FORMATS 0x00000100
+MA_API void ma_pcm_s24_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_s24_to_s32__reference(dst, src, count, ditherMode);
+#else
+ # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
+ if (ma_has_avx2()) {
+ ma_pcm_s24_to_s32__avx2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
+ if (ma_has_sse2()) {
+ ma_pcm_s24_to_s32__sse2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
+ if (ma_has_neon()) {
+ ma_pcm_s24_to_s32__neon(dst, src, count, ditherMode);
+ } else
+ #endif
+ {
+ ma_pcm_s24_to_s32__optimized(dst, src, count, ditherMode);
+ }
+#endif
+}
-typedef int ma_pa_source_flags_t;
-#define MA_PA_SOURCE_NOFLAGS 0x00000000
-#define MA_PA_SOURCE_HW_VOLUME_CTRL 0x00000001
-#define MA_PA_SOURCE_LATENCY 0x00000002
-#define MA_PA_SOURCE_HARDWARE 0x00000004
-#define MA_PA_SOURCE_NETWORK 0x00000008
-#define MA_PA_SOURCE_HW_MUTE_CTRL 0x00000010
-#define MA_PA_SOURCE_DECIBEL_VOLUME 0x00000020
-#define MA_PA_SOURCE_DYNAMIC_LATENCY 0x00000040
-#define MA_PA_SOURCE_FLAT_VOLUME 0x00000080
-typedef int ma_pa_context_state_t;
-#define MA_PA_CONTEXT_UNCONNECTED 0
-#define MA_PA_CONTEXT_CONNECTING 1
-#define MA_PA_CONTEXT_AUTHORIZING 2
-#define MA_PA_CONTEXT_SETTING_NAME 3
-#define MA_PA_CONTEXT_READY 4
-#define MA_PA_CONTEXT_FAILED 5
-#define MA_PA_CONTEXT_TERMINATED 6
+static MA_INLINE void ma_pcm_s24_to_f32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ float* dst_f32 = (float*)dst;
+ const ma_uint8* src_s24 = (const ma_uint8*)src;
-typedef int ma_pa_stream_state_t;
-#define MA_PA_STREAM_UNCONNECTED 0
-#define MA_PA_STREAM_CREATING 1
-#define MA_PA_STREAM_READY 2
-#define MA_PA_STREAM_FAILED 3
-#define MA_PA_STREAM_TERMINATED 4
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ float x = (float)(((ma_int32)(((ma_uint32)(src_s24[i*3+0]) << 8) | ((ma_uint32)(src_s24[i*3+1]) << 16) | ((ma_uint32)(src_s24[i*3+2])) << 24)) >> 8);
-typedef int ma_pa_operation_state_t;
-#define MA_PA_OPERATION_RUNNING 0
-#define MA_PA_OPERATION_DONE 1
-#define MA_PA_OPERATION_CANCELLED 2
+#if 0
+ /* The accurate way. */
+ x = x + 8388608.0f; /* -8388608..8388607 to 0..16777215 */
+ x = x * 0.00000011920929665621f; /* 0..16777215 to 0..2 */
+ x = x - 1; /* 0..2 to -1..1 */
+#else
+ /* The fast way. */
+ x = x * 0.00000011920928955078125f; /* -8388608..8388607 to -1..0.999969482421875 */
+#endif
-typedef int ma_pa_sink_state_t;
-#define MA_PA_SINK_INVALID_STATE -1
-#define MA_PA_SINK_RUNNING 0
-#define MA_PA_SINK_IDLE 1
-#define MA_PA_SINK_SUSPENDED 2
+ dst_f32[i] = x;
+ }
-typedef int ma_pa_source_state_t;
-#define MA_PA_SOURCE_INVALID_STATE -1
-#define MA_PA_SOURCE_RUNNING 0
-#define MA_PA_SOURCE_IDLE 1
-#define MA_PA_SOURCE_SUSPENDED 2
+ (void)ditherMode;
+}
-typedef int ma_pa_seek_mode_t;
-#define MA_PA_SEEK_RELATIVE 0
-#define MA_PA_SEEK_ABSOLUTE 1
-#define MA_PA_SEEK_RELATIVE_ON_READ 2
-#define MA_PA_SEEK_RELATIVE_END 3
+static MA_INLINE void ma_pcm_s24_to_f32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s24_to_f32__reference(dst, src, count, ditherMode);
+}
-typedef int ma_pa_channel_position_t;
-#define MA_PA_CHANNEL_POSITION_INVALID -1
-#define MA_PA_CHANNEL_POSITION_MONO 0
-#define MA_PA_CHANNEL_POSITION_FRONT_LEFT 1
-#define MA_PA_CHANNEL_POSITION_FRONT_RIGHT 2
-#define MA_PA_CHANNEL_POSITION_FRONT_CENTER 3
-#define MA_PA_CHANNEL_POSITION_REAR_CENTER 4
-#define MA_PA_CHANNEL_POSITION_REAR_LEFT 5
-#define MA_PA_CHANNEL_POSITION_REAR_RIGHT 6
-#define MA_PA_CHANNEL_POSITION_LFE 7
-#define MA_PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER 8
-#define MA_PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER 9
-#define MA_PA_CHANNEL_POSITION_SIDE_LEFT 10
-#define MA_PA_CHANNEL_POSITION_SIDE_RIGHT 11
-#define MA_PA_CHANNEL_POSITION_AUX0 12
-#define MA_PA_CHANNEL_POSITION_AUX1 13
-#define MA_PA_CHANNEL_POSITION_AUX2 14
-#define MA_PA_CHANNEL_POSITION_AUX3 15
-#define MA_PA_CHANNEL_POSITION_AUX4 16
-#define MA_PA_CHANNEL_POSITION_AUX5 17
-#define MA_PA_CHANNEL_POSITION_AUX6 18
-#define MA_PA_CHANNEL_POSITION_AUX7 19
-#define MA_PA_CHANNEL_POSITION_AUX8 20
-#define MA_PA_CHANNEL_POSITION_AUX9 21
-#define MA_PA_CHANNEL_POSITION_AUX10 22
-#define MA_PA_CHANNEL_POSITION_AUX11 23
-#define MA_PA_CHANNEL_POSITION_AUX12 24
-#define MA_PA_CHANNEL_POSITION_AUX13 25
-#define MA_PA_CHANNEL_POSITION_AUX14 26
-#define MA_PA_CHANNEL_POSITION_AUX15 27
-#define MA_PA_CHANNEL_POSITION_AUX16 28
-#define MA_PA_CHANNEL_POSITION_AUX17 29
-#define MA_PA_CHANNEL_POSITION_AUX18 30
-#define MA_PA_CHANNEL_POSITION_AUX19 31
-#define MA_PA_CHANNEL_POSITION_AUX20 32
-#define MA_PA_CHANNEL_POSITION_AUX21 33
-#define MA_PA_CHANNEL_POSITION_AUX22 34
-#define MA_PA_CHANNEL_POSITION_AUX23 35
-#define MA_PA_CHANNEL_POSITION_AUX24 36
-#define MA_PA_CHANNEL_POSITION_AUX25 37
-#define MA_PA_CHANNEL_POSITION_AUX26 38
-#define MA_PA_CHANNEL_POSITION_AUX27 39
-#define MA_PA_CHANNEL_POSITION_AUX28 40
-#define MA_PA_CHANNEL_POSITION_AUX29 41
-#define MA_PA_CHANNEL_POSITION_AUX30 42
-#define MA_PA_CHANNEL_POSITION_AUX31 43
-#define MA_PA_CHANNEL_POSITION_TOP_CENTER 44
-#define MA_PA_CHANNEL_POSITION_TOP_FRONT_LEFT 45
-#define MA_PA_CHANNEL_POSITION_TOP_FRONT_RIGHT 46
-#define MA_PA_CHANNEL_POSITION_TOP_FRONT_CENTER 47
-#define MA_PA_CHANNEL_POSITION_TOP_REAR_LEFT 48
-#define MA_PA_CHANNEL_POSITION_TOP_REAR_RIGHT 49
-#define MA_PA_CHANNEL_POSITION_TOP_REAR_CENTER 50
-#define MA_PA_CHANNEL_POSITION_LEFT MA_PA_CHANNEL_POSITION_FRONT_LEFT
-#define MA_PA_CHANNEL_POSITION_RIGHT MA_PA_CHANNEL_POSITION_FRONT_RIGHT
-#define MA_PA_CHANNEL_POSITION_CENTER MA_PA_CHANNEL_POSITION_FRONT_CENTER
-#define MA_PA_CHANNEL_POSITION_SUBWOOFER MA_PA_CHANNEL_POSITION_LFE
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE void ma_pcm_s24_to_f32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s24_to_f32__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE void ma_pcm_s24_to_f32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s24_to_f32__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE void ma_pcm_s24_to_f32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s24_to_f32__optimized(dst, src, count, ditherMode);
+}
+#endif
-typedef int ma_pa_channel_map_def_t;
-#define MA_PA_CHANNEL_MAP_AIFF 0
-#define MA_PA_CHANNEL_MAP_ALSA 1
-#define MA_PA_CHANNEL_MAP_AUX 2
-#define MA_PA_CHANNEL_MAP_WAVEEX 3
-#define MA_PA_CHANNEL_MAP_OSS 4
-#define MA_PA_CHANNEL_MAP_DEFAULT MA_PA_CHANNEL_MAP_AIFF
+MA_API void ma_pcm_s24_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_s24_to_f32__reference(dst, src, count, ditherMode);
+#else
+ # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
+ if (ma_has_avx2()) {
+ ma_pcm_s24_to_f32__avx2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
+ if (ma_has_sse2()) {
+ ma_pcm_s24_to_f32__sse2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
+ if (ma_has_neon()) {
+ ma_pcm_s24_to_f32__neon(dst, src, count, ditherMode);
+ } else
+ #endif
+ {
+ ma_pcm_s24_to_f32__optimized(dst, src, count, ditherMode);
+ }
+#endif
+}
-typedef int ma_pa_sample_format_t;
-#define MA_PA_SAMPLE_INVALID -1
-#define MA_PA_SAMPLE_U8 0
-#define MA_PA_SAMPLE_ALAW 1
-#define MA_PA_SAMPLE_ULAW 2
-#define MA_PA_SAMPLE_S16LE 3
-#define MA_PA_SAMPLE_S16BE 4
-#define MA_PA_SAMPLE_FLOAT32LE 5
-#define MA_PA_SAMPLE_FLOAT32BE 6
-#define MA_PA_SAMPLE_S32LE 7
-#define MA_PA_SAMPLE_S32BE 8
-#define MA_PA_SAMPLE_S24LE 9
-#define MA_PA_SAMPLE_S24BE 10
-#define MA_PA_SAMPLE_S24_32LE 11
-#define MA_PA_SAMPLE_S24_32BE 12
-typedef struct ma_pa_mainloop ma_pa_mainloop;
-typedef struct ma_pa_threaded_mainloop ma_pa_threaded_mainloop;
-typedef struct ma_pa_mainloop_api ma_pa_mainloop_api;
-typedef struct ma_pa_context ma_pa_context;
-typedef struct ma_pa_operation ma_pa_operation;
-typedef struct ma_pa_stream ma_pa_stream;
-typedef struct ma_pa_spawn_api ma_pa_spawn_api;
+static MA_INLINE void ma_pcm_interleave_s24__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
+{
+ ma_uint8* dst8 = (ma_uint8*)dst;
+ const ma_uint8** src8 = (const ma_uint8**)src;
-typedef struct
+ ma_uint64 iFrame;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ dst8[iFrame*3*channels + iChannel*3 + 0] = src8[iChannel][iFrame*3 + 0];
+ dst8[iFrame*3*channels + iChannel*3 + 1] = src8[iChannel][iFrame*3 + 1];
+ dst8[iFrame*3*channels + iChannel*3 + 2] = src8[iChannel][iFrame*3 + 2];
+ }
+ }
+}
+
+static MA_INLINE void ma_pcm_interleave_s24__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
{
- ma_uint32 maxlength;
- ma_uint32 tlength;
- ma_uint32 prebuf;
- ma_uint32 minreq;
- ma_uint32 fragsize;
-} ma_pa_buffer_attr;
+ ma_pcm_interleave_s24__reference(dst, src, frameCount, channels);
+}
-typedef struct
+MA_API void ma_pcm_interleave_s24(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
{
- ma_uint8 channels;
- ma_pa_channel_position_t map[MA_PA_CHANNELS_MAX];
-} ma_pa_channel_map;
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_interleave_s24__reference(dst, src, frameCount, channels);
+#else
+ ma_pcm_interleave_s24__optimized(dst, src, frameCount, channels);
+#endif
+}
-typedef struct
+
+static MA_INLINE void ma_pcm_deinterleave_s24__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
{
- ma_uint8 channels;
- ma_uint32 values[MA_PA_CHANNELS_MAX];
-} ma_pa_cvolume;
+ ma_uint8** dst8 = (ma_uint8**)dst;
+ const ma_uint8* src8 = (const ma_uint8*)src;
-typedef struct
+ ma_uint32 iFrame;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ dst8[iChannel][iFrame*3 + 0] = src8[iFrame*3*channels + iChannel*3 + 0];
+ dst8[iChannel][iFrame*3 + 1] = src8[iFrame*3*channels + iChannel*3 + 1];
+ dst8[iChannel][iFrame*3 + 2] = src8[iFrame*3*channels + iChannel*3 + 2];
+ }
+ }
+}
+
+static MA_INLINE void ma_pcm_deinterleave_s24__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
{
- ma_pa_sample_format_t format;
- ma_uint32 rate;
- ma_uint8 channels;
-} ma_pa_sample_spec;
+ ma_pcm_deinterleave_s24__reference(dst, src, frameCount, channels);
+}
-typedef struct
+MA_API void ma_pcm_deinterleave_s24(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
{
- const char* name;
- ma_uint32 index;
- const char* description;
- ma_pa_sample_spec sample_spec;
- ma_pa_channel_map channel_map;
- ma_uint32 owner_module;
- ma_pa_cvolume volume;
- int mute;
- ma_uint32 monitor_source;
- const char* monitor_source_name;
- ma_uint64 latency;
- const char* driver;
- ma_pa_sink_flags_t flags;
- void* proplist;
- ma_uint64 configured_latency;
- ma_uint32 base_volume;
- ma_pa_sink_state_t state;
- ma_uint32 n_volume_steps;
- ma_uint32 card;
- ma_uint32 n_ports;
- void** ports;
- void* active_port;
- ma_uint8 n_formats;
- void** formats;
-} ma_pa_sink_info;
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_deinterleave_s24__reference(dst, src, frameCount, channels);
+#else
+ ma_pcm_deinterleave_s24__optimized(dst, src, frameCount, channels);
+#endif
+}
-typedef struct
+
+
+/* s32 */
+static MA_INLINE void ma_pcm_s32_to_u8__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- const char *name;
- ma_uint32 index;
- const char *description;
- ma_pa_sample_spec sample_spec;
- ma_pa_channel_map channel_map;
- ma_uint32 owner_module;
- ma_pa_cvolume volume;
- int mute;
- ma_uint32 monitor_of_sink;
- const char *monitor_of_sink_name;
- ma_uint64 latency;
- const char *driver;
- ma_pa_source_flags_t flags;
- void* proplist;
- ma_uint64 configured_latency;
- ma_uint32 base_volume;
- ma_pa_source_state_t state;
- ma_uint32 n_volume_steps;
- ma_uint32 card;
- ma_uint32 n_ports;
- void** ports;
- void* active_port;
- ma_uint8 n_formats;
- void** formats;
-} ma_pa_source_info;
+ ma_uint8* dst_u8 = (ma_uint8*)dst;
+ const ma_int32* src_s32 = (const ma_int32*)src;
-typedef void (* ma_pa_context_notify_cb_t)(ma_pa_context* c, void* userdata);
-typedef void (* ma_pa_sink_info_cb_t) (ma_pa_context* c, const ma_pa_sink_info* i, int eol, void* userdata);
-typedef void (* ma_pa_source_info_cb_t) (ma_pa_context* c, const ma_pa_source_info* i, int eol, void* userdata);
-typedef void (* ma_pa_stream_success_cb_t)(ma_pa_stream* s, int success, void* userdata);
-typedef void (* ma_pa_stream_request_cb_t)(ma_pa_stream* s, size_t nbytes, void* userdata);
-typedef void (* ma_pa_stream_notify_cb_t) (ma_pa_stream* s, void* userdata);
-typedef void (* ma_pa_free_cb_t) (void* p);
-#endif
+ if (ditherMode == ma_dither_mode_none) {
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ ma_int32 x = src_s32[i];
+ x = x >> 24;
+ x = x + 128;
+ dst_u8[i] = (ma_uint8)x;
+ }
+ } else {
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ ma_int32 x = src_s32[i];
+ /* Dither. Don't overflow. */
+ ma_int32 dither = ma_dither_s32(ditherMode, -0x800000, 0x7FFFFF);
+ if ((ma_int64)x + dither <= 0x7FFFFFFF) {
+ x = x + dither;
+ } else {
+ x = 0x7FFFFFFF;
+ }
-typedef ma_pa_mainloop* (* ma_pa_mainloop_new_proc) (void);
-typedef void (* ma_pa_mainloop_free_proc) (ma_pa_mainloop* m);
-typedef void (* ma_pa_mainloop_quit_proc) (ma_pa_mainloop* m, int retval);
-typedef ma_pa_mainloop_api* (* ma_pa_mainloop_get_api_proc) (ma_pa_mainloop* m);
-typedef int (* ma_pa_mainloop_iterate_proc) (ma_pa_mainloop* m, int block, int* retval);
-typedef void (* ma_pa_mainloop_wakeup_proc) (ma_pa_mainloop* m);
-typedef ma_pa_threaded_mainloop* (* ma_pa_threaded_mainloop_new_proc) (void);
-typedef void (* ma_pa_threaded_mainloop_free_proc) (ma_pa_threaded_mainloop* m);
-typedef int (* ma_pa_threaded_mainloop_start_proc) (ma_pa_threaded_mainloop* m);
-typedef void (* ma_pa_threaded_mainloop_stop_proc) (ma_pa_threaded_mainloop* m);
-typedef void (* ma_pa_threaded_mainloop_lock_proc) (ma_pa_threaded_mainloop* m);
-typedef void (* ma_pa_threaded_mainloop_unlock_proc) (ma_pa_threaded_mainloop* m);
-typedef void (* ma_pa_threaded_mainloop_wait_proc) (ma_pa_threaded_mainloop* m);
-typedef void (* ma_pa_threaded_mainloop_signal_proc) (ma_pa_threaded_mainloop* m, int wait_for_accept);
-typedef void (* ma_pa_threaded_mainloop_accept_proc) (ma_pa_threaded_mainloop* m);
-typedef int (* ma_pa_threaded_mainloop_get_retval_proc) (ma_pa_threaded_mainloop* m);
-typedef ma_pa_mainloop_api* (* ma_pa_threaded_mainloop_get_api_proc) (ma_pa_threaded_mainloop* m);
-typedef int (* ma_pa_threaded_mainloop_in_thread_proc) (ma_pa_threaded_mainloop* m);
-typedef void (* ma_pa_threaded_mainloop_set_name_proc) (ma_pa_threaded_mainloop* m, const char* name);
-typedef ma_pa_context* (* ma_pa_context_new_proc) (ma_pa_mainloop_api* mainloop, const char* name);
-typedef void (* ma_pa_context_unref_proc) (ma_pa_context* c);
-typedef int (* ma_pa_context_connect_proc) (ma_pa_context* c, const char* server, ma_pa_context_flags_t flags, const ma_pa_spawn_api* api);
-typedef void (* ma_pa_context_disconnect_proc) (ma_pa_context* c);
-typedef void (* ma_pa_context_set_state_callback_proc) (ma_pa_context* c, ma_pa_context_notify_cb_t cb, void* userdata);
-typedef ma_pa_context_state_t (* ma_pa_context_get_state_proc) (ma_pa_context* c);
-typedef ma_pa_operation* (* ma_pa_context_get_sink_info_list_proc) (ma_pa_context* c, ma_pa_sink_info_cb_t cb, void* userdata);
-typedef ma_pa_operation* (* ma_pa_context_get_source_info_list_proc) (ma_pa_context* c, ma_pa_source_info_cb_t cb, void* userdata);
-typedef ma_pa_operation* (* ma_pa_context_get_sink_info_by_name_proc) (ma_pa_context* c, const char* name, ma_pa_sink_info_cb_t cb, void* userdata);
-typedef ma_pa_operation* (* ma_pa_context_get_source_info_by_name_proc)(ma_pa_context* c, const char* name, ma_pa_source_info_cb_t cb, void* userdata);
-typedef void (* ma_pa_operation_unref_proc) (ma_pa_operation* o);
-typedef ma_pa_operation_state_t (* ma_pa_operation_get_state_proc) (ma_pa_operation* o);
-typedef ma_pa_channel_map* (* ma_pa_channel_map_init_extend_proc) (ma_pa_channel_map* m, unsigned channels, ma_pa_channel_map_def_t def);
-typedef int (* ma_pa_channel_map_valid_proc) (const ma_pa_channel_map* m);
-typedef int (* ma_pa_channel_map_compatible_proc) (const ma_pa_channel_map* m, const ma_pa_sample_spec* ss);
-typedef ma_pa_stream* (* ma_pa_stream_new_proc) (ma_pa_context* c, const char* name, const ma_pa_sample_spec* ss, const ma_pa_channel_map* map);
-typedef void (* ma_pa_stream_unref_proc) (ma_pa_stream* s);
-typedef int (* ma_pa_stream_connect_playback_proc) (ma_pa_stream* s, const char* dev, const ma_pa_buffer_attr* attr, ma_pa_stream_flags_t flags, const ma_pa_cvolume* volume, ma_pa_stream* sync_stream);
-typedef int (* ma_pa_stream_connect_record_proc) (ma_pa_stream* s, const char* dev, const ma_pa_buffer_attr* attr, ma_pa_stream_flags_t flags);
-typedef int (* ma_pa_stream_disconnect_proc) (ma_pa_stream* s);
-typedef ma_pa_stream_state_t (* ma_pa_stream_get_state_proc) (ma_pa_stream* s);
-typedef const ma_pa_sample_spec* (* ma_pa_stream_get_sample_spec_proc) (ma_pa_stream* s);
-typedef const ma_pa_channel_map* (* ma_pa_stream_get_channel_map_proc) (ma_pa_stream* s);
-typedef const ma_pa_buffer_attr* (* ma_pa_stream_get_buffer_attr_proc) (ma_pa_stream* s);
-typedef ma_pa_operation* (* ma_pa_stream_set_buffer_attr_proc) (ma_pa_stream* s, const ma_pa_buffer_attr* attr, ma_pa_stream_success_cb_t cb, void* userdata);
-typedef const char* (* ma_pa_stream_get_device_name_proc) (ma_pa_stream* s);
-typedef void (* ma_pa_stream_set_write_callback_proc) (ma_pa_stream* s, ma_pa_stream_request_cb_t cb, void* userdata);
-typedef void (* ma_pa_stream_set_read_callback_proc) (ma_pa_stream* s, ma_pa_stream_request_cb_t cb, void* userdata);
-typedef void (* ma_pa_stream_set_suspended_callback_proc) (ma_pa_stream* s, ma_pa_stream_notify_cb_t cb, void* userdata);
-typedef int (* ma_pa_stream_is_suspended_proc) (const ma_pa_stream* s);
-typedef ma_pa_operation* (* ma_pa_stream_flush_proc) (ma_pa_stream* s, ma_pa_stream_success_cb_t cb, void* userdata);
-typedef ma_pa_operation* (* ma_pa_stream_drain_proc) (ma_pa_stream* s, ma_pa_stream_success_cb_t cb, void* userdata);
-typedef int (* ma_pa_stream_is_corked_proc) (ma_pa_stream* s);
-typedef ma_pa_operation* (* ma_pa_stream_cork_proc) (ma_pa_stream* s, int b, ma_pa_stream_success_cb_t cb, void* userdata);
-typedef ma_pa_operation* (* ma_pa_stream_trigger_proc) (ma_pa_stream* s, ma_pa_stream_success_cb_t cb, void* userdata);
-typedef int (* ma_pa_stream_begin_write_proc) (ma_pa_stream* s, void** data, size_t* nbytes);
-typedef int (* ma_pa_stream_write_proc) (ma_pa_stream* s, const void* data, size_t nbytes, ma_pa_free_cb_t free_cb, int64_t offset, ma_pa_seek_mode_t seek);
-typedef int (* ma_pa_stream_peek_proc) (ma_pa_stream* s, const void** data, size_t* nbytes);
-typedef int (* ma_pa_stream_drop_proc) (ma_pa_stream* s);
-typedef size_t (* ma_pa_stream_writable_size_proc) (ma_pa_stream* s);
-typedef size_t (* ma_pa_stream_readable_size_proc) (ma_pa_stream* s);
+ x = x >> 24;
+ x = x + 128;
+ dst_u8[i] = (ma_uint8)x;
+ }
+ }
+}
-typedef struct
+static MA_INLINE void ma_pcm_s32_to_u8__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- ma_uint32 count;
- ma_uint32 capacity;
- ma_device_info* pInfo;
-} ma_pulse_device_enum_data;
+ ma_pcm_s32_to_u8__reference(dst, src, count, ditherMode);
+}
-static ma_result ma_result_from_pulse(int result)
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE void ma_pcm_s32_to_u8__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- if (result < 0) {
- return MA_ERROR;
- }
+ ma_pcm_s32_to_u8__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE void ma_pcm_s32_to_u8__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s32_to_u8__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE void ma_pcm_s32_to_u8__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s32_to_u8__optimized(dst, src, count, ditherMode);
+}
+#endif
- switch (result) {
- case MA_PA_OK: return MA_SUCCESS;
- case MA_PA_ERR_ACCESS: return MA_ACCESS_DENIED;
- case MA_PA_ERR_INVALID: return MA_INVALID_ARGS;
- case MA_PA_ERR_NOENTITY: return MA_NO_DEVICE;
- default: return MA_ERROR;
- }
+MA_API void ma_pcm_s32_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_s32_to_u8__reference(dst, src, count, ditherMode);
+#else
+ # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
+ if (ma_has_avx2()) {
+ ma_pcm_s32_to_u8__avx2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
+ if (ma_has_sse2()) {
+ ma_pcm_s32_to_u8__sse2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
+ if (ma_has_neon()) {
+ ma_pcm_s32_to_u8__neon(dst, src, count, ditherMode);
+ } else
+ #endif
+ {
+ ma_pcm_s32_to_u8__optimized(dst, src, count, ditherMode);
+ }
+#endif
}
-#if 0
-static ma_pa_sample_format_t ma_format_to_pulse(ma_format format)
+
+static MA_INLINE void ma_pcm_s32_to_s16__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- if (ma_is_little_endian()) {
- switch (format) {
- case ma_format_s16: return MA_PA_SAMPLE_S16LE;
- case ma_format_s24: return MA_PA_SAMPLE_S24LE;
- case ma_format_s32: return MA_PA_SAMPLE_S32LE;
- case ma_format_f32: return MA_PA_SAMPLE_FLOAT32LE;
- default: break;
+ ma_int16* dst_s16 = (ma_int16*)dst;
+ const ma_int32* src_s32 = (const ma_int32*)src;
+
+ if (ditherMode == ma_dither_mode_none) {
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ ma_int32 x = src_s32[i];
+ x = x >> 16;
+ dst_s16[i] = (ma_int16)x;
}
} else {
- switch (format) {
- case ma_format_s16: return MA_PA_SAMPLE_S16BE;
- case ma_format_s24: return MA_PA_SAMPLE_S24BE;
- case ma_format_s32: return MA_PA_SAMPLE_S32BE;
- case ma_format_f32: return MA_PA_SAMPLE_FLOAT32BE;
- default: break;
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ ma_int32 x = src_s32[i];
+
+ /* Dither. Don't overflow. */
+ ma_int32 dither = ma_dither_s32(ditherMode, -0x8000, 0x7FFF);
+ if ((ma_int64)x + dither <= 0x7FFFFFFF) {
+ x = x + dither;
+ } else {
+ x = 0x7FFFFFFF;
+ }
+
+ x = x >> 16;
+ dst_s16[i] = (ma_int16)x;
}
}
+}
- /* Endian agnostic. */
- switch (format) {
- case ma_format_u8: return MA_PA_SAMPLE_U8;
- default: return MA_PA_SAMPLE_INVALID;
- }
+static MA_INLINE void ma_pcm_s32_to_s16__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s32_to_s16__reference(dst, src, count, ditherMode);
+}
+
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE void ma_pcm_s32_to_s16__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s32_to_s16__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE void ma_pcm_s32_to_s16__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s32_to_s16__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE void ma_pcm_s32_to_s16__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s32_to_s16__optimized(dst, src, count, ditherMode);
}
#endif
-static ma_format ma_format_from_pulse(ma_pa_sample_format_t format)
+MA_API void ma_pcm_s32_to_s16(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- if (ma_is_little_endian()) {
- switch (format) {
- case MA_PA_SAMPLE_S16LE: return ma_format_s16;
- case MA_PA_SAMPLE_S24LE: return ma_format_s24;
- case MA_PA_SAMPLE_S32LE: return ma_format_s32;
- case MA_PA_SAMPLE_FLOAT32LE: return ma_format_f32;
- default: break;
- }
- } else {
- switch (format) {
- case MA_PA_SAMPLE_S16BE: return ma_format_s16;
- case MA_PA_SAMPLE_S24BE: return ma_format_s24;
- case MA_PA_SAMPLE_S32BE: return ma_format_s32;
- case MA_PA_SAMPLE_FLOAT32BE: return ma_format_f32;
- default: break;
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_s32_to_s16__reference(dst, src, count, ditherMode);
+#else
+ # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
+ if (ma_has_avx2()) {
+ ma_pcm_s32_to_s16__avx2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
+ if (ma_has_sse2()) {
+ ma_pcm_s32_to_s16__sse2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
+ if (ma_has_neon()) {
+ ma_pcm_s32_to_s16__neon(dst, src, count, ditherMode);
+ } else
+ #endif
+ {
+ ma_pcm_s32_to_s16__optimized(dst, src, count, ditherMode);
}
- }
+#endif
+}
- /* Endian agnostic. */
- switch (format) {
- case MA_PA_SAMPLE_U8: return ma_format_u8;
- default: return ma_format_unknown;
+
+static MA_INLINE void ma_pcm_s32_to_s24__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_uint8* dst_s24 = (ma_uint8*)dst;
+ const ma_int32* src_s32 = (const ma_int32*)src;
+
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ ma_uint32 x = (ma_uint32)src_s32[i];
+ dst_s24[i*3+0] = (ma_uint8)((x & 0x0000FF00) >> 8);
+ dst_s24[i*3+1] = (ma_uint8)((x & 0x00FF0000) >> 16);
+ dst_s24[i*3+2] = (ma_uint8)((x & 0xFF000000) >> 24);
}
+
+ (void)ditherMode; /* No dithering for s32 -> s24. */
}
-static ma_channel ma_channel_position_from_pulse(ma_pa_channel_position_t position)
+static MA_INLINE void ma_pcm_s32_to_s24__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- switch (position)
- {
- case MA_PA_CHANNEL_POSITION_INVALID: return MA_CHANNEL_NONE;
- case MA_PA_CHANNEL_POSITION_MONO: return MA_CHANNEL_MONO;
- case MA_PA_CHANNEL_POSITION_FRONT_LEFT: return MA_CHANNEL_FRONT_LEFT;
- case MA_PA_CHANNEL_POSITION_FRONT_RIGHT: return MA_CHANNEL_FRONT_RIGHT;
- case MA_PA_CHANNEL_POSITION_FRONT_CENTER: return MA_CHANNEL_FRONT_CENTER;
- case MA_PA_CHANNEL_POSITION_REAR_CENTER: return MA_CHANNEL_BACK_CENTER;
- case MA_PA_CHANNEL_POSITION_REAR_LEFT: return MA_CHANNEL_BACK_LEFT;
- case MA_PA_CHANNEL_POSITION_REAR_RIGHT: return MA_CHANNEL_BACK_RIGHT;
- case MA_PA_CHANNEL_POSITION_LFE: return MA_CHANNEL_LFE;
- case MA_PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER: return MA_CHANNEL_FRONT_LEFT_CENTER;
- case MA_PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER: return MA_CHANNEL_FRONT_RIGHT_CENTER;
- case MA_PA_CHANNEL_POSITION_SIDE_LEFT: return MA_CHANNEL_SIDE_LEFT;
- case MA_PA_CHANNEL_POSITION_SIDE_RIGHT: return MA_CHANNEL_SIDE_RIGHT;
- case MA_PA_CHANNEL_POSITION_AUX0: return MA_CHANNEL_AUX_0;
- case MA_PA_CHANNEL_POSITION_AUX1: return MA_CHANNEL_AUX_1;
- case MA_PA_CHANNEL_POSITION_AUX2: return MA_CHANNEL_AUX_2;
- case MA_PA_CHANNEL_POSITION_AUX3: return MA_CHANNEL_AUX_3;
- case MA_PA_CHANNEL_POSITION_AUX4: return MA_CHANNEL_AUX_4;
- case MA_PA_CHANNEL_POSITION_AUX5: return MA_CHANNEL_AUX_5;
- case MA_PA_CHANNEL_POSITION_AUX6: return MA_CHANNEL_AUX_6;
- case MA_PA_CHANNEL_POSITION_AUX7: return MA_CHANNEL_AUX_7;
- case MA_PA_CHANNEL_POSITION_AUX8: return MA_CHANNEL_AUX_8;
- case MA_PA_CHANNEL_POSITION_AUX9: return MA_CHANNEL_AUX_9;
- case MA_PA_CHANNEL_POSITION_AUX10: return MA_CHANNEL_AUX_10;
- case MA_PA_CHANNEL_POSITION_AUX11: return MA_CHANNEL_AUX_11;
- case MA_PA_CHANNEL_POSITION_AUX12: return MA_CHANNEL_AUX_12;
- case MA_PA_CHANNEL_POSITION_AUX13: return MA_CHANNEL_AUX_13;
- case MA_PA_CHANNEL_POSITION_AUX14: return MA_CHANNEL_AUX_14;
- case MA_PA_CHANNEL_POSITION_AUX15: return MA_CHANNEL_AUX_15;
- case MA_PA_CHANNEL_POSITION_AUX16: return MA_CHANNEL_AUX_16;
- case MA_PA_CHANNEL_POSITION_AUX17: return MA_CHANNEL_AUX_17;
- case MA_PA_CHANNEL_POSITION_AUX18: return MA_CHANNEL_AUX_18;
- case MA_PA_CHANNEL_POSITION_AUX19: return MA_CHANNEL_AUX_19;
- case MA_PA_CHANNEL_POSITION_AUX20: return MA_CHANNEL_AUX_20;
- case MA_PA_CHANNEL_POSITION_AUX21: return MA_CHANNEL_AUX_21;
- case MA_PA_CHANNEL_POSITION_AUX22: return MA_CHANNEL_AUX_22;
- case MA_PA_CHANNEL_POSITION_AUX23: return MA_CHANNEL_AUX_23;
- case MA_PA_CHANNEL_POSITION_AUX24: return MA_CHANNEL_AUX_24;
- case MA_PA_CHANNEL_POSITION_AUX25: return MA_CHANNEL_AUX_25;
- case MA_PA_CHANNEL_POSITION_AUX26: return MA_CHANNEL_AUX_26;
- case MA_PA_CHANNEL_POSITION_AUX27: return MA_CHANNEL_AUX_27;
- case MA_PA_CHANNEL_POSITION_AUX28: return MA_CHANNEL_AUX_28;
- case MA_PA_CHANNEL_POSITION_AUX29: return MA_CHANNEL_AUX_29;
- case MA_PA_CHANNEL_POSITION_AUX30: return MA_CHANNEL_AUX_30;
- case MA_PA_CHANNEL_POSITION_AUX31: return MA_CHANNEL_AUX_31;
- case MA_PA_CHANNEL_POSITION_TOP_CENTER: return MA_CHANNEL_TOP_CENTER;
- case MA_PA_CHANNEL_POSITION_TOP_FRONT_LEFT: return MA_CHANNEL_TOP_FRONT_LEFT;
- case MA_PA_CHANNEL_POSITION_TOP_FRONT_RIGHT: return MA_CHANNEL_TOP_FRONT_RIGHT;
- case MA_PA_CHANNEL_POSITION_TOP_FRONT_CENTER: return MA_CHANNEL_TOP_FRONT_CENTER;
- case MA_PA_CHANNEL_POSITION_TOP_REAR_LEFT: return MA_CHANNEL_TOP_BACK_LEFT;
- case MA_PA_CHANNEL_POSITION_TOP_REAR_RIGHT: return MA_CHANNEL_TOP_BACK_RIGHT;
- case MA_PA_CHANNEL_POSITION_TOP_REAR_CENTER: return MA_CHANNEL_TOP_BACK_CENTER;
- default: return MA_CHANNEL_NONE;
- }
+ ma_pcm_s32_to_s24__reference(dst, src, count, ditherMode);
}
-#if 0
-static ma_pa_channel_position_t ma_channel_position_to_pulse(ma_channel position)
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE void ma_pcm_s32_to_s24__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- switch (position)
- {
- case MA_CHANNEL_NONE: return MA_PA_CHANNEL_POSITION_INVALID;
- case MA_CHANNEL_FRONT_LEFT: return MA_PA_CHANNEL_POSITION_FRONT_LEFT;
- case MA_CHANNEL_FRONT_RIGHT: return MA_PA_CHANNEL_POSITION_FRONT_RIGHT;
- case MA_CHANNEL_FRONT_CENTER: return MA_PA_CHANNEL_POSITION_FRONT_CENTER;
- case MA_CHANNEL_LFE: return MA_PA_CHANNEL_POSITION_LFE;
- case MA_CHANNEL_BACK_LEFT: return MA_PA_CHANNEL_POSITION_REAR_LEFT;
- case MA_CHANNEL_BACK_RIGHT: return MA_PA_CHANNEL_POSITION_REAR_RIGHT;
- case MA_CHANNEL_FRONT_LEFT_CENTER: return MA_PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER;
- case MA_CHANNEL_FRONT_RIGHT_CENTER: return MA_PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER;
- case MA_CHANNEL_BACK_CENTER: return MA_PA_CHANNEL_POSITION_REAR_CENTER;
- case MA_CHANNEL_SIDE_LEFT: return MA_PA_CHANNEL_POSITION_SIDE_LEFT;
- case MA_CHANNEL_SIDE_RIGHT: return MA_PA_CHANNEL_POSITION_SIDE_RIGHT;
- case MA_CHANNEL_TOP_CENTER: return MA_PA_CHANNEL_POSITION_TOP_CENTER;
- case MA_CHANNEL_TOP_FRONT_LEFT: return MA_PA_CHANNEL_POSITION_TOP_FRONT_LEFT;
- case MA_CHANNEL_TOP_FRONT_CENTER: return MA_PA_CHANNEL_POSITION_TOP_FRONT_CENTER;
- case MA_CHANNEL_TOP_FRONT_RIGHT: return MA_PA_CHANNEL_POSITION_TOP_FRONT_RIGHT;
- case MA_CHANNEL_TOP_BACK_LEFT: return MA_PA_CHANNEL_POSITION_TOP_REAR_LEFT;
- case MA_CHANNEL_TOP_BACK_CENTER: return MA_PA_CHANNEL_POSITION_TOP_REAR_CENTER;
- case MA_CHANNEL_TOP_BACK_RIGHT: return MA_PA_CHANNEL_POSITION_TOP_REAR_RIGHT;
- case MA_CHANNEL_19: return MA_PA_CHANNEL_POSITION_AUX18;
- case MA_CHANNEL_20: return MA_PA_CHANNEL_POSITION_AUX19;
- case MA_CHANNEL_21: return MA_PA_CHANNEL_POSITION_AUX20;
- case MA_CHANNEL_22: return MA_PA_CHANNEL_POSITION_AUX21;
- case MA_CHANNEL_23: return MA_PA_CHANNEL_POSITION_AUX22;
- case MA_CHANNEL_24: return MA_PA_CHANNEL_POSITION_AUX23;
- case MA_CHANNEL_25: return MA_PA_CHANNEL_POSITION_AUX24;
- case MA_CHANNEL_26: return MA_PA_CHANNEL_POSITION_AUX25;
- case MA_CHANNEL_27: return MA_PA_CHANNEL_POSITION_AUX26;
- case MA_CHANNEL_28: return MA_PA_CHANNEL_POSITION_AUX27;
- case MA_CHANNEL_29: return MA_PA_CHANNEL_POSITION_AUX28;
- case MA_CHANNEL_30: return MA_PA_CHANNEL_POSITION_AUX29;
- case MA_CHANNEL_31: return MA_PA_CHANNEL_POSITION_AUX30;
- case MA_CHANNEL_32: return MA_PA_CHANNEL_POSITION_AUX31;
- default: return (ma_pa_channel_position_t)position;
- }
+ ma_pcm_s32_to_s24__optimized(dst, src, count, ditherMode);
}
#endif
-
-static ma_result ma_wait_for_operation__pulse(ma_context* pContext, ma_pa_operation* pOP)
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE void ma_pcm_s32_to_s24__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- int resultPA;
- ma_pa_operation_state_t state;
-
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pOP != NULL);
+ ma_pcm_s32_to_s24__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE void ma_pcm_s32_to_s24__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s32_to_s24__optimized(dst, src, count, ditherMode);
+}
+#endif
- for (;;) {
- state = ((ma_pa_operation_get_state_proc)pContext->pulse.pa_operation_get_state)(pOP);
- if (state != MA_PA_OPERATION_RUNNING) {
- break; /* Done. */
+MA_API void ma_pcm_s32_to_s24(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_s32_to_s24__reference(dst, src, count, ditherMode);
+#else
+ # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
+ if (ma_has_avx2()) {
+ ma_pcm_s32_to_s24__avx2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
+ if (ma_has_sse2()) {
+ ma_pcm_s32_to_s24__sse2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
+ if (ma_has_neon()) {
+ ma_pcm_s32_to_s24__neon(dst, src, count, ditherMode);
+ } else
+ #endif
+ {
+ ma_pcm_s32_to_s24__optimized(dst, src, count, ditherMode);
}
+#endif
+}
- resultPA = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pContext->pulse.pMainLoop, 1, NULL);
- if (resultPA < 0) {
- return ma_result_from_pulse(resultPA);
- }
- }
- return MA_SUCCESS;
+MA_API void ma_pcm_s32_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ (void)ditherMode;
+
+ ma_copy_memory_64(dst, src, count * sizeof(ma_int32));
}
-static ma_result ma_wait_for_operation_and_unref__pulse(ma_context* pContext, ma_pa_operation* pOP)
+
+static MA_INLINE void ma_pcm_s32_to_f32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- ma_result result;
+ float* dst_f32 = (float*)dst;
+ const ma_int32* src_s32 = (const ma_int32*)src;
- if (pOP == NULL) {
- return MA_INVALID_ARGS;
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ double x = src_s32[i];
+
+#if 0
+ x = x + 2147483648.0;
+ x = x * 0.0000000004656612873077392578125;
+ x = x - 1;
+#else
+ x = x / 2147483648.0;
+#endif
+
+ dst_f32[i] = (float)x;
}
- result = ma_wait_for_operation__pulse(pContext, pOP);
- ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP);
+ (void)ditherMode; /* No dithering for s32 -> f32. */
+}
- return result;
+static MA_INLINE void ma_pcm_s32_to_f32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s32_to_f32__reference(dst, src, count, ditherMode);
}
-static ma_result ma_context_wait_for_pa_context_to_connect__pulse(ma_context* pContext)
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE void ma_pcm_s32_to_f32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- int resultPA;
- ma_pa_context_state_t state;
+ ma_pcm_s32_to_f32__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE void ma_pcm_s32_to_f32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s32_to_f32__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE void ma_pcm_s32_to_f32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_s32_to_f32__optimized(dst, src, count, ditherMode);
+}
+#endif
- for (;;) {
- state = ((ma_pa_context_get_state_proc)pContext->pulse.pa_context_get_state)((ma_pa_context*)pContext->pulse.pPulseContext);
- if (state == MA_PA_CONTEXT_READY) {
- break; /* Done. */
+MA_API void ma_pcm_s32_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_s32_to_f32__reference(dst, src, count, ditherMode);
+#else
+ # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
+ if (ma_has_avx2()) {
+ ma_pcm_s32_to_f32__avx2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
+ if (ma_has_sse2()) {
+ ma_pcm_s32_to_f32__sse2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
+ if (ma_has_neon()) {
+ ma_pcm_s32_to_f32__neon(dst, src, count, ditherMode);
+ } else
+ #endif
+ {
+ ma_pcm_s32_to_f32__optimized(dst, src, count, ditherMode);
}
+#endif
+}
- if (state == MA_PA_CONTEXT_FAILED || state == MA_PA_CONTEXT_TERMINATED) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[PulseAudio] An error occurred while connecting the PulseAudio context.", MA_ERROR);
- }
- resultPA = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pContext->pulse.pMainLoop, 1, NULL);
- if (resultPA < 0) {
- return ma_result_from_pulse(resultPA);
+static MA_INLINE void ma_pcm_interleave_s32__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
+{
+ ma_int32* dst_s32 = (ma_int32*)dst;
+ const ma_int32** src_s32 = (const ma_int32**)src;
+
+ ma_uint64 iFrame;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ dst_s32[iFrame*channels + iChannel] = src_s32[iChannel][iFrame];
}
}
+}
- /* Should never get here. */
- return MA_SUCCESS;
+static MA_INLINE void ma_pcm_interleave_s32__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
+{
+ ma_pcm_interleave_s32__reference(dst, src, frameCount, channels);
}
-static ma_result ma_context_wait_for_pa_stream_to_connect__pulse(ma_context* pContext, ma_pa_stream* pStream)
+MA_API void ma_pcm_interleave_s32(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
{
- int resultPA;
- ma_pa_stream_state_t state;
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_interleave_s32__reference(dst, src, frameCount, channels);
+#else
+ ma_pcm_interleave_s32__optimized(dst, src, frameCount, channels);
+#endif
+}
- for (;;) {
- state = ((ma_pa_stream_get_state_proc)pContext->pulse.pa_stream_get_state)(pStream);
- if (state == MA_PA_STREAM_READY) {
- break; /* Done. */
- }
- if (state == MA_PA_STREAM_FAILED || state == MA_PA_STREAM_TERMINATED) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[PulseAudio] An error occurred while connecting the PulseAudio stream.", MA_ERROR);
- }
+static MA_INLINE void ma_pcm_deinterleave_s32__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
+{
+ ma_int32** dst_s32 = (ma_int32**)dst;
+ const ma_int32* src_s32 = (const ma_int32*)src;
- resultPA = ((ma_pa_mainloop_iterate_proc)pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pContext->pulse.pMainLoop, 1, NULL);
- if (resultPA < 0) {
- return ma_result_from_pulse(resultPA);
+ ma_uint64 iFrame;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ dst_s32[iChannel][iFrame] = src_s32[iFrame*channels + iChannel];
}
}
+}
- return MA_SUCCESS;
+static MA_INLINE void ma_pcm_deinterleave_s32__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
+{
+ ma_pcm_deinterleave_s32__reference(dst, src, frameCount, channels);
+}
+
+MA_API void ma_pcm_deinterleave_s32(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_deinterleave_s32__reference(dst, src, frameCount, channels);
+#else
+ ma_pcm_deinterleave_s32__optimized(dst, src, frameCount, channels);
+#endif
}
-static void ma_device_sink_info_callback(ma_pa_context* pPulseContext, const ma_pa_sink_info* pInfo, int endOfList, void* pUserData)
+/* f32 */
+static MA_INLINE void ma_pcm_f32_to_u8__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- ma_pa_sink_info* pInfoOut;
+ ma_uint64 i;
- if (endOfList > 0) {
- return;
+ ma_uint8* dst_u8 = (ma_uint8*)dst;
+ const float* src_f32 = (const float*)src;
+
+ float ditherMin = 0;
+ float ditherMax = 0;
+ if (ditherMode != ma_dither_mode_none) {
+ ditherMin = 1.0f / -128;
+ ditherMax = 1.0f / 127;
}
- pInfoOut = (ma_pa_sink_info*)pUserData;
- MA_ASSERT(pInfoOut != NULL);
+ for (i = 0; i < count; i += 1) {
+ float x = src_f32[i];
+ x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax);
+ x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */
+ x = x + 1; /* -1..1 to 0..2 */
+ x = x * 127.5f; /* 0..2 to 0..255 */
- *pInfoOut = *pInfo;
+ dst_u8[i] = (ma_uint8)x;
+ }
+}
- (void)pPulseContext; /* Unused. */
+static MA_INLINE void ma_pcm_f32_to_u8__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_f32_to_u8__reference(dst, src, count, ditherMode);
}
-static void ma_device_source_info_callback(ma_pa_context* pPulseContext, const ma_pa_source_info* pInfo, int endOfList, void* pUserData)
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE void ma_pcm_f32_to_u8__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- ma_pa_source_info* pInfoOut;
+ ma_pcm_f32_to_u8__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE void ma_pcm_f32_to_u8__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_f32_to_u8__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE void ma_pcm_f32_to_u8__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_f32_to_u8__optimized(dst, src, count, ditherMode);
+}
+#endif
- if (endOfList > 0) {
- return;
- }
+MA_API void ma_pcm_f32_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_f32_to_u8__reference(dst, src, count, ditherMode);
+#else
+ # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
+ if (ma_has_avx2()) {
+ ma_pcm_f32_to_u8__avx2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
+ if (ma_has_sse2()) {
+ ma_pcm_f32_to_u8__sse2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
+ if (ma_has_neon()) {
+ ma_pcm_f32_to_u8__neon(dst, src, count, ditherMode);
+ } else
+ #endif
+ {
+ ma_pcm_f32_to_u8__optimized(dst, src, count, ditherMode);
+ }
+#endif
+}
- pInfoOut = (ma_pa_source_info*)pUserData;
- MA_ASSERT(pInfoOut != NULL);
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+static MA_INLINE void ma_pcm_f32_to_s16__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_uint64 i;
- *pInfoOut = *pInfo;
+ ma_int16* dst_s16 = (ma_int16*)dst;
+ const float* src_f32 = (const float*)src;
- (void)pPulseContext; /* Unused. */
-}
+ float ditherMin = 0;
+ float ditherMax = 0;
+ if (ditherMode != ma_dither_mode_none) {
+ ditherMin = 1.0f / -32768;
+ ditherMax = 1.0f / 32767;
+ }
-static void ma_device_sink_name_callback(ma_pa_context* pPulseContext, const ma_pa_sink_info* pInfo, int endOfList, void* pUserData)
-{
- ma_device* pDevice;
+ for (i = 0; i < count; i += 1) {
+ float x = src_f32[i];
+ x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax);
+ x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */
+
+#if 0
+ /* The accurate way. */
+ x = x + 1; /* -1..1 to 0..2 */
+ x = x * 32767.5f; /* 0..2 to 0..65535 */
+ x = x - 32768.0f; /* 0...65535 to -32768..32767 */
+#else
+ /* The fast way. */
+ x = x * 32767.0f; /* -1..1 to -32767..32767 */
+#endif
- if (endOfList > 0) {
- return;
+ dst_s16[i] = (ma_int16)x;
}
-
- pDevice = (ma_device*)pUserData;
- MA_ASSERT(pDevice != NULL);
-
- ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), pInfo->description, (size_t)-1);
-
- (void)pPulseContext; /* Unused. */
}
-
-static void ma_device_source_name_callback(ma_pa_context* pPulseContext, const ma_pa_source_info* pInfo, int endOfList, void* pUserData)
+#else
+static MA_INLINE void ma_pcm_f32_to_s16__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- ma_device* pDevice;
+ ma_uint64 i;
+ ma_uint64 i4;
+ ma_uint64 count4;
- if (endOfList > 0) {
- return;
+ ma_int16* dst_s16 = (ma_int16*)dst;
+ const float* src_f32 = (const float*)src;
+
+ float ditherMin = 0;
+ float ditherMax = 0;
+ if (ditherMode != ma_dither_mode_none) {
+ ditherMin = 1.0f / -32768;
+ ditherMax = 1.0f / 32767;
}
- pDevice = (ma_device*)pUserData;
- MA_ASSERT(pDevice != NULL);
+ /* Unrolled. */
+ i = 0;
+ count4 = count >> 2;
+ for (i4 = 0; i4 < count4; i4 += 1) {
+ float d0 = ma_dither_f32(ditherMode, ditherMin, ditherMax);
+ float d1 = ma_dither_f32(ditherMode, ditherMin, ditherMax);
+ float d2 = ma_dither_f32(ditherMode, ditherMin, ditherMax);
+ float d3 = ma_dither_f32(ditherMode, ditherMin, ditherMax);
- ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), pInfo->description, (size_t)-1);
+ float x0 = src_f32[i+0];
+ float x1 = src_f32[i+1];
+ float x2 = src_f32[i+2];
+ float x3 = src_f32[i+3];
- (void)pPulseContext; /* Unused. */
-}
+ x0 = x0 + d0;
+ x1 = x1 + d1;
+ x2 = x2 + d2;
+ x3 = x3 + d3;
+ x0 = ((x0 < -1) ? -1 : ((x0 > 1) ? 1 : x0));
+ x1 = ((x1 < -1) ? -1 : ((x1 > 1) ? 1 : x1));
+ x2 = ((x2 < -1) ? -1 : ((x2 > 1) ? 1 : x2));
+ x3 = ((x3 < -1) ? -1 : ((x3 > 1) ? 1 : x3));
-static ma_result ma_context_get_sink_info__pulse(ma_context* pContext, const char* pDeviceName, ma_pa_sink_info* pSinkInfo)
-{
- ma_pa_operation* pOP;
+ x0 = x0 * 32767.0f;
+ x1 = x1 * 32767.0f;
+ x2 = x2 * 32767.0f;
+ x3 = x3 * 32767.0f;
- pOP = ((ma_pa_context_get_sink_info_by_name_proc)pContext->pulse.pa_context_get_sink_info_by_name)((ma_pa_context*)pContext->pulse.pPulseContext, pDeviceName, ma_device_sink_info_callback, pSinkInfo);
- if (pOP == NULL) {
- return MA_ERROR;
- }
+ dst_s16[i+0] = (ma_int16)x0;
+ dst_s16[i+1] = (ma_int16)x1;
+ dst_s16[i+2] = (ma_int16)x2;
+ dst_s16[i+3] = (ma_int16)x3;
- return ma_wait_for_operation_and_unref__pulse(pContext, pOP);
-}
+ i += 4;
+ }
-static ma_result ma_context_get_source_info__pulse(ma_context* pContext, const char* pDeviceName, ma_pa_source_info* pSourceInfo)
-{
- ma_pa_operation* pOP;
+ /* Leftover. */
+ for (; i < count; i += 1) {
+ float x = src_f32[i];
+ x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax);
+ x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */
+ x = x * 32767.0f; /* -1..1 to -32767..32767 */
- pOP = ((ma_pa_context_get_source_info_by_name_proc)pContext->pulse.pa_context_get_source_info_by_name)((ma_pa_context*)pContext->pulse.pPulseContext, pDeviceName, ma_device_source_info_callback, pSourceInfo);
- if (pOP == NULL) {
- return MA_ERROR;
+ dst_s16[i] = (ma_int16)x;
}
-
- return ma_wait_for_operation_and_unref__pulse(pContext, pOP);;
}
-static ma_result ma_context_get_default_device_index__pulse(ma_context* pContext, ma_device_type deviceType, ma_uint32* pIndex)
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE void ma_pcm_f32_to_s16__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- ma_result result;
-
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pIndex != NULL);
+ ma_uint64 i;
+ ma_uint64 i8;
+ ma_uint64 count8;
+ ma_int16* dst_s16;
+ const float* src_f32;
+ float ditherMin;
+ float ditherMax;
- if (pIndex != NULL) {
- *pIndex = (ma_uint32)-1;
+ /* Both the input and output buffers need to be aligned to 16 bytes. */
+ if ((((ma_uintptr)dst & 15) != 0) || (((ma_uintptr)src & 15) != 0)) {
+ ma_pcm_f32_to_s16__optimized(dst, src, count, ditherMode);
+ return;
}
- if (deviceType == ma_device_type_playback) {
- ma_pa_sink_info sinkInfo;
- result = ma_context_get_sink_info__pulse(pContext, NULL, &sinkInfo);
- if (result != MA_SUCCESS) {
- return result;
- }
+ dst_s16 = (ma_int16*)dst;
+ src_f32 = (const float*)src;
- if (pIndex != NULL) {
- *pIndex = sinkInfo.index;
- }
+ ditherMin = 0;
+ ditherMax = 0;
+ if (ditherMode != ma_dither_mode_none) {
+ ditherMin = 1.0f / -32768;
+ ditherMax = 1.0f / 32767;
}
- if (deviceType == ma_device_type_capture) {
- ma_pa_source_info sourceInfo;
- result = ma_context_get_source_info__pulse(pContext, NULL, &sourceInfo);
- if (result != MA_SUCCESS) {
- return result;
- }
+ i = 0;
- if (pIndex != NULL) {
- *pIndex = sourceInfo.index;
- }
- }
+ /* SSE2. SSE allows us to output 8 s16's at a time which means our loop is unrolled 8 times. */
+ count8 = count >> 3;
+ for (i8 = 0; i8 < count8; i8 += 1) {
+ __m128 d0;
+ __m128 d1;
+ __m128 x0;
+ __m128 x1;
- return MA_SUCCESS;
-}
+ if (ditherMode == ma_dither_mode_none) {
+ d0 = _mm_set1_ps(0);
+ d1 = _mm_set1_ps(0);
+ } else if (ditherMode == ma_dither_mode_rectangle) {
+ d0 = _mm_set_ps(
+ ma_dither_f32_rectangle(ditherMin, ditherMax),
+ ma_dither_f32_rectangle(ditherMin, ditherMax),
+ ma_dither_f32_rectangle(ditherMin, ditherMax),
+ ma_dither_f32_rectangle(ditherMin, ditherMax)
+ );
+ d1 = _mm_set_ps(
+ ma_dither_f32_rectangle(ditherMin, ditherMax),
+ ma_dither_f32_rectangle(ditherMin, ditherMax),
+ ma_dither_f32_rectangle(ditherMin, ditherMax),
+ ma_dither_f32_rectangle(ditherMin, ditherMax)
+ );
+ } else {
+ d0 = _mm_set_ps(
+ ma_dither_f32_triangle(ditherMin, ditherMax),
+ ma_dither_f32_triangle(ditherMin, ditherMax),
+ ma_dither_f32_triangle(ditherMin, ditherMax),
+ ma_dither_f32_triangle(ditherMin, ditherMax)
+ );
+ d1 = _mm_set_ps(
+ ma_dither_f32_triangle(ditherMin, ditherMax),
+ ma_dither_f32_triangle(ditherMin, ditherMax),
+ ma_dither_f32_triangle(ditherMin, ditherMax),
+ ma_dither_f32_triangle(ditherMin, ditherMax)
+ );
+ }
+ x0 = *((__m128*)(src_f32 + i) + 0);
+ x1 = *((__m128*)(src_f32 + i) + 1);
-typedef struct
-{
- ma_context* pContext;
- ma_enum_devices_callback_proc callback;
- void* pUserData;
- ma_bool32 isTerminated;
- ma_uint32 defaultDeviceIndexPlayback;
- ma_uint32 defaultDeviceIndexCapture;
-} ma_context_enumerate_devices_callback_data__pulse;
+ x0 = _mm_add_ps(x0, d0);
+ x1 = _mm_add_ps(x1, d1);
-static void ma_context_enumerate_devices_sink_callback__pulse(ma_pa_context* pPulseContext, const ma_pa_sink_info* pSinkInfo, int endOfList, void* pUserData)
-{
- ma_context_enumerate_devices_callback_data__pulse* pData = (ma_context_enumerate_devices_callback_data__pulse*)pUserData;
- ma_device_info deviceInfo;
+ x0 = _mm_mul_ps(x0, _mm_set1_ps(32767.0f));
+ x1 = _mm_mul_ps(x1, _mm_set1_ps(32767.0f));
- MA_ASSERT(pData != NULL);
+ _mm_stream_si128(((__m128i*)(dst_s16 + i)), _mm_packs_epi32(_mm_cvttps_epi32(x0), _mm_cvttps_epi32(x1)));
- if (endOfList || pData->isTerminated) {
- return;
+ i += 8;
}
- MA_ZERO_OBJECT(&deviceInfo);
-
- /* The name from PulseAudio is the ID for miniaudio. */
- if (pSinkInfo->name != NULL) {
- ma_strncpy_s(deviceInfo.id.pulse, sizeof(deviceInfo.id.pulse), pSinkInfo->name, (size_t)-1);
- }
- /* The description from PulseAudio is the name for miniaudio. */
- if (pSinkInfo->description != NULL) {
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), pSinkInfo->description, (size_t)-1);
- }
+ /* Leftover. */
+ for (; i < count; i += 1) {
+ float x = src_f32[i];
+ x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax);
+ x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */
+ x = x * 32767.0f; /* -1..1 to -32767..32767 */
- if (pSinkInfo->index == pData->defaultDeviceIndexPlayback) {
- deviceInfo.isDefault = MA_TRUE;
+ dst_s16[i] = (ma_int16)x;
}
-
- pData->isTerminated = !pData->callback(pData->pContext, ma_device_type_playback, &deviceInfo, pData->pUserData);
-
- (void)pPulseContext; /* Unused. */
}
+#endif /* SSE2 */
-static void ma_context_enumerate_devices_source_callback__pulse(ma_pa_context* pPulseContext, const ma_pa_source_info* pSourceInfo, int endOfList, void* pUserData)
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE void ma_pcm_f32_to_s16__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- ma_context_enumerate_devices_callback_data__pulse* pData = (ma_context_enumerate_devices_callback_data__pulse*)pUserData;
- ma_device_info deviceInfo;
-
- MA_ASSERT(pData != NULL);
+ ma_uint64 i;
+ ma_uint64 i16;
+ ma_uint64 count16;
+ ma_int16* dst_s16;
+ const float* src_f32;
+ float ditherMin;
+ float ditherMax;
- if (endOfList || pData->isTerminated) {
+ /* Both the input and output buffers need to be aligned to 32 bytes. */
+ if ((((ma_uintptr)dst & 31) != 0) || (((ma_uintptr)src & 31) != 0)) {
+ ma_pcm_f32_to_s16__optimized(dst, src, count, ditherMode);
return;
}
- MA_ZERO_OBJECT(&deviceInfo);
-
- /* The name from PulseAudio is the ID for miniaudio. */
- if (pSourceInfo->name != NULL) {
- ma_strncpy_s(deviceInfo.id.pulse, sizeof(deviceInfo.id.pulse), pSourceInfo->name, (size_t)-1);
- }
-
- /* The description from PulseAudio is the name for miniaudio. */
- if (pSourceInfo->description != NULL) {
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), pSourceInfo->description, (size_t)-1);
- }
+ dst_s16 = (ma_int16*)dst;
+ src_f32 = (const float*)src;
- if (pSourceInfo->index == pData->defaultDeviceIndexCapture) {
- deviceInfo.isDefault = MA_TRUE;
+ ditherMin = 0;
+ ditherMax = 0;
+ if (ditherMode != ma_dither_mode_none) {
+ ditherMin = 1.0f / -32768;
+ ditherMax = 1.0f / 32767;
}
- pData->isTerminated = !pData->callback(pData->pContext, ma_device_type_capture, &deviceInfo, pData->pUserData);
+ i = 0;
- (void)pPulseContext; /* Unused. */
-}
+ /* AVX2. AVX2 allows us to output 16 s16's at a time which means our loop is unrolled 16 times. */
+ count16 = count >> 4;
+ for (i16 = 0; i16 < count16; i16 += 1) {
+ __m256 d0;
+ __m256 d1;
+ __m256 x0;
+ __m256 x1;
+ __m256i i0;
+ __m256i i1;
+ __m256i p0;
+ __m256i p1;
+ __m256i r;
-static ma_result ma_context_enumerate_devices__pulse(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
-{
- ma_result result = MA_SUCCESS;
- ma_context_enumerate_devices_callback_data__pulse callbackData;
- ma_pa_operation* pOP = NULL;
+ if (ditherMode == ma_dither_mode_none) {
+ d0 = _mm256_set1_ps(0);
+ d1 = _mm256_set1_ps(0);
+ } else if (ditherMode == ma_dither_mode_rectangle) {
+ d0 = _mm256_set_ps(
+ ma_dither_f32_rectangle(ditherMin, ditherMax),
+ ma_dither_f32_rectangle(ditherMin, ditherMax),
+ ma_dither_f32_rectangle(ditherMin, ditherMax),
+ ma_dither_f32_rectangle(ditherMin, ditherMax),
+ ma_dither_f32_rectangle(ditherMin, ditherMax),
+ ma_dither_f32_rectangle(ditherMin, ditherMax),
+ ma_dither_f32_rectangle(ditherMin, ditherMax),
+ ma_dither_f32_rectangle(ditherMin, ditherMax)
+ );
+ d1 = _mm256_set_ps(
+ ma_dither_f32_rectangle(ditherMin, ditherMax),
+ ma_dither_f32_rectangle(ditherMin, ditherMax),
+ ma_dither_f32_rectangle(ditherMin, ditherMax),
+ ma_dither_f32_rectangle(ditherMin, ditherMax),
+ ma_dither_f32_rectangle(ditherMin, ditherMax),
+ ma_dither_f32_rectangle(ditherMin, ditherMax),
+ ma_dither_f32_rectangle(ditherMin, ditherMax),
+ ma_dither_f32_rectangle(ditherMin, ditherMax)
+ );
+ } else {
+ d0 = _mm256_set_ps(
+ ma_dither_f32_triangle(ditherMin, ditherMax),
+ ma_dither_f32_triangle(ditherMin, ditherMax),
+ ma_dither_f32_triangle(ditherMin, ditherMax),
+ ma_dither_f32_triangle(ditherMin, ditherMax),
+ ma_dither_f32_triangle(ditherMin, ditherMax),
+ ma_dither_f32_triangle(ditherMin, ditherMax),
+ ma_dither_f32_triangle(ditherMin, ditherMax),
+ ma_dither_f32_triangle(ditherMin, ditherMax)
+ );
+ d1 = _mm256_set_ps(
+ ma_dither_f32_triangle(ditherMin, ditherMax),
+ ma_dither_f32_triangle(ditherMin, ditherMax),
+ ma_dither_f32_triangle(ditherMin, ditherMax),
+ ma_dither_f32_triangle(ditherMin, ditherMax),
+ ma_dither_f32_triangle(ditherMin, ditherMax),
+ ma_dither_f32_triangle(ditherMin, ditherMax),
+ ma_dither_f32_triangle(ditherMin, ditherMax),
+ ma_dither_f32_triangle(ditherMin, ditherMax)
+ );
+ }
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(callback != NULL);
+ x0 = *((__m256*)(src_f32 + i) + 0);
+ x1 = *((__m256*)(src_f32 + i) + 1);
- callbackData.pContext = pContext;
- callbackData.callback = callback;
- callbackData.pUserData = pUserData;
- callbackData.isTerminated = MA_FALSE;
- callbackData.defaultDeviceIndexPlayback = (ma_uint32)-1;
- callbackData.defaultDeviceIndexCapture = (ma_uint32)-1;
+ x0 = _mm256_add_ps(x0, d0);
+ x1 = _mm256_add_ps(x1, d1);
- /* We need to get the index of the default devices. */
- ma_context_get_default_device_index__pulse(pContext, ma_device_type_playback, &callbackData.defaultDeviceIndexPlayback);
- ma_context_get_default_device_index__pulse(pContext, ma_device_type_capture, &callbackData.defaultDeviceIndexCapture);
+ x0 = _mm256_mul_ps(x0, _mm256_set1_ps(32767.0f));
+ x1 = _mm256_mul_ps(x1, _mm256_set1_ps(32767.0f));
- /* Playback. */
- if (!callbackData.isTerminated) {
- pOP = ((ma_pa_context_get_sink_info_list_proc)pContext->pulse.pa_context_get_sink_info_list)((ma_pa_context*)(pContext->pulse.pPulseContext), ma_context_enumerate_devices_sink_callback__pulse, &callbackData);
- if (pOP == NULL) {
- result = MA_ERROR;
- goto done;
- }
+ /* Computing the final result is a little more complicated for AVX2 than SSE2. */
+ i0 = _mm256_cvttps_epi32(x0);
+ i1 = _mm256_cvttps_epi32(x1);
+ p0 = _mm256_permute2x128_si256(i0, i1, 0 | 32);
+ p1 = _mm256_permute2x128_si256(i0, i1, 1 | 48);
+ r = _mm256_packs_epi32(p0, p1);
- result = ma_wait_for_operation__pulse(pContext, pOP);
- ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP);
+ _mm256_stream_si256(((__m256i*)(dst_s16 + i)), r);
- if (result != MA_SUCCESS) {
- goto done;
- }
+ i += 16;
}
- /* Capture. */
- if (!callbackData.isTerminated) {
- pOP = ((ma_pa_context_get_source_info_list_proc)pContext->pulse.pa_context_get_source_info_list)((ma_pa_context*)(pContext->pulse.pPulseContext), ma_context_enumerate_devices_source_callback__pulse, &callbackData);
- if (pOP == NULL) {
- result = MA_ERROR;
- goto done;
- }
-
- result = ma_wait_for_operation__pulse(pContext, pOP);
- ((ma_pa_operation_unref_proc)pContext->pulse.pa_operation_unref)(pOP);
+ /* Leftover. */
+ for (; i < count; i += 1) {
+ float x = src_f32[i];
+ x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax);
+ x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */
+ x = x * 32767.0f; /* -1..1 to -32767..32767 */
- if (result != MA_SUCCESS) {
- goto done;
- }
+ dst_s16[i] = (ma_int16)x;
}
-
-done:
- return result;
}
+#endif /* AVX2 */
-
-typedef struct
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE void ma_pcm_f32_to_s16__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- ma_device_info* pDeviceInfo;
- ma_uint32 defaultDeviceIndex;
- ma_bool32 foundDevice;
-} ma_context_get_device_info_callback_data__pulse;
+ ma_uint64 i;
+ ma_uint64 i8;
+ ma_uint64 count8;
+ ma_int16* dst_s16;
+ const float* src_f32;
+ float ditherMin;
+ float ditherMax;
-static void ma_context_get_device_info_sink_callback__pulse(ma_pa_context* pPulseContext, const ma_pa_sink_info* pInfo, int endOfList, void* pUserData)
-{
- ma_context_get_device_info_callback_data__pulse* pData = (ma_context_get_device_info_callback_data__pulse*)pUserData;
+ if (!ma_has_neon()) {
+ return ma_pcm_f32_to_s16__optimized(dst, src, count, ditherMode);
+ }
- if (endOfList > 0) {
+ /* Both the input and output buffers need to be aligned to 16 bytes. */
+ if ((((ma_uintptr)dst & 15) != 0) || (((ma_uintptr)src & 15) != 0)) {
+ ma_pcm_f32_to_s16__optimized(dst, src, count, ditherMode);
return;
}
- MA_ASSERT(pData != NULL);
- pData->foundDevice = MA_TRUE;
-
- if (pInfo->name != NULL) {
- ma_strncpy_s(pData->pDeviceInfo->id.pulse, sizeof(pData->pDeviceInfo->id.pulse), pInfo->name, (size_t)-1);
- }
+ dst_s16 = (ma_int16*)dst;
+ src_f32 = (const float*)src;
- if (pInfo->description != NULL) {
- ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), pInfo->description, (size_t)-1);
+ ditherMin = 0;
+ ditherMax = 0;
+ if (ditherMode != ma_dither_mode_none) {
+ ditherMin = 1.0f / -32768;
+ ditherMax = 1.0f / 32767;
}
- /*
- We're just reporting a single data format here. I think technically PulseAudio might support
- all formats, but I don't trust that PulseAudio will do *anything* right, so I'm just going to
- report the "native" device format.
- */
- pData->pDeviceInfo->nativeDataFormats[0].format = ma_format_from_pulse(pInfo->sample_spec.format);
- pData->pDeviceInfo->nativeDataFormats[0].channels = pInfo->sample_spec.channels;
- pData->pDeviceInfo->nativeDataFormats[0].sampleRate = pInfo->sample_spec.rate;
- pData->pDeviceInfo->nativeDataFormats[0].flags = 0;
- pData->pDeviceInfo->nativeDataFormatCount = 1;
+ i = 0;
- if (pData->defaultDeviceIndex == pInfo->index) {
- pData->pDeviceInfo->isDefault = MA_TRUE;
- }
+ /* NEON. NEON allows us to output 8 s16's at a time which means our loop is unrolled 8 times. */
+ count8 = count >> 3;
+ for (i8 = 0; i8 < count8; i8 += 1) {
+ float32x4_t d0;
+ float32x4_t d1;
+ float32x4_t x0;
+ float32x4_t x1;
+ int32x4_t i0;
+ int32x4_t i1;
- (void)pPulseContext; /* Unused. */
-}
+ if (ditherMode == ma_dither_mode_none) {
+ d0 = vmovq_n_f32(0);
+ d1 = vmovq_n_f32(0);
+ } else if (ditherMode == ma_dither_mode_rectangle) {
+ float d0v[4];
+ d0v[0] = ma_dither_f32_rectangle(ditherMin, ditherMax);
+ d0v[1] = ma_dither_f32_rectangle(ditherMin, ditherMax);
+ d0v[2] = ma_dither_f32_rectangle(ditherMin, ditherMax);
+ d0v[3] = ma_dither_f32_rectangle(ditherMin, ditherMax);
+ d0 = vld1q_f32(d0v);
-static void ma_context_get_device_info_source_callback__pulse(ma_pa_context* pPulseContext, const ma_pa_source_info* pInfo, int endOfList, void* pUserData)
-{
- ma_context_get_device_info_callback_data__pulse* pData = (ma_context_get_device_info_callback_data__pulse*)pUserData;
+ float d1v[4];
+ d1v[0] = ma_dither_f32_rectangle(ditherMin, ditherMax);
+ d1v[1] = ma_dither_f32_rectangle(ditherMin, ditherMax);
+ d1v[2] = ma_dither_f32_rectangle(ditherMin, ditherMax);
+ d1v[3] = ma_dither_f32_rectangle(ditherMin, ditherMax);
+ d1 = vld1q_f32(d1v);
+ } else {
+ float d0v[4];
+ d0v[0] = ma_dither_f32_triangle(ditherMin, ditherMax);
+ d0v[1] = ma_dither_f32_triangle(ditherMin, ditherMax);
+ d0v[2] = ma_dither_f32_triangle(ditherMin, ditherMax);
+ d0v[3] = ma_dither_f32_triangle(ditherMin, ditherMax);
+ d0 = vld1q_f32(d0v);
- if (endOfList > 0) {
- return;
- }
+ float d1v[4];
+ d1v[0] = ma_dither_f32_triangle(ditherMin, ditherMax);
+ d1v[1] = ma_dither_f32_triangle(ditherMin, ditherMax);
+ d1v[2] = ma_dither_f32_triangle(ditherMin, ditherMax);
+ d1v[3] = ma_dither_f32_triangle(ditherMin, ditherMax);
+ d1 = vld1q_f32(d1v);
+ }
- MA_ASSERT(pData != NULL);
- pData->foundDevice = MA_TRUE;
+ x0 = *((float32x4_t*)(src_f32 + i) + 0);
+ x1 = *((float32x4_t*)(src_f32 + i) + 1);
- if (pInfo->name != NULL) {
- ma_strncpy_s(pData->pDeviceInfo->id.pulse, sizeof(pData->pDeviceInfo->id.pulse), pInfo->name, (size_t)-1);
- }
+ x0 = vaddq_f32(x0, d0);
+ x1 = vaddq_f32(x1, d1);
- if (pInfo->description != NULL) {
- ma_strncpy_s(pData->pDeviceInfo->name, sizeof(pData->pDeviceInfo->name), pInfo->description, (size_t)-1);
- }
+ x0 = vmulq_n_f32(x0, 32767.0f);
+ x1 = vmulq_n_f32(x1, 32767.0f);
- /*
- We're just reporting a single data format here. I think technically PulseAudio might support
- all formats, but I don't trust that PulseAudio will do *anything* right, so I'm just going to
- report the "native" device format.
- */
- pData->pDeviceInfo->nativeDataFormats[0].format = ma_format_from_pulse(pInfo->sample_spec.format);
- pData->pDeviceInfo->nativeDataFormats[0].channels = pInfo->sample_spec.channels;
- pData->pDeviceInfo->nativeDataFormats[0].sampleRate = pInfo->sample_spec.rate;
- pData->pDeviceInfo->nativeDataFormats[0].flags = 0;
- pData->pDeviceInfo->nativeDataFormatCount = 1;
+ i0 = vcvtq_s32_f32(x0);
+ i1 = vcvtq_s32_f32(x1);
+ *((int16x8_t*)(dst_s16 + i)) = vcombine_s16(vqmovn_s32(i0), vqmovn_s32(i1));
- if (pData->defaultDeviceIndex == pInfo->index) {
- pData->pDeviceInfo->isDefault = MA_TRUE;
+ i += 8;
}
- (void)pPulseContext; /* Unused. */
+
+ /* Leftover. */
+ for (; i < count; i += 1) {
+ float x = src_f32[i];
+ x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax);
+ x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */
+ x = x * 32767.0f; /* -1..1 to -32767..32767 */
+
+ dst_s16[i] = (ma_int16)x;
+ }
}
+#endif /* Neon */
+#endif /* MA_USE_REFERENCE_CONVERSION_APIS */
-static ma_result ma_context_get_device_info__pulse(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+MA_API void ma_pcm_f32_to_s16(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- ma_result result = MA_SUCCESS;
- ma_context_get_device_info_callback_data__pulse callbackData;
- ma_pa_operation* pOP = NULL;
- const char* pDeviceName = NULL;
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_f32_to_s16__reference(dst, src, count, ditherMode);
+#else
+ # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
+ if (ma_has_avx2()) {
+ ma_pcm_f32_to_s16__avx2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
+ if (ma_has_sse2()) {
+ ma_pcm_f32_to_s16__sse2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
+ if (ma_has_neon()) {
+ ma_pcm_f32_to_s16__neon(dst, src, count, ditherMode);
+ } else
+ #endif
+ {
+ ma_pcm_f32_to_s16__optimized(dst, src, count, ditherMode);
+ }
+#endif
+}
- MA_ASSERT(pContext != NULL);
- callbackData.pDeviceInfo = pDeviceInfo;
- callbackData.foundDevice = MA_FALSE;
+static MA_INLINE void ma_pcm_f32_to_s24__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_uint8* dst_s24 = (ma_uint8*)dst;
+ const float* src_f32 = (const float*)src;
- if (pDeviceID != NULL) {
- pDeviceName = pDeviceID->pulse;
- } else {
- pDeviceName = NULL;
- }
+ ma_uint64 i;
+ for (i = 0; i < count; i += 1) {
+ ma_int32 r;
+ float x = src_f32[i];
+ x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */
- result = ma_context_get_default_device_index__pulse(pContext, deviceType, &callbackData.defaultDeviceIndex);
+#if 0
+ /* The accurate way. */
+ x = x + 1; /* -1..1 to 0..2 */
+ x = x * 8388607.5f; /* 0..2 to 0..16777215 */
+ x = x - 8388608.0f; /* 0..16777215 to -8388608..8388607 */
+#else
+ /* The fast way. */
+ x = x * 8388607.0f; /* -1..1 to -8388607..8388607 */
+#endif
- if (deviceType == ma_device_type_playback) {
- pOP = ((ma_pa_context_get_sink_info_by_name_proc)pContext->pulse.pa_context_get_sink_info_by_name)((ma_pa_context*)(pContext->pulse.pPulseContext), pDeviceName, ma_context_get_device_info_sink_callback__pulse, &callbackData);
- } else {
- pOP = ((ma_pa_context_get_source_info_by_name_proc)pContext->pulse.pa_context_get_source_info_by_name)((ma_pa_context*)(pContext->pulse.pPulseContext), pDeviceName, ma_context_get_device_info_source_callback__pulse, &callbackData);
+ r = (ma_int32)x;
+ dst_s24[(i*3)+0] = (ma_uint8)((r & 0x0000FF) >> 0);
+ dst_s24[(i*3)+1] = (ma_uint8)((r & 0x00FF00) >> 8);
+ dst_s24[(i*3)+2] = (ma_uint8)((r & 0xFF0000) >> 16);
}
- if (pOP != NULL) {
- ma_wait_for_operation_and_unref__pulse(pContext, pOP);
- } else {
- result = MA_ERROR;
- goto done;
- }
+ (void)ditherMode; /* No dithering for f32 -> s24. */
+}
- if (!callbackData.foundDevice) {
- result = MA_NO_DEVICE;
- goto done;
- }
+static MA_INLINE void ma_pcm_f32_to_s24__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_f32_to_s24__reference(dst, src, count, ditherMode);
+}
-done:
- return result;
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE void ma_pcm_f32_to_s24__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_f32_to_s24__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE void ma_pcm_f32_to_s24__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_f32_to_s24__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE void ma_pcm_f32_to_s24__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_f32_to_s24__optimized(dst, src, count, ditherMode);
}
+#endif
-static ma_result ma_device_uninit__pulse(ma_device* pDevice)
+MA_API void ma_pcm_f32_to_s24(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- ma_context* pContext;
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_f32_to_s24__reference(dst, src, count, ditherMode);
+#else
+ # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
+ if (ma_has_avx2()) {
+ ma_pcm_f32_to_s24__avx2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
+ if (ma_has_sse2()) {
+ ma_pcm_f32_to_s24__sse2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
+ if (ma_has_neon()) {
+ ma_pcm_f32_to_s24__neon(dst, src, count, ditherMode);
+ } else
+ #endif
+ {
+ ma_pcm_f32_to_s24__optimized(dst, src, count, ditherMode);
+ }
+#endif
+}
- MA_ASSERT(pDevice != NULL);
- pContext = pDevice->pContext;
- MA_ASSERT(pContext != NULL);
+static MA_INLINE void ma_pcm_f32_to_s32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_int32* dst_s32 = (ma_int32*)dst;
+ const float* src_f32 = (const float*)src;
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- ((ma_pa_stream_disconnect_proc)pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
- ((ma_pa_stream_unref_proc)pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
- }
+ ma_uint32 i;
+ for (i = 0; i < count; i += 1) {
+ double x = src_f32[i];
+ x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- ((ma_pa_stream_disconnect_proc)pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
- ((ma_pa_stream_unref_proc)pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
- }
+#if 0
+ /* The accurate way. */
+ x = x + 1; /* -1..1 to 0..2 */
+ x = x * 2147483647.5; /* 0..2 to 0..4294967295 */
+ x = x - 2147483648.0; /* 0...4294967295 to -2147483648..2147483647 */
+#else
+ /* The fast way. */
+ x = x * 2147483647.0; /* -1..1 to -2147483647..2147483647 */
+#endif
- if (pDevice->type == ma_device_type_duplex) {
- ma_duplex_rb_uninit(&pDevice->duplexRB);
+ dst_s32[i] = (ma_int32)x;
}
- return MA_SUCCESS;
+ (void)ditherMode; /* No dithering for f32 -> s32. */
}
-static ma_pa_buffer_attr ma_device__pa_buffer_attr_new(ma_uint32 periodSizeInFrames, ma_uint32 periods, const ma_pa_sample_spec* ss)
+static MA_INLINE void ma_pcm_f32_to_s32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- ma_pa_buffer_attr attr;
- attr.maxlength = periodSizeInFrames * periods * ma_get_bytes_per_frame(ma_format_from_pulse(ss->format), ss->channels);
- attr.tlength = attr.maxlength / periods;
- attr.prebuf = (ma_uint32)-1;
- attr.minreq = (ma_uint32)-1;
- attr.fragsize = attr.maxlength / periods;
-
- return attr;
+ ma_pcm_f32_to_s32__reference(dst, src, count, ditherMode);
}
-static ma_pa_stream* ma_context__pa_stream_new__pulse(ma_context* pContext, const char* pStreamName, const ma_pa_sample_spec* ss, const ma_pa_channel_map* cmap)
+#if defined(MA_SUPPORT_SSE2)
+static MA_INLINE void ma_pcm_f32_to_s32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- static int g_StreamCounter = 0;
- char actualStreamName[256];
-
- if (pStreamName != NULL) {
- ma_strncpy_s(actualStreamName, sizeof(actualStreamName), pStreamName, (size_t)-1);
- } else {
- ma_strcpy_s(actualStreamName, sizeof(actualStreamName), "miniaudio:");
- ma_itoa_s(g_StreamCounter, actualStreamName + 8, sizeof(actualStreamName)-8, 10); /* 8 = strlen("miniaudio:") */
- }
- g_StreamCounter += 1;
-
- return ((ma_pa_stream_new_proc)pContext->pulse.pa_stream_new)((ma_pa_context*)pContext->pulse.pPulseContext, actualStreamName, ss, cmap);
+ ma_pcm_f32_to_s32__optimized(dst, src, count, ditherMode);
}
+#endif
+#if defined(MA_SUPPORT_AVX2)
+static MA_INLINE void ma_pcm_f32_to_s32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_f32_to_s32__optimized(dst, src, count, ditherMode);
+}
+#endif
+#if defined(MA_SUPPORT_NEON)
+static MA_INLINE void ma_pcm_f32_to_s32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ ma_pcm_f32_to_s32__optimized(dst, src, count, ditherMode);
+}
+#endif
-
-static void ma_device_on_read__pulse(ma_pa_stream* pStream, size_t byteCount, void* pUserData)
+MA_API void ma_pcm_f32_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
- ma_device* pDevice = (ma_device*)pUserData;
- ma_uint32 bpf;
- ma_uint32 deviceState;
- ma_uint64 frameCount;
- ma_uint64 framesProcessed;
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_f32_to_s32__reference(dst, src, count, ditherMode);
+#else
+ # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
+ if (ma_has_avx2()) {
+ ma_pcm_f32_to_s32__avx2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
+ if (ma_has_sse2()) {
+ ma_pcm_f32_to_s32__sse2(dst, src, count, ditherMode);
+ } else
+ #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
+ if (ma_has_neon()) {
+ ma_pcm_f32_to_s32__neon(dst, src, count, ditherMode);
+ } else
+ #endif
+ {
+ ma_pcm_f32_to_s32__optimized(dst, src, count, ditherMode);
+ }
+#endif
+}
- MA_ASSERT(pDevice != NULL);
- /*
- Don't do anything if the device isn't initialized yet. Yes, this can happen because PulseAudio
- can fire this callback before the stream has even started. Ridiculous.
- */
- deviceState = ma_device_get_state(pDevice);
- if (deviceState != MA_STATE_STARTING && deviceState != MA_STATE_STARTED) {
- return;
- }
+MA_API void ma_pcm_f32_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+{
+ (void)ditherMode;
- bpf = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
- MA_ASSERT(bpf > 0);
+ ma_copy_memory_64(dst, src, count * sizeof(float));
+}
- frameCount = byteCount / bpf;
- framesProcessed = 0;
- while (ma_device_get_state(pDevice) == MA_STATE_STARTED && framesProcessed < frameCount) {
- const void* pMappedPCMFrames;
- size_t bytesMapped;
- ma_uint64 framesMapped;
+static void ma_pcm_interleave_f32__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
+{
+ float* dst_f32 = (float*)dst;
+ const float** src_f32 = (const float**)src;
- int pulseResult = ((ma_pa_stream_peek_proc)pDevice->pContext->pulse.pa_stream_peek)(pStream, &pMappedPCMFrames, &bytesMapped);
- if (pulseResult < 0) {
- break; /* Failed to map. Abort. */
+ ma_uint64 iFrame;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ dst_f32[iFrame*channels + iChannel] = src_f32[iChannel][iFrame];
}
+ }
+}
- framesMapped = bytesMapped / bpf;
- if (framesMapped > 0) {
- if (pMappedPCMFrames != NULL) {
- ma_device_handle_backend_data_callback(pDevice, NULL, pMappedPCMFrames, framesMapped);
- } else {
- /* It's a hole. */
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[PulseAudio] ma_device_on_read__pulse: Hole.\n");
- }
+static void ma_pcm_interleave_f32__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
+{
+ ma_pcm_interleave_f32__reference(dst, src, frameCount, channels);
+}
- pulseResult = ((ma_pa_stream_drop_proc)pDevice->pContext->pulse.pa_stream_drop)(pStream);
- if (pulseResult < 0) {
- break; /* Failed to drop the buffer. */
- }
+MA_API void ma_pcm_interleave_f32(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_interleave_f32__reference(dst, src, frameCount, channels);
+#else
+ ma_pcm_interleave_f32__optimized(dst, src, frameCount, channels);
+#endif
+}
- framesProcessed += framesMapped;
- } else {
- /* Nothing was mapped. Just abort. */
- break;
+static void ma_pcm_deinterleave_f32__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
+{
+ float** dst_f32 = (float**)dst;
+ const float* src_f32 = (const float*)src;
+
+ ma_uint64 iFrame;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ dst_f32[iChannel][iFrame] = src_f32[iFrame*channels + iChannel];
}
}
}
-static ma_result ma_device_write_to_stream__pulse(ma_device* pDevice, ma_pa_stream* pStream, ma_uint64* pFramesProcessed)
+static void ma_pcm_deinterleave_f32__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
{
- ma_result result = MA_SUCCESS;
- ma_uint64 framesProcessed = 0;
- size_t bytesMapped;
- ma_uint32 bpf;
- ma_uint32 deviceState;
+ ma_pcm_deinterleave_f32__reference(dst, src, frameCount, channels);
+}
- MA_ASSERT(pDevice != NULL);
- MA_ASSERT(pStream != NULL);
+MA_API void ma_pcm_deinterleave_f32(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
+{
+#ifdef MA_USE_REFERENCE_CONVERSION_APIS
+ ma_pcm_deinterleave_f32__reference(dst, src, frameCount, channels);
+#else
+ ma_pcm_deinterleave_f32__optimized(dst, src, frameCount, channels);
+#endif
+}
- bpf = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
- MA_ASSERT(bpf > 0);
- deviceState = ma_device_get_state(pDevice);
+MA_API void ma_pcm_convert(void* pOut, ma_format formatOut, const void* pIn, ma_format formatIn, ma_uint64 sampleCount, ma_dither_mode ditherMode)
+{
+ if (formatOut == formatIn) {
+ ma_copy_memory_64(pOut, pIn, sampleCount * ma_get_bytes_per_sample(formatOut));
+ return;
+ }
- bytesMapped = ((ma_pa_stream_writable_size_proc)pDevice->pContext->pulse.pa_stream_writable_size)(pStream);
- if (bytesMapped != (size_t)-1) {
- if (bytesMapped > 0) {
- ma_uint64 framesMapped;
- void* pMappedPCMFrames;
- int pulseResult = ((ma_pa_stream_begin_write_proc)pDevice->pContext->pulse.pa_stream_begin_write)(pStream, &pMappedPCMFrames, &bytesMapped);
- if (pulseResult < 0) {
- result = ma_result_from_pulse(pulseResult);
- goto done;
+ switch (formatIn)
+ {
+ case ma_format_u8:
+ {
+ switch (formatOut)
+ {
+ case ma_format_s16: ma_pcm_u8_to_s16(pOut, pIn, sampleCount, ditherMode); return;
+ case ma_format_s24: ma_pcm_u8_to_s24(pOut, pIn, sampleCount, ditherMode); return;
+ case ma_format_s32: ma_pcm_u8_to_s32(pOut, pIn, sampleCount, ditherMode); return;
+ case ma_format_f32: ma_pcm_u8_to_f32(pOut, pIn, sampleCount, ditherMode); return;
+ default: break;
}
+ } break;
- framesMapped = bytesMapped / bpf;
+ case ma_format_s16:
+ {
+ switch (formatOut)
+ {
+ case ma_format_u8: ma_pcm_s16_to_u8( pOut, pIn, sampleCount, ditherMode); return;
+ case ma_format_s24: ma_pcm_s16_to_s24(pOut, pIn, sampleCount, ditherMode); return;
+ case ma_format_s32: ma_pcm_s16_to_s32(pOut, pIn, sampleCount, ditherMode); return;
+ case ma_format_f32: ma_pcm_s16_to_f32(pOut, pIn, sampleCount, ditherMode); return;
+ default: break;
+ }
+ } break;
- if (deviceState == MA_STATE_STARTED || deviceState == MA_STATE_STARTING) { /* Check for starting state just in case this is being used to do the initial fill. */
- ma_device_handle_backend_data_callback(pDevice, pMappedPCMFrames, NULL, framesMapped);
- } else {
- /* Device is not started. Write silence. */
- ma_silence_pcm_frames(pMappedPCMFrames, framesMapped, pDevice->playback.format, pDevice->playback.channels);
+ case ma_format_s24:
+ {
+ switch (formatOut)
+ {
+ case ma_format_u8: ma_pcm_s24_to_u8( pOut, pIn, sampleCount, ditherMode); return;
+ case ma_format_s16: ma_pcm_s24_to_s16(pOut, pIn, sampleCount, ditherMode); return;
+ case ma_format_s32: ma_pcm_s24_to_s32(pOut, pIn, sampleCount, ditherMode); return;
+ case ma_format_f32: ma_pcm_s24_to_f32(pOut, pIn, sampleCount, ditherMode); return;
+ default: break;
}
+ } break;
- pulseResult = ((ma_pa_stream_write_proc)pDevice->pContext->pulse.pa_stream_write)(pStream, pMappedPCMFrames, bytesMapped, NULL, 0, MA_PA_SEEK_RELATIVE);
- if (pulseResult < 0) {
- result = ma_result_from_pulse(pulseResult);
- goto done; /* Failed to write data to stream. */
+ case ma_format_s32:
+ {
+ switch (formatOut)
+ {
+ case ma_format_u8: ma_pcm_s32_to_u8( pOut, pIn, sampleCount, ditherMode); return;
+ case ma_format_s16: ma_pcm_s32_to_s16(pOut, pIn, sampleCount, ditherMode); return;
+ case ma_format_s24: ma_pcm_s32_to_s24(pOut, pIn, sampleCount, ditherMode); return;
+ case ma_format_f32: ma_pcm_s32_to_f32(pOut, pIn, sampleCount, ditherMode); return;
+ default: break;
}
+ } break;
- framesProcessed += framesMapped;
- } else {
- result = MA_ERROR; /* No data available. Abort. */
- goto done;
- }
- } else {
- result = MA_ERROR; /* Failed to retrieve the writable size. Abort. */
- goto done;
- }
+ case ma_format_f32:
+ {
+ switch (formatOut)
+ {
+ case ma_format_u8: ma_pcm_f32_to_u8( pOut, pIn, sampleCount, ditherMode); return;
+ case ma_format_s16: ma_pcm_f32_to_s16(pOut, pIn, sampleCount, ditherMode); return;
+ case ma_format_s24: ma_pcm_f32_to_s24(pOut, pIn, sampleCount, ditherMode); return;
+ case ma_format_s32: ma_pcm_f32_to_s32(pOut, pIn, sampleCount, ditherMode); return;
+ default: break;
+ }
+ } break;
-done:
- if (pFramesProcessed != NULL) {
- *pFramesProcessed = framesProcessed;
+ default: break;
}
-
- return result;
}
-static void ma_device_on_write__pulse(ma_pa_stream* pStream, size_t byteCount, void* pUserData)
+MA_API void ma_convert_pcm_frames_format(void* pOut, ma_format formatOut, const void* pIn, ma_format formatIn, ma_uint64 frameCount, ma_uint32 channels, ma_dither_mode ditherMode)
{
- ma_device* pDevice = (ma_device*)pUserData;
- ma_uint32 bpf;
- ma_uint64 frameCount;
- ma_uint64 framesProcessed;
- ma_uint32 deviceState;
- ma_result result;
-
- MA_ASSERT(pDevice != NULL);
+ ma_pcm_convert(pOut, formatOut, pIn, formatIn, frameCount * channels, ditherMode);
+}
- /*
- Don't do anything if the device isn't initialized yet. Yes, this can happen because PulseAudio
- can fire this callback before the stream has even started. Ridiculous.
- */
- deviceState = ma_device_get_state(pDevice);
- if (deviceState != MA_STATE_STARTING && deviceState != MA_STATE_STARTED) {
- return;
+MA_API void ma_deinterleave_pcm_frames(ma_format format, ma_uint32 channels, ma_uint64 frameCount, const void* pInterleavedPCMFrames, void** ppDeinterleavedPCMFrames)
+{
+ if (pInterleavedPCMFrames == NULL || ppDeinterleavedPCMFrames == NULL) {
+ return; /* Invalid args. */
}
- bpf = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
- MA_ASSERT(bpf > 0);
+ /* For efficiency we do this per format. */
+ switch (format) {
+ case ma_format_s16:
+ {
+ const ma_int16* pSrcS16 = (const ma_int16*)pInterleavedPCMFrames;
+ ma_uint64 iPCMFrame;
+ for (iPCMFrame = 0; iPCMFrame < frameCount; ++iPCMFrame) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < channels; ++iChannel) {
+ ma_int16* pDstS16 = (ma_int16*)ppDeinterleavedPCMFrames[iChannel];
+ pDstS16[iPCMFrame] = pSrcS16[iPCMFrame*channels+iChannel];
+ }
+ }
+ } break;
- frameCount = byteCount / bpf;
- framesProcessed = 0;
+ case ma_format_f32:
+ {
+ const float* pSrcF32 = (const float*)pInterleavedPCMFrames;
+ ma_uint64 iPCMFrame;
+ for (iPCMFrame = 0; iPCMFrame < frameCount; ++iPCMFrame) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < channels; ++iChannel) {
+ float* pDstF32 = (float*)ppDeinterleavedPCMFrames[iChannel];
+ pDstF32[iPCMFrame] = pSrcF32[iPCMFrame*channels+iChannel];
+ }
+ }
+ } break;
- while (framesProcessed < frameCount) {
- ma_uint64 framesProcessedThisIteration;
+ default:
+ {
+ ma_uint32 sampleSizeInBytes = ma_get_bytes_per_sample(format);
+ ma_uint64 iPCMFrame;
+ for (iPCMFrame = 0; iPCMFrame < frameCount; ++iPCMFrame) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < channels; ++iChannel) {
+ void* pDst = ma_offset_ptr(ppDeinterleavedPCMFrames[iChannel], iPCMFrame*sampleSizeInBytes);
+ const void* pSrc = ma_offset_ptr(pInterleavedPCMFrames, (iPCMFrame*channels+iChannel)*sampleSizeInBytes);
+ memcpy(pDst, pSrc, sampleSizeInBytes);
+ }
+ }
+ } break;
+ }
+}
- /* Don't keep trying to process frames if the device isn't started. */
- deviceState = ma_device_get_state(pDevice);
- if (deviceState != MA_STATE_STARTING && deviceState != MA_STATE_STARTED) {
- break;
- }
+MA_API void ma_interleave_pcm_frames(ma_format format, ma_uint32 channels, ma_uint64 frameCount, const void** ppDeinterleavedPCMFrames, void* pInterleavedPCMFrames)
+{
+ switch (format)
+ {
+ case ma_format_s16:
+ {
+ ma_int16* pDstS16 = (ma_int16*)pInterleavedPCMFrames;
+ ma_uint64 iPCMFrame;
+ for (iPCMFrame = 0; iPCMFrame < frameCount; ++iPCMFrame) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < channels; ++iChannel) {
+ const ma_int16* pSrcS16 = (const ma_int16*)ppDeinterleavedPCMFrames[iChannel];
+ pDstS16[iPCMFrame*channels+iChannel] = pSrcS16[iPCMFrame];
+ }
+ }
+ } break;
- result = ma_device_write_to_stream__pulse(pDevice, pStream, &framesProcessedThisIteration);
- if (result != MA_SUCCESS) {
- break;
- }
+ case ma_format_f32:
+ {
+ float* pDstF32 = (float*)pInterleavedPCMFrames;
+ ma_uint64 iPCMFrame;
+ for (iPCMFrame = 0; iPCMFrame < frameCount; ++iPCMFrame) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < channels; ++iChannel) {
+ const float* pSrcF32 = (const float*)ppDeinterleavedPCMFrames[iChannel];
+ pDstF32[iPCMFrame*channels+iChannel] = pSrcF32[iPCMFrame];
+ }
+ }
+ } break;
- framesProcessed += framesProcessedThisIteration;
+ default:
+ {
+ ma_uint32 sampleSizeInBytes = ma_get_bytes_per_sample(format);
+ ma_uint64 iPCMFrame;
+ for (iPCMFrame = 0; iPCMFrame < frameCount; ++iPCMFrame) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < channels; ++iChannel) {
+ void* pDst = ma_offset_ptr(pInterleavedPCMFrames, (iPCMFrame*channels+iChannel)*sampleSizeInBytes);
+ const void* pSrc = ma_offset_ptr(ppDeinterleavedPCMFrames[iChannel], iPCMFrame*sampleSizeInBytes);
+ memcpy(pDst, pSrc, sampleSizeInBytes);
+ }
+ }
+ } break;
}
}
-static void ma_device_on_suspended__pulse(ma_pa_stream* pStream, void* pUserData)
-{
- ma_device* pDevice = (ma_device*)pUserData;
- int suspended;
- (void)pStream;
+/**************************************************************************************************************************************************************
- suspended = ((ma_pa_stream_is_suspended_proc)pDevice->pContext->pulse.pa_stream_is_suspended)(pStream);
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[Pulse] Device suspended state changed. pa_stream_is_suspended() returned %d.\n", suspended);
+Biquad Filter
- if (suspended < 0) {
- return;
- }
+**************************************************************************************************************************************************************/
+#ifndef MA_BIQUAD_FIXED_POINT_SHIFT
+#define MA_BIQUAD_FIXED_POINT_SHIFT 14
+#endif
- if (suspended == 1) {
- ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[Pulse] Device suspended state changed. Suspended.\n");
-
- if (pDevice->onStop) {
- pDevice->onStop(pDevice);
- }
- } else {
- ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[Pulse] Device suspended state changed. Resumed.\n");
- }
+static ma_int32 ma_biquad_float_to_fp(double x)
+{
+ return (ma_int32)(x * (1 << MA_BIQUAD_FIXED_POINT_SHIFT));
}
-static ma_result ma_device_init__pulse(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+MA_API ma_biquad_config ma_biquad_config_init(ma_format format, ma_uint32 channels, double b0, double b1, double b2, double a0, double a1, double a2)
{
- /*
- Notes for PulseAudio:
+ ma_biquad_config config;
- - We're always using native format/channels/rate regardless of whether or not PulseAudio
- supports the format directly through their own data conversion system. I'm doing this to
- reduce as much variability from the PulseAudio side as possible because it's seems to be
- extremely unreliable at everything it does.
+ MA_ZERO_OBJECT(&config);
+ config.format = format;
+ config.channels = channels;
+ config.b0 = b0;
+ config.b1 = b1;
+ config.b2 = b2;
+ config.a0 = a0;
+ config.a1 = a1;
+ config.a2 = a2;
- - When both the period size in frames and milliseconds are 0, we default to miniaudio's
- default buffer sizes rather than leaving it up to PulseAudio because I don't trust
- PulseAudio to give us any kind of reasonable latency by default.
+ return config;
+}
- - Do not ever, *ever* forget to use MA_PA_STREAM_ADJUST_LATENCY. If you don't specify this
- flag, capture mode will just not work properly until you open another PulseAudio app.
- */
- ma_result result = MA_SUCCESS;
- int error = 0;
- const char* devPlayback = NULL;
- const char* devCapture = NULL;
- ma_format format = ma_format_unknown;
- ma_uint32 channels = 0;
- ma_uint32 sampleRate = 0;
- ma_pa_sink_info sinkInfo;
- ma_pa_source_info sourceInfo;
- ma_pa_sample_spec ss;
- ma_pa_channel_map cmap;
- ma_pa_buffer_attr attr;
- const ma_pa_sample_spec* pActualSS = NULL;
- const ma_pa_channel_map* pActualCMap = NULL;
- const ma_pa_buffer_attr* pActualAttr = NULL;
- ma_uint32 iChannel;
- ma_pa_stream_flags_t streamFlags;
+typedef struct
+{
+ size_t sizeInBytes;
+ size_t r1Offset;
+ size_t r2Offset;
+} ma_biquad_heap_layout;
- MA_ASSERT(pDevice != NULL);
- MA_ZERO_OBJECT(&pDevice->pulse);
+static ma_result ma_biquad_get_heap_layout(const ma_biquad_config* pConfig, ma_biquad_heap_layout* pHeapLayout)
+{
+ MA_ASSERT(pHeapLayout != NULL);
- if (pConfig->deviceType == ma_device_type_loopback) {
- return MA_DEVICE_TYPE_NOT_SUPPORTED;
- }
+ MA_ZERO_OBJECT(pHeapLayout);
- /* No exclusive mode with the PulseAudio backend. */
- if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive) ||
- ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive)) {
- return MA_SHARE_MODE_NOT_SUPPORTED;
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- if (pDescriptorPlayback->pDeviceID != NULL) {
- devPlayback = pDescriptorPlayback->pDeviceID->pulse;
- }
-
- format = pDescriptorPlayback->format;
- channels = pDescriptorPlayback->channels;
- sampleRate = pDescriptorPlayback->sampleRate;
+ if (pConfig->channels == 0) {
+ return MA_INVALID_ARGS;
}
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
- if (pDescriptorCapture->pDeviceID != NULL) {
- devCapture = pDescriptorCapture->pDeviceID->pulse;
- }
+ pHeapLayout->sizeInBytes = 0;
- format = pDescriptorCapture->format;
- channels = pDescriptorCapture->channels;
- sampleRate = pDescriptorCapture->sampleRate;
- }
+ /* R0 */
+ pHeapLayout->r1Offset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += sizeof(ma_biquad_coefficient) * pConfig->channels;
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
- result = ma_context_get_source_info__pulse(pDevice->pContext, devCapture, &sourceInfo);
- if (result != MA_SUCCESS) {
- ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to retrieve source info for capture device.", result);
- goto on_error0;
- }
+ /* R1 */
+ pHeapLayout->r2Offset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += sizeof(ma_biquad_coefficient) * pConfig->channels;
- ss = sourceInfo.sample_spec;
- cmap = sourceInfo.channel_map;
+ /* Make sure allocation size is aligned. */
+ pHeapLayout->sizeInBytes = ma_align_64(pHeapLayout->sizeInBytes);
- if (ma_format_from_pulse(ss.format) == ma_format_unknown) {
- if (ma_is_little_endian()) {
- ss.format = MA_PA_SAMPLE_FLOAT32LE;
- } else {
- ss.format = MA_PA_SAMPLE_FLOAT32BE;
- }
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[PulseAudio] WARNING: sample_spec.format not supported by miniaudio. Defaulting to PA_SAMPLE_RATE_FLOAT32\n");
- }
- if (ss.rate == 0) {
- ss.rate = MA_DEFAULT_SAMPLE_RATE;
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[PulseAudio] WARNING: sample_spec.rate = 0. Defaulting to %d\n", ss.rate);
- }
- if (ss.channels == 0) {
- ss.channels = MA_DEFAULT_CHANNELS;
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[PulseAudio] WARNING: sample_spec.channels = 0. Defaulting to %d\n", ss.channels);
- }
+ return MA_SUCCESS;
+}
- /* We now have enough information to calculate our actual period size in frames. */
- pDescriptorCapture->periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptorCapture, ss.rate, pConfig->performanceProfile);
+MA_API ma_result ma_biquad_get_heap_size(const ma_biquad_config* pConfig, size_t* pHeapSizeInBytes)
+{
+ ma_result result;
+ ma_biquad_heap_layout heapLayout;
- attr = ma_device__pa_buffer_attr_new(pDescriptorCapture->periodSizeInFrames, pDescriptorCapture->periodCount, &ss);
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[PulseAudio] Capture attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; periodSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsize, pDescriptorCapture->periodSizeInFrames);
+ if (pHeapSizeInBytes == NULL) {
+ return MA_INVALID_ARGS;
+ }
- pDevice->pulse.pStreamCapture = ma_context__pa_stream_new__pulse(pDevice->pContext, pConfig->pulse.pStreamNameCapture, &ss, &cmap);
- if (pDevice->pulse.pStreamCapture == NULL) {
- result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create PulseAudio capture stream.", MA_FAILED_TO_OPEN_BACKEND_DEVICE);
- goto on_error0;
- }
+ *pHeapSizeInBytes = 0;
+ result = ma_biquad_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- /* The callback needs to be set before connecting the stream. */
- ((ma_pa_stream_set_read_callback_proc)pDevice->pContext->pulse.pa_stream_set_read_callback)((ma_pa_stream*)pDevice->pulse.pStreamCapture, ma_device_on_read__pulse, pDevice);
+ *pHeapSizeInBytes = heapLayout.sizeInBytes;
- /* State callback for checking when the device has been corked. */
- ((ma_pa_stream_set_suspended_callback_proc)pDevice->pContext->pulse.pa_stream_set_suspended_callback)((ma_pa_stream*)pDevice->pulse.pStreamCapture, ma_device_on_suspended__pulse, pDevice);
+ return MA_SUCCESS;
+}
+MA_API ma_result ma_biquad_init_preallocated(const ma_biquad_config* pConfig, void* pHeap, ma_biquad* pBQ)
+{
+ ma_result result;
+ ma_biquad_heap_layout heapLayout;
- /* Connect after we've got all of our internal state set up. */
- streamFlags = MA_PA_STREAM_START_CORKED | MA_PA_STREAM_ADJUST_LATENCY | MA_PA_STREAM_FIX_FORMAT | MA_PA_STREAM_FIX_RATE | MA_PA_STREAM_FIX_CHANNELS;
- if (devCapture != NULL) {
- streamFlags |= MA_PA_STREAM_DONT_MOVE;
- }
+ if (pBQ == NULL) {
+ return MA_INVALID_ARGS;
+ }
- error = ((ma_pa_stream_connect_record_proc)pDevice->pContext->pulse.pa_stream_connect_record)((ma_pa_stream*)pDevice->pulse.pStreamCapture, devCapture, &attr, streamFlags);
- if (error != MA_PA_OK) {
- result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to connect PulseAudio capture stream.", ma_result_from_pulse(error));
- goto on_error1;
- }
+ MA_ZERO_OBJECT(pBQ);
- result = ma_context_wait_for_pa_stream_to_connect__pulse(pDevice->pContext, (ma_pa_stream*)pDevice->pulse.pStreamCapture);
- if (result != MA_SUCCESS) {
- goto on_error2;
- }
+ result = ma_biquad_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- /* Internal format. */
- pActualSS = ((ma_pa_stream_get_sample_spec_proc)pDevice->pContext->pulse.pa_stream_get_sample_spec)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
- if (pActualSS != NULL) {
- ss = *pActualSS;
- }
+ pBQ->_pHeap = pHeap;
+ MA_ZERO_MEMORY(pHeap, heapLayout.sizeInBytes);
- pDescriptorCapture->format = ma_format_from_pulse(ss.format);
- pDescriptorCapture->channels = ss.channels;
- pDescriptorCapture->sampleRate = ss.rate;
+ pBQ->pR1 = (ma_biquad_coefficient*)ma_offset_ptr(pHeap, heapLayout.r1Offset);
+ pBQ->pR2 = (ma_biquad_coefficient*)ma_offset_ptr(pHeap, heapLayout.r2Offset);
- /* Internal channel map. */
- pActualCMap = ((ma_pa_stream_get_channel_map_proc)pDevice->pContext->pulse.pa_stream_get_channel_map)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
- if (pActualCMap != NULL) {
- cmap = *pActualCMap;
- }
+ return ma_biquad_reinit(pConfig, pBQ);
+}
- for (iChannel = 0; iChannel < pDescriptorCapture->channels; ++iChannel) {
- pDescriptorCapture->channelMap[iChannel] = ma_channel_position_from_pulse(cmap.map[iChannel]);
- }
+MA_API ma_result ma_biquad_init(const ma_biquad_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_biquad* pBQ)
+{
+ ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
+ result = ma_biquad_get_heap_size(pConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- /* Buffer. */
- pActualAttr = ((ma_pa_stream_get_buffer_attr_proc)pDevice->pContext->pulse.pa_stream_get_buffer_attr)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
- if (pActualAttr != NULL) {
- attr = *pActualAttr;
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
}
+ } else {
+ pHeap = NULL;
+ }
- pDescriptorCapture->periodCount = attr.maxlength / attr.fragsize;
- pDescriptorCapture->periodSizeInFrames = attr.maxlength / ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels) / pDescriptorCapture->periodCount;
+ result = ma_biquad_init_preallocated(pConfig, pHeap, pBQ);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
+ }
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[PulseAudio] Capture actual attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; periodSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsize, pDescriptorCapture->periodSizeInFrames);
+ pBQ->_ownsHeap = MA_TRUE;
+ return MA_SUCCESS;
+}
+MA_API void ma_biquad_uninit(ma_biquad* pBQ, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pBQ == NULL) {
+ return;
+ }
- /* Name. */
- devCapture = ((ma_pa_stream_get_device_name_proc)pDevice->pContext->pulse.pa_stream_get_device_name)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
- if (devCapture != NULL) {
- ma_pa_operation* pOP = ((ma_pa_context_get_source_info_by_name_proc)pDevice->pContext->pulse.pa_context_get_source_info_by_name)((ma_pa_context*)pDevice->pContext->pulse.pPulseContext, devCapture, ma_device_source_name_callback, pDevice);
- ma_wait_for_operation_and_unref__pulse(pDevice->pContext, pOP);
- }
+ if (pBQ->_ownsHeap) {
+ ma_free(pBQ->_pHeap, pAllocationCallbacks);
}
+}
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- result = ma_context_get_sink_info__pulse(pDevice->pContext, devPlayback, &sinkInfo);
- if (result != MA_SUCCESS) {
- ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to retrieve sink info for playback device.", result);
- goto on_error2;
- }
+MA_API ma_result ma_biquad_reinit(const ma_biquad_config* pConfig, ma_biquad* pBQ)
+{
+ if (pBQ == NULL || pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
- ss = sinkInfo.sample_spec;
- cmap = sinkInfo.channel_map;
+ if (pConfig->a0 == 0) {
+ return MA_INVALID_ARGS; /* Division by zero. */
+ }
- if (ma_format_from_pulse(ss.format) == ma_format_unknown) {
- if (ma_is_little_endian()) {
- ss.format = MA_PA_SAMPLE_FLOAT32LE;
- } else {
- ss.format = MA_PA_SAMPLE_FLOAT32BE;
- }
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[PulseAudio] WARNING: sample_spec.format not supported by miniaudio. Defaulting to PA_SAMPLE_RATE_FLOAT32\n");
- }
- if (ss.rate == 0) {
- ss.rate = MA_DEFAULT_SAMPLE_RATE;
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[PulseAudio] WARNING: sample_spec.rate = 0. Defaulting to %d\n", ss.rate);
- }
- if (ss.channels == 0) {
- ss.channels = MA_DEFAULT_CHANNELS;
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[PulseAudio] WARNING: sample_spec.channels = 0. Defaulting to %d\n", ss.channels);
- }
+ /* Only supporting f32 and s16. */
+ if (pConfig->format != ma_format_f32 && pConfig->format != ma_format_s16) {
+ return MA_INVALID_ARGS;
+ }
- /* We now have enough information to calculate the actual buffer size in frames. */
- pDescriptorPlayback->periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptorPlayback, ss.rate, pConfig->performanceProfile);
+ /* The format cannot be changed after initialization. */
+ if (pBQ->format != ma_format_unknown && pBQ->format != pConfig->format) {
+ return MA_INVALID_OPERATION;
+ }
- attr = ma_device__pa_buffer_attr_new(pDescriptorPlayback->periodSizeInFrames, pDescriptorPlayback->periodCount, &ss);
+ /* The channel count cannot be changed after initialization. */
+ if (pBQ->channels != 0 && pBQ->channels != pConfig->channels) {
+ return MA_INVALID_OPERATION;
+ }
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[PulseAudio] Playback attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; periodSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsize, pDescriptorPlayback->periodSizeInFrames);
- pDevice->pulse.pStreamPlayback = ma_context__pa_stream_new__pulse(pDevice->pContext, pConfig->pulse.pStreamNamePlayback, &ss, &cmap);
- if (pDevice->pulse.pStreamPlayback == NULL) {
- result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create PulseAudio playback stream.", MA_FAILED_TO_OPEN_BACKEND_DEVICE);
- goto on_error2;
- }
+ pBQ->format = pConfig->format;
+ pBQ->channels = pConfig->channels;
+ /* Normalize. */
+ if (pConfig->format == ma_format_f32) {
+ pBQ->b0.f32 = (float)(pConfig->b0 / pConfig->a0);
+ pBQ->b1.f32 = (float)(pConfig->b1 / pConfig->a0);
+ pBQ->b2.f32 = (float)(pConfig->b2 / pConfig->a0);
+ pBQ->a1.f32 = (float)(pConfig->a1 / pConfig->a0);
+ pBQ->a2.f32 = (float)(pConfig->a2 / pConfig->a0);
+ } else {
+ pBQ->b0.s32 = ma_biquad_float_to_fp(pConfig->b0 / pConfig->a0);
+ pBQ->b1.s32 = ma_biquad_float_to_fp(pConfig->b1 / pConfig->a0);
+ pBQ->b2.s32 = ma_biquad_float_to_fp(pConfig->b2 / pConfig->a0);
+ pBQ->a1.s32 = ma_biquad_float_to_fp(pConfig->a1 / pConfig->a0);
+ pBQ->a2.s32 = ma_biquad_float_to_fp(pConfig->a2 / pConfig->a0);
+ }
- /*
- Note that this callback will be fired as soon as the stream is connected, even though it's started as corked. The callback needs to handle a
- device state of MA_STATE_UNINITIALIZED.
- */
- ((ma_pa_stream_set_write_callback_proc)pDevice->pContext->pulse.pa_stream_set_write_callback)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, ma_device_on_write__pulse, pDevice);
+ return MA_SUCCESS;
+}
- /* State callback for checking when the device has been corked. */
- ((ma_pa_stream_set_suspended_callback_proc)pDevice->pContext->pulse.pa_stream_set_suspended_callback)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, ma_device_on_suspended__pulse, pDevice);
+static MA_INLINE void ma_biquad_process_pcm_frame_f32__direct_form_2_transposed(ma_biquad* pBQ, float* pY, const float* pX)
+{
+ ma_uint32 c;
+ const ma_uint32 channels = pBQ->channels;
+ const float b0 = pBQ->b0.f32;
+ const float b1 = pBQ->b1.f32;
+ const float b2 = pBQ->b2.f32;
+ const float a1 = pBQ->a1.f32;
+ const float a2 = pBQ->a2.f32;
+ MA_ASSUME(channels > 0);
+ for (c = 0; c < channels; c += 1) {
+ float r1 = pBQ->pR1[c].f32;
+ float r2 = pBQ->pR2[c].f32;
+ float x = pX[c];
+ float y;
- /* Connect after we've got all of our internal state set up. */
- streamFlags = MA_PA_STREAM_START_CORKED | MA_PA_STREAM_ADJUST_LATENCY | MA_PA_STREAM_FIX_FORMAT | MA_PA_STREAM_FIX_RATE | MA_PA_STREAM_FIX_CHANNELS;
- if (devPlayback != NULL) {
- streamFlags |= MA_PA_STREAM_DONT_MOVE;
- }
+ y = b0*x + r1;
+ r1 = b1*x - a1*y + r2;
+ r2 = b2*x - a2*y;
- error = ((ma_pa_stream_connect_playback_proc)pDevice->pContext->pulse.pa_stream_connect_playback)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, devPlayback, &attr, streamFlags, NULL, NULL);
- if (error != MA_PA_OK) {
- result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to connect PulseAudio playback stream.", ma_result_from_pulse(error));
- goto on_error3;
- }
+ pY[c] = y;
+ pBQ->pR1[c].f32 = r1;
+ pBQ->pR2[c].f32 = r2;
+ }
+}
- result = ma_context_wait_for_pa_stream_to_connect__pulse(pDevice->pContext, (ma_pa_stream*)pDevice->pulse.pStreamPlayback);
- if (result != MA_SUCCESS) {
- goto on_error3;
- }
+static MA_INLINE void ma_biquad_process_pcm_frame_f32(ma_biquad* pBQ, float* pY, const float* pX)
+{
+ ma_biquad_process_pcm_frame_f32__direct_form_2_transposed(pBQ, pY, pX);
+}
+static MA_INLINE void ma_biquad_process_pcm_frame_s16__direct_form_2_transposed(ma_biquad* pBQ, ma_int16* pY, const ma_int16* pX)
+{
+ ma_uint32 c;
+ const ma_uint32 channels = pBQ->channels;
+ const ma_int32 b0 = pBQ->b0.s32;
+ const ma_int32 b1 = pBQ->b1.s32;
+ const ma_int32 b2 = pBQ->b2.s32;
+ const ma_int32 a1 = pBQ->a1.s32;
+ const ma_int32 a2 = pBQ->a2.s32;
- /* Internal format. */
- pActualSS = ((ma_pa_stream_get_sample_spec_proc)pDevice->pContext->pulse.pa_stream_get_sample_spec)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
- if (pActualSS != NULL) {
- ss = *pActualSS;
- }
+ MA_ASSUME(channels > 0);
+ for (c = 0; c < channels; c += 1) {
+ ma_int32 r1 = pBQ->pR1[c].s32;
+ ma_int32 r2 = pBQ->pR2[c].s32;
+ ma_int32 x = pX[c];
+ ma_int32 y;
- pDescriptorPlayback->format = ma_format_from_pulse(ss.format);
- pDescriptorPlayback->channels = ss.channels;
- pDescriptorPlayback->sampleRate = ss.rate;
+ y = (b0*x + r1) >> MA_BIQUAD_FIXED_POINT_SHIFT;
+ r1 = (b1*x - a1*y + r2);
+ r2 = (b2*x - a2*y);
- /* Internal channel map. */
- pActualCMap = ((ma_pa_stream_get_channel_map_proc)pDevice->pContext->pulse.pa_stream_get_channel_map)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
- if (pActualCMap != NULL) {
- cmap = *pActualCMap;
- }
+ pY[c] = (ma_int16)ma_clamp(y, -32768, 32767);
+ pBQ->pR1[c].s32 = r1;
+ pBQ->pR2[c].s32 = r2;
+ }
+}
- for (iChannel = 0; iChannel < pDescriptorPlayback->channels; ++iChannel) {
- pDescriptorPlayback->channelMap[iChannel] = ma_channel_position_from_pulse(cmap.map[iChannel]);
- }
+static MA_INLINE void ma_biquad_process_pcm_frame_s16(ma_biquad* pBQ, ma_int16* pY, const ma_int16* pX)
+{
+ ma_biquad_process_pcm_frame_s16__direct_form_2_transposed(pBQ, pY, pX);
+}
+MA_API ma_result ma_biquad_process_pcm_frames(ma_biquad* pBQ, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+{
+ ma_uint32 n;
- /* Buffer. */
- pActualAttr = ((ma_pa_stream_get_buffer_attr_proc)pDevice->pContext->pulse.pa_stream_get_buffer_attr)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
- if (pActualAttr != NULL) {
- attr = *pActualAttr;
- }
+ if (pBQ == NULL || pFramesOut == NULL || pFramesIn == NULL) {
+ return MA_INVALID_ARGS;
+ }
- pDescriptorPlayback->periodCount = attr.maxlength / attr.tlength;
- pDescriptorPlayback->periodSizeInFrames = attr.maxlength / ma_get_bytes_per_frame(pDescriptorPlayback->format, pDescriptorPlayback->channels) / pDescriptorPlayback->periodCount;
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[PulseAudio] Playback actual attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; internalPeriodSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsize, pDescriptorPlayback->periodSizeInFrames);
+ /* Note that the logic below needs to support in-place filtering. That is, it must support the case where pFramesOut and pFramesIn are the same. */
+ if (pBQ->format == ma_format_f32) {
+ /* */ float* pY = ( float*)pFramesOut;
+ const float* pX = (const float*)pFramesIn;
- /* Name. */
- devPlayback = ((ma_pa_stream_get_device_name_proc)pDevice->pContext->pulse.pa_stream_get_device_name)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
- if (devPlayback != NULL) {
- ma_pa_operation* pOP = ((ma_pa_context_get_sink_info_by_name_proc)pDevice->pContext->pulse.pa_context_get_sink_info_by_name)((ma_pa_context*)pDevice->pContext->pulse.pPulseContext, devPlayback, ma_device_sink_name_callback, pDevice);
- ma_wait_for_operation_and_unref__pulse(pDevice->pContext, pOP);
+ for (n = 0; n < frameCount; n += 1) {
+ ma_biquad_process_pcm_frame_f32__direct_form_2_transposed(pBQ, pY, pX);
+ pY += pBQ->channels;
+ pX += pBQ->channels;
}
- }
-
+ } else if (pBQ->format == ma_format_s16) {
+ /* */ ma_int16* pY = ( ma_int16*)pFramesOut;
+ const ma_int16* pX = (const ma_int16*)pFramesIn;
- /*
- We need a ring buffer for handling duplex mode. We can use the main duplex ring buffer in the main
- part of the ma_device struct. We cannot, however, depend on ma_device_init() initializing this for
- us later on because that will only do it if it's a fully asynchronous backend - i.e. the
- onDeviceDataLoop callback is NULL, which is not the case for PulseAudio.
- */
- if (pConfig->deviceType == ma_device_type_duplex) {
- result = ma_duplex_rb_init(format, channels, sampleRate, pDescriptorCapture->sampleRate, pDescriptorCapture->periodSizeInFrames, &pDevice->pContext->allocationCallbacks, &pDevice->duplexRB);
- if (result != MA_SUCCESS) {
- result = ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to initialize ring buffer.", result);
- goto on_error4;
+ for (n = 0; n < frameCount; n += 1) {
+ ma_biquad_process_pcm_frame_s16__direct_form_2_transposed(pBQ, pY, pX);
+ pY += pBQ->channels;
+ pX += pBQ->channels;
}
+ } else {
+ MA_ASSERT(MA_FALSE);
+ return MA_INVALID_ARGS; /* Format not supported. Should never hit this because it's checked in ma_biquad_init() and ma_biquad_reinit(). */
}
return MA_SUCCESS;
+}
-
-on_error4:
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- ((ma_pa_stream_disconnect_proc)pDevice->pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
- }
-on_error3:
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- ((ma_pa_stream_unref_proc)pDevice->pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
- }
-on_error2:
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
- ((ma_pa_stream_disconnect_proc)pDevice->pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
- }
-on_error1:
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
- ((ma_pa_stream_unref_proc)pDevice->pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
+MA_API ma_uint32 ma_biquad_get_latency(const ma_biquad* pBQ)
+{
+ if (pBQ == NULL) {
+ return 0;
}
-on_error0:
- return result;
-}
+ return 2;
+}
-static void ma_pulse_operation_complete_callback(ma_pa_stream* pStream, int success, void* pUserData)
-{
- ma_bool32* pIsSuccessful = (ma_bool32*)pUserData;
- MA_ASSERT(pIsSuccessful != NULL);
- *pIsSuccessful = (ma_bool32)success;
+/**************************************************************************************************************************************************************
- (void)pStream; /* Unused. */
-}
+Low-Pass Filter
-static ma_result ma_device__cork_stream__pulse(ma_device* pDevice, ma_device_type deviceType, int cork)
+**************************************************************************************************************************************************************/
+MA_API ma_lpf1_config ma_lpf1_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency)
{
- ma_context* pContext = pDevice->pContext;
- ma_bool32 wasSuccessful;
- ma_pa_stream* pStream;
- ma_pa_operation* pOP;
- ma_result result;
-
- /* This should not be called with a duplex device type. */
- if (deviceType == ma_device_type_duplex) {
- return MA_INVALID_ARGS;
- }
+ ma_lpf1_config config;
- wasSuccessful = MA_FALSE;
+ MA_ZERO_OBJECT(&config);
+ config.format = format;
+ config.channels = channels;
+ config.sampleRate = sampleRate;
+ config.cutoffFrequency = cutoffFrequency;
+ config.q = 0.5;
- pStream = (ma_pa_stream*)((deviceType == ma_device_type_capture) ? pDevice->pulse.pStreamCapture : pDevice->pulse.pStreamPlayback);
- MA_ASSERT(pStream != NULL);
+ return config;
+}
- pOP = ((ma_pa_stream_cork_proc)pContext->pulse.pa_stream_cork)(pStream, cork, ma_pulse_operation_complete_callback, &wasSuccessful);
- if (pOP == NULL) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to cork PulseAudio stream.", (cork == 0) ? MA_FAILED_TO_START_BACKEND_DEVICE : MA_FAILED_TO_STOP_BACKEND_DEVICE);
- }
+MA_API ma_lpf2_config ma_lpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q)
+{
+ ma_lpf2_config config;
- result = ma_wait_for_operation_and_unref__pulse(pDevice->pContext, pOP);
- if (result != MA_SUCCESS) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] An error occurred while waiting for the PulseAudio stream to cork.", result);
- }
+ MA_ZERO_OBJECT(&config);
+ config.format = format;
+ config.channels = channels;
+ config.sampleRate = sampleRate;
+ config.cutoffFrequency = cutoffFrequency;
+ config.q = q;
- if (!wasSuccessful) {
- if (cork) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to stop PulseAudio stream.", MA_FAILED_TO_STOP_BACKEND_DEVICE);
- } else {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to start PulseAudio stream.", MA_FAILED_TO_START_BACKEND_DEVICE);
- }
+ /* Q cannot be 0 or else it'll result in a division by 0. In this case just default to 0.707107. */
+ if (config.q == 0) {
+ config.q = 0.707107;
}
- return MA_SUCCESS;
+ return config;
}
-static ma_result ma_device_start__pulse(ma_device* pDevice)
-{
- ma_result result;
- MA_ASSERT(pDevice != NULL);
+typedef struct
+{
+ size_t sizeInBytes;
+ size_t r1Offset;
+} ma_lpf1_heap_layout;
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- result = ma_device__cork_stream__pulse(pDevice, ma_device_type_capture, 0);
- if (result != MA_SUCCESS) {
- return result;
- }
- }
+static ma_result ma_lpf1_get_heap_layout(const ma_lpf1_config* pConfig, ma_lpf1_heap_layout* pHeapLayout)
+{
+ MA_ASSERT(pHeapLayout != NULL);
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- /* We need to fill some data before uncorking. Not doing this will result in the write callback never getting fired. */
- result = ma_device_write_to_stream__pulse(pDevice, (ma_pa_stream*)(pDevice->pulse.pStreamPlayback), NULL);
- if (result != MA_SUCCESS) {
- return result; /* Failed to write data. Not sure what to do here... Just aborting. */
- }
+ MA_ZERO_OBJECT(pHeapLayout);
- result = ma_device__cork_stream__pulse(pDevice, ma_device_type_playback, 0);
- if (result != MA_SUCCESS) {
- return result;
- }
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
+ if (pConfig->channels == 0) {
+ return MA_INVALID_ARGS;
+ }
+
+ pHeapLayout->sizeInBytes = 0;
+
+ /* R1 */
+ pHeapLayout->r1Offset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += sizeof(ma_biquad_coefficient) * pConfig->channels;
+
+ /* Make sure allocation size is aligned. */
+ pHeapLayout->sizeInBytes = ma_align_64(pHeapLayout->sizeInBytes);
+
return MA_SUCCESS;
}
-static ma_result ma_device_stop__pulse(ma_device* pDevice)
+MA_API ma_result ma_lpf1_get_heap_size(const ma_lpf1_config* pConfig, size_t* pHeapSizeInBytes)
{
ma_result result;
- ma_bool32 wasSuccessful;
-
- MA_ASSERT(pDevice != NULL);
+ ma_lpf1_heap_layout heapLayout;
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- result = ma_device__cork_stream__pulse(pDevice, ma_device_type_capture, 1);
- if (result != MA_SUCCESS) {
- return result;
- }
+ if (pHeapSizeInBytes == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- /* The stream needs to be drained if it's a playback device. */
- ma_pa_operation* pOP = ((ma_pa_stream_drain_proc)pDevice->pContext->pulse.pa_stream_drain)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, ma_pulse_operation_complete_callback, &wasSuccessful);
- ma_wait_for_operation_and_unref__pulse(pDevice->pContext, pOP);
-
- result = ma_device__cork_stream__pulse(pDevice, ma_device_type_playback, 1);
- if (result != MA_SUCCESS) {
- return result;
- }
+ result = ma_lpf1_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
}
+ *pHeapSizeInBytes = heapLayout.sizeInBytes;
+
return MA_SUCCESS;
}
-static ma_result ma_device_data_loop__pulse(ma_device* pDevice)
+MA_API ma_result ma_lpf1_init_preallocated(const ma_lpf1_config* pConfig, void* pHeap, ma_lpf1* pLPF)
{
- int resultPA;
+ ma_result result;
+ ma_lpf1_heap_layout heapLayout;
- MA_ASSERT(pDevice != NULL);
+ if (pLPF == NULL) {
+ return MA_INVALID_ARGS;
+ }
- /* NOTE: Don't start the device here. It'll be done at a higher level. */
+ MA_ZERO_OBJECT(pLPF);
- /*
- All data is handled through callbacks. All we need to do is iterate over the main loop and let
- the callbacks deal with it.
- */
- while (ma_device_get_state(pDevice) == MA_STATE_STARTED) {
- resultPA = ((ma_pa_mainloop_iterate_proc)pDevice->pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pDevice->pContext->pulse.pMainLoop, 1, NULL);
- if (resultPA < 0) {
- break;
- }
+ result = ma_lpf1_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
}
- /* NOTE: Don't stop the device here. It'll be done at a higher level. */
- return MA_SUCCESS;
-}
-
-static ma_result ma_device_data_loop_wakeup__pulse(ma_device* pDevice)
-{
- MA_ASSERT(pDevice != NULL);
+ pLPF->_pHeap = pHeap;
+ MA_ZERO_MEMORY(pHeap, heapLayout.sizeInBytes);
- ((ma_pa_mainloop_wakeup_proc)pDevice->pContext->pulse.pa_mainloop_wakeup)((ma_pa_mainloop*)pDevice->pContext->pulse.pMainLoop);
+ pLPF->pR1 = (ma_biquad_coefficient*)ma_offset_ptr(pHeap, heapLayout.r1Offset);
- return MA_SUCCESS;
+ return ma_lpf1_reinit(pConfig, pLPF);
}
-static ma_result ma_context_uninit__pulse(ma_context* pContext)
+MA_API ma_result ma_lpf1_init(const ma_lpf1_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_lpf1* pLPF)
{
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pContext->backend == ma_backend_pulseaudio);
+ ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
- ((ma_pa_context_disconnect_proc)pContext->pulse.pa_context_disconnect)((ma_pa_context*)pContext->pulse.pPulseContext);
- ((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)((ma_pa_context*)pContext->pulse.pPulseContext);
- ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)((ma_pa_mainloop*)pContext->pulse.pMainLoop);
+ result = ma_lpf1_get_heap_size(pConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
-#ifndef MA_NO_RUNTIME_LINKING
- ma_dlclose(pContext, pContext->pulse.pulseSO);
-#endif
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+ } else {
+ pHeap = NULL;
+ }
+
+ result = ma_lpf1_init_preallocated(pConfig, pHeap, pLPF);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
+ }
+ pLPF->_ownsHeap = MA_TRUE;
return MA_SUCCESS;
}
-static ma_result ma_context_init__pulse(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+MA_API void ma_lpf1_uninit(ma_lpf1* pLPF, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_result result;
-#ifndef MA_NO_RUNTIME_LINKING
- const char* libpulseNames[] = {
- "libpulse.so",
- "libpulse.so.0"
- };
- size_t i;
-
- for (i = 0; i < ma_countof(libpulseNames); ++i) {
- pContext->pulse.pulseSO = ma_dlopen(pContext, libpulseNames[i]);
- if (pContext->pulse.pulseSO != NULL) {
- break;
- }
+ if (pLPF == NULL) {
+ return;
}
- if (pContext->pulse.pulseSO == NULL) {
- return MA_NO_BACKEND;
+ if (pLPF->_ownsHeap) {
+ ma_free(pLPF->_pHeap, pAllocationCallbacks);
}
+}
- pContext->pulse.pa_mainloop_new = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_new");
- pContext->pulse.pa_mainloop_free = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_free");
- pContext->pulse.pa_mainloop_quit = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_quit");
- pContext->pulse.pa_mainloop_get_api = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_get_api");
- pContext->pulse.pa_mainloop_iterate = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_iterate");
- pContext->pulse.pa_mainloop_wakeup = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_mainloop_wakeup");
- pContext->pulse.pa_threaded_mainloop_new = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_new");
- pContext->pulse.pa_threaded_mainloop_free = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_free");
- pContext->pulse.pa_threaded_mainloop_start = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_start");
- pContext->pulse.pa_threaded_mainloop_stop = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_stop");
- pContext->pulse.pa_threaded_mainloop_lock = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_lock");
- pContext->pulse.pa_threaded_mainloop_unlock = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_unlock");
- pContext->pulse.pa_threaded_mainloop_wait = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_wait");
- pContext->pulse.pa_threaded_mainloop_signal = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_signal");
- pContext->pulse.pa_threaded_mainloop_accept = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_accept");
- pContext->pulse.pa_threaded_mainloop_get_retval = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_get_retval");
- pContext->pulse.pa_threaded_mainloop_get_api = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_get_api");
- pContext->pulse.pa_threaded_mainloop_in_thread = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_in_thread");
- pContext->pulse.pa_threaded_mainloop_set_name = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_threaded_mainloop_set_name");
- pContext->pulse.pa_context_new = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_new");
- pContext->pulse.pa_context_unref = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_unref");
- pContext->pulse.pa_context_connect = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_connect");
- pContext->pulse.pa_context_disconnect = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_disconnect");
- pContext->pulse.pa_context_set_state_callback = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_set_state_callback");
- pContext->pulse.pa_context_get_state = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_get_state");
- pContext->pulse.pa_context_get_sink_info_list = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_get_sink_info_list");
- pContext->pulse.pa_context_get_source_info_list = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_get_source_info_list");
- pContext->pulse.pa_context_get_sink_info_by_name = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_get_sink_info_by_name");
- pContext->pulse.pa_context_get_source_info_by_name = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_context_get_source_info_by_name");
- pContext->pulse.pa_operation_unref = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_operation_unref");
- pContext->pulse.pa_operation_get_state = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_operation_get_state");
- pContext->pulse.pa_channel_map_init_extend = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_channel_map_init_extend");
- pContext->pulse.pa_channel_map_valid = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_channel_map_valid");
- pContext->pulse.pa_channel_map_compatible = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_channel_map_compatible");
- pContext->pulse.pa_stream_new = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_new");
- pContext->pulse.pa_stream_unref = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_unref");
- pContext->pulse.pa_stream_connect_playback = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_connect_playback");
- pContext->pulse.pa_stream_connect_record = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_connect_record");
- pContext->pulse.pa_stream_disconnect = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_disconnect");
- pContext->pulse.pa_stream_get_state = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_get_state");
- pContext->pulse.pa_stream_get_sample_spec = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_get_sample_spec");
- pContext->pulse.pa_stream_get_channel_map = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_get_channel_map");
- pContext->pulse.pa_stream_get_buffer_attr = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_get_buffer_attr");
- pContext->pulse.pa_stream_set_buffer_attr = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_set_buffer_attr");
- pContext->pulse.pa_stream_get_device_name = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_get_device_name");
- pContext->pulse.pa_stream_set_write_callback = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_set_write_callback");
- pContext->pulse.pa_stream_set_read_callback = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_set_read_callback");
- pContext->pulse.pa_stream_set_suspended_callback = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_set_suspended_callback");
- pContext->pulse.pa_stream_is_suspended = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_is_suspended");
- pContext->pulse.pa_stream_flush = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_flush");
- pContext->pulse.pa_stream_drain = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_drain");
- pContext->pulse.pa_stream_is_corked = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_is_corked");
- pContext->pulse.pa_stream_cork = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_cork");
- pContext->pulse.pa_stream_trigger = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_trigger");
- pContext->pulse.pa_stream_begin_write = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_begin_write");
- pContext->pulse.pa_stream_write = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_write");
- pContext->pulse.pa_stream_peek = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_peek");
- pContext->pulse.pa_stream_drop = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_drop");
- pContext->pulse.pa_stream_writable_size = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_writable_size");
- pContext->pulse.pa_stream_readable_size = (ma_proc)ma_dlsym(pContext, pContext->pulse.pulseSO, "pa_stream_readable_size");
-#else
- /* This strange assignment system is just for type safety. */
- ma_pa_mainloop_new_proc _pa_mainloop_new = pa_mainloop_new;
- ma_pa_mainloop_free_proc _pa_mainloop_free = pa_mainloop_free;
- ma_pa_mainloop_quit_proc _pa_mainloop_quit = pa_mainloop_quit;
- ma_pa_mainloop_get_api_proc _pa_mainloop_get_api = pa_mainloop_get_api;
- ma_pa_mainloop_iterate_proc _pa_mainloop_iterate = pa_mainloop_iterate;
- ma_pa_mainloop_wakeup_proc _pa_mainloop_wakeup = pa_mainloop_wakeup;
- ma_pa_threaded_mainloop_new_proc _pa_threaded_mainloop_new = pa_threaded_mainloop_new;
- ma_pa_threaded_mainloop_free_proc _pa_threaded_mainloop_free = pa_threaded_mainloop_free;
- ma_pa_threaded_mainloop_start_proc _pa_threaded_mainloop_start = pa_threaded_mainloop_start;
- ma_pa_threaded_mainloop_stop_proc _pa_threaded_mainloop_stop = pa_threaded_mainloop_stop;
- ma_pa_threaded_mainloop_lock_proc _pa_threaded_mainloop_lock = pa_threaded_mainloop_lock;
- ma_pa_threaded_mainloop_unlock_proc _pa_threaded_mainloop_unlock = pa_threaded_mainloop_unlock;
- ma_pa_threaded_mainloop_wait_proc _pa_threaded_mainloop_wait = pa_threaded_mainloop_wait;
- ma_pa_threaded_mainloop_signal_proc _pa_threaded_mainloop_signal = pa_threaded_mainloop_signal;
- ma_pa_threaded_mainloop_accept_proc _pa_threaded_mainloop_accept = pa_threaded_mainloop_accept;
- ma_pa_threaded_mainloop_get_retval_proc _pa_threaded_mainloop_get_retval = pa_threaded_mainloop_get_retval;
- ma_pa_threaded_mainloop_get_api_proc _pa_threaded_mainloop_get_api = pa_threaded_mainloop_get_api;
- ma_pa_threaded_mainloop_in_thread_proc _pa_threaded_mainloop_in_thread = pa_threaded_mainloop_in_thread;
- ma_pa_threaded_mainloop_set_name_proc _pa_threaded_mainloop_set_name = pa_threaded_mainloop_set_name;
- ma_pa_context_new_proc _pa_context_new = pa_context_new;
- ma_pa_context_unref_proc _pa_context_unref = pa_context_unref;
- ma_pa_context_connect_proc _pa_context_connect = pa_context_connect;
- ma_pa_context_disconnect_proc _pa_context_disconnect = pa_context_disconnect;
- ma_pa_context_set_state_callback_proc _pa_context_set_state_callback = pa_context_set_state_callback;
- ma_pa_context_get_state_proc _pa_context_get_state = pa_context_get_state;
- ma_pa_context_get_sink_info_list_proc _pa_context_get_sink_info_list = pa_context_get_sink_info_list;
- ma_pa_context_get_source_info_list_proc _pa_context_get_source_info_list = pa_context_get_source_info_list;
- ma_pa_context_get_sink_info_by_name_proc _pa_context_get_sink_info_by_name = pa_context_get_sink_info_by_name;
- ma_pa_context_get_source_info_by_name_proc _pa_context_get_source_info_by_name= pa_context_get_source_info_by_name;
- ma_pa_operation_unref_proc _pa_operation_unref = pa_operation_unref;
- ma_pa_operation_get_state_proc _pa_operation_get_state = pa_operation_get_state;
- ma_pa_channel_map_init_extend_proc _pa_channel_map_init_extend = pa_channel_map_init_extend;
- ma_pa_channel_map_valid_proc _pa_channel_map_valid = pa_channel_map_valid;
- ma_pa_channel_map_compatible_proc _pa_channel_map_compatible = pa_channel_map_compatible;
- ma_pa_stream_new_proc _pa_stream_new = pa_stream_new;
- ma_pa_stream_unref_proc _pa_stream_unref = pa_stream_unref;
- ma_pa_stream_connect_playback_proc _pa_stream_connect_playback = pa_stream_connect_playback;
- ma_pa_stream_connect_record_proc _pa_stream_connect_record = pa_stream_connect_record;
- ma_pa_stream_disconnect_proc _pa_stream_disconnect = pa_stream_disconnect;
- ma_pa_stream_get_state_proc _pa_stream_get_state = pa_stream_get_state;
- ma_pa_stream_get_sample_spec_proc _pa_stream_get_sample_spec = pa_stream_get_sample_spec;
- ma_pa_stream_get_channel_map_proc _pa_stream_get_channel_map = pa_stream_get_channel_map;
- ma_pa_stream_get_buffer_attr_proc _pa_stream_get_buffer_attr = pa_stream_get_buffer_attr;
- ma_pa_stream_set_buffer_attr_proc _pa_stream_set_buffer_attr = pa_stream_set_buffer_attr;
- ma_pa_stream_get_device_name_proc _pa_stream_get_device_name = pa_stream_get_device_name;
- ma_pa_stream_set_write_callback_proc _pa_stream_set_write_callback = pa_stream_set_write_callback;
- ma_pa_stream_set_read_callback_proc _pa_stream_set_read_callback = pa_stream_set_read_callback;
- ma_pa_stream_set_suspended_callback_proc _pa_stream_set_suspended_callback = pa_stream_set_suspended_callback;
- ma_pa_stream_is_suspended_proc _pa_stream_is_suspended = pa_stream_is_suspended;
- ma_pa_stream_flush_proc _pa_stream_flush = pa_stream_flush;
- ma_pa_stream_drain_proc _pa_stream_drain = pa_stream_drain;
- ma_pa_stream_is_corked_proc _pa_stream_is_corked = pa_stream_is_corked;
- ma_pa_stream_cork_proc _pa_stream_cork = pa_stream_cork;
- ma_pa_stream_trigger_proc _pa_stream_trigger = pa_stream_trigger;
- ma_pa_stream_begin_write_proc _pa_stream_begin_write = pa_stream_begin_write;
- ma_pa_stream_write_proc _pa_stream_write = pa_stream_write;
- ma_pa_stream_peek_proc _pa_stream_peek = pa_stream_peek;
- ma_pa_stream_drop_proc _pa_stream_drop = pa_stream_drop;
- ma_pa_stream_writable_size_proc _pa_stream_writable_size = pa_stream_writable_size;
- ma_pa_stream_readable_size_proc _pa_stream_readable_size = pa_stream_readable_size;
-
- pContext->pulse.pa_mainloop_new = (ma_proc)_pa_mainloop_new;
- pContext->pulse.pa_mainloop_free = (ma_proc)_pa_mainloop_free;
- pContext->pulse.pa_mainloop_quit = (ma_proc)_pa_mainloop_quit;
- pContext->pulse.pa_mainloop_get_api = (ma_proc)_pa_mainloop_get_api;
- pContext->pulse.pa_mainloop_iterate = (ma_proc)_pa_mainloop_iterate;
- pContext->pulse.pa_mainloop_wakeup = (ma_proc)_pa_mainloop_wakeup;
- pContext->pulse.pa_threaded_mainloop_new = (ma_proc)_pa_threaded_mainloop_new;
- pContext->pulse.pa_threaded_mainloop_free = (ma_proc)_pa_threaded_mainloop_free;
- pContext->pulse.pa_threaded_mainloop_start = (ma_proc)_pa_threaded_mainloop_start;
- pContext->pulse.pa_threaded_mainloop_stop = (ma_proc)_pa_threaded_mainloop_stop;
- pContext->pulse.pa_threaded_mainloop_lock = (ma_proc)_pa_threaded_mainloop_lock;
- pContext->pulse.pa_threaded_mainloop_unlock = (ma_proc)_pa_threaded_mainloop_unlock;
- pContext->pulse.pa_threaded_mainloop_wait = (ma_proc)_pa_threaded_mainloop_wait;
- pContext->pulse.pa_threaded_mainloop_signal = (ma_proc)_pa_threaded_mainloop_signal;
- pContext->pulse.pa_threaded_mainloop_accept = (ma_proc)_pa_threaded_mainloop_accept;
- pContext->pulse.pa_threaded_mainloop_get_retval = (ma_proc)_pa_threaded_mainloop_get_retval;
- pContext->pulse.pa_threaded_mainloop_get_api = (ma_proc)_pa_threaded_mainloop_get_api;
- pContext->pulse.pa_threaded_mainloop_in_thread = (ma_proc)_pa_threaded_mainloop_in_thread;
- pContext->pulse.pa_threaded_mainloop_set_name = (ma_proc)_pa_threaded_mainloop_set_name;
- pContext->pulse.pa_context_new = (ma_proc)_pa_context_new;
- pContext->pulse.pa_context_unref = (ma_proc)_pa_context_unref;
- pContext->pulse.pa_context_connect = (ma_proc)_pa_context_connect;
- pContext->pulse.pa_context_disconnect = (ma_proc)_pa_context_disconnect;
- pContext->pulse.pa_context_set_state_callback = (ma_proc)_pa_context_set_state_callback;
- pContext->pulse.pa_context_get_state = (ma_proc)_pa_context_get_state;
- pContext->pulse.pa_context_get_sink_info_list = (ma_proc)_pa_context_get_sink_info_list;
- pContext->pulse.pa_context_get_source_info_list = (ma_proc)_pa_context_get_source_info_list;
- pContext->pulse.pa_context_get_sink_info_by_name = (ma_proc)_pa_context_get_sink_info_by_name;
- pContext->pulse.pa_context_get_source_info_by_name = (ma_proc)_pa_context_get_source_info_by_name;
- pContext->pulse.pa_operation_unref = (ma_proc)_pa_operation_unref;
- pContext->pulse.pa_operation_get_state = (ma_proc)_pa_operation_get_state;
- pContext->pulse.pa_channel_map_init_extend = (ma_proc)_pa_channel_map_init_extend;
- pContext->pulse.pa_channel_map_valid = (ma_proc)_pa_channel_map_valid;
- pContext->pulse.pa_channel_map_compatible = (ma_proc)_pa_channel_map_compatible;
- pContext->pulse.pa_stream_new = (ma_proc)_pa_stream_new;
- pContext->pulse.pa_stream_unref = (ma_proc)_pa_stream_unref;
- pContext->pulse.pa_stream_connect_playback = (ma_proc)_pa_stream_connect_playback;
- pContext->pulse.pa_stream_connect_record = (ma_proc)_pa_stream_connect_record;
- pContext->pulse.pa_stream_disconnect = (ma_proc)_pa_stream_disconnect;
- pContext->pulse.pa_stream_get_state = (ma_proc)_pa_stream_get_state;
- pContext->pulse.pa_stream_get_sample_spec = (ma_proc)_pa_stream_get_sample_spec;
- pContext->pulse.pa_stream_get_channel_map = (ma_proc)_pa_stream_get_channel_map;
- pContext->pulse.pa_stream_get_buffer_attr = (ma_proc)_pa_stream_get_buffer_attr;
- pContext->pulse.pa_stream_set_buffer_attr = (ma_proc)_pa_stream_set_buffer_attr;
- pContext->pulse.pa_stream_get_device_name = (ma_proc)_pa_stream_get_device_name;
- pContext->pulse.pa_stream_set_write_callback = (ma_proc)_pa_stream_set_write_callback;
- pContext->pulse.pa_stream_set_read_callback = (ma_proc)_pa_stream_set_read_callback;
- pContext->pulse.pa_stream_set_suspended_callback = (ma_proc)_pa_stream_set_suspended_callback;
- pContext->pulse.pa_stream_is_suspended = (ma_proc)_pa_stream_is_suspended;
- pContext->pulse.pa_stream_flush = (ma_proc)_pa_stream_flush;
- pContext->pulse.pa_stream_drain = (ma_proc)_pa_stream_drain;
- pContext->pulse.pa_stream_is_corked = (ma_proc)_pa_stream_is_corked;
- pContext->pulse.pa_stream_cork = (ma_proc)_pa_stream_cork;
- pContext->pulse.pa_stream_trigger = (ma_proc)_pa_stream_trigger;
- pContext->pulse.pa_stream_begin_write = (ma_proc)_pa_stream_begin_write;
- pContext->pulse.pa_stream_write = (ma_proc)_pa_stream_write;
- pContext->pulse.pa_stream_peek = (ma_proc)_pa_stream_peek;
- pContext->pulse.pa_stream_drop = (ma_proc)_pa_stream_drop;
- pContext->pulse.pa_stream_writable_size = (ma_proc)_pa_stream_writable_size;
- pContext->pulse.pa_stream_readable_size = (ma_proc)_pa_stream_readable_size;
-#endif
+MA_API ma_result ma_lpf1_reinit(const ma_lpf1_config* pConfig, ma_lpf1* pLPF)
+{
+ double a;
- /* The PulseAudio context maps well to miniaudio's notion of a context. The pa_context object will be initialized as part of the ma_context. */
- pContext->pulse.pMainLoop = ((ma_pa_mainloop_new_proc)pContext->pulse.pa_mainloop_new)();
- if (pContext->pulse.pMainLoop == NULL) {
- result = ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create mainloop.", MA_FAILED_TO_INIT_BACKEND);
- #ifndef MA_NO_RUNTIME_LINKING
- ma_dlclose(pContext, pContext->pulse.pulseSO);
- #endif
- return result;
+ if (pLPF == NULL || pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- pContext->pulse.pPulseContext = ((ma_pa_context_new_proc)pContext->pulse.pa_context_new)(((ma_pa_mainloop_get_api_proc)pContext->pulse.pa_mainloop_get_api)((ma_pa_mainloop*)pContext->pulse.pMainLoop), pConfig->pulse.pApplicationName);
- if (pContext->pulse.pPulseContext == NULL) {
- result = ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create PulseAudio context.", MA_FAILED_TO_INIT_BACKEND);
- ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)((ma_pa_mainloop*)(pContext->pulse.pMainLoop));
- #ifndef MA_NO_RUNTIME_LINKING
- ma_dlclose(pContext, pContext->pulse.pulseSO);
- #endif
- return result;
+ /* Only supporting f32 and s16. */
+ if (pConfig->format != ma_format_f32 && pConfig->format != ma_format_s16) {
+ return MA_INVALID_ARGS;
}
- /* Now we need to connect to the context. Everything is asynchronous so we need to wait for it to connect before returning. */
- result = ma_result_from_pulse(((ma_pa_context_connect_proc)pContext->pulse.pa_context_connect)((ma_pa_context*)pContext->pulse.pPulseContext, pConfig->pulse.pServerName, (pConfig->pulse.tryAutoSpawn) ? 0 : MA_PA_CONTEXT_NOAUTOSPAWN, NULL));
- if (result != MA_SUCCESS) {
- ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to connect PulseAudio context.", result);
- ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)((ma_pa_mainloop*)(pContext->pulse.pMainLoop));
- #ifndef MA_NO_RUNTIME_LINKING
- ma_dlclose(pContext, pContext->pulse.pulseSO);
- #endif
- return result;
+ /* The format cannot be changed after initialization. */
+ if (pLPF->format != ma_format_unknown && pLPF->format != pConfig->format) {
+ return MA_INVALID_OPERATION;
}
- /* Since ma_context_init() runs synchronously we need to wait for the PulseAudio context to connect before we return. */
- result = ma_context_wait_for_pa_context_to_connect__pulse(pContext);
- if (result != MA_SUCCESS) {
- ((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)((ma_pa_mainloop*)(pContext->pulse.pMainLoop));
- #ifndef MA_NO_RUNTIME_LINKING
- ma_dlclose(pContext, pContext->pulse.pulseSO);
- #endif
- return result;
+ /* The channel count cannot be changed after initialization. */
+ if (pLPF->channels != 0 && pLPF->channels != pConfig->channels) {
+ return MA_INVALID_OPERATION;
}
+ pLPF->format = pConfig->format;
+ pLPF->channels = pConfig->channels;
- /* With pa_mainloop we run a synchronous backend, but we implement our own main loop. */
- pCallbacks->onContextInit = ma_context_init__pulse;
- pCallbacks->onContextUninit = ma_context_uninit__pulse;
- pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__pulse;
- pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__pulse;
- pCallbacks->onDeviceInit = ma_device_init__pulse;
- pCallbacks->onDeviceUninit = ma_device_uninit__pulse;
- pCallbacks->onDeviceStart = ma_device_start__pulse;
- pCallbacks->onDeviceStop = ma_device_stop__pulse;
- pCallbacks->onDeviceRead = NULL; /* Not used because we're implementing onDeviceDataLoop. */
- pCallbacks->onDeviceWrite = NULL; /* Not used because we're implementing onDeviceDataLoop. */
- pCallbacks->onDeviceDataLoop = ma_device_data_loop__pulse;
- pCallbacks->onDeviceDataLoopWakeup = ma_device_data_loop_wakeup__pulse;
+ a = ma_expd(-2 * MA_PI_D * pConfig->cutoffFrequency / pConfig->sampleRate);
+ if (pConfig->format == ma_format_f32) {
+ pLPF->a.f32 = (float)a;
+ } else {
+ pLPF->a.s32 = ma_biquad_float_to_fp(a);
+ }
return MA_SUCCESS;
}
-#endif
+static MA_INLINE void ma_lpf1_process_pcm_frame_f32(ma_lpf1* pLPF, float* pY, const float* pX)
+{
+ ma_uint32 c;
+ const ma_uint32 channels = pLPF->channels;
+ const float a = pLPF->a.f32;
+ const float b = 1 - a;
-/******************************************************************************
+ MA_ASSUME(channels > 0);
+ for (c = 0; c < channels; c += 1) {
+ float r1 = pLPF->pR1[c].f32;
+ float x = pX[c];
+ float y;
-JACK Backend
+ y = b*x + a*r1;
-******************************************************************************/
-#ifdef MA_HAS_JACK
+ pY[c] = y;
+ pLPF->pR1[c].f32 = y;
+ }
+}
-/* It is assumed jack.h is available when compile-time linking is being used. */
-#ifdef MA_NO_RUNTIME_LINKING
-#include
+static MA_INLINE void ma_lpf1_process_pcm_frame_s16(ma_lpf1* pLPF, ma_int16* pY, const ma_int16* pX)
+{
+ ma_uint32 c;
+ const ma_uint32 channels = pLPF->channels;
+ const ma_int32 a = pLPF->a.s32;
+ const ma_int32 b = ((1 << MA_BIQUAD_FIXED_POINT_SHIFT) - a);
-typedef jack_nframes_t ma_jack_nframes_t;
-typedef jack_options_t ma_jack_options_t;
-typedef jack_status_t ma_jack_status_t;
-typedef jack_client_t ma_jack_client_t;
-typedef jack_port_t ma_jack_port_t;
-typedef JackProcessCallback ma_JackProcessCallback;
-typedef JackBufferSizeCallback ma_JackBufferSizeCallback;
-typedef JackShutdownCallback ma_JackShutdownCallback;
-#define MA_JACK_DEFAULT_AUDIO_TYPE JACK_DEFAULT_AUDIO_TYPE
-#define ma_JackNoStartServer JackNoStartServer
-#define ma_JackPortIsInput JackPortIsInput
-#define ma_JackPortIsOutput JackPortIsOutput
-#define ma_JackPortIsPhysical JackPortIsPhysical
-#else
-typedef ma_uint32 ma_jack_nframes_t;
-typedef int ma_jack_options_t;
-typedef int ma_jack_status_t;
-typedef struct ma_jack_client_t ma_jack_client_t;
-typedef struct ma_jack_port_t ma_jack_port_t;
-typedef int (* ma_JackProcessCallback) (ma_jack_nframes_t nframes, void* arg);
-typedef int (* ma_JackBufferSizeCallback)(ma_jack_nframes_t nframes, void* arg);
-typedef void (* ma_JackShutdownCallback) (void* arg);
-#define MA_JACK_DEFAULT_AUDIO_TYPE "32 bit float mono audio"
-#define ma_JackNoStartServer 1
-#define ma_JackPortIsInput 1
-#define ma_JackPortIsOutput 2
-#define ma_JackPortIsPhysical 4
-#endif
+ MA_ASSUME(channels > 0);
+ for (c = 0; c < channels; c += 1) {
+ ma_int32 r1 = pLPF->pR1[c].s32;
+ ma_int32 x = pX[c];
+ ma_int32 y;
-typedef ma_jack_client_t* (* ma_jack_client_open_proc) (const char* client_name, ma_jack_options_t options, ma_jack_status_t* status, ...);
-typedef int (* ma_jack_client_close_proc) (ma_jack_client_t* client);
-typedef int (* ma_jack_client_name_size_proc) (void);
-typedef int (* ma_jack_set_process_callback_proc) (ma_jack_client_t* client, ma_JackProcessCallback process_callback, void* arg);
-typedef int (* ma_jack_set_buffer_size_callback_proc)(ma_jack_client_t* client, ma_JackBufferSizeCallback bufsize_callback, void* arg);
-typedef void (* ma_jack_on_shutdown_proc) (ma_jack_client_t* client, ma_JackShutdownCallback function, void* arg);
-typedef ma_jack_nframes_t (* ma_jack_get_sample_rate_proc) (ma_jack_client_t* client);
-typedef ma_jack_nframes_t (* ma_jack_get_buffer_size_proc) (ma_jack_client_t* client);
-typedef const char** (* ma_jack_get_ports_proc) (ma_jack_client_t* client, const char* port_name_pattern, const char* type_name_pattern, unsigned long flags);
-typedef int (* ma_jack_activate_proc) (ma_jack_client_t* client);
-typedef int (* ma_jack_deactivate_proc) (ma_jack_client_t* client);
-typedef int (* ma_jack_connect_proc) (ma_jack_client_t* client, const char* source_port, const char* destination_port);
-typedef ma_jack_port_t* (* ma_jack_port_register_proc) (ma_jack_client_t* client, const char* port_name, const char* port_type, unsigned long flags, unsigned long buffer_size);
-typedef const char* (* ma_jack_port_name_proc) (const ma_jack_port_t* port);
-typedef void* (* ma_jack_port_get_buffer_proc) (ma_jack_port_t* port, ma_jack_nframes_t nframes);
-typedef void (* ma_jack_free_proc) (void* ptr);
+ y = (b*x + a*r1) >> MA_BIQUAD_FIXED_POINT_SHIFT;
-static ma_result ma_context_open_client__jack(ma_context* pContext, ma_jack_client_t** ppClient)
-{
- size_t maxClientNameSize;
- char clientName[256];
- ma_jack_status_t status;
- ma_jack_client_t* pClient;
+ pY[c] = (ma_int16)y;
+ pLPF->pR1[c].s32 = (ma_int32)y;
+ }
+}
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(ppClient != NULL);
+MA_API ma_result ma_lpf1_process_pcm_frames(ma_lpf1* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+{
+ ma_uint32 n;
- if (ppClient) {
- *ppClient = NULL;
+ if (pLPF == NULL || pFramesOut == NULL || pFramesIn == NULL) {
+ return MA_INVALID_ARGS;
}
- maxClientNameSize = ((ma_jack_client_name_size_proc)pContext->jack.jack_client_name_size)(); /* Includes null terminator. */
- ma_strncpy_s(clientName, ma_min(sizeof(clientName), maxClientNameSize), (pContext->jack.pClientName != NULL) ? pContext->jack.pClientName : "miniaudio", (size_t)-1);
+ /* Note that the logic below needs to support in-place filtering. That is, it must support the case where pFramesOut and pFramesIn are the same. */
- pClient = ((ma_jack_client_open_proc)pContext->jack.jack_client_open)(clientName, (pContext->jack.tryStartServer) ? 0 : ma_JackNoStartServer, &status, NULL);
- if (pClient == NULL) {
- return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
- }
+ if (pLPF->format == ma_format_f32) {
+ /* */ float* pY = ( float*)pFramesOut;
+ const float* pX = (const float*)pFramesIn;
- if (ppClient) {
- *ppClient = pClient;
+ for (n = 0; n < frameCount; n += 1) {
+ ma_lpf1_process_pcm_frame_f32(pLPF, pY, pX);
+ pY += pLPF->channels;
+ pX += pLPF->channels;
+ }
+ } else if (pLPF->format == ma_format_s16) {
+ /* */ ma_int16* pY = ( ma_int16*)pFramesOut;
+ const ma_int16* pX = (const ma_int16*)pFramesIn;
+
+ for (n = 0; n < frameCount; n += 1) {
+ ma_lpf1_process_pcm_frame_s16(pLPF, pY, pX);
+ pY += pLPF->channels;
+ pX += pLPF->channels;
+ }
+ } else {
+ MA_ASSERT(MA_FALSE);
+ return MA_INVALID_ARGS; /* Format not supported. Should never hit this because it's checked in ma_biquad_init() and ma_biquad_reinit(). */
}
return MA_SUCCESS;
}
+MA_API ma_uint32 ma_lpf1_get_latency(const ma_lpf1* pLPF)
+{
+ if (pLPF == NULL) {
+ return 0;
+ }
+
+ return 1;
+}
-static ma_result ma_context_enumerate_devices__jack(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+
+static MA_INLINE ma_biquad_config ma_lpf2__get_biquad_config(const ma_lpf2_config* pConfig)
{
- ma_bool32 cbResult = MA_TRUE;
+ ma_biquad_config bqConfig;
+ double q;
+ double w;
+ double s;
+ double c;
+ double a;
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(callback != NULL);
+ MA_ASSERT(pConfig != NULL);
- /* Playback. */
- if (cbResult) {
- ma_device_info deviceInfo;
- MA_ZERO_OBJECT(&deviceInfo);
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
- deviceInfo.isDefault = MA_TRUE; /* JACK only uses default devices. */
- cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
- }
+ q = pConfig->q;
+ w = 2 * MA_PI_D * pConfig->cutoffFrequency / pConfig->sampleRate;
+ s = ma_sind(w);
+ c = ma_cosd(w);
+ a = s / (2*q);
- /* Capture. */
- if (cbResult) {
- ma_device_info deviceInfo;
- MA_ZERO_OBJECT(&deviceInfo);
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
- deviceInfo.isDefault = MA_TRUE; /* JACK only uses default devices. */
- cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
- }
+ bqConfig.b0 = (1 - c) / 2;
+ bqConfig.b1 = 1 - c;
+ bqConfig.b2 = (1 - c) / 2;
+ bqConfig.a0 = 1 + a;
+ bqConfig.a1 = -2 * c;
+ bqConfig.a2 = 1 - a;
- (void)cbResult; /* For silencing a static analysis warning. */
+ bqConfig.format = pConfig->format;
+ bqConfig.channels = pConfig->channels;
+
+ return bqConfig;
+}
+
+MA_API ma_result ma_lpf2_get_heap_size(const ma_lpf2_config* pConfig, size_t* pHeapSizeInBytes)
+{
+ ma_biquad_config bqConfig;
+ bqConfig = ma_lpf2__get_biquad_config(pConfig);
- return MA_SUCCESS;
+ return ma_biquad_get_heap_size(&bqConfig, pHeapSizeInBytes);
}
-static ma_result ma_context_get_device_info__jack(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+MA_API ma_result ma_lpf2_init_preallocated(const ma_lpf2_config* pConfig, void* pHeap, ma_lpf2* pLPF)
{
- ma_jack_client_t* pClient;
ma_result result;
- const char** ppPorts;
+ ma_biquad_config bqConfig;
- MA_ASSERT(pContext != NULL);
+ if (pLPF == NULL) {
+ return MA_INVALID_ARGS;
+ }
- if (pDeviceID != NULL && pDeviceID->jack != 0) {
- return MA_NO_DEVICE; /* Don't know the device. */
+ MA_ZERO_OBJECT(pLPF);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- /* Name / Description */
- if (deviceType == ma_device_type_playback) {
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
- } else {
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
+ bqConfig = ma_lpf2__get_biquad_config(pConfig);
+ result = ma_biquad_init_preallocated(&bqConfig, pHeap, &pLPF->bq);
+ if (result != MA_SUCCESS) {
+ return result;
}
- /* Jack only uses default devices. */
- pDeviceInfo->isDefault = MA_TRUE;
+ return MA_SUCCESS;
+}
- /* Jack only supports f32 and has a specific channel count and sample rate. */
- pDeviceInfo->nativeDataFormats[0].format = ma_format_f32;
+MA_API ma_result ma_lpf2_init(const ma_lpf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_lpf2* pLPF)
+{
+ ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
- /* The channel count and sample rate can only be determined by opening the device. */
- result = ma_context_open_client__jack(pContext, &pClient);
+ result = ma_lpf2_get_heap_size(pConfig, &heapSizeInBytes);
if (result != MA_SUCCESS) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[JACK] Failed to open client.", result);
+ return result;
}
- pDeviceInfo->nativeDataFormats[0].sampleRate = ((ma_jack_get_sample_rate_proc)pContext->jack.jack_get_sample_rate)((ma_jack_client_t*)pClient);
- pDeviceInfo->nativeDataFormats[0].channels = 0;
-
- ppPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ((deviceType == ma_device_type_playback) ? ma_JackPortIsInput : ma_JackPortIsOutput));
- if (ppPorts == NULL) {
- ((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pClient);
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[JACK] Failed to query physical ports.", MA_FAILED_TO_OPEN_BACKEND_DEVICE);
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+ } else {
+ pHeap = NULL;
}
- while (ppPorts[pDeviceInfo->nativeDataFormats[0].channels] != NULL) {
- pDeviceInfo->nativeDataFormats[0].channels += 1;
+ result = ma_lpf2_init_preallocated(pConfig, pHeap, pLPF);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
}
- pDeviceInfo->nativeDataFormats[0].flags = 0;
- pDeviceInfo->nativeDataFormatCount = 1;
-
- ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppPorts);
- ((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pClient);
-
- (void)pContext;
+ pLPF->bq._ownsHeap = MA_TRUE; /* <-- This will cause the biquad to take ownership of the heap and free it when it's uninitialized. */
return MA_SUCCESS;
}
-
-static ma_result ma_device_uninit__jack(ma_device* pDevice)
+MA_API void ma_lpf2_uninit(ma_lpf2* pLPF, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_context* pContext;
-
- MA_ASSERT(pDevice != NULL);
+ if (pLPF == NULL) {
+ return;
+ }
- pContext = pDevice->pContext;
- MA_ASSERT(pContext != NULL);
+ ma_biquad_uninit(&pLPF->bq, pAllocationCallbacks); /* <-- This will free the heap allocation. */
+}
- if (pDevice->jack.pClient != NULL) {
- ((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pDevice->jack.pClient);
- }
+MA_API ma_result ma_lpf2_reinit(const ma_lpf2_config* pConfig, ma_lpf2* pLPF)
+{
+ ma_result result;
+ ma_biquad_config bqConfig;
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- ma__free_from_callbacks(pDevice->jack.pIntermediaryBufferCapture, &pDevice->pContext->allocationCallbacks);
+ if (pLPF == NULL || pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- ma__free_from_callbacks(pDevice->jack.pIntermediaryBufferPlayback, &pDevice->pContext->allocationCallbacks);
+ bqConfig = ma_lpf2__get_biquad_config(pConfig);
+ result = ma_biquad_reinit(&bqConfig, &pLPF->bq);
+ if (result != MA_SUCCESS) {
+ return result;
}
return MA_SUCCESS;
}
-static void ma_device__jack_shutdown_callback(void* pUserData)
+static MA_INLINE void ma_lpf2_process_pcm_frame_s16(ma_lpf2* pLPF, ma_int16* pFrameOut, const ma_int16* pFrameIn)
{
- /* JACK died. Stop the device. */
- ma_device* pDevice = (ma_device*)pUserData;
- MA_ASSERT(pDevice != NULL);
-
- ma_device_stop(pDevice);
+ ma_biquad_process_pcm_frame_s16(&pLPF->bq, pFrameOut, pFrameIn);
}
-static int ma_device__jack_buffer_size_callback(ma_jack_nframes_t frameCount, void* pUserData)
+static MA_INLINE void ma_lpf2_process_pcm_frame_f32(ma_lpf2* pLPF, float* pFrameOut, const float* pFrameIn)
{
- ma_device* pDevice = (ma_device*)pUserData;
- MA_ASSERT(pDevice != NULL);
-
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- size_t newBufferSize = frameCount * (pDevice->capture.internalChannels * ma_get_bytes_per_sample(pDevice->capture.internalFormat));
- float* pNewBuffer = (float*)ma__calloc_from_callbacks(newBufferSize, &pDevice->pContext->allocationCallbacks);
- if (pNewBuffer == NULL) {
- return MA_OUT_OF_MEMORY;
- }
-
- ma__free_from_callbacks(pDevice->jack.pIntermediaryBufferCapture, &pDevice->pContext->allocationCallbacks);
-
- pDevice->jack.pIntermediaryBufferCapture = pNewBuffer;
- pDevice->playback.internalPeriodSizeInFrames = frameCount;
- }
-
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- size_t newBufferSize = frameCount * (pDevice->playback.internalChannels * ma_get_bytes_per_sample(pDevice->playback.internalFormat));
- float* pNewBuffer = (float*)ma__calloc_from_callbacks(newBufferSize, &pDevice->pContext->allocationCallbacks);
- if (pNewBuffer == NULL) {
- return MA_OUT_OF_MEMORY;
- }
-
- ma__free_from_callbacks(pDevice->jack.pIntermediaryBufferPlayback, &pDevice->pContext->allocationCallbacks);
+ ma_biquad_process_pcm_frame_f32(&pLPF->bq, pFrameOut, pFrameIn);
+}
- pDevice->jack.pIntermediaryBufferPlayback = pNewBuffer;
- pDevice->playback.internalPeriodSizeInFrames = frameCount;
+MA_API ma_result ma_lpf2_process_pcm_frames(ma_lpf2* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+{
+ if (pLPF == NULL) {
+ return MA_INVALID_ARGS;
}
- return 0;
+ return ma_biquad_process_pcm_frames(&pLPF->bq, pFramesOut, pFramesIn, frameCount);
}
-static int ma_device__jack_process_callback(ma_jack_nframes_t frameCount, void* pUserData)
+MA_API ma_uint32 ma_lpf2_get_latency(const ma_lpf2* pLPF)
{
- ma_device* pDevice;
- ma_context* pContext;
- ma_uint32 iChannel;
+ if (pLPF == NULL) {
+ return 0;
+ }
- pDevice = (ma_device*)pUserData;
- MA_ASSERT(pDevice != NULL);
+ return ma_biquad_get_latency(&pLPF->bq);
+}
- pContext = pDevice->pContext;
- MA_ASSERT(pContext != NULL);
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- /* Channels need to be interleaved. */
- for (iChannel = 0; iChannel < pDevice->capture.internalChannels; ++iChannel) {
- const float* pSrc = (const float*)((ma_jack_port_get_buffer_proc)pContext->jack.jack_port_get_buffer)((ma_jack_port_t*)pDevice->jack.pPortsCapture[iChannel], frameCount);
- if (pSrc != NULL) {
- float* pDst = pDevice->jack.pIntermediaryBufferCapture + iChannel;
- ma_jack_nframes_t iFrame;
- for (iFrame = 0; iFrame < frameCount; ++iFrame) {
- *pDst = *pSrc;
+MA_API ma_lpf_config ma_lpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order)
+{
+ ma_lpf_config config;
- pDst += pDevice->capture.internalChannels;
- pSrc += 1;
- }
- }
- }
+ MA_ZERO_OBJECT(&config);
+ config.format = format;
+ config.channels = channels;
+ config.sampleRate = sampleRate;
+ config.cutoffFrequency = cutoffFrequency;
+ config.order = ma_min(order, MA_MAX_FILTER_ORDER);
- ma_device_handle_backend_data_callback(pDevice, NULL, pDevice->jack.pIntermediaryBufferCapture, frameCount);
- }
+ return config;
+}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- ma_device_handle_backend_data_callback(pDevice, pDevice->jack.pIntermediaryBufferPlayback, NULL, frameCount);
- /* Channels need to be deinterleaved. */
- for (iChannel = 0; iChannel < pDevice->playback.internalChannels; ++iChannel) {
- float* pDst = (float*)((ma_jack_port_get_buffer_proc)pContext->jack.jack_port_get_buffer)((ma_jack_port_t*)pDevice->jack.pPortsPlayback[iChannel], frameCount);
- if (pDst != NULL) {
- const float* pSrc = pDevice->jack.pIntermediaryBufferPlayback + iChannel;
- ma_jack_nframes_t iFrame;
- for (iFrame = 0; iFrame < frameCount; ++iFrame) {
- *pDst = *pSrc;
+typedef struct
+{
+ size_t sizeInBytes;
+ size_t lpf1Offset;
+ size_t lpf2Offset; /* Offset of the first second order filter. Subsequent filters will come straight after, and will each have the same heap size. */
+} ma_lpf_heap_layout;
- pDst += 1;
- pSrc += pDevice->playback.internalChannels;
- }
- }
- }
- }
+static void ma_lpf_calculate_sub_lpf_counts(ma_uint32 order, ma_uint32* pLPF1Count, ma_uint32* pLPF2Count)
+{
+ MA_ASSERT(pLPF1Count != NULL);
+ MA_ASSERT(pLPF2Count != NULL);
- return 0;
+ *pLPF1Count = order % 2;
+ *pLPF2Count = order / 2;
}
-static ma_result ma_device_init__jack(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+static ma_result ma_lpf_get_heap_layout(const ma_lpf_config* pConfig, ma_lpf_heap_layout* pHeapLayout)
{
ma_result result;
- ma_uint32 periodSizeInFrames;
-
- MA_ASSERT(pConfig != NULL);
- MA_ASSERT(pDevice != NULL);
+ ma_uint32 lpf1Count;
+ ma_uint32 lpf2Count;
+ ma_uint32 ilpf1;
+ ma_uint32 ilpf2;
- if (pConfig->deviceType == ma_device_type_loopback) {
- return MA_DEVICE_TYPE_NOT_SUPPORTED;
- }
+ MA_ASSERT(pHeapLayout != NULL);
- /* Only supporting default devices with JACK. */
- if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->pDeviceID != NULL && pDescriptorPlayback->pDeviceID->jack != 0) ||
- ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->pDeviceID != NULL && pDescriptorCapture->pDeviceID->jack != 0)) {
- return MA_NO_DEVICE;
- }
+ MA_ZERO_OBJECT(pHeapLayout);
- /* No exclusive mode with the JACK backend. */
- if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) ||
- ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) {
- return MA_SHARE_MODE_NOT_SUPPORTED;
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- /* Open the client. */
- result = ma_context_open_client__jack(pDevice->pContext, (ma_jack_client_t**)&pDevice->jack.pClient);
- if (result != MA_SUCCESS) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to open client.", result);
+ if (pConfig->channels == 0) {
+ return MA_INVALID_ARGS;
}
- /* Callbacks. */
- if (((ma_jack_set_process_callback_proc)pDevice->pContext->jack.jack_set_process_callback)((ma_jack_client_t*)pDevice->jack.pClient, ma_device__jack_process_callback, pDevice) != 0) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to set process callback.", MA_FAILED_TO_OPEN_BACKEND_DEVICE);
- }
- if (((ma_jack_set_buffer_size_callback_proc)pDevice->pContext->jack.jack_set_buffer_size_callback)((ma_jack_client_t*)pDevice->jack.pClient, ma_device__jack_buffer_size_callback, pDevice) != 0) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to set buffer size callback.", MA_FAILED_TO_OPEN_BACKEND_DEVICE);
+ if (pConfig->order > MA_MAX_FILTER_ORDER) {
+ return MA_INVALID_ARGS;
}
- ((ma_jack_on_shutdown_proc)pDevice->pContext->jack.jack_on_shutdown)((ma_jack_client_t*)pDevice->jack.pClient, ma_device__jack_shutdown_callback, pDevice);
-
-
- /* The buffer size in frames can change. */
- periodSizeInFrames = ((ma_jack_get_buffer_size_proc)pDevice->pContext->jack.jack_get_buffer_size)((ma_jack_client_t*)pDevice->jack.pClient);
+ ma_lpf_calculate_sub_lpf_counts(pConfig->order, &lpf1Count, &lpf2Count);
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
- const char** ppPorts;
+ pHeapLayout->sizeInBytes = 0;
- pDescriptorCapture->format = ma_format_f32;
- pDescriptorCapture->channels = 0;
- pDescriptorCapture->sampleRate = ((ma_jack_get_sample_rate_proc)pDevice->pContext->jack.jack_get_sample_rate)((ma_jack_client_t*)pDevice->jack.pClient);
- ma_get_standard_channel_map(ma_standard_channel_map_alsa, pDescriptorCapture->channels, pDescriptorCapture->channelMap);
+ /* LPF 1 */
+ pHeapLayout->lpf1Offset = pHeapLayout->sizeInBytes;
+ for (ilpf1 = 0; ilpf1 < lpf1Count; ilpf1 += 1) {
+ size_t lpf1HeapSizeInBytes;
+ ma_lpf1_config lpf1Config = ma_lpf1_config_init(pConfig->format, pConfig->channels, pConfig->sampleRate, pConfig->cutoffFrequency);
- ppPorts = ((ma_jack_get_ports_proc)pDevice->pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ma_JackPortIsOutput);
- if (ppPorts == NULL) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to query physical ports.", MA_FAILED_TO_OPEN_BACKEND_DEVICE);
+ result = ma_lpf1_get_heap_size(&lpf1Config, &lpf1HeapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
}
- while (ppPorts[pDescriptorCapture->channels] != NULL) {
- char name[64];
- ma_strcpy_s(name, sizeof(name), "capture");
- ma_itoa_s((int)pDescriptorCapture->channels, name+7, sizeof(name)-7, 10); /* 7 = length of "capture" */
+ pHeapLayout->sizeInBytes += sizeof(ma_lpf1) + lpf1HeapSizeInBytes;
+ }
- pDevice->jack.pPortsCapture[pDescriptorCapture->channels] = ((ma_jack_port_register_proc)pDevice->pContext->jack.jack_port_register)((ma_jack_client_t*)pDevice->jack.pClient, name, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsInput, 0);
- if (pDevice->jack.pPortsCapture[pDescriptorCapture->channels] == NULL) {
- ((ma_jack_free_proc)pDevice->pContext->jack.jack_free)((void*)ppPorts);
- ma_device_uninit__jack(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to register ports.", MA_FAILED_TO_OPEN_BACKEND_DEVICE);
- }
+ /* LPF 2*/
+ pHeapLayout->lpf2Offset = pHeapLayout->sizeInBytes;
+ for (ilpf2 = 0; ilpf2 < lpf2Count; ilpf2 += 1) {
+ size_t lpf2HeapSizeInBytes;
+ ma_lpf2_config lpf2Config = ma_lpf2_config_init(pConfig->format, pConfig->channels, pConfig->sampleRate, pConfig->cutoffFrequency, 0.707107); /* <-- The "q" parameter does not matter for the purpose of calculating the heap size. */
- pDescriptorCapture->channels += 1;
+ result = ma_lpf2_get_heap_size(&lpf2Config, &lpf2HeapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
}
- ((ma_jack_free_proc)pDevice->pContext->jack.jack_free)((void*)ppPorts);
-
- pDescriptorCapture->periodSizeInFrames = periodSizeInFrames;
- pDescriptorCapture->periodCount = 1; /* There's no notion of a period in JACK. Just set to 1. */
-
- pDevice->jack.pIntermediaryBufferCapture = (float*)ma__calloc_from_callbacks(pDescriptorCapture->periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels), &pDevice->pContext->allocationCallbacks);
- if (pDevice->jack.pIntermediaryBufferCapture == NULL) {
- ma_device_uninit__jack(pDevice);
- return MA_OUT_OF_MEMORY;
- }
+ pHeapLayout->sizeInBytes += sizeof(ma_lpf2) + lpf2HeapSizeInBytes;
}
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- const char** ppPorts;
-
- pDescriptorPlayback->format = ma_format_f32;
- pDescriptorPlayback->channels = 0;
- pDescriptorPlayback->sampleRate = ((ma_jack_get_sample_rate_proc)pDevice->pContext->jack.jack_get_sample_rate)((ma_jack_client_t*)pDevice->jack.pClient);
- ma_get_standard_channel_map(ma_standard_channel_map_alsa, pDescriptorPlayback->channels, pDescriptorPlayback->channelMap);
-
- ppPorts = ((ma_jack_get_ports_proc)pDevice->pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ma_JackPortIsInput);
- if (ppPorts == NULL) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to query physical ports.", MA_FAILED_TO_OPEN_BACKEND_DEVICE);
- }
-
- while (ppPorts[pDescriptorPlayback->channels] != NULL) {
- char name[64];
- ma_strcpy_s(name, sizeof(name), "playback");
- ma_itoa_s((int)pDescriptorPlayback->channels, name+8, sizeof(name)-8, 10); /* 8 = length of "playback" */
+ /* Make sure allocation size is aligned. */
+ pHeapLayout->sizeInBytes = ma_align_64(pHeapLayout->sizeInBytes);
- pDevice->jack.pPortsPlayback[pDescriptorPlayback->channels] = ((ma_jack_port_register_proc)pDevice->pContext->jack.jack_port_register)((ma_jack_client_t*)pDevice->jack.pClient, name, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsOutput, 0);
- if (pDevice->jack.pPortsPlayback[pDescriptorPlayback->channels] == NULL) {
- ((ma_jack_free_proc)pDevice->pContext->jack.jack_free)((void*)ppPorts);
- ma_device_uninit__jack(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to register ports.", MA_FAILED_TO_OPEN_BACKEND_DEVICE);
- }
+ return MA_SUCCESS;
+}
- pDescriptorPlayback->channels += 1;
- }
+static ma_result ma_lpf_reinit__internal(const ma_lpf_config* pConfig, void* pHeap, ma_lpf* pLPF, ma_bool32 isNew)
+{
+ ma_result result;
+ ma_uint32 lpf1Count;
+ ma_uint32 lpf2Count;
+ ma_uint32 ilpf1;
+ ma_uint32 ilpf2;
+ ma_lpf_heap_layout heapLayout; /* Only used if isNew is true. */
- ((ma_jack_free_proc)pDevice->pContext->jack.jack_free)((void*)ppPorts);
+ if (pLPF == NULL || pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
- pDescriptorPlayback->periodSizeInFrames = periodSizeInFrames;
- pDescriptorPlayback->periodCount = 1; /* There's no notion of a period in JACK. Just set to 1. */
+ /* Only supporting f32 and s16. */
+ if (pConfig->format != ma_format_f32 && pConfig->format != ma_format_s16) {
+ return MA_INVALID_ARGS;
+ }
- pDevice->jack.pIntermediaryBufferPlayback = (float*)ma__calloc_from_callbacks(pDescriptorPlayback->periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorPlayback->format, pDescriptorPlayback->channels), &pDevice->pContext->allocationCallbacks);
- if (pDevice->jack.pIntermediaryBufferPlayback == NULL) {
- ma_device_uninit__jack(pDevice);
- return MA_OUT_OF_MEMORY;
- }
+ /* The format cannot be changed after initialization. */
+ if (pLPF->format != ma_format_unknown && pLPF->format != pConfig->format) {
+ return MA_INVALID_OPERATION;
}
- return MA_SUCCESS;
-}
+ /* The channel count cannot be changed after initialization. */
+ if (pLPF->channels != 0 && pLPF->channels != pConfig->channels) {
+ return MA_INVALID_OPERATION;
+ }
+ if (pConfig->order > MA_MAX_FILTER_ORDER) {
+ return MA_INVALID_ARGS;
+ }
-static ma_result ma_device_start__jack(ma_device* pDevice)
-{
- ma_context* pContext = pDevice->pContext;
- int resultJACK;
- size_t i;
+ ma_lpf_calculate_sub_lpf_counts(pConfig->order, &lpf1Count, &lpf2Count);
- resultJACK = ((ma_jack_activate_proc)pContext->jack.jack_activate)((ma_jack_client_t*)pDevice->jack.pClient);
- if (resultJACK != 0) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to activate the JACK client.", MA_FAILED_TO_START_BACKEND_DEVICE);
+ /* The filter order can't change between reinits. */
+ if (!isNew) {
+ if (pLPF->lpf1Count != lpf1Count || pLPF->lpf2Count != lpf2Count) {
+ return MA_INVALID_OPERATION;
+ }
}
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- const char** ppServerPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ma_JackPortIsOutput);
- if (ppServerPorts == NULL) {
- ((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to retrieve physical ports.", MA_ERROR);
+ if (isNew) {
+ result = ma_lpf_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
}
- for (i = 0; ppServerPorts[i] != NULL; ++i) {
- const char* pServerPort = ppServerPorts[i];
- const char* pClientPort = ((ma_jack_port_name_proc)pContext->jack.jack_port_name)((ma_jack_port_t*)pDevice->jack.pPortsCapture[i]);
-
- resultJACK = ((ma_jack_connect_proc)pContext->jack.jack_connect)((ma_jack_client_t*)pDevice->jack.pClient, pServerPort, pClientPort);
- if (resultJACK != 0) {
- ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts);
- ((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to connect ports.", MA_ERROR);
- }
- }
+ pLPF->_pHeap = pHeap;
+ MA_ZERO_MEMORY(pHeap, heapLayout.sizeInBytes);
- ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts);
+ pLPF->pLPF1 = (ma_lpf1*)ma_offset_ptr(pHeap, heapLayout.lpf1Offset);
+ pLPF->pLPF2 = (ma_lpf2*)ma_offset_ptr(pHeap, heapLayout.lpf2Offset);
+ } else {
+ MA_ZERO_OBJECT(&heapLayout); /* To silence a compiler warning. */
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- const char** ppServerPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ma_JackPortIsInput);
- if (ppServerPorts == NULL) {
- ((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to retrieve physical ports.", MA_ERROR);
- }
+ for (ilpf1 = 0; ilpf1 < lpf1Count; ilpf1 += 1) {
+ ma_lpf1_config lpf1Config = ma_lpf1_config_init(pConfig->format, pConfig->channels, pConfig->sampleRate, pConfig->cutoffFrequency);
- for (i = 0; ppServerPorts[i] != NULL; ++i) {
- const char* pServerPort = ppServerPorts[i];
- const char* pClientPort = ((ma_jack_port_name_proc)pContext->jack.jack_port_name)((ma_jack_port_t*)pDevice->jack.pPortsPlayback[i]);
+ if (isNew) {
+ size_t lpf1HeapSizeInBytes;
- resultJACK = ((ma_jack_connect_proc)pContext->jack.jack_connect)((ma_jack_client_t*)pDevice->jack.pClient, pClientPort, pServerPort);
- if (resultJACK != 0) {
- ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts);
- ((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] Failed to connect ports.", MA_ERROR);
+ result = ma_lpf1_get_heap_size(&lpf1Config, &lpf1HeapSizeInBytes);
+ if (result == MA_SUCCESS) {
+ result = ma_lpf1_init_preallocated(&lpf1Config, ma_offset_ptr(pHeap, heapLayout.lpf1Offset + (sizeof(ma_lpf1) * lpf1Count) + (ilpf1 * lpf1HeapSizeInBytes)), &pLPF->pLPF1[ilpf1]);
}
+ } else {
+ result = ma_lpf1_reinit(&lpf1Config, &pLPF->pLPF1[ilpf1]);
}
- ((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts);
- }
-
- return MA_SUCCESS;
-}
-
-static ma_result ma_device_stop__jack(ma_device* pDevice)
-{
- ma_context* pContext = pDevice->pContext;
- ma_stop_proc onStop;
+ if (result != MA_SUCCESS) {
+ ma_uint32 jlpf1;
- if (((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient) != 0) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[JACK] An error occurred when deactivating the JACK client.", MA_ERROR);
- }
+ for (jlpf1 = 0; jlpf1 < ilpf1; jlpf1 += 1) {
+ ma_lpf1_uninit(&pLPF->pLPF1[jlpf1], NULL); /* No need for allocation callbacks here since we used a preallocated heap allocation. */
+ }
- onStop = pDevice->onStop;
- if (onStop) {
- onStop(pDevice);
+ return result;
+ }
}
- return MA_SUCCESS;
-}
-
-
-static ma_result ma_context_uninit__jack(ma_context* pContext)
-{
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pContext->backend == ma_backend_jack);
-
- ma_free(pContext->jack.pClientName, &pContext->allocationCallbacks);
- pContext->jack.pClientName = NULL;
+ for (ilpf2 = 0; ilpf2 < lpf2Count; ilpf2 += 1) {
+ ma_lpf2_config lpf2Config;
+ double q;
+ double a;
-#ifndef MA_NO_RUNTIME_LINKING
- ma_dlclose(pContext, pContext->jack.jackSO);
-#endif
+ /* Tempting to use 0.707107, but won't result in a Butterworth filter if the order is > 2. */
+ if (lpf1Count == 1) {
+ a = (1 + ilpf2*1) * (MA_PI_D/(pConfig->order*1)); /* Odd order. */
+ } else {
+ a = (1 + ilpf2*2) * (MA_PI_D/(pConfig->order*2)); /* Even order. */
+ }
+ q = 1 / (2*ma_cosd(a));
- return MA_SUCCESS;
-}
+ lpf2Config = ma_lpf2_config_init(pConfig->format, pConfig->channels, pConfig->sampleRate, pConfig->cutoffFrequency, q);
-static ma_result ma_context_init__jack(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
-{
-#ifndef MA_NO_RUNTIME_LINKING
- const char* libjackNames[] = {
-#ifdef MA_WIN32
- "libjack.dll",
- "libjack64.dll"
-#else
- "libjack.so",
- "libjack.so.0"
-#endif
- };
- size_t i;
+ if (isNew) {
+ size_t lpf2HeapSizeInBytes;
- for (i = 0; i < ma_countof(libjackNames); ++i) {
- pContext->jack.jackSO = ma_dlopen(pContext, libjackNames[i]);
- if (pContext->jack.jackSO != NULL) {
- break;
+ result = ma_lpf2_get_heap_size(&lpf2Config, &lpf2HeapSizeInBytes);
+ if (result == MA_SUCCESS) {
+ result = ma_lpf2_init_preallocated(&lpf2Config, ma_offset_ptr(pHeap, heapLayout.lpf2Offset + (sizeof(ma_lpf2) * lpf2Count) + (ilpf2 * lpf2HeapSizeInBytes)), &pLPF->pLPF2[ilpf2]);
+ }
+ } else {
+ result = ma_lpf2_reinit(&lpf2Config, &pLPF->pLPF2[ilpf2]);
}
- }
-
- if (pContext->jack.jackSO == NULL) {
- return MA_NO_BACKEND;
- }
- pContext->jack.jack_client_open = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_client_open");
- pContext->jack.jack_client_close = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_client_close");
- pContext->jack.jack_client_name_size = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_client_name_size");
- pContext->jack.jack_set_process_callback = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_set_process_callback");
- pContext->jack.jack_set_buffer_size_callback = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_set_buffer_size_callback");
- pContext->jack.jack_on_shutdown = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_on_shutdown");
- pContext->jack.jack_get_sample_rate = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_get_sample_rate");
- pContext->jack.jack_get_buffer_size = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_get_buffer_size");
- pContext->jack.jack_get_ports = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_get_ports");
- pContext->jack.jack_activate = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_activate");
- pContext->jack.jack_deactivate = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_deactivate");
- pContext->jack.jack_connect = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_connect");
- pContext->jack.jack_port_register = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_port_register");
- pContext->jack.jack_port_name = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_port_name");
- pContext->jack.jack_port_get_buffer = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_port_get_buffer");
- pContext->jack.jack_free = (ma_proc)ma_dlsym(pContext, pContext->jack.jackSO, "jack_free");
-#else
- /*
- This strange assignment system is here just to ensure type safety of miniaudio's function pointer
- types. If anything differs slightly the compiler should throw a warning.
- */
- ma_jack_client_open_proc _jack_client_open = jack_client_open;
- ma_jack_client_close_proc _jack_client_close = jack_client_close;
- ma_jack_client_name_size_proc _jack_client_name_size = jack_client_name_size;
- ma_jack_set_process_callback_proc _jack_set_process_callback = jack_set_process_callback;
- ma_jack_set_buffer_size_callback_proc _jack_set_buffer_size_callback = jack_set_buffer_size_callback;
- ma_jack_on_shutdown_proc _jack_on_shutdown = jack_on_shutdown;
- ma_jack_get_sample_rate_proc _jack_get_sample_rate = jack_get_sample_rate;
- ma_jack_get_buffer_size_proc _jack_get_buffer_size = jack_get_buffer_size;
- ma_jack_get_ports_proc _jack_get_ports = jack_get_ports;
- ma_jack_activate_proc _jack_activate = jack_activate;
- ma_jack_deactivate_proc _jack_deactivate = jack_deactivate;
- ma_jack_connect_proc _jack_connect = jack_connect;
- ma_jack_port_register_proc _jack_port_register = jack_port_register;
- ma_jack_port_name_proc _jack_port_name = jack_port_name;
- ma_jack_port_get_buffer_proc _jack_port_get_buffer = jack_port_get_buffer;
- ma_jack_free_proc _jack_free = jack_free;
+ if (result != MA_SUCCESS) {
+ ma_uint32 jlpf1;
+ ma_uint32 jlpf2;
- pContext->jack.jack_client_open = (ma_proc)_jack_client_open;
- pContext->jack.jack_client_close = (ma_proc)_jack_client_close;
- pContext->jack.jack_client_name_size = (ma_proc)_jack_client_name_size;
- pContext->jack.jack_set_process_callback = (ma_proc)_jack_set_process_callback;
- pContext->jack.jack_set_buffer_size_callback = (ma_proc)_jack_set_buffer_size_callback;
- pContext->jack.jack_on_shutdown = (ma_proc)_jack_on_shutdown;
- pContext->jack.jack_get_sample_rate = (ma_proc)_jack_get_sample_rate;
- pContext->jack.jack_get_buffer_size = (ma_proc)_jack_get_buffer_size;
- pContext->jack.jack_get_ports = (ma_proc)_jack_get_ports;
- pContext->jack.jack_activate = (ma_proc)_jack_activate;
- pContext->jack.jack_deactivate = (ma_proc)_jack_deactivate;
- pContext->jack.jack_connect = (ma_proc)_jack_connect;
- pContext->jack.jack_port_register = (ma_proc)_jack_port_register;
- pContext->jack.jack_port_name = (ma_proc)_jack_port_name;
- pContext->jack.jack_port_get_buffer = (ma_proc)_jack_port_get_buffer;
- pContext->jack.jack_free = (ma_proc)_jack_free;
-#endif
+ for (jlpf1 = 0; jlpf1 < lpf1Count; jlpf1 += 1) {
+ ma_lpf1_uninit(&pLPF->pLPF1[jlpf1], NULL); /* No need for allocation callbacks here since we used a preallocated heap allocation. */
+ }
- if (pConfig->jack.pClientName != NULL) {
- pContext->jack.pClientName = ma_copy_string(pConfig->jack.pClientName, &pContext->allocationCallbacks);
- }
- pContext->jack.tryStartServer = pConfig->jack.tryStartServer;
+ for (jlpf2 = 0; jlpf2 < ilpf2; jlpf2 += 1) {
+ ma_lpf2_uninit(&pLPF->pLPF2[jlpf2], NULL); /* No need for allocation callbacks here since we used a preallocated heap allocation. */
+ }
- /*
- Getting here means the JACK library is installed, but it doesn't necessarily mean it's usable. We need to quickly test this by connecting
- a temporary client.
- */
- {
- ma_jack_client_t* pDummyClient;
- ma_result result = ma_context_open_client__jack(pContext, &pDummyClient);
- if (result != MA_SUCCESS) {
- ma_free(pContext->jack.pClientName, &pContext->allocationCallbacks);
- #ifndef MA_NO_RUNTIME_LINKING
- ma_dlclose(pContext, pContext->jack.jackSO);
- #endif
- return MA_NO_BACKEND;
+ return result;
}
-
- ((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pDummyClient);
}
-
- pCallbacks->onContextInit = ma_context_init__jack;
- pCallbacks->onContextUninit = ma_context_uninit__jack;
- pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__jack;
- pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__jack;
- pCallbacks->onDeviceInit = ma_device_init__jack;
- pCallbacks->onDeviceUninit = ma_device_uninit__jack;
- pCallbacks->onDeviceStart = ma_device_start__jack;
- pCallbacks->onDeviceStop = ma_device_stop__jack;
- pCallbacks->onDeviceRead = NULL; /* Not used because JACK is asynchronous. */
- pCallbacks->onDeviceWrite = NULL; /* Not used because JACK is asynchronous. */
- pCallbacks->onDeviceDataLoop = NULL; /* Not used because JACK is asynchronous. */
+ pLPF->lpf1Count = lpf1Count;
+ pLPF->lpf2Count = lpf2Count;
+ pLPF->format = pConfig->format;
+ pLPF->channels = pConfig->channels;
+ pLPF->sampleRate = pConfig->sampleRate;
return MA_SUCCESS;
}
-#endif /* JACK */
-
+MA_API ma_result ma_lpf_get_heap_size(const ma_lpf_config* pConfig, size_t* pHeapSizeInBytes)
+{
+ ma_result result;
+ ma_lpf_heap_layout heapLayout;
-/******************************************************************************
-
-Core Audio Backend
+ if (pHeapSizeInBytes == NULL) {
+ return MA_INVALID_ARGS;
+ }
-References
-==========
-- Technical Note TN2091: Device input using the HAL Output Audio Unit
- https://developer.apple.com/library/archive/technotes/tn2091/_index.html
+ *pHeapSizeInBytes = 0;
-******************************************************************************/
-#ifdef MA_HAS_COREAUDIO
-#include
+ result = ma_lpf_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
-#if defined(TARGET_OS_IPHONE) && TARGET_OS_IPHONE == 1
- #define MA_APPLE_MOBILE
- #if defined(TARGET_OS_TV) && TARGET_OS_TV == 1
- #define MA_APPLE_TV
- #endif
- #if defined(TARGET_OS_WATCH) && TARGET_OS_WATCH == 1
- #define MA_APPLE_WATCH
- #endif
-#else
- #define MA_APPLE_DESKTOP
-#endif
+ *pHeapSizeInBytes = heapLayout.sizeInBytes;
-#if defined(MA_APPLE_DESKTOP)
-#include
-#else
-#include
-#endif
+ return result;
+}
-#include
+MA_API ma_result ma_lpf_init_preallocated(const ma_lpf_config* pConfig, void* pHeap, ma_lpf* pLPF)
+{
+ if (pLPF == NULL) {
+ return MA_INVALID_ARGS;
+ }
-/* CoreFoundation */
-typedef Boolean (* ma_CFStringGetCString_proc)(CFStringRef theString, char* buffer, CFIndex bufferSize, CFStringEncoding encoding);
-typedef void (* ma_CFRelease_proc)(CFTypeRef cf);
+ MA_ZERO_OBJECT(pLPF);
-/* CoreAudio */
-#if defined(MA_APPLE_DESKTOP)
-typedef OSStatus (* ma_AudioObjectGetPropertyData_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, UInt32 inQualifierDataSize, const void* inQualifierData, UInt32* ioDataSize, void* outData);
-typedef OSStatus (* ma_AudioObjectGetPropertyDataSize_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, UInt32 inQualifierDataSize, const void* inQualifierData, UInt32* outDataSize);
-typedef OSStatus (* ma_AudioObjectSetPropertyData_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, UInt32 inQualifierDataSize, const void* inQualifierData, UInt32 inDataSize, const void* inData);
-typedef OSStatus (* ma_AudioObjectAddPropertyListener_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, AudioObjectPropertyListenerProc inListener, void* inClientData);
-typedef OSStatus (* ma_AudioObjectRemovePropertyListener_proc)(AudioObjectID inObjectID, const AudioObjectPropertyAddress* inAddress, AudioObjectPropertyListenerProc inListener, void* inClientData);
-#endif
+ return ma_lpf_reinit__internal(pConfig, pHeap, pLPF, /*isNew*/MA_TRUE);
+}
-/* AudioToolbox */
-typedef AudioComponent (* ma_AudioComponentFindNext_proc)(AudioComponent inComponent, const AudioComponentDescription* inDesc);
-typedef OSStatus (* ma_AudioComponentInstanceDispose_proc)(AudioComponentInstance inInstance);
-typedef OSStatus (* ma_AudioComponentInstanceNew_proc)(AudioComponent inComponent, AudioComponentInstance* outInstance);
-typedef OSStatus (* ma_AudioOutputUnitStart_proc)(AudioUnit inUnit);
-typedef OSStatus (* ma_AudioOutputUnitStop_proc)(AudioUnit inUnit);
-typedef OSStatus (* ma_AudioUnitAddPropertyListener_proc)(AudioUnit inUnit, AudioUnitPropertyID inID, AudioUnitPropertyListenerProc inProc, void* inProcUserData);
-typedef OSStatus (* ma_AudioUnitGetPropertyInfo_proc)(AudioUnit inUnit, AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, UInt32* outDataSize, Boolean* outWriteable);
-typedef OSStatus (* ma_AudioUnitGetProperty_proc)(AudioUnit inUnit, AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, void* outData, UInt32* ioDataSize);
-typedef OSStatus (* ma_AudioUnitSetProperty_proc)(AudioUnit inUnit, AudioUnitPropertyID inID, AudioUnitScope inScope, AudioUnitElement inElement, const void* inData, UInt32 inDataSize);
-typedef OSStatus (* ma_AudioUnitInitialize_proc)(AudioUnit inUnit);
-typedef OSStatus (* ma_AudioUnitRender_proc)(AudioUnit inUnit, AudioUnitRenderActionFlags* ioActionFlags, const AudioTimeStamp* inTimeStamp, UInt32 inOutputBusNumber, UInt32 inNumberFrames, AudioBufferList* ioData);
+MA_API ma_result ma_lpf_init(const ma_lpf_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_lpf* pLPF)
+{
+ ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
+ result = ma_lpf_get_heap_size(pConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
-#define MA_COREAUDIO_OUTPUT_BUS 0
-#define MA_COREAUDIO_INPUT_BUS 1
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+ } else {
+ pHeap = NULL;
+ }
-#if defined(MA_APPLE_DESKTOP)
-static ma_result ma_device_reinit_internal__coreaudio(ma_device* pDevice, ma_device_type deviceType, ma_bool32 disposePreviousAudioUnit);
-#endif
+ result = ma_lpf_init_preallocated(pConfig, pHeap, pLPF);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
+ }
-/*
-Core Audio
+ pLPF->_ownsHeap = MA_TRUE;
+ return MA_SUCCESS;
+}
-So far, Core Audio has been the worst backend to work with due to being both unintuitive and having almost no documentation
-apart from comments in the headers (which admittedly are quite good). For my own purposes, and for anybody out there whose
-needing to figure out how this darn thing works, I'm going to outline a few things here.
+MA_API void ma_lpf_uninit(ma_lpf* pLPF, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ ma_uint32 ilpf1;
+ ma_uint32 ilpf2;
-Since miniaudio is a fairly low-level API, one of the things it needs is control over specific devices, and it needs to be
-able to identify whether or not it can be used as playback and/or capture. The AudioObject API is the only one I've seen
-that supports this level of detail. There was some public domain sample code I stumbled across that used the AudioComponent
-and AudioUnit APIs, but I couldn't see anything that gave low-level control over device selection and capabilities (the
-distinction between playback and capture in particular). Therefore, miniaudio is using the AudioObject API.
+ if (pLPF == NULL) {
+ return;
+ }
-Most (all?) functions in the AudioObject API take a AudioObjectID as it's input. This is the device identifier. When
-retrieving global information, such as the device list, you use kAudioObjectSystemObject. When retrieving device-specific
-data, you pass in the ID for that device. In order to retrieve device-specific IDs you need to enumerate over each of the
-devices. This is done using the AudioObjectGetPropertyDataSize() and AudioObjectGetPropertyData() APIs which seem to be
-the central APIs for retrieving information about the system and specific devices.
+ for (ilpf1 = 0; ilpf1 < pLPF->lpf1Count; ilpf1 += 1) {
+ ma_lpf1_uninit(&pLPF->pLPF1[ilpf1], pAllocationCallbacks);
+ }
-To use the AudioObjectGetPropertyData() API you need to use the notion of a property address. A property address is a
-structure with three variables and is used to identify which property you are getting or setting. The first is the "selector"
-which is basically the specific property that you're wanting to retrieve or set. The second is the "scope", which is
-typically set to kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyScopeInput for input-specific properties and
-kAudioObjectPropertyScopeOutput for output-specific properties. The last is the "element" which is always set to
-kAudioObjectPropertyElementMaster in miniaudio's case. I don't know of any cases where this would be set to anything different.
+ for (ilpf2 = 0; ilpf2 < pLPF->lpf2Count; ilpf2 += 1) {
+ ma_lpf2_uninit(&pLPF->pLPF2[ilpf2], pAllocationCallbacks);
+ }
-Back to the earlier issue of device retrieval, you first use the AudioObjectGetPropertyDataSize() API to retrieve the size
-of the raw data which is just a list of AudioDeviceID's. You use the kAudioObjectSystemObject AudioObjectID, and a property
-address with the kAudioHardwarePropertyDevices selector and the kAudioObjectPropertyScopeGlobal scope. Once you have the
-size, allocate a block of memory of that size and then call AudioObjectGetPropertyData(). The data is just a list of
-AudioDeviceID's so just do "dataSize/sizeof(AudioDeviceID)" to know the device count.
-*/
+ if (pLPF->_ownsHeap) {
+ ma_free(pLPF->_pHeap, pAllocationCallbacks);
+ }
+}
-static ma_result ma_result_from_OSStatus(OSStatus status)
+MA_API ma_result ma_lpf_reinit(const ma_lpf_config* pConfig, ma_lpf* pLPF)
{
- switch (status)
- {
- case noErr: return MA_SUCCESS;
- #if defined(MA_APPLE_DESKTOP)
- case kAudioHardwareNotRunningError: return MA_DEVICE_NOT_STARTED;
- case kAudioHardwareUnspecifiedError: return MA_ERROR;
- case kAudioHardwareUnknownPropertyError: return MA_INVALID_ARGS;
- case kAudioHardwareBadPropertySizeError: return MA_INVALID_OPERATION;
- case kAudioHardwareIllegalOperationError: return MA_INVALID_OPERATION;
- case kAudioHardwareBadObjectError: return MA_INVALID_ARGS;
- case kAudioHardwareBadDeviceError: return MA_INVALID_ARGS;
- case kAudioHardwareBadStreamError: return MA_INVALID_ARGS;
- case kAudioHardwareUnsupportedOperationError: return MA_INVALID_OPERATION;
- case kAudioDeviceUnsupportedFormatError: return MA_FORMAT_NOT_SUPPORTED;
- case kAudioDevicePermissionsError: return MA_ACCESS_DENIED;
- #endif
- default: return MA_ERROR;
- }
+ return ma_lpf_reinit__internal(pConfig, NULL, pLPF, /*isNew*/MA_FALSE);
}
-#if 0
-static ma_channel ma_channel_from_AudioChannelBitmap(AudioChannelBitmap bit)
+static MA_INLINE void ma_lpf_process_pcm_frame_f32(ma_lpf* pLPF, float* pY, const void* pX)
{
- switch (bit)
- {
- case kAudioChannelBit_Left: return MA_CHANNEL_LEFT;
- case kAudioChannelBit_Right: return MA_CHANNEL_RIGHT;
- case kAudioChannelBit_Center: return MA_CHANNEL_FRONT_CENTER;
- case kAudioChannelBit_LFEScreen: return MA_CHANNEL_LFE;
- case kAudioChannelBit_LeftSurround: return MA_CHANNEL_BACK_LEFT;
- case kAudioChannelBit_RightSurround: return MA_CHANNEL_BACK_RIGHT;
- case kAudioChannelBit_LeftCenter: return MA_CHANNEL_FRONT_LEFT_CENTER;
- case kAudioChannelBit_RightCenter: return MA_CHANNEL_FRONT_RIGHT_CENTER;
- case kAudioChannelBit_CenterSurround: return MA_CHANNEL_BACK_CENTER;
- case kAudioChannelBit_LeftSurroundDirect: return MA_CHANNEL_SIDE_LEFT;
- case kAudioChannelBit_RightSurroundDirect: return MA_CHANNEL_SIDE_RIGHT;
- case kAudioChannelBit_TopCenterSurround: return MA_CHANNEL_TOP_CENTER;
- case kAudioChannelBit_VerticalHeightLeft: return MA_CHANNEL_TOP_FRONT_LEFT;
- case kAudioChannelBit_VerticalHeightCenter: return MA_CHANNEL_TOP_FRONT_CENTER;
- case kAudioChannelBit_VerticalHeightRight: return MA_CHANNEL_TOP_FRONT_RIGHT;
- case kAudioChannelBit_TopBackLeft: return MA_CHANNEL_TOP_BACK_LEFT;
- case kAudioChannelBit_TopBackCenter: return MA_CHANNEL_TOP_BACK_CENTER;
- case kAudioChannelBit_TopBackRight: return MA_CHANNEL_TOP_BACK_RIGHT;
- default: return MA_CHANNEL_NONE;
+ ma_uint32 ilpf1;
+ ma_uint32 ilpf2;
+
+ MA_ASSERT(pLPF->format == ma_format_f32);
+
+ MA_COPY_MEMORY(pY, pX, ma_get_bytes_per_frame(pLPF->format, pLPF->channels));
+
+ for (ilpf1 = 0; ilpf1 < pLPF->lpf1Count; ilpf1 += 1) {
+ ma_lpf1_process_pcm_frame_f32(&pLPF->pLPF1[ilpf1], pY, pY);
+ }
+
+ for (ilpf2 = 0; ilpf2 < pLPF->lpf2Count; ilpf2 += 1) {
+ ma_lpf2_process_pcm_frame_f32(&pLPF->pLPF2[ilpf2], pY, pY);
}
}
-#endif
-static ma_result ma_format_from_AudioStreamBasicDescription(const AudioStreamBasicDescription* pDescription, ma_format* pFormatOut)
+static MA_INLINE void ma_lpf_process_pcm_frame_s16(ma_lpf* pLPF, ma_int16* pY, const ma_int16* pX)
{
- MA_ASSERT(pDescription != NULL);
- MA_ASSERT(pFormatOut != NULL);
+ ma_uint32 ilpf1;
+ ma_uint32 ilpf2;
- *pFormatOut = ma_format_unknown; /* Safety. */
+ MA_ASSERT(pLPF->format == ma_format_s16);
- /* There's a few things miniaudio doesn't support. */
- if (pDescription->mFormatID != kAudioFormatLinearPCM) {
- return MA_FORMAT_NOT_SUPPORTED;
+ MA_COPY_MEMORY(pY, pX, ma_get_bytes_per_frame(pLPF->format, pLPF->channels));
+
+ for (ilpf1 = 0; ilpf1 < pLPF->lpf1Count; ilpf1 += 1) {
+ ma_lpf1_process_pcm_frame_s16(&pLPF->pLPF1[ilpf1], pY, pY);
}
- /* We don't support any non-packed formats that are aligned high. */
- if ((pDescription->mFormatFlags & kLinearPCMFormatFlagIsAlignedHigh) != 0) {
- return MA_FORMAT_NOT_SUPPORTED;
+ for (ilpf2 = 0; ilpf2 < pLPF->lpf2Count; ilpf2 += 1) {
+ ma_lpf2_process_pcm_frame_s16(&pLPF->pLPF2[ilpf2], pY, pY);
}
+}
- /* Only supporting native-endian. */
- if ((ma_is_little_endian() && (pDescription->mFormatFlags & kAudioFormatFlagIsBigEndian) != 0) || (ma_is_big_endian() && (pDescription->mFormatFlags & kAudioFormatFlagIsBigEndian) == 0)) {
- return MA_FORMAT_NOT_SUPPORTED;
+MA_API ma_result ma_lpf_process_pcm_frames(ma_lpf* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+{
+ ma_result result;
+ ma_uint32 ilpf1;
+ ma_uint32 ilpf2;
+
+ if (pLPF == NULL) {
+ return MA_INVALID_ARGS;
}
- /* We are not currently supporting non-interleaved formats (this will be added in a future version of miniaudio). */
- /*if ((pDescription->mFormatFlags & kAudioFormatFlagIsNonInterleaved) != 0) {
- return MA_FORMAT_NOT_SUPPORTED;
- }*/
+ /* Faster path for in-place. */
+ if (pFramesOut == pFramesIn) {
+ for (ilpf1 = 0; ilpf1 < pLPF->lpf1Count; ilpf1 += 1) {
+ result = ma_lpf1_process_pcm_frames(&pLPF->pLPF1[ilpf1], pFramesOut, pFramesOut, frameCount);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ }
- if ((pDescription->mFormatFlags & kLinearPCMFormatFlagIsFloat) != 0) {
- if (pDescription->mBitsPerChannel == 32) {
- *pFormatOut = ma_format_f32;
- return MA_SUCCESS;
+ for (ilpf2 = 0; ilpf2 < pLPF->lpf2Count; ilpf2 += 1) {
+ result = ma_lpf2_process_pcm_frames(&pLPF->pLPF2[ilpf2], pFramesOut, pFramesOut, frameCount);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
}
- } else {
- if ((pDescription->mFormatFlags & kLinearPCMFormatFlagIsSignedInteger) != 0) {
- if (pDescription->mBitsPerChannel == 16) {
- *pFormatOut = ma_format_s16;
- return MA_SUCCESS;
- } else if (pDescription->mBitsPerChannel == 24) {
- if (pDescription->mBytesPerFrame == (pDescription->mBitsPerChannel/8 * pDescription->mChannelsPerFrame)) {
- *pFormatOut = ma_format_s24;
- return MA_SUCCESS;
- } else {
- if (pDescription->mBytesPerFrame/pDescription->mChannelsPerFrame == sizeof(ma_int32)) {
- /* TODO: Implement ma_format_s24_32. */
- /**pFormatOut = ma_format_s24_32;*/
- /*return MA_SUCCESS;*/
- return MA_FORMAT_NOT_SUPPORTED;
- }
- }
- } else if (pDescription->mBitsPerChannel == 32) {
- *pFormatOut = ma_format_s32;
- return MA_SUCCESS;
+ }
+
+ /* Slightly slower path for copying. */
+ if (pFramesOut != pFramesIn) {
+ ma_uint32 iFrame;
+
+ /* */ if (pLPF->format == ma_format_f32) {
+ /* */ float* pFramesOutF32 = ( float*)pFramesOut;
+ const float* pFramesInF32 = (const float*)pFramesIn;
+
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_lpf_process_pcm_frame_f32(pLPF, pFramesOutF32, pFramesInF32);
+ pFramesOutF32 += pLPF->channels;
+ pFramesInF32 += pLPF->channels;
}
- } else {
- if (pDescription->mBitsPerChannel == 8) {
- *pFormatOut = ma_format_u8;
- return MA_SUCCESS;
+ } else if (pLPF->format == ma_format_s16) {
+ /* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut;
+ const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn;
+
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_lpf_process_pcm_frame_s16(pLPF, pFramesOutS16, pFramesInS16);
+ pFramesOutS16 += pLPF->channels;
+ pFramesInS16 += pLPF->channels;
}
+ } else {
+ MA_ASSERT(MA_FALSE);
+ return MA_INVALID_OPERATION; /* Should never hit this. */
}
}
- /* Getting here means the format is not supported. */
- return MA_FORMAT_NOT_SUPPORTED;
+ return MA_SUCCESS;
}
-#if defined(MA_APPLE_DESKTOP)
-static ma_channel ma_channel_from_AudioChannelLabel(AudioChannelLabel label)
+MA_API ma_uint32 ma_lpf_get_latency(const ma_lpf* pLPF)
{
- switch (label)
- {
- case kAudioChannelLabel_Unknown: return MA_CHANNEL_NONE;
- case kAudioChannelLabel_Unused: return MA_CHANNEL_NONE;
- case kAudioChannelLabel_UseCoordinates: return MA_CHANNEL_NONE;
- case kAudioChannelLabel_Left: return MA_CHANNEL_LEFT;
- case kAudioChannelLabel_Right: return MA_CHANNEL_RIGHT;
- case kAudioChannelLabel_Center: return MA_CHANNEL_FRONT_CENTER;
- case kAudioChannelLabel_LFEScreen: return MA_CHANNEL_LFE;
- case kAudioChannelLabel_LeftSurround: return MA_CHANNEL_BACK_LEFT;
- case kAudioChannelLabel_RightSurround: return MA_CHANNEL_BACK_RIGHT;
- case kAudioChannelLabel_LeftCenter: return MA_CHANNEL_FRONT_LEFT_CENTER;
- case kAudioChannelLabel_RightCenter: return MA_CHANNEL_FRONT_RIGHT_CENTER;
- case kAudioChannelLabel_CenterSurround: return MA_CHANNEL_BACK_CENTER;
- case kAudioChannelLabel_LeftSurroundDirect: return MA_CHANNEL_SIDE_LEFT;
- case kAudioChannelLabel_RightSurroundDirect: return MA_CHANNEL_SIDE_RIGHT;
- case kAudioChannelLabel_TopCenterSurround: return MA_CHANNEL_TOP_CENTER;
- case kAudioChannelLabel_VerticalHeightLeft: return MA_CHANNEL_TOP_FRONT_LEFT;
- case kAudioChannelLabel_VerticalHeightCenter: return MA_CHANNEL_TOP_FRONT_CENTER;
- case kAudioChannelLabel_VerticalHeightRight: return MA_CHANNEL_TOP_FRONT_RIGHT;
- case kAudioChannelLabel_TopBackLeft: return MA_CHANNEL_TOP_BACK_LEFT;
- case kAudioChannelLabel_TopBackCenter: return MA_CHANNEL_TOP_BACK_CENTER;
- case kAudioChannelLabel_TopBackRight: return MA_CHANNEL_TOP_BACK_RIGHT;
- case kAudioChannelLabel_RearSurroundLeft: return MA_CHANNEL_BACK_LEFT;
- case kAudioChannelLabel_RearSurroundRight: return MA_CHANNEL_BACK_RIGHT;
- case kAudioChannelLabel_LeftWide: return MA_CHANNEL_SIDE_LEFT;
- case kAudioChannelLabel_RightWide: return MA_CHANNEL_SIDE_RIGHT;
- case kAudioChannelLabel_LFE2: return MA_CHANNEL_LFE;
- case kAudioChannelLabel_LeftTotal: return MA_CHANNEL_LEFT;
- case kAudioChannelLabel_RightTotal: return MA_CHANNEL_RIGHT;
- case kAudioChannelLabel_HearingImpaired: return MA_CHANNEL_NONE;
- case kAudioChannelLabel_Narration: return MA_CHANNEL_MONO;
- case kAudioChannelLabel_Mono: return MA_CHANNEL_MONO;
- case kAudioChannelLabel_DialogCentricMix: return MA_CHANNEL_MONO;
- case kAudioChannelLabel_CenterSurroundDirect: return MA_CHANNEL_BACK_CENTER;
- case kAudioChannelLabel_Haptic: return MA_CHANNEL_NONE;
- case kAudioChannelLabel_Ambisonic_W: return MA_CHANNEL_NONE;
- case kAudioChannelLabel_Ambisonic_X: return MA_CHANNEL_NONE;
- case kAudioChannelLabel_Ambisonic_Y: return MA_CHANNEL_NONE;
- case kAudioChannelLabel_Ambisonic_Z: return MA_CHANNEL_NONE;
- case kAudioChannelLabel_MS_Mid: return MA_CHANNEL_LEFT;
- case kAudioChannelLabel_MS_Side: return MA_CHANNEL_RIGHT;
- case kAudioChannelLabel_XY_X: return MA_CHANNEL_LEFT;
- case kAudioChannelLabel_XY_Y: return MA_CHANNEL_RIGHT;
- case kAudioChannelLabel_HeadphonesLeft: return MA_CHANNEL_LEFT;
- case kAudioChannelLabel_HeadphonesRight: return MA_CHANNEL_RIGHT;
- case kAudioChannelLabel_ClickTrack: return MA_CHANNEL_NONE;
- case kAudioChannelLabel_ForeignLanguage: return MA_CHANNEL_NONE;
- case kAudioChannelLabel_Discrete: return MA_CHANNEL_NONE;
- case kAudioChannelLabel_Discrete_0: return MA_CHANNEL_AUX_0;
- case kAudioChannelLabel_Discrete_1: return MA_CHANNEL_AUX_1;
- case kAudioChannelLabel_Discrete_2: return MA_CHANNEL_AUX_2;
- case kAudioChannelLabel_Discrete_3: return MA_CHANNEL_AUX_3;
- case kAudioChannelLabel_Discrete_4: return MA_CHANNEL_AUX_4;
- case kAudioChannelLabel_Discrete_5: return MA_CHANNEL_AUX_5;
- case kAudioChannelLabel_Discrete_6: return MA_CHANNEL_AUX_6;
- case kAudioChannelLabel_Discrete_7: return MA_CHANNEL_AUX_7;
- case kAudioChannelLabel_Discrete_8: return MA_CHANNEL_AUX_8;
- case kAudioChannelLabel_Discrete_9: return MA_CHANNEL_AUX_9;
- case kAudioChannelLabel_Discrete_10: return MA_CHANNEL_AUX_10;
- case kAudioChannelLabel_Discrete_11: return MA_CHANNEL_AUX_11;
- case kAudioChannelLabel_Discrete_12: return MA_CHANNEL_AUX_12;
- case kAudioChannelLabel_Discrete_13: return MA_CHANNEL_AUX_13;
- case kAudioChannelLabel_Discrete_14: return MA_CHANNEL_AUX_14;
- case kAudioChannelLabel_Discrete_15: return MA_CHANNEL_AUX_15;
- case kAudioChannelLabel_Discrete_65535: return MA_CHANNEL_NONE;
-
- #if 0 /* Introduced in a later version of macOS. */
- case kAudioChannelLabel_HOA_ACN: return MA_CHANNEL_NONE;
- case kAudioChannelLabel_HOA_ACN_0: return MA_CHANNEL_AUX_0;
- case kAudioChannelLabel_HOA_ACN_1: return MA_CHANNEL_AUX_1;
- case kAudioChannelLabel_HOA_ACN_2: return MA_CHANNEL_AUX_2;
- case kAudioChannelLabel_HOA_ACN_3: return MA_CHANNEL_AUX_3;
- case kAudioChannelLabel_HOA_ACN_4: return MA_CHANNEL_AUX_4;
- case kAudioChannelLabel_HOA_ACN_5: return MA_CHANNEL_AUX_5;
- case kAudioChannelLabel_HOA_ACN_6: return MA_CHANNEL_AUX_6;
- case kAudioChannelLabel_HOA_ACN_7: return MA_CHANNEL_AUX_7;
- case kAudioChannelLabel_HOA_ACN_8: return MA_CHANNEL_AUX_8;
- case kAudioChannelLabel_HOA_ACN_9: return MA_CHANNEL_AUX_9;
- case kAudioChannelLabel_HOA_ACN_10: return MA_CHANNEL_AUX_10;
- case kAudioChannelLabel_HOA_ACN_11: return MA_CHANNEL_AUX_11;
- case kAudioChannelLabel_HOA_ACN_12: return MA_CHANNEL_AUX_12;
- case kAudioChannelLabel_HOA_ACN_13: return MA_CHANNEL_AUX_13;
- case kAudioChannelLabel_HOA_ACN_14: return MA_CHANNEL_AUX_14;
- case kAudioChannelLabel_HOA_ACN_15: return MA_CHANNEL_AUX_15;
- case kAudioChannelLabel_HOA_ACN_65024: return MA_CHANNEL_NONE;
- #endif
-
- default: return MA_CHANNEL_NONE;
+ if (pLPF == NULL) {
+ return 0;
}
+
+ return pLPF->lpf2Count*2 + pLPF->lpf1Count;
}
-static ma_result ma_get_channel_map_from_AudioChannelLayout(AudioChannelLayout* pChannelLayout, ma_channel* pChannelMap, size_t channelMapCap)
-{
- MA_ASSERT(pChannelLayout != NULL);
- if (pChannelLayout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelDescriptions) {
- UInt32 iChannel;
- for (iChannel = 0; iChannel < pChannelLayout->mNumberChannelDescriptions && iChannel < channelMapCap; ++iChannel) {
- pChannelMap[iChannel] = ma_channel_from_AudioChannelLabel(pChannelLayout->mChannelDescriptions[iChannel].mChannelLabel);
- }
- } else
-#if 0
- if (pChannelLayout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelBitmap) {
- /* This is the same kind of system that's used by Windows audio APIs. */
- UInt32 iChannel = 0;
- UInt32 iBit;
- AudioChannelBitmap bitmap = pChannelLayout->mChannelBitmap;
- for (iBit = 0; iBit < 32 && iChannel < channelMapCap; ++iBit) {
- AudioChannelBitmap bit = bitmap & (1 << iBit);
- if (bit != 0) {
- pChannelMap[iChannel++] = ma_channel_from_AudioChannelBit(bit);
- }
- }
- } else
-#endif
- {
- /*
- Need to use the tag to determine the channel map. For now I'm just assuming a default channel map, but later on this should
- be updated to determine the mapping based on the tag.
- */
- UInt32 channelCount;
+/**************************************************************************************************************************************************************
- /* Our channel map retrieval APIs below take 32-bit integers, so we'll want to clamp the channel map capacity. */
- if (channelMapCap > 0xFFFFFFFF) {
- channelMapCap = 0xFFFFFFFF;
- }
+High-Pass Filtering
- channelCount = ma_min(AudioChannelLayoutTag_GetNumberOfChannels(pChannelLayout->mChannelLayoutTag), (UInt32)channelMapCap);
+**************************************************************************************************************************************************************/
+MA_API ma_hpf1_config ma_hpf1_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency)
+{
+ ma_hpf1_config config;
- switch (pChannelLayout->mChannelLayoutTag)
- {
- case kAudioChannelLayoutTag_Mono:
- case kAudioChannelLayoutTag_Stereo:
- case kAudioChannelLayoutTag_StereoHeadphones:
- case kAudioChannelLayoutTag_MatrixStereo:
- case kAudioChannelLayoutTag_MidSide:
- case kAudioChannelLayoutTag_XY:
- case kAudioChannelLayoutTag_Binaural:
- case kAudioChannelLayoutTag_Ambisonic_B_Format:
- {
- ma_get_standard_channel_map(ma_standard_channel_map_default, channelCount, pChannelMap);
- } break;
+ MA_ZERO_OBJECT(&config);
+ config.format = format;
+ config.channels = channels;
+ config.sampleRate = sampleRate;
+ config.cutoffFrequency = cutoffFrequency;
- case kAudioChannelLayoutTag_Octagonal:
- {
- pChannelMap[7] = MA_CHANNEL_SIDE_RIGHT;
- pChannelMap[6] = MA_CHANNEL_SIDE_LEFT;
- } /* Intentional fallthrough. */
- case kAudioChannelLayoutTag_Hexagonal:
- {
- pChannelMap[5] = MA_CHANNEL_BACK_CENTER;
- } /* Intentional fallthrough. */
- case kAudioChannelLayoutTag_Pentagonal:
- {
- pChannelMap[4] = MA_CHANNEL_FRONT_CENTER;
- } /* Intentional fallghrough. */
- case kAudioChannelLayoutTag_Quadraphonic:
- {
- pChannelMap[3] = MA_CHANNEL_BACK_RIGHT;
- pChannelMap[2] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[1] = MA_CHANNEL_RIGHT;
- pChannelMap[0] = MA_CHANNEL_LEFT;
- } break;
+ return config;
+}
- /* TODO: Add support for more tags here. */
+MA_API ma_hpf2_config ma_hpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q)
+{
+ ma_hpf2_config config;
- default:
- {
- ma_get_standard_channel_map(ma_standard_channel_map_default, channelCount, pChannelMap);
- } break;
- }
+ MA_ZERO_OBJECT(&config);
+ config.format = format;
+ config.channels = channels;
+ config.sampleRate = sampleRate;
+ config.cutoffFrequency = cutoffFrequency;
+ config.q = q;
+
+ /* Q cannot be 0 or else it'll result in a division by 0. In this case just default to 0.707107. */
+ if (config.q == 0) {
+ config.q = 0.707107;
}
- return MA_SUCCESS;
+ return config;
}
-static ma_result ma_get_device_object_ids__coreaudio(ma_context* pContext, UInt32* pDeviceCount, AudioObjectID** ppDeviceObjectIDs) /* NOTE: Free the returned buffer with ma_free(). */
-{
- AudioObjectPropertyAddress propAddressDevices;
- UInt32 deviceObjectsDataSize;
- OSStatus status;
- AudioObjectID* pDeviceObjectIDs;
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pDeviceCount != NULL);
- MA_ASSERT(ppDeviceObjectIDs != NULL);
+typedef struct
+{
+ size_t sizeInBytes;
+ size_t r1Offset;
+} ma_hpf1_heap_layout;
- /* Safety. */
- *pDeviceCount = 0;
- *ppDeviceObjectIDs = NULL;
+static ma_result ma_hpf1_get_heap_layout(const ma_hpf1_config* pConfig, ma_hpf1_heap_layout* pHeapLayout)
+{
+ MA_ASSERT(pHeapLayout != NULL);
- propAddressDevices.mSelector = kAudioHardwarePropertyDevices;
- propAddressDevices.mScope = kAudioObjectPropertyScopeGlobal;
- propAddressDevices.mElement = kAudioObjectPropertyElementMaster;
+ MA_ZERO_OBJECT(pHeapLayout);
- status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(kAudioObjectSystemObject, &propAddressDevices, 0, NULL, &deviceObjectsDataSize);
- if (status != noErr) {
- return ma_result_from_OSStatus(status);
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- pDeviceObjectIDs = (AudioObjectID*)ma_malloc(deviceObjectsDataSize, &pContext->allocationCallbacks);
- if (pDeviceObjectIDs == NULL) {
- return MA_OUT_OF_MEMORY;
+ if (pConfig->channels == 0) {
+ return MA_INVALID_ARGS;
}
- status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(kAudioObjectSystemObject, &propAddressDevices, 0, NULL, &deviceObjectsDataSize, pDeviceObjectIDs);
- if (status != noErr) {
- ma_free(pDeviceObjectIDs, &pContext->allocationCallbacks);
- return ma_result_from_OSStatus(status);
- }
+ pHeapLayout->sizeInBytes = 0;
+
+ /* R1 */
+ pHeapLayout->r1Offset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += sizeof(ma_biquad_coefficient) * pConfig->channels;
- *pDeviceCount = deviceObjectsDataSize / sizeof(AudioObjectID);
- *ppDeviceObjectIDs = pDeviceObjectIDs;
+ /* Make sure allocation size is aligned. */
+ pHeapLayout->sizeInBytes = ma_align_64(pHeapLayout->sizeInBytes);
return MA_SUCCESS;
}
-static ma_result ma_get_AudioObject_uid_as_CFStringRef(ma_context* pContext, AudioObjectID objectID, CFStringRef* pUID)
+MA_API ma_result ma_hpf1_get_heap_size(const ma_hpf1_config* pConfig, size_t* pHeapSizeInBytes)
{
- AudioObjectPropertyAddress propAddress;
- UInt32 dataSize;
- OSStatus status;
-
- MA_ASSERT(pContext != NULL);
+ ma_result result;
+ ma_hpf1_heap_layout heapLayout;
- propAddress.mSelector = kAudioDevicePropertyDeviceUID;
- propAddress.mScope = kAudioObjectPropertyScopeGlobal;
- propAddress.mElement = kAudioObjectPropertyElementMaster;
+ if (pHeapSizeInBytes == NULL) {
+ return MA_INVALID_ARGS;
+ }
- dataSize = sizeof(*pUID);
- status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(objectID, &propAddress, 0, NULL, &dataSize, pUID);
- if (status != noErr) {
- return ma_result_from_OSStatus(status);
+ result = ma_hpf1_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
}
+ *pHeapSizeInBytes = heapLayout.sizeInBytes;
+
return MA_SUCCESS;
}
-static ma_result ma_get_AudioObject_uid(ma_context* pContext, AudioObjectID objectID, size_t bufferSize, char* bufferOut)
+MA_API ma_result ma_hpf1_init_preallocated(const ma_hpf1_config* pConfig, void* pHeap, ma_hpf1* pLPF)
{
- CFStringRef uid;
ma_result result;
+ ma_hpf1_heap_layout heapLayout;
- MA_ASSERT(pContext != NULL);
+ if (pLPF == NULL) {
+ return MA_INVALID_ARGS;
+ }
- result = ma_get_AudioObject_uid_as_CFStringRef(pContext, objectID, &uid);
+ MA_ZERO_OBJECT(pLPF);
+
+ result = ma_hpf1_get_heap_layout(pConfig, &heapLayout);
if (result != MA_SUCCESS) {
return result;
}
- if (!((ma_CFStringGetCString_proc)pContext->coreaudio.CFStringGetCString)(uid, bufferOut, bufferSize, kCFStringEncodingUTF8)) {
- return MA_ERROR;
- }
+ pLPF->_pHeap = pHeap;
+ MA_ZERO_MEMORY(pHeap, heapLayout.sizeInBytes);
- ((ma_CFRelease_proc)pContext->coreaudio.CFRelease)(uid);
- return MA_SUCCESS;
+ pLPF->pR1 = (ma_biquad_coefficient*)ma_offset_ptr(pHeap, heapLayout.r1Offset);
+
+ return ma_hpf1_reinit(pConfig, pLPF);
}
-static ma_result ma_get_AudioObject_name(ma_context* pContext, AudioObjectID objectID, size_t bufferSize, char* bufferOut)
+MA_API ma_result ma_hpf1_init(const ma_hpf1_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hpf1* pLPF)
{
- AudioObjectPropertyAddress propAddress;
- CFStringRef deviceName = NULL;
- UInt32 dataSize;
- OSStatus status;
-
- MA_ASSERT(pContext != NULL);
+ ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
- propAddress.mSelector = kAudioDevicePropertyDeviceNameCFString;
- propAddress.mScope = kAudioObjectPropertyScopeGlobal;
- propAddress.mElement = kAudioObjectPropertyElementMaster;
+ result = ma_hpf1_get_heap_size(pConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- dataSize = sizeof(deviceName);
- status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(objectID, &propAddress, 0, NULL, &dataSize, &deviceName);
- if (status != noErr) {
- return ma_result_from_OSStatus(status);
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+ } else {
+ pHeap = NULL;
}
- if (!((ma_CFStringGetCString_proc)pContext->coreaudio.CFStringGetCString)(deviceName, bufferOut, bufferSize, kCFStringEncodingUTF8)) {
- return MA_ERROR;
+ result = ma_hpf1_init_preallocated(pConfig, pHeap, pLPF);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
}
- ((ma_CFRelease_proc)pContext->coreaudio.CFRelease)(deviceName);
+ pLPF->_ownsHeap = MA_TRUE;
return MA_SUCCESS;
}
-static ma_bool32 ma_does_AudioObject_support_scope(ma_context* pContext, AudioObjectID deviceObjectID, AudioObjectPropertyScope scope)
+MA_API void ma_hpf1_uninit(ma_hpf1* pHPF, const ma_allocation_callbacks* pAllocationCallbacks)
{
- AudioObjectPropertyAddress propAddress;
- UInt32 dataSize;
- OSStatus status;
- AudioBufferList* pBufferList;
- ma_bool32 isSupported;
+ if (pHPF == NULL) {
+ return;
+ }
- MA_ASSERT(pContext != NULL);
+ if (pHPF->_ownsHeap) {
+ ma_free(pHPF->_pHeap, pAllocationCallbacks);
+ }
+}
- /* To know whether or not a device is an input device we need ot look at the stream configuration. If it has an output channel it's a playback device. */
- propAddress.mSelector = kAudioDevicePropertyStreamConfiguration;
- propAddress.mScope = scope;
- propAddress.mElement = kAudioObjectPropertyElementMaster;
+MA_API ma_result ma_hpf1_reinit(const ma_hpf1_config* pConfig, ma_hpf1* pHPF)
+{
+ double a;
- status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(deviceObjectID, &propAddress, 0, NULL, &dataSize);
- if (status != noErr) {
- return MA_FALSE;
+ if (pHPF == NULL || pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- pBufferList = (AudioBufferList*)ma__malloc_from_callbacks(dataSize, &pContext->allocationCallbacks);
- if (pBufferList == NULL) {
- return MA_FALSE; /* Out of memory. */
+ /* Only supporting f32 and s16. */
+ if (pConfig->format != ma_format_f32 && pConfig->format != ma_format_s16) {
+ return MA_INVALID_ARGS;
}
- status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pBufferList);
- if (status != noErr) {
- ma__free_from_callbacks(pBufferList, &pContext->allocationCallbacks);
- return MA_FALSE;
+ /* The format cannot be changed after initialization. */
+ if (pHPF->format != ma_format_unknown && pHPF->format != pConfig->format) {
+ return MA_INVALID_OPERATION;
}
- isSupported = MA_FALSE;
- if (pBufferList->mNumberBuffers > 0) {
- isSupported = MA_TRUE;
+ /* The channel count cannot be changed after initialization. */
+ if (pHPF->channels != 0 && pHPF->channels != pConfig->channels) {
+ return MA_INVALID_OPERATION;
}
- ma__free_from_callbacks(pBufferList, &pContext->allocationCallbacks);
- return isSupported;
-}
+ pHPF->format = pConfig->format;
+ pHPF->channels = pConfig->channels;
-static ma_bool32 ma_does_AudioObject_support_playback(ma_context* pContext, AudioObjectID deviceObjectID)
-{
- return ma_does_AudioObject_support_scope(pContext, deviceObjectID, kAudioObjectPropertyScopeOutput);
+ a = ma_expd(-2 * MA_PI_D * pConfig->cutoffFrequency / pConfig->sampleRate);
+ if (pConfig->format == ma_format_f32) {
+ pHPF->a.f32 = (float)a;
+ } else {
+ pHPF->a.s32 = ma_biquad_float_to_fp(a);
+ }
+
+ return MA_SUCCESS;
}
-static ma_bool32 ma_does_AudioObject_support_capture(ma_context* pContext, AudioObjectID deviceObjectID)
+static MA_INLINE void ma_hpf1_process_pcm_frame_f32(ma_hpf1* pHPF, float* pY, const float* pX)
{
- return ma_does_AudioObject_support_scope(pContext, deviceObjectID, kAudioObjectPropertyScopeInput);
-}
+ ma_uint32 c;
+ const ma_uint32 channels = pHPF->channels;
+ const float a = 1 - pHPF->a.f32;
+ const float b = 1 - a;
+
+ MA_ASSUME(channels > 0);
+ for (c = 0; c < channels; c += 1) {
+ float r1 = pHPF->pR1[c].f32;
+ float x = pX[c];
+ float y;
+ y = b*x - a*r1;
-static ma_result ma_get_AudioObject_stream_descriptions(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, UInt32* pDescriptionCount, AudioStreamRangedDescription** ppDescriptions) /* NOTE: Free the returned pointer with ma_free(). */
+ pY[c] = y;
+ pHPF->pR1[c].f32 = y;
+ }
+}
+
+static MA_INLINE void ma_hpf1_process_pcm_frame_s16(ma_hpf1* pHPF, ma_int16* pY, const ma_int16* pX)
{
- AudioObjectPropertyAddress propAddress;
- UInt32 dataSize;
- OSStatus status;
- AudioStreamRangedDescription* pDescriptions;
+ ma_uint32 c;
+ const ma_uint32 channels = pHPF->channels;
+ const ma_int32 a = ((1 << MA_BIQUAD_FIXED_POINT_SHIFT) - pHPF->a.s32);
+ const ma_int32 b = ((1 << MA_BIQUAD_FIXED_POINT_SHIFT) - a);
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pDescriptionCount != NULL);
- MA_ASSERT(ppDescriptions != NULL);
+ MA_ASSUME(channels > 0);
+ for (c = 0; c < channels; c += 1) {
+ ma_int32 r1 = pHPF->pR1[c].s32;
+ ma_int32 x = pX[c];
+ ma_int32 y;
- /*
- TODO: Experiment with kAudioStreamPropertyAvailablePhysicalFormats instead of (or in addition to) kAudioStreamPropertyAvailableVirtualFormats. My
- MacBook Pro uses s24/32 format, however, which miniaudio does not currently support.
- */
- propAddress.mSelector = kAudioStreamPropertyAvailableVirtualFormats; /*kAudioStreamPropertyAvailablePhysicalFormats;*/
- propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput;
- propAddress.mElement = kAudioObjectPropertyElementMaster;
+ y = (b*x - a*r1) >> MA_BIQUAD_FIXED_POINT_SHIFT;
- status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(deviceObjectID, &propAddress, 0, NULL, &dataSize);
- if (status != noErr) {
- return ma_result_from_OSStatus(status);
+ pY[c] = (ma_int16)y;
+ pHPF->pR1[c].s32 = (ma_int32)y;
}
+}
- pDescriptions = (AudioStreamRangedDescription*)ma_malloc(dataSize, &pContext->allocationCallbacks);
- if (pDescriptions == NULL) {
- return MA_OUT_OF_MEMORY;
+MA_API ma_result ma_hpf1_process_pcm_frames(ma_hpf1* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+{
+ ma_uint32 n;
+
+ if (pHPF == NULL || pFramesOut == NULL || pFramesIn == NULL) {
+ return MA_INVALID_ARGS;
}
- status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pDescriptions);
- if (status != noErr) {
- ma_free(pDescriptions, &pContext->allocationCallbacks);
- return ma_result_from_OSStatus(status);
+ /* Note that the logic below needs to support in-place filtering. That is, it must support the case where pFramesOut and pFramesIn are the same. */
+
+ if (pHPF->format == ma_format_f32) {
+ /* */ float* pY = ( float*)pFramesOut;
+ const float* pX = (const float*)pFramesIn;
+
+ for (n = 0; n < frameCount; n += 1) {
+ ma_hpf1_process_pcm_frame_f32(pHPF, pY, pX);
+ pY += pHPF->channels;
+ pX += pHPF->channels;
+ }
+ } else if (pHPF->format == ma_format_s16) {
+ /* */ ma_int16* pY = ( ma_int16*)pFramesOut;
+ const ma_int16* pX = (const ma_int16*)pFramesIn;
+
+ for (n = 0; n < frameCount; n += 1) {
+ ma_hpf1_process_pcm_frame_s16(pHPF, pY, pX);
+ pY += pHPF->channels;
+ pX += pHPF->channels;
+ }
+ } else {
+ MA_ASSERT(MA_FALSE);
+ return MA_INVALID_ARGS; /* Format not supported. Should never hit this because it's checked in ma_biquad_init() and ma_biquad_reinit(). */
}
- *pDescriptionCount = dataSize / sizeof(*pDescriptions);
- *ppDescriptions = pDescriptions;
return MA_SUCCESS;
}
+MA_API ma_uint32 ma_hpf1_get_latency(const ma_hpf1* pHPF)
+{
+ if (pHPF == NULL) {
+ return 0;
+ }
-static ma_result ma_get_AudioObject_channel_layout(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, AudioChannelLayout** ppChannelLayout) /* NOTE: Free the returned pointer with ma_free(). */
+ return 1;
+}
+
+
+static MA_INLINE ma_biquad_config ma_hpf2__get_biquad_config(const ma_hpf2_config* pConfig)
{
- AudioObjectPropertyAddress propAddress;
- UInt32 dataSize;
- OSStatus status;
- AudioChannelLayout* pChannelLayout;
+ ma_biquad_config bqConfig;
+ double q;
+ double w;
+ double s;
+ double c;
+ double a;
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(ppChannelLayout != NULL);
+ MA_ASSERT(pConfig != NULL);
- *ppChannelLayout = NULL; /* Safety. */
+ q = pConfig->q;
+ w = 2 * MA_PI_D * pConfig->cutoffFrequency / pConfig->sampleRate;
+ s = ma_sind(w);
+ c = ma_cosd(w);
+ a = s / (2*q);
- propAddress.mSelector = kAudioDevicePropertyPreferredChannelLayout;
- propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput;
- propAddress.mElement = kAudioObjectPropertyElementMaster;
+ bqConfig.b0 = (1 + c) / 2;
+ bqConfig.b1 = -(1 + c);
+ bqConfig.b2 = (1 + c) / 2;
+ bqConfig.a0 = 1 + a;
+ bqConfig.a1 = -2 * c;
+ bqConfig.a2 = 1 - a;
- status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(deviceObjectID, &propAddress, 0, NULL, &dataSize);
- if (status != noErr) {
- return ma_result_from_OSStatus(status);
- }
+ bqConfig.format = pConfig->format;
+ bqConfig.channels = pConfig->channels;
- pChannelLayout = (AudioChannelLayout*)ma_malloc(dataSize, &pContext->allocationCallbacks);
- if (pChannelLayout == NULL) {
- return MA_OUT_OF_MEMORY;
- }
+ return bqConfig;
+}
- status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pChannelLayout);
- if (status != noErr) {
- ma_free(pChannelLayout, &pContext->allocationCallbacks);
- return ma_result_from_OSStatus(status);
- }
+MA_API ma_result ma_hpf2_get_heap_size(const ma_hpf2_config* pConfig, size_t* pHeapSizeInBytes)
+{
+ ma_biquad_config bqConfig;
+ bqConfig = ma_hpf2__get_biquad_config(pConfig);
- *ppChannelLayout = pChannelLayout;
- return MA_SUCCESS;
+ return ma_biquad_get_heap_size(&bqConfig, pHeapSizeInBytes);
}
-static ma_result ma_get_AudioObject_channel_count(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32* pChannelCount)
+MA_API ma_result ma_hpf2_init_preallocated(const ma_hpf2_config* pConfig, void* pHeap, ma_hpf2* pHPF)
{
- AudioChannelLayout* pChannelLayout;
ma_result result;
+ ma_biquad_config bqConfig;
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pChannelCount != NULL);
+ if (pHPF == NULL) {
+ return MA_INVALID_ARGS;
+ }
- *pChannelCount = 0; /* Safety. */
+ MA_ZERO_OBJECT(pHPF);
- result = ma_get_AudioObject_channel_layout(pContext, deviceObjectID, deviceType, &pChannelLayout);
- if (result != MA_SUCCESS) {
- return result;
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pChannelLayout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelDescriptions) {
- *pChannelCount = pChannelLayout->mNumberChannelDescriptions;
- } else if (pChannelLayout->mChannelLayoutTag == kAudioChannelLayoutTag_UseChannelBitmap) {
- *pChannelCount = ma_count_set_bits(pChannelLayout->mChannelBitmap);
- } else {
- *pChannelCount = AudioChannelLayoutTag_GetNumberOfChannels(pChannelLayout->mChannelLayoutTag);
+ bqConfig = ma_hpf2__get_biquad_config(pConfig);
+ result = ma_biquad_init_preallocated(&bqConfig, pHeap, &pHPF->bq);
+ if (result != MA_SUCCESS) {
+ return result;
}
- ma_free(pChannelLayout, &pContext->allocationCallbacks);
return MA_SUCCESS;
}
-#if 0
-static ma_result ma_get_AudioObject_channel_map(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_channel* pChannelMap, size_t channelMapCap)
+MA_API ma_result ma_hpf2_init(const ma_hpf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hpf2* pHPF)
{
- AudioChannelLayout* pChannelLayout;
ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
- MA_ASSERT(pContext != NULL);
-
- result = ma_get_AudioObject_channel_layout(pContext, deviceObjectID, deviceType, &pChannelLayout);
+ result = ma_hpf2_get_heap_size(pConfig, &heapSizeInBytes);
if (result != MA_SUCCESS) {
- return result; /* Rather than always failing here, would it be more robust to simply assume a default? */
+ return result;
}
- result = ma_get_channel_map_from_AudioChannelLayout(pChannelLayout, pChannelMap, channelMapCap);
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+ } else {
+ pHeap = NULL;
+ }
+
+ result = ma_hpf2_init_preallocated(pConfig, pHeap, pHPF);
if (result != MA_SUCCESS) {
- ma_free(pChannelLayout, &pContext->allocationCallbacks);
+ ma_free(pHeap, pAllocationCallbacks);
return result;
}
- ma_free(pChannelLayout, &pContext->allocationCallbacks);
- return result;
+ pHPF->bq._ownsHeap = MA_TRUE; /* <-- This will cause the biquad to take ownership of the heap and free it when it's uninitialized. */
+ return MA_SUCCESS;
}
-#endif
-static ma_result ma_get_AudioObject_sample_rates(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, UInt32* pSampleRateRangesCount, AudioValueRange** ppSampleRateRanges) /* NOTE: Free the returned pointer with ma_free(). */
+MA_API void ma_hpf2_uninit(ma_hpf2* pHPF, const ma_allocation_callbacks* pAllocationCallbacks)
{
- AudioObjectPropertyAddress propAddress;
- UInt32 dataSize;
- OSStatus status;
- AudioValueRange* pSampleRateRanges;
-
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pSampleRateRangesCount != NULL);
- MA_ASSERT(ppSampleRateRanges != NULL);
-
- /* Safety. */
- *pSampleRateRangesCount = 0;
- *ppSampleRateRanges = NULL;
-
- propAddress.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
- propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput;
- propAddress.mElement = kAudioObjectPropertyElementMaster;
-
- status = ((ma_AudioObjectGetPropertyDataSize_proc)pContext->coreaudio.AudioObjectGetPropertyDataSize)(deviceObjectID, &propAddress, 0, NULL, &dataSize);
- if (status != noErr) {
- return ma_result_from_OSStatus(status);
- }
-
- pSampleRateRanges = (AudioValueRange*)ma_malloc(dataSize, &pContext->allocationCallbacks);
- if (pSampleRateRanges == NULL) {
- return MA_OUT_OF_MEMORY;
- }
-
- status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, pSampleRateRanges);
- if (status != noErr) {
- ma_free(pSampleRateRanges, &pContext->allocationCallbacks);
- return ma_result_from_OSStatus(status);
+ if (pHPF == NULL) {
+ return;
}
- *pSampleRateRangesCount = dataSize / sizeof(*pSampleRateRanges);
- *ppSampleRateRanges = pSampleRateRanges;
- return MA_SUCCESS;
+ ma_biquad_uninit(&pHPF->bq, pAllocationCallbacks); /* <-- This will free the heap allocation. */
}
-#if 0
-static ma_result ma_get_AudioObject_get_closest_sample_rate(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32 sampleRateIn, ma_uint32* pSampleRateOut)
+MA_API ma_result ma_hpf2_reinit(const ma_hpf2_config* pConfig, ma_hpf2* pHPF)
{
- UInt32 sampleRateRangeCount;
- AudioValueRange* pSampleRateRanges;
ma_result result;
+ ma_biquad_config bqConfig;
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pSampleRateOut != NULL);
-
- *pSampleRateOut = 0; /* Safety. */
+ if (pHPF == NULL || pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
- result = ma_get_AudioObject_sample_rates(pContext, deviceObjectID, deviceType, &sampleRateRangeCount, &pSampleRateRanges);
+ bqConfig = ma_hpf2__get_biquad_config(pConfig);
+ result = ma_biquad_reinit(&bqConfig, &pHPF->bq);
if (result != MA_SUCCESS) {
return result;
}
- if (sampleRateRangeCount == 0) {
- ma_free(pSampleRateRanges, &pContext->allocationCallbacks);
- return MA_ERROR; /* Should never hit this case should we? */
- }
-
- if (sampleRateIn == 0) {
- /* Search in order of miniaudio's preferred priority. */
- UInt32 iMALSampleRate;
- for (iMALSampleRate = 0; iMALSampleRate < ma_countof(g_maStandardSampleRatePriorities); ++iMALSampleRate) {
- ma_uint32 malSampleRate = g_maStandardSampleRatePriorities[iMALSampleRate];
- UInt32 iCASampleRate;
- for (iCASampleRate = 0; iCASampleRate < sampleRateRangeCount; ++iCASampleRate) {
- AudioValueRange caSampleRate = pSampleRateRanges[iCASampleRate];
- if (caSampleRate.mMinimum <= malSampleRate && caSampleRate.mMaximum >= malSampleRate) {
- *pSampleRateOut = malSampleRate;
- ma_free(pSampleRateRanges, &pContext->allocationCallbacks);
- return MA_SUCCESS;
- }
- }
- }
-
- /*
- If we get here it means none of miniaudio's standard sample rates matched any of the supported sample rates from the device. In this
- case we just fall back to the first one reported by Core Audio.
- */
- MA_ASSERT(sampleRateRangeCount > 0);
-
- *pSampleRateOut = pSampleRateRanges[0].mMinimum;
- ma_free(pSampleRateRanges, &pContext->allocationCallbacks);
- return MA_SUCCESS;
- } else {
- /* Find the closest match to this sample rate. */
- UInt32 currentAbsoluteDifference = INT32_MAX;
- UInt32 iCurrentClosestRange = (UInt32)-1;
- UInt32 iRange;
- for (iRange = 0; iRange < sampleRateRangeCount; ++iRange) {
- if (pSampleRateRanges[iRange].mMinimum <= sampleRateIn && pSampleRateRanges[iRange].mMaximum >= sampleRateIn) {
- *pSampleRateOut = sampleRateIn;
- ma_free(pSampleRateRanges, &pContext->allocationCallbacks);
- return MA_SUCCESS;
- } else {
- UInt32 absoluteDifference;
- if (pSampleRateRanges[iRange].mMinimum > sampleRateIn) {
- absoluteDifference = pSampleRateRanges[iRange].mMinimum - sampleRateIn;
- } else {
- absoluteDifference = sampleRateIn - pSampleRateRanges[iRange].mMaximum;
- }
+ return MA_SUCCESS;
+}
- if (currentAbsoluteDifference > absoluteDifference) {
- currentAbsoluteDifference = absoluteDifference;
- iCurrentClosestRange = iRange;
- }
- }
- }
+static MA_INLINE void ma_hpf2_process_pcm_frame_s16(ma_hpf2* pHPF, ma_int16* pFrameOut, const ma_int16* pFrameIn)
+{
+ ma_biquad_process_pcm_frame_s16(&pHPF->bq, pFrameOut, pFrameIn);
+}
- MA_ASSERT(iCurrentClosestRange != (UInt32)-1);
+static MA_INLINE void ma_hpf2_process_pcm_frame_f32(ma_hpf2* pHPF, float* pFrameOut, const float* pFrameIn)
+{
+ ma_biquad_process_pcm_frame_f32(&pHPF->bq, pFrameOut, pFrameIn);
+}
- *pSampleRateOut = pSampleRateRanges[iCurrentClosestRange].mMinimum;
- ma_free(pSampleRateRanges, &pContext->allocationCallbacks);
- return MA_SUCCESS;
+MA_API ma_result ma_hpf2_process_pcm_frames(ma_hpf2* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+{
+ if (pHPF == NULL) {
+ return MA_INVALID_ARGS;
}
- /* Should never get here, but it would mean we weren't able to find any suitable sample rates. */
- /*ma_free(pSampleRateRanges, &pContext->allocationCallbacks);*/
- /*return MA_ERROR;*/
+ return ma_biquad_process_pcm_frames(&pHPF->bq, pFramesOut, pFramesIn, frameCount);
}
-#endif
-static ma_result ma_get_AudioObject_closest_buffer_size_in_frames(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32 bufferSizeInFramesIn, ma_uint32* pBufferSizeInFramesOut)
+MA_API ma_uint32 ma_hpf2_get_latency(const ma_hpf2* pHPF)
{
- AudioObjectPropertyAddress propAddress;
- AudioValueRange bufferSizeRange;
- UInt32 dataSize;
- OSStatus status;
-
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pBufferSizeInFramesOut != NULL);
-
- *pBufferSizeInFramesOut = 0; /* Safety. */
-
- propAddress.mSelector = kAudioDevicePropertyBufferFrameSizeRange;
- propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput;
- propAddress.mElement = kAudioObjectPropertyElementMaster;
-
- dataSize = sizeof(bufferSizeRange);
- status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, &bufferSizeRange);
- if (status != noErr) {
- return ma_result_from_OSStatus(status);
- }
-
- /* This is just a clamp. */
- if (bufferSizeInFramesIn < bufferSizeRange.mMinimum) {
- *pBufferSizeInFramesOut = (ma_uint32)bufferSizeRange.mMinimum;
- } else if (bufferSizeInFramesIn > bufferSizeRange.mMaximum) {
- *pBufferSizeInFramesOut = (ma_uint32)bufferSizeRange.mMaximum;
- } else {
- *pBufferSizeInFramesOut = bufferSizeInFramesIn;
+ if (pHPF == NULL) {
+ return 0;
}
- return MA_SUCCESS;
+ return ma_biquad_get_latency(&pHPF->bq);
}
-static ma_result ma_set_AudioObject_buffer_size_in_frames(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32* pPeriodSizeInOut)
+
+MA_API ma_hpf_config ma_hpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order)
{
- ma_result result;
- ma_uint32 chosenBufferSizeInFrames;
- AudioObjectPropertyAddress propAddress;
- UInt32 dataSize;
- OSStatus status;
+ ma_hpf_config config;
- MA_ASSERT(pContext != NULL);
+ MA_ZERO_OBJECT(&config);
+ config.format = format;
+ config.channels = channels;
+ config.sampleRate = sampleRate;
+ config.cutoffFrequency = cutoffFrequency;
+ config.order = ma_min(order, MA_MAX_FILTER_ORDER);
- result = ma_get_AudioObject_closest_buffer_size_in_frames(pContext, deviceObjectID, deviceType, *pPeriodSizeInOut, &chosenBufferSizeInFrames);
- if (result != MA_SUCCESS) {
- return result;
- }
+ return config;
+}
- /* Try setting the size of the buffer... If this fails we just use whatever is currently set. */
- propAddress.mSelector = kAudioDevicePropertyBufferFrameSize;
- propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput;
- propAddress.mElement = kAudioObjectPropertyElementMaster;
- ((ma_AudioObjectSetPropertyData_proc)pContext->coreaudio.AudioObjectSetPropertyData)(deviceObjectID, &propAddress, 0, NULL, sizeof(chosenBufferSizeInFrames), &chosenBufferSizeInFrames);
+typedef struct
+{
+ size_t sizeInBytes;
+ size_t hpf1Offset;
+ size_t hpf2Offset; /* Offset of the first second order filter. Subsequent filters will come straight after, and will each have the same heap size. */
+} ma_hpf_heap_layout;
- /* Get the actual size of the buffer. */
- dataSize = sizeof(*pPeriodSizeInOut);
- status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, &chosenBufferSizeInFrames);
- if (status != noErr) {
- return ma_result_from_OSStatus(status);
- }
+static void ma_hpf_calculate_sub_hpf_counts(ma_uint32 order, ma_uint32* pHPF1Count, ma_uint32* pHPF2Count)
+{
+ MA_ASSERT(pHPF1Count != NULL);
+ MA_ASSERT(pHPF2Count != NULL);
- *pPeriodSizeInOut = chosenBufferSizeInFrames;
- return MA_SUCCESS;
+ *pHPF1Count = order % 2;
+ *pHPF2Count = order / 2;
}
-static ma_result ma_find_default_AudioObjectID(ma_context* pContext, ma_device_type deviceType, AudioObjectID* pDeviceObjectID)
+static ma_result ma_hpf_get_heap_layout(const ma_hpf_config* pConfig, ma_hpf_heap_layout* pHeapLayout)
{
- AudioObjectPropertyAddress propAddressDefaultDevice;
- UInt32 defaultDeviceObjectIDSize = sizeof(AudioObjectID);
- AudioObjectID defaultDeviceObjectID;
- OSStatus status;
+ ma_result result;
+ ma_uint32 hpf1Count;
+ ma_uint32 hpf2Count;
+ ma_uint32 ihpf1;
+ ma_uint32 ihpf2;
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pDeviceObjectID != NULL);
+ MA_ASSERT(pHeapLayout != NULL);
- /* Safety. */
- *pDeviceObjectID = 0;
+ MA_ZERO_OBJECT(pHeapLayout);
- propAddressDefaultDevice.mScope = kAudioObjectPropertyScopeGlobal;
- propAddressDefaultDevice.mElement = kAudioObjectPropertyElementMaster;
- if (deviceType == ma_device_type_playback) {
- propAddressDefaultDevice.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
- } else {
- propAddressDefaultDevice.mSelector = kAudioHardwarePropertyDefaultInputDevice;
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- defaultDeviceObjectIDSize = sizeof(AudioObjectID);
- status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(kAudioObjectSystemObject, &propAddressDefaultDevice, 0, NULL, &defaultDeviceObjectIDSize, &defaultDeviceObjectID);
- if (status == noErr) {
- *pDeviceObjectID = defaultDeviceObjectID;
- return MA_SUCCESS;
+ if (pConfig->channels == 0) {
+ return MA_INVALID_ARGS;
}
- /* If we get here it means we couldn't find the device. */
- return MA_NO_DEVICE;
-}
+ if (pConfig->order > MA_MAX_FILTER_ORDER) {
+ return MA_INVALID_ARGS;
+ }
-static ma_result ma_find_AudioObjectID(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, AudioObjectID* pDeviceObjectID)
-{
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pDeviceObjectID != NULL);
+ ma_hpf_calculate_sub_hpf_counts(pConfig->order, &hpf1Count, &hpf2Count);
- /* Safety. */
- *pDeviceObjectID = 0;
+ pHeapLayout->sizeInBytes = 0;
- if (pDeviceID == NULL) {
- /* Default device. */
- return ma_find_default_AudioObjectID(pContext, deviceType, pDeviceObjectID);
- } else {
- /* Explicit device. */
- UInt32 deviceCount;
- AudioObjectID* pDeviceObjectIDs;
- ma_result result;
- UInt32 iDevice;
+ /* HPF 1 */
+ pHeapLayout->hpf1Offset = pHeapLayout->sizeInBytes;
+ for (ihpf1 = 0; ihpf1 < hpf1Count; ihpf1 += 1) {
+ size_t hpf1HeapSizeInBytes;
+ ma_hpf1_config hpf1Config = ma_hpf1_config_init(pConfig->format, pConfig->channels, pConfig->sampleRate, pConfig->cutoffFrequency);
- result = ma_get_device_object_ids__coreaudio(pContext, &deviceCount, &pDeviceObjectIDs);
+ result = ma_hpf1_get_heap_size(&hpf1Config, &hpf1HeapSizeInBytes);
if (result != MA_SUCCESS) {
return result;
}
- for (iDevice = 0; iDevice < deviceCount; ++iDevice) {
- AudioObjectID deviceObjectID = pDeviceObjectIDs[iDevice];
+ pHeapLayout->sizeInBytes += sizeof(ma_hpf1) + hpf1HeapSizeInBytes;
+ }
- char uid[256];
- if (ma_get_AudioObject_uid(pContext, deviceObjectID, sizeof(uid), uid) != MA_SUCCESS) {
- continue;
- }
+ /* HPF 2*/
+ pHeapLayout->hpf2Offset = pHeapLayout->sizeInBytes;
+ for (ihpf2 = 0; ihpf2 < hpf2Count; ihpf2 += 1) {
+ size_t hpf2HeapSizeInBytes;
+ ma_hpf2_config hpf2Config = ma_hpf2_config_init(pConfig->format, pConfig->channels, pConfig->sampleRate, pConfig->cutoffFrequency, 0.707107); /* <-- The "q" parameter does not matter for the purpose of calculating the heap size. */
- if (deviceType == ma_device_type_playback) {
- if (ma_does_AudioObject_support_playback(pContext, deviceObjectID)) {
- if (strcmp(uid, pDeviceID->coreaudio) == 0) {
- *pDeviceObjectID = deviceObjectID;
- ma_free(pDeviceObjectIDs, &pContext->allocationCallbacks);
- return MA_SUCCESS;
- }
- }
- } else {
- if (ma_does_AudioObject_support_capture(pContext, deviceObjectID)) {
- if (strcmp(uid, pDeviceID->coreaudio) == 0) {
- *pDeviceObjectID = deviceObjectID;
- ma_free(pDeviceObjectIDs, &pContext->allocationCallbacks);
- return MA_SUCCESS;
- }
- }
- }
+ result = ma_hpf2_get_heap_size(&hpf2Config, &hpf2HeapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
}
- ma_free(pDeviceObjectIDs, &pContext->allocationCallbacks);
+ pHeapLayout->sizeInBytes += sizeof(ma_hpf2) + hpf2HeapSizeInBytes;
}
- /* If we get here it means we couldn't find the device. */
- return MA_NO_DEVICE;
-}
+ /* Make sure allocation size is aligned. */
+ pHeapLayout->sizeInBytes = ma_align_64(pHeapLayout->sizeInBytes);
+ return MA_SUCCESS;
+}
-static ma_result ma_find_best_format__coreaudio(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_format format, ma_uint32 channels, ma_uint32 sampleRate, const AudioStreamBasicDescription* pOrigFormat, AudioStreamBasicDescription* pFormat)
+static ma_result ma_hpf_reinit__internal(const ma_hpf_config* pConfig, void* pHeap, ma_hpf* pHPF, ma_bool32 isNew)
{
- UInt32 deviceFormatDescriptionCount;
- AudioStreamRangedDescription* pDeviceFormatDescriptions;
ma_result result;
- ma_uint32 desiredSampleRate;
- ma_uint32 desiredChannelCount;
- ma_format desiredFormat;
- AudioStreamBasicDescription bestDeviceFormatSoFar;
- ma_bool32 hasSupportedFormat;
- UInt32 iFormat;
+ ma_uint32 hpf1Count;
+ ma_uint32 hpf2Count;
+ ma_uint32 ihpf1;
+ ma_uint32 ihpf2;
+ ma_hpf_heap_layout heapLayout; /* Only used if isNew is true. */
- result = ma_get_AudioObject_stream_descriptions(pContext, deviceObjectID, deviceType, &deviceFormatDescriptionCount, &pDeviceFormatDescriptions);
- if (result != MA_SUCCESS) {
- return result;
+ if (pHPF == NULL || pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- desiredSampleRate = sampleRate;
- if (desiredSampleRate == 0) {
- desiredSampleRate = pOrigFormat->mSampleRate;
+ /* Only supporting f32 and s16. */
+ if (pConfig->format != ma_format_f32 && pConfig->format != ma_format_s16) {
+ return MA_INVALID_ARGS;
}
- desiredChannelCount = channels;
- if (desiredChannelCount == 0) {
- desiredChannelCount = pOrigFormat->mChannelsPerFrame;
+ /* The format cannot be changed after initialization. */
+ if (pHPF->format != ma_format_unknown && pHPF->format != pConfig->format) {
+ return MA_INVALID_OPERATION;
}
- desiredFormat = format;
- if (desiredFormat == ma_format_unknown) {
- result = ma_format_from_AudioStreamBasicDescription(pOrigFormat, &desiredFormat);
- if (result != MA_SUCCESS || desiredFormat == ma_format_unknown) {
- desiredFormat = g_maFormatPriorities[0];
- }
+ /* The channel count cannot be changed after initialization. */
+ if (pHPF->channels != 0 && pHPF->channels != pConfig->channels) {
+ return MA_INVALID_OPERATION;
}
- /*
- If we get here it means we don't have an exact match to what the client is asking for. We'll need to find the closest one. The next
- loop will check for formats that have the same sample rate to what we're asking for. If there is, we prefer that one in all cases.
- */
- MA_ZERO_OBJECT(&bestDeviceFormatSoFar);
+ if (pConfig->order > MA_MAX_FILTER_ORDER) {
+ return MA_INVALID_ARGS;
+ }
- hasSupportedFormat = MA_FALSE;
- for (iFormat = 0; iFormat < deviceFormatDescriptionCount; ++iFormat) {
- ma_format format;
- ma_result formatResult = ma_format_from_AudioStreamBasicDescription(&pDeviceFormatDescriptions[iFormat].mFormat, &format);
- if (formatResult == MA_SUCCESS && format != ma_format_unknown) {
- hasSupportedFormat = MA_TRUE;
- bestDeviceFormatSoFar = pDeviceFormatDescriptions[iFormat].mFormat;
- break;
+ ma_hpf_calculate_sub_hpf_counts(pConfig->order, &hpf1Count, &hpf2Count);
+
+ /* The filter order can't change between reinits. */
+ if (!isNew) {
+ if (pHPF->hpf1Count != hpf1Count || pHPF->hpf2Count != hpf2Count) {
+ return MA_INVALID_OPERATION;
}
}
- if (!hasSupportedFormat) {
- ma_free(pDeviceFormatDescriptions, &pContext->allocationCallbacks);
- return MA_FORMAT_NOT_SUPPORTED;
+ if (isNew) {
+ result = ma_hpf_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ pHPF->_pHeap = pHeap;
+ MA_ZERO_MEMORY(pHeap, heapLayout.sizeInBytes);
+
+ pHPF->pHPF1 = (ma_hpf1*)ma_offset_ptr(pHeap, heapLayout.hpf1Offset);
+ pHPF->pHPF2 = (ma_hpf2*)ma_offset_ptr(pHeap, heapLayout.hpf2Offset);
+ } else {
+ MA_ZERO_OBJECT(&heapLayout); /* To silence a compiler warning. */
}
+ for (ihpf1 = 0; ihpf1 < hpf1Count; ihpf1 += 1) {
+ ma_hpf1_config hpf1Config = ma_hpf1_config_init(pConfig->format, pConfig->channels, pConfig->sampleRate, pConfig->cutoffFrequency);
- for (iFormat = 0; iFormat < deviceFormatDescriptionCount; ++iFormat) {
- AudioStreamBasicDescription thisDeviceFormat = pDeviceFormatDescriptions[iFormat].mFormat;
- ma_format thisSampleFormat;
- ma_result formatResult;
- ma_format bestSampleFormatSoFar;
+ if (isNew) {
+ size_t hpf1HeapSizeInBytes;
- /* If the format is not supported by miniaudio we need to skip this one entirely. */
- formatResult = ma_format_from_AudioStreamBasicDescription(&pDeviceFormatDescriptions[iFormat].mFormat, &thisSampleFormat);
- if (formatResult != MA_SUCCESS || thisSampleFormat == ma_format_unknown) {
- continue; /* The format is not supported by miniaudio. Skip. */
+ result = ma_hpf1_get_heap_size(&hpf1Config, &hpf1HeapSizeInBytes);
+ if (result == MA_SUCCESS) {
+ result = ma_hpf1_init_preallocated(&hpf1Config, ma_offset_ptr(pHeap, heapLayout.hpf1Offset + (sizeof(ma_hpf1) * hpf1Count) + (ihpf1 * hpf1HeapSizeInBytes)), &pHPF->pHPF1[ihpf1]);
+ }
+ } else {
+ result = ma_hpf1_reinit(&hpf1Config, &pHPF->pHPF1[ihpf1]);
}
- ma_format_from_AudioStreamBasicDescription(&bestDeviceFormatSoFar, &bestSampleFormatSoFar);
+ if (result != MA_SUCCESS) {
+ ma_uint32 jhpf1;
- /* Getting here means the format is supported by miniaudio which makes this format a candidate. */
- if (thisDeviceFormat.mSampleRate != desiredSampleRate) {
- /*
- The sample rate does not match, but this format could still be usable, although it's a very low priority. If the best format
- so far has an equal sample rate we can just ignore this one.
- */
- if (bestDeviceFormatSoFar.mSampleRate == desiredSampleRate) {
- continue; /* The best sample rate so far has the same sample rate as what we requested which means it's still the best so far. Skip this format. */
- } else {
- /* In this case, neither the best format so far nor this one have the same sample rate. Check the channel count next. */
- if (thisDeviceFormat.mChannelsPerFrame != desiredChannelCount) {
- /* This format has a different sample rate _and_ a different channel count. */
- if (bestDeviceFormatSoFar.mChannelsPerFrame == desiredChannelCount) {
- continue; /* No change to the best format. */
- } else {
- /*
- Both this format and the best so far have different sample rates and different channel counts. Whichever has the
- best format is the new best.
- */
- if (ma_get_format_priority_index(thisSampleFormat) < ma_get_format_priority_index(bestSampleFormatSoFar)) {
- bestDeviceFormatSoFar = thisDeviceFormat;
- continue;
- } else {
- continue; /* No change to the best format. */
- }
- }
- } else {
- /* This format has a different sample rate but the desired channel count. */
- if (bestDeviceFormatSoFar.mChannelsPerFrame == desiredChannelCount) {
- /* Both this format and the best so far have the desired channel count. Whichever has the best format is the new best. */
- if (ma_get_format_priority_index(thisSampleFormat) < ma_get_format_priority_index(bestSampleFormatSoFar)) {
- bestDeviceFormatSoFar = thisDeviceFormat;
- continue;
- } else {
- continue; /* No change to the best format for now. */
- }
- } else {
- /* This format has the desired channel count, but the best so far does not. We have a new best. */
- bestDeviceFormatSoFar = thisDeviceFormat;
- continue;
- }
- }
+ for (jhpf1 = 0; jhpf1 < ihpf1; jhpf1 += 1) {
+ ma_hpf1_uninit(&pHPF->pHPF1[jhpf1], NULL); /* No need for allocation callbacks here since we used a preallocated heap allocation. */
}
+
+ return result;
+ }
+ }
+
+ for (ihpf2 = 0; ihpf2 < hpf2Count; ihpf2 += 1) {
+ ma_hpf2_config hpf2Config;
+ double q;
+ double a;
+
+ /* Tempting to use 0.707107, but won't result in a Butterworth filter if the order is > 2. */
+ if (hpf1Count == 1) {
+ a = (1 + ihpf2*1) * (MA_PI_D/(pConfig->order*1)); /* Odd order. */
} else {
- /*
- The sample rates match which makes this format a very high priority contender. If the best format so far has a different
- sample rate it needs to be replaced with this one.
- */
- if (bestDeviceFormatSoFar.mSampleRate != desiredSampleRate) {
- bestDeviceFormatSoFar = thisDeviceFormat;
- continue;
- } else {
- /* In this case both this format and the best format so far have the same sample rate. Check the channel count next. */
- if (thisDeviceFormat.mChannelsPerFrame == desiredChannelCount) {
- /*
- In this case this format has the same channel count as what the client is requesting. If the best format so far has
- a different count, this one becomes the new best.
- */
- if (bestDeviceFormatSoFar.mChannelsPerFrame != desiredChannelCount) {
- bestDeviceFormatSoFar = thisDeviceFormat;
- continue;
- } else {
- /* In this case both this format and the best so far have the ideal sample rate and channel count. Check the format. */
- if (thisSampleFormat == desiredFormat) {
- bestDeviceFormatSoFar = thisDeviceFormat;
- break; /* Found the exact match. */
- } else {
- /* The formats are different. The new best format is the one with the highest priority format according to miniaudio. */
- if (ma_get_format_priority_index(thisSampleFormat) < ma_get_format_priority_index(bestSampleFormatSoFar)) {
- bestDeviceFormatSoFar = thisDeviceFormat;
- continue;
- } else {
- continue; /* No change to the best format for now. */
- }
- }
- }
- } else {
- /*
- In this case the channel count is different to what the client has requested. If the best so far has the same channel
- count as the requested count then it remains the best.
- */
- if (bestDeviceFormatSoFar.mChannelsPerFrame == desiredChannelCount) {
- continue;
- } else {
- /*
- This is the case where both have the same sample rate (good) but different channel counts. Right now both have about
- the same priority, but we need to compare the format now.
- */
- if (thisSampleFormat == bestSampleFormatSoFar) {
- if (ma_get_format_priority_index(thisSampleFormat) < ma_get_format_priority_index(bestSampleFormatSoFar)) {
- bestDeviceFormatSoFar = thisDeviceFormat;
- continue;
- } else {
- continue; /* No change to the best format for now. */
- }
- }
- }
- }
+ a = (1 + ihpf2*2) * (MA_PI_D/(pConfig->order*2)); /* Even order. */
+ }
+ q = 1 / (2*ma_cosd(a));
+
+ hpf2Config = ma_hpf2_config_init(pConfig->format, pConfig->channels, pConfig->sampleRate, pConfig->cutoffFrequency, q);
+
+ if (isNew) {
+ size_t hpf2HeapSizeInBytes;
+
+ result = ma_hpf2_get_heap_size(&hpf2Config, &hpf2HeapSizeInBytes);
+ if (result == MA_SUCCESS) {
+ result = ma_hpf2_init_preallocated(&hpf2Config, ma_offset_ptr(pHeap, heapLayout.hpf2Offset + (sizeof(ma_hpf2) * hpf2Count) + (ihpf2 * hpf2HeapSizeInBytes)), &pHPF->pHPF2[ihpf2]);
+ }
+ } else {
+ result = ma_hpf2_reinit(&hpf2Config, &pHPF->pHPF2[ihpf2]);
+ }
+
+ if (result != MA_SUCCESS) {
+ ma_uint32 jhpf1;
+ ma_uint32 jhpf2;
+
+ for (jhpf1 = 0; jhpf1 < hpf1Count; jhpf1 += 1) {
+ ma_hpf1_uninit(&pHPF->pHPF1[jhpf1], NULL); /* No need for allocation callbacks here since we used a preallocated heap allocation. */
+ }
+
+ for (jhpf2 = 0; jhpf2 < ihpf2; jhpf2 += 1) {
+ ma_hpf2_uninit(&pHPF->pHPF2[jhpf2], NULL); /* No need for allocation callbacks here since we used a preallocated heap allocation. */
}
+
+ return result;
}
}
- *pFormat = bestDeviceFormatSoFar;
+ pHPF->hpf1Count = hpf1Count;
+ pHPF->hpf2Count = hpf2Count;
+ pHPF->format = pConfig->format;
+ pHPF->channels = pConfig->channels;
+ pHPF->sampleRate = pConfig->sampleRate;
- ma_free(pDeviceFormatDescriptions, &pContext->allocationCallbacks);
return MA_SUCCESS;
}
-static ma_result ma_get_AudioUnit_channel_map(ma_context* pContext, AudioUnit audioUnit, ma_device_type deviceType, ma_channel* pChannelMap, size_t channelMapCap)
+MA_API ma_result ma_hpf_get_heap_size(const ma_hpf_config* pConfig, size_t* pHeapSizeInBytes)
{
- AudioUnitScope deviceScope;
- AudioUnitElement deviceBus;
- UInt32 channelLayoutSize;
- OSStatus status;
- AudioChannelLayout* pChannelLayout;
ma_result result;
+ ma_hpf_heap_layout heapLayout;
- MA_ASSERT(pContext != NULL);
-
- if (deviceType == ma_device_type_playback) {
- deviceScope = kAudioUnitScope_Input;
- deviceBus = MA_COREAUDIO_OUTPUT_BUS;
- } else {
- deviceScope = kAudioUnitScope_Output;
- deviceBus = MA_COREAUDIO_INPUT_BUS;
- }
-
- status = ((ma_AudioUnitGetPropertyInfo_proc)pContext->coreaudio.AudioUnitGetPropertyInfo)(audioUnit, kAudioUnitProperty_AudioChannelLayout, deviceScope, deviceBus, &channelLayoutSize, NULL);
- if (status != noErr) {
- return ma_result_from_OSStatus(status);
- }
-
- pChannelLayout = (AudioChannelLayout*)ma__malloc_from_callbacks(channelLayoutSize, &pContext->allocationCallbacks);
- if (pChannelLayout == NULL) {
- return MA_OUT_OF_MEMORY;
+ if (pHeapSizeInBytes == NULL) {
+ return MA_INVALID_ARGS;
}
- status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(audioUnit, kAudioUnitProperty_AudioChannelLayout, deviceScope, deviceBus, pChannelLayout, &channelLayoutSize);
- if (status != noErr) {
- ma__free_from_callbacks(pChannelLayout, &pContext->allocationCallbacks);
- return ma_result_from_OSStatus(status);
- }
+ *pHeapSizeInBytes = 0;
- result = ma_get_channel_map_from_AudioChannelLayout(pChannelLayout, pChannelMap, channelMapCap);
+ result = ma_hpf_get_heap_layout(pConfig, &heapLayout);
if (result != MA_SUCCESS) {
- ma__free_from_callbacks(pChannelLayout, &pContext->allocationCallbacks);
return result;
}
- ma__free_from_callbacks(pChannelLayout, &pContext->allocationCallbacks);
- return MA_SUCCESS;
-}
-#endif /* MA_APPLE_DESKTOP */
+ *pHeapSizeInBytes = heapLayout.sizeInBytes;
+ return result;
+}
-#if !defined(MA_APPLE_DESKTOP)
-static void ma_AVAudioSessionPortDescription_to_device_info(AVAudioSessionPortDescription* pPortDesc, ma_device_info* pInfo)
+MA_API ma_result ma_hpf_init_preallocated(const ma_hpf_config* pConfig, void* pHeap, ma_hpf* pLPF)
{
- MA_ZERO_OBJECT(pInfo);
- ma_strncpy_s(pInfo->name, sizeof(pInfo->name), [pPortDesc.portName UTF8String], (size_t)-1);
- ma_strncpy_s(pInfo->id.coreaudio, sizeof(pInfo->id.coreaudio), [pPortDesc.UID UTF8String], (size_t)-1);
+ if (pLPF == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ MA_ZERO_OBJECT(pLPF);
+
+ return ma_hpf_reinit__internal(pConfig, pHeap, pLPF, /*isNew*/MA_TRUE);
}
-#endif
-static ma_result ma_context_enumerate_devices__coreaudio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+MA_API ma_result ma_hpf_init(const ma_hpf_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hpf* pHPF)
{
-#if defined(MA_APPLE_DESKTOP)
- UInt32 deviceCount;
- AudioObjectID* pDeviceObjectIDs;
- AudioObjectID defaultDeviceObjectIDPlayback;
- AudioObjectID defaultDeviceObjectIDCapture;
ma_result result;
- UInt32 iDevice;
-
- ma_find_default_AudioObjectID(pContext, ma_device_type_playback, &defaultDeviceObjectIDPlayback); /* OK if this fails. */
- ma_find_default_AudioObjectID(pContext, ma_device_type_capture, &defaultDeviceObjectIDCapture); /* OK if this fails. */
+ size_t heapSizeInBytes;
+ void* pHeap;
- result = ma_get_device_object_ids__coreaudio(pContext, &deviceCount, &pDeviceObjectIDs);
+ result = ma_hpf_get_heap_size(pConfig, &heapSizeInBytes);
if (result != MA_SUCCESS) {
return result;
}
- for (iDevice = 0; iDevice < deviceCount; ++iDevice) {
- AudioObjectID deviceObjectID = pDeviceObjectIDs[iDevice];
- ma_device_info info;
-
- MA_ZERO_OBJECT(&info);
- if (ma_get_AudioObject_uid(pContext, deviceObjectID, sizeof(info.id.coreaudio), info.id.coreaudio) != MA_SUCCESS) {
- continue;
- }
- if (ma_get_AudioObject_name(pContext, deviceObjectID, sizeof(info.name), info.name) != MA_SUCCESS) {
- continue;
- }
-
- if (ma_does_AudioObject_support_playback(pContext, deviceObjectID)) {
- if (deviceObjectID == defaultDeviceObjectIDPlayback) {
- info.isDefault = MA_TRUE;
- }
-
- if (!callback(pContext, ma_device_type_playback, &info, pUserData)) {
- break;
- }
- }
- if (ma_does_AudioObject_support_capture(pContext, deviceObjectID)) {
- if (deviceObjectID == defaultDeviceObjectIDCapture) {
- info.isDefault = MA_TRUE;
- }
-
- if (!callback(pContext, ma_device_type_capture, &info, pUserData)) {
- break;
- }
- }
- }
-
- ma_free(pDeviceObjectIDs, &pContext->allocationCallbacks);
-#else
- ma_device_info info;
- NSArray *pInputs = [[[AVAudioSession sharedInstance] currentRoute] inputs];
- NSArray *pOutputs = [[[AVAudioSession sharedInstance] currentRoute] outputs];
-
- for (AVAudioSessionPortDescription* pPortDesc in pOutputs) {
- ma_AVAudioSessionPortDescription_to_device_info(pPortDesc, &info);
- if (!callback(pContext, ma_device_type_playback, &info, pUserData)) {
- return MA_SUCCESS;
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
}
+ } else {
+ pHeap = NULL;
}
- for (AVAudioSessionPortDescription* pPortDesc in pInputs) {
- ma_AVAudioSessionPortDescription_to_device_info(pPortDesc, &info);
- if (!callback(pContext, ma_device_type_capture, &info, pUserData)) {
- return MA_SUCCESS;
- }
+ result = ma_hpf_init_preallocated(pConfig, pHeap, pHPF);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
}
-#endif
+ pHPF->_ownsHeap = MA_TRUE;
return MA_SUCCESS;
}
-static ma_result ma_context_get_device_info__coreaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+MA_API void ma_hpf_uninit(ma_hpf* pHPF, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_result result;
-
- MA_ASSERT(pContext != NULL);
-
-#if defined(MA_APPLE_DESKTOP)
- /* Desktop */
- {
- AudioObjectID deviceObjectID;
- AudioObjectID defaultDeviceObjectID;
- UInt32 streamDescriptionCount;
- AudioStreamRangedDescription* pStreamDescriptions;
- UInt32 iStreamDescription;
- UInt32 sampleRateRangeCount;
- AudioValueRange* pSampleRateRanges;
-
- ma_find_default_AudioObjectID(pContext, deviceType, &defaultDeviceObjectID); /* OK if this fails. */
+ ma_uint32 ihpf1;
+ ma_uint32 ihpf2;
- result = ma_find_AudioObjectID(pContext, deviceType, pDeviceID, &deviceObjectID);
- if (result != MA_SUCCESS) {
- return result;
- }
+ if (pHPF == NULL) {
+ return;
+ }
- result = ma_get_AudioObject_uid(pContext, deviceObjectID, sizeof(pDeviceInfo->id.coreaudio), pDeviceInfo->id.coreaudio);
- if (result != MA_SUCCESS) {
- return result;
- }
+ for (ihpf1 = 0; ihpf1 < pHPF->hpf1Count; ihpf1 += 1) {
+ ma_hpf1_uninit(&pHPF->pHPF1[ihpf1], pAllocationCallbacks);
+ }
- result = ma_get_AudioObject_name(pContext, deviceObjectID, sizeof(pDeviceInfo->name), pDeviceInfo->name);
- if (result != MA_SUCCESS) {
- return result;
- }
+ for (ihpf2 = 0; ihpf2 < pHPF->hpf2Count; ihpf2 += 1) {
+ ma_hpf2_uninit(&pHPF->pHPF2[ihpf2], pAllocationCallbacks);
+ }
- if (deviceObjectID == defaultDeviceObjectID) {
- pDeviceInfo->isDefault = MA_TRUE;
- }
+ if (pHPF->_ownsHeap) {
+ ma_free(pHPF->_pHeap, pAllocationCallbacks);
+ }
+}
- /*
- There could be a large number of permutations here. Fortunately there is only a single channel count
- being reported which reduces this quite a bit. For sample rates we're only reporting those that are
- one of miniaudio's recognized "standard" rates. If there are still more formats than can fit into
- our fixed sized array we'll just need to truncate them. This is unlikely and will probably only happen
- if some driver performs software data conversion and therefore reports every possible format and
- sample rate.
- */
- pDeviceInfo->nativeDataFormatCount = 0;
+MA_API ma_result ma_hpf_reinit(const ma_hpf_config* pConfig, ma_hpf* pHPF)
+{
+ return ma_hpf_reinit__internal(pConfig, NULL, pHPF, /*isNew*/MA_FALSE);
+}
- /* Formats. */
- {
- ma_format uniqueFormats[ma_format_count];
- ma_uint32 uniqueFormatCount = 0;
- ma_uint32 channels;
+MA_API ma_result ma_hpf_process_pcm_frames(ma_hpf* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+{
+ ma_result result;
+ ma_uint32 ihpf1;
+ ma_uint32 ihpf2;
- /* Channels. */
- result = ma_get_AudioObject_channel_count(pContext, deviceObjectID, deviceType, &channels);
- if (result != MA_SUCCESS) {
- return result;
- }
+ if (pHPF == NULL) {
+ return MA_INVALID_ARGS;
+ }
- /* Formats. */
- result = ma_get_AudioObject_stream_descriptions(pContext, deviceObjectID, deviceType, &streamDescriptionCount, &pStreamDescriptions);
+ /* Faster path for in-place. */
+ if (pFramesOut == pFramesIn) {
+ for (ihpf1 = 0; ihpf1 < pHPF->hpf1Count; ihpf1 += 1) {
+ result = ma_hpf1_process_pcm_frames(&pHPF->pHPF1[ihpf1], pFramesOut, pFramesOut, frameCount);
if (result != MA_SUCCESS) {
return result;
}
+ }
- for (iStreamDescription = 0; iStreamDescription < streamDescriptionCount; ++iStreamDescription) {
- ma_format format;
- ma_bool32 hasFormatBeenHandled = MA_FALSE;
- ma_uint32 iOutputFormat;
- ma_uint32 iSampleRate;
-
- result = ma_format_from_AudioStreamBasicDescription(&pStreamDescriptions[iStreamDescription].mFormat, &format);
- if (result != MA_SUCCESS) {
- continue;
- }
-
- MA_ASSERT(format != ma_format_unknown);
-
- /* Make sure the format isn't already in the output list. */
- for (iOutputFormat = 0; iOutputFormat < uniqueFormatCount; ++iOutputFormat) {
- if (uniqueFormats[iOutputFormat] == format) {
- hasFormatBeenHandled = MA_TRUE;
- break;
- }
- }
-
- /* If we've already handled this format just skip it. */
- if (hasFormatBeenHandled) {
- continue;
- }
+ for (ihpf2 = 0; ihpf2 < pHPF->hpf2Count; ihpf2 += 1) {
+ result = ma_hpf2_process_pcm_frames(&pHPF->pHPF2[ihpf2], pFramesOut, pFramesOut, frameCount);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ }
+ }
- uniqueFormats[uniqueFormatCount] = format;
- uniqueFormatCount += 1;
+ /* Slightly slower path for copying. */
+ if (pFramesOut != pFramesIn) {
+ ma_uint32 iFrame;
- /* Sample Rates */
- result = ma_get_AudioObject_sample_rates(pContext, deviceObjectID, deviceType, &sampleRateRangeCount, &pSampleRateRanges);
- if (result != MA_SUCCESS) {
- return result;
- }
+ /* */ if (pHPF->format == ma_format_f32) {
+ /* */ float* pFramesOutF32 = ( float*)pFramesOut;
+ const float* pFramesInF32 = (const float*)pFramesIn;
- /*
- Annoyingly Core Audio reports a sample rate range. We just get all the standard rates that are
- between this range.
- */
- for (iSampleRate = 0; iSampleRate < sampleRateRangeCount; ++iSampleRate) {
- ma_uint32 iStandardSampleRate;
- for (iStandardSampleRate = 0; iStandardSampleRate < ma_countof(g_maStandardSampleRatePriorities); iStandardSampleRate += 1) {
- ma_uint32 standardSampleRate = g_maStandardSampleRatePriorities[iStandardSampleRate];
- if (standardSampleRate >= pSampleRateRanges[iSampleRate].mMinimum && standardSampleRate <= pSampleRateRanges[iSampleRate].mMaximum) {
- /* We have a new data format. Add it to the list. */
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = format;
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = channels;
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = standardSampleRate;
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = 0;
- pDeviceInfo->nativeDataFormatCount += 1;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ MA_COPY_MEMORY(pFramesOutF32, pFramesInF32, ma_get_bytes_per_frame(pHPF->format, pHPF->channels));
- if (pDeviceInfo->nativeDataFormatCount >= ma_countof(pDeviceInfo->nativeDataFormats)) {
- break; /* No more room for any more formats. */
- }
- }
- }
+ for (ihpf1 = 0; ihpf1 < pHPF->hpf1Count; ihpf1 += 1) {
+ ma_hpf1_process_pcm_frame_f32(&pHPF->pHPF1[ihpf1], pFramesOutF32, pFramesOutF32);
}
- ma_free(pSampleRateRanges, &pContext->allocationCallbacks);
-
- if (pDeviceInfo->nativeDataFormatCount >= ma_countof(pDeviceInfo->nativeDataFormats)) {
- break; /* No more room for any more formats. */
+ for (ihpf2 = 0; ihpf2 < pHPF->hpf2Count; ihpf2 += 1) {
+ ma_hpf2_process_pcm_frame_f32(&pHPF->pHPF2[ihpf2], pFramesOutF32, pFramesOutF32);
}
+
+ pFramesOutF32 += pHPF->channels;
+ pFramesInF32 += pHPF->channels;
}
+ } else if (pHPF->format == ma_format_s16) {
+ /* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut;
+ const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn;
- ma_free(pStreamDescriptions, &pContext->allocationCallbacks);
- }
- }
-#else
- /* Mobile */
- {
- AudioComponentDescription desc;
- AudioComponent component;
- AudioUnit audioUnit;
- OSStatus status;
- AudioUnitScope formatScope;
- AudioUnitElement formatElement;
- AudioStreamBasicDescription bestFormat;
- UInt32 propSize;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ MA_COPY_MEMORY(pFramesOutS16, pFramesInS16, ma_get_bytes_per_frame(pHPF->format, pHPF->channels));
- /* We want to ensure we use a consistent device name to device enumeration. */
- if (pDeviceID != NULL) {
- ma_bool32 found = MA_FALSE;
- if (deviceType == ma_device_type_playback) {
- NSArray *pOutputs = [[[AVAudioSession sharedInstance] currentRoute] outputs];
- for (AVAudioSessionPortDescription* pPortDesc in pOutputs) {
- if (strcmp(pDeviceID->coreaudio, [pPortDesc.UID UTF8String]) == 0) {
- ma_AVAudioSessionPortDescription_to_device_info(pPortDesc, pDeviceInfo);
- found = MA_TRUE;
- break;
- }
+ for (ihpf1 = 0; ihpf1 < pHPF->hpf1Count; ihpf1 += 1) {
+ ma_hpf1_process_pcm_frame_s16(&pHPF->pHPF1[ihpf1], pFramesOutS16, pFramesOutS16);
}
- } else {
- NSArray *pInputs = [[[AVAudioSession sharedInstance] currentRoute] inputs];
- for (AVAudioSessionPortDescription* pPortDesc in pInputs) {
- if (strcmp(pDeviceID->coreaudio, [pPortDesc.UID UTF8String]) == 0) {
- ma_AVAudioSessionPortDescription_to_device_info(pPortDesc, pDeviceInfo);
- found = MA_TRUE;
- break;
- }
+
+ for (ihpf2 = 0; ihpf2 < pHPF->hpf2Count; ihpf2 += 1) {
+ ma_hpf2_process_pcm_frame_s16(&pHPF->pHPF2[ihpf2], pFramesOutS16, pFramesOutS16);
}
- }
- if (!found) {
- return MA_DOES_NOT_EXIST;
+ pFramesOutS16 += pHPF->channels;
+ pFramesInS16 += pHPF->channels;
}
} else {
- if (deviceType == ma_device_type_playback) {
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
- } else {
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
- }
+ MA_ASSERT(MA_FALSE);
+ return MA_INVALID_OPERATION; /* Should never hit this. */
}
+ }
+ return MA_SUCCESS;
+}
- /*
- Retrieving device information is more annoying on mobile than desktop. For simplicity I'm locking this down to whatever format is
- reported on a temporary I/O unit. The problem, however, is that this doesn't return a value for the sample rate which we need to
- retrieve from the AVAudioSession shared instance.
- */
- desc.componentType = kAudioUnitType_Output;
- desc.componentSubType = kAudioUnitSubType_RemoteIO;
- desc.componentManufacturer = kAudioUnitManufacturer_Apple;
- desc.componentFlags = 0;
- desc.componentFlagsMask = 0;
+MA_API ma_uint32 ma_hpf_get_latency(const ma_hpf* pHPF)
+{
+ if (pHPF == NULL) {
+ return 0;
+ }
- component = ((ma_AudioComponentFindNext_proc)pContext->coreaudio.AudioComponentFindNext)(NULL, &desc);
- if (component == NULL) {
- return MA_FAILED_TO_INIT_BACKEND;
- }
+ return pHPF->hpf2Count*2 + pHPF->hpf1Count;
+}
- status = ((ma_AudioComponentInstanceNew_proc)pContext->coreaudio.AudioComponentInstanceNew)(component, &audioUnit);
- if (status != noErr) {
- return ma_result_from_OSStatus(status);
- }
- formatScope = (deviceType == ma_device_type_playback) ? kAudioUnitScope_Input : kAudioUnitScope_Output;
- formatElement = (deviceType == ma_device_type_playback) ? MA_COREAUDIO_OUTPUT_BUS : MA_COREAUDIO_INPUT_BUS;
+/**************************************************************************************************************************************************************
- propSize = sizeof(bestFormat);
- status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, &propSize);
- if (status != noErr) {
- ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(audioUnit);
- return ma_result_from_OSStatus(status);
- }
+Band-Pass Filtering
- ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(audioUnit);
- audioUnit = NULL;
+**************************************************************************************************************************************************************/
+MA_API ma_bpf2_config ma_bpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q)
+{
+ ma_bpf2_config config;
- /* Only a single format is being reported for iOS. */
- pDeviceInfo->nativeDataFormatCount = 1;
+ MA_ZERO_OBJECT(&config);
+ config.format = format;
+ config.channels = channels;
+ config.sampleRate = sampleRate;
+ config.cutoffFrequency = cutoffFrequency;
+ config.q = q;
- result = ma_format_from_AudioStreamBasicDescription(&bestFormat, &pDeviceInfo->nativeDataFormats[0].format);
- if (result != MA_SUCCESS) {
- return result;
- }
+ /* Q cannot be 0 or else it'll result in a division by 0. In this case just default to 0.707107. */
+ if (config.q == 0) {
+ config.q = 0.707107;
+ }
- pDeviceInfo->nativeDataFormats[0].channels = bestFormat.mChannelsPerFrame;
+ return config;
+}
- /*
- It looks like Apple are wanting to push the whole AVAudioSession thing. Thus, we need to use that to determine device settings. To do
- this we just get the shared instance and inspect.
- */
- @autoreleasepool {
- AVAudioSession* pAudioSession = [AVAudioSession sharedInstance];
- MA_ASSERT(pAudioSession != NULL);
- pDeviceInfo->nativeDataFormats[0].sampleRate = (ma_uint32)pAudioSession.sampleRate;
- }
- }
-#endif
+static MA_INLINE ma_biquad_config ma_bpf2__get_biquad_config(const ma_bpf2_config* pConfig)
+{
+ ma_biquad_config bqConfig;
+ double q;
+ double w;
+ double s;
+ double c;
+ double a;
- (void)pDeviceInfo; /* Unused. */
- return MA_SUCCESS;
+ MA_ASSERT(pConfig != NULL);
+
+ q = pConfig->q;
+ w = 2 * MA_PI_D * pConfig->cutoffFrequency / pConfig->sampleRate;
+ s = ma_sind(w);
+ c = ma_cosd(w);
+ a = s / (2*q);
+
+ bqConfig.b0 = q * a;
+ bqConfig.b1 = 0;
+ bqConfig.b2 = -q * a;
+ bqConfig.a0 = 1 + a;
+ bqConfig.a1 = -2 * c;
+ bqConfig.a2 = 1 - a;
+
+ bqConfig.format = pConfig->format;
+ bqConfig.channels = pConfig->channels;
+
+ return bqConfig;
}
-static AudioBufferList* ma_allocate_AudioBufferList__coreaudio(ma_uint32 sizeInFrames, ma_format format, ma_uint32 channels, ma_stream_layout layout, const ma_allocation_callbacks* pAllocationCallbacks)
+MA_API ma_result ma_bpf2_get_heap_size(const ma_bpf2_config* pConfig, size_t* pHeapSizeInBytes)
{
- AudioBufferList* pBufferList;
- UInt32 audioBufferSizeInBytes;
- size_t allocationSize;
+ ma_biquad_config bqConfig;
+ bqConfig = ma_bpf2__get_biquad_config(pConfig);
- MA_ASSERT(sizeInFrames > 0);
- MA_ASSERT(format != ma_format_unknown);
- MA_ASSERT(channels > 0);
+ return ma_biquad_get_heap_size(&bqConfig, pHeapSizeInBytes);
+}
- allocationSize = sizeof(AudioBufferList) - sizeof(AudioBuffer); /* Subtract sizeof(AudioBuffer) because that part is dynamically sized. */
- if (layout == ma_stream_layout_interleaved) {
- /* Interleaved case. This is the simple case because we just have one buffer. */
- allocationSize += sizeof(AudioBuffer) * 1;
- } else {
- /* Non-interleaved case. This is the more complex case because there's more than one buffer. */
- allocationSize += sizeof(AudioBuffer) * channels;
+MA_API ma_result ma_bpf2_init_preallocated(const ma_bpf2_config* pConfig, void* pHeap, ma_bpf2* pBPF)
+{
+ ma_result result;
+ ma_biquad_config bqConfig;
+
+ if (pBPF == NULL) {
+ return MA_INVALID_ARGS;
}
- allocationSize += sizeInFrames * ma_get_bytes_per_frame(format, channels);
+ MA_ZERO_OBJECT(pBPF);
- pBufferList = (AudioBufferList*)ma__malloc_from_callbacks(allocationSize, pAllocationCallbacks);
- if (pBufferList == NULL) {
- return NULL;
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- audioBufferSizeInBytes = (UInt32)(sizeInFrames * ma_get_bytes_per_sample(format));
-
- if (layout == ma_stream_layout_interleaved) {
- pBufferList->mNumberBuffers = 1;
- pBufferList->mBuffers[0].mNumberChannels = channels;
- pBufferList->mBuffers[0].mDataByteSize = audioBufferSizeInBytes * channels;
- pBufferList->mBuffers[0].mData = (ma_uint8*)pBufferList + sizeof(AudioBufferList);
- } else {
- ma_uint32 iBuffer;
- pBufferList->mNumberBuffers = channels;
- for (iBuffer = 0; iBuffer < pBufferList->mNumberBuffers; ++iBuffer) {
- pBufferList->mBuffers[iBuffer].mNumberChannels = 1;
- pBufferList->mBuffers[iBuffer].mDataByteSize = audioBufferSizeInBytes;
- pBufferList->mBuffers[iBuffer].mData = (ma_uint8*)pBufferList + ((sizeof(AudioBufferList) - sizeof(AudioBuffer)) + (sizeof(AudioBuffer) * channels)) + (audioBufferSizeInBytes * iBuffer);
- }
+ bqConfig = ma_bpf2__get_biquad_config(pConfig);
+ result = ma_biquad_init_preallocated(&bqConfig, pHeap, &pBPF->bq);
+ if (result != MA_SUCCESS) {
+ return result;
}
- return pBufferList;
+ return MA_SUCCESS;
}
-static ma_result ma_device_realloc_AudioBufferList__coreaudio(ma_device* pDevice, ma_uint32 sizeInFrames, ma_format format, ma_uint32 channels, ma_stream_layout layout)
+MA_API ma_result ma_bpf2_init(const ma_bpf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_bpf2* pBPF)
{
- MA_ASSERT(pDevice != NULL);
- MA_ASSERT(format != ma_format_unknown);
- MA_ASSERT(channels > 0);
+ ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
- /* Only resize the buffer if necessary. */
- if (pDevice->coreaudio.audioBufferCapInFrames < sizeInFrames) {
- AudioBufferList* pNewAudioBufferList;
+ result = ma_bpf2_get_heap_size(pConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- pNewAudioBufferList = ma_allocate_AudioBufferList__coreaudio(sizeInFrames, format, channels, layout, &pDevice->pContext->allocationCallbacks);
- if (pNewAudioBufferList != NULL) {
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
return MA_OUT_OF_MEMORY;
}
+ } else {
+ pHeap = NULL;
+ }
- /* At this point we'll have a new AudioBufferList and we can free the old one. */
- ma__free_from_callbacks(pDevice->coreaudio.pAudioBufferList, &pDevice->pContext->allocationCallbacks);
- pDevice->coreaudio.pAudioBufferList = pNewAudioBufferList;
- pDevice->coreaudio.audioBufferCapInFrames = sizeInFrames;
+ result = ma_bpf2_init_preallocated(pConfig, pHeap, pBPF);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
}
- /* Getting here means the capacity of the audio is fine. */
+ pBPF->bq._ownsHeap = MA_TRUE; /* <-- This will cause the biquad to take ownership of the heap and free it when it's uninitialized. */
return MA_SUCCESS;
}
-
-static OSStatus ma_on_output__coreaudio(void* pUserData, AudioUnitRenderActionFlags* pActionFlags, const AudioTimeStamp* pTimeStamp, UInt32 busNumber, UInt32 frameCount, AudioBufferList* pBufferList)
+MA_API void ma_bpf2_uninit(ma_bpf2* pBPF, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_device* pDevice = (ma_device*)pUserData;
- ma_stream_layout layout;
+ if (pBPF == NULL) {
+ return;
+ }
- MA_ASSERT(pDevice != NULL);
+ ma_biquad_uninit(&pBPF->bq, pAllocationCallbacks); /* <-- This will free the heap allocation. */
+}
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "INFO: Output Callback: busNumber=%d, frameCount=%d, mNumberBuffers=%d\n", busNumber, frameCount, pBufferList->mNumberBuffers);
+MA_API ma_result ma_bpf2_reinit(const ma_bpf2_config* pConfig, ma_bpf2* pBPF)
+{
+ ma_result result;
+ ma_biquad_config bqConfig;
- /* We need to check whether or not we are outputting interleaved or non-interleaved samples. The way we do this is slightly different for each type. */
- layout = ma_stream_layout_interleaved;
- if (pBufferList->mBuffers[0].mNumberChannels != pDevice->playback.internalChannels) {
- layout = ma_stream_layout_deinterleaved;
+ if (pBPF == NULL || pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- if (layout == ma_stream_layout_interleaved) {
- /* For now we can assume everything is interleaved. */
- UInt32 iBuffer;
- for (iBuffer = 0; iBuffer < pBufferList->mNumberBuffers; ++iBuffer) {
- if (pBufferList->mBuffers[iBuffer].mNumberChannels == pDevice->playback.internalChannels) {
- ma_uint32 frameCountForThisBuffer = pBufferList->mBuffers[iBuffer].mDataByteSize / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
- if (frameCountForThisBuffer > 0) {
- ma_device_handle_backend_data_callback(pDevice, pBufferList->mBuffers[iBuffer].mData, NULL, frameCountForThisBuffer);
- }
+ bqConfig = ma_bpf2__get_biquad_config(pConfig);
+ result = ma_biquad_reinit(&bqConfig, &pBPF->bq);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", frameCount, pBufferList->mBuffers[iBuffer].mNumberChannels, pBufferList->mBuffers[iBuffer].mDataByteSize);
- } else {
- /*
- This case is where the number of channels in the output buffer do not match our internal channels. It could mean that it's
- not interleaved, in which case we can't handle right now since miniaudio does not yet support non-interleaved streams. We just
- output silence here.
- */
- MA_ZERO_MEMORY(pBufferList->mBuffers[iBuffer].mData, pBufferList->mBuffers[iBuffer].mDataByteSize);
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " WARNING: Outputting silence. frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", frameCount, pBufferList->mBuffers[iBuffer].mNumberChannels, pBufferList->mBuffers[iBuffer].mDataByteSize);
- }
- }
- } else {
- /* This is the deinterleaved case. We need to update each buffer in groups of internalChannels. This assumes each buffer is the same size. */
- MA_ASSERT(pDevice->playback.internalChannels <= MA_MAX_CHANNELS); /* This should heve been validated at initialization time. */
+ return MA_SUCCESS;
+}
- /*
- For safety we'll check that the internal channels is a multiple of the buffer count. If it's not it means something
- very strange has happened and we're not going to support it.
- */
- if ((pBufferList->mNumberBuffers % pDevice->playback.internalChannels) == 0) {
- ma_uint8 tempBuffer[4096];
- UInt32 iBuffer;
+static MA_INLINE void ma_bpf2_process_pcm_frame_s16(ma_bpf2* pBPF, ma_int16* pFrameOut, const ma_int16* pFrameIn)
+{
+ ma_biquad_process_pcm_frame_s16(&pBPF->bq, pFrameOut, pFrameIn);
+}
- for (iBuffer = 0; iBuffer < pBufferList->mNumberBuffers; iBuffer += pDevice->playback.internalChannels) {
- ma_uint32 frameCountPerBuffer = pBufferList->mBuffers[iBuffer].mDataByteSize / ma_get_bytes_per_sample(pDevice->playback.internalFormat);
- ma_uint32 framesRemaining = frameCountPerBuffer;
+static MA_INLINE void ma_bpf2_process_pcm_frame_f32(ma_bpf2* pBPF, float* pFrameOut, const float* pFrameIn)
+{
+ ma_biquad_process_pcm_frame_f32(&pBPF->bq, pFrameOut, pFrameIn);
+}
- while (framesRemaining > 0) {
- void* ppDeinterleavedBuffers[MA_MAX_CHANNELS];
- ma_uint32 iChannel;
- ma_uint32 framesToRead = sizeof(tempBuffer) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
- if (framesToRead > framesRemaining) {
- framesToRead = framesRemaining;
- }
+MA_API ma_result ma_bpf2_process_pcm_frames(ma_bpf2* pBPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+{
+ if (pBPF == NULL) {
+ return MA_INVALID_ARGS;
+ }
- ma_device_handle_backend_data_callback(pDevice, tempBuffer, NULL, framesToRead);
+ return ma_biquad_process_pcm_frames(&pBPF->bq, pFramesOut, pFramesIn, frameCount);
+}
- for (iChannel = 0; iChannel < pDevice->playback.internalChannels; ++iChannel) {
- ppDeinterleavedBuffers[iChannel] = (void*)ma_offset_ptr(pBufferList->mBuffers[iBuffer+iChannel].mData, (frameCountPerBuffer - framesRemaining) * ma_get_bytes_per_sample(pDevice->playback.internalFormat));
- }
+MA_API ma_uint32 ma_bpf2_get_latency(const ma_bpf2* pBPF)
+{
+ if (pBPF == NULL) {
+ return 0;
+ }
- ma_deinterleave_pcm_frames(pDevice->playback.internalFormat, pDevice->playback.internalChannels, framesToRead, tempBuffer, ppDeinterleavedBuffers);
+ return ma_biquad_get_latency(&pBPF->bq);
+}
- framesRemaining -= framesToRead;
- }
- }
- }
- }
- (void)pActionFlags;
- (void)pTimeStamp;
- (void)busNumber;
- (void)frameCount;
+MA_API ma_bpf_config ma_bpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order)
+{
+ ma_bpf_config config;
- return noErr;
+ MA_ZERO_OBJECT(&config);
+ config.format = format;
+ config.channels = channels;
+ config.sampleRate = sampleRate;
+ config.cutoffFrequency = cutoffFrequency;
+ config.order = ma_min(order, MA_MAX_FILTER_ORDER);
+
+ return config;
}
-static OSStatus ma_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFlags* pActionFlags, const AudioTimeStamp* pTimeStamp, UInt32 busNumber, UInt32 frameCount, AudioBufferList* pUnusedBufferList)
+
+typedef struct
+{
+ size_t sizeInBytes;
+ size_t bpf2Offset;
+} ma_bpf_heap_layout;
+
+static ma_result ma_bpf_get_heap_layout(const ma_bpf_config* pConfig, ma_bpf_heap_layout* pHeapLayout)
{
- ma_device* pDevice = (ma_device*)pUserData;
- AudioBufferList* pRenderedBufferList;
ma_result result;
- ma_stream_layout layout;
- ma_uint32 iBuffer;
- OSStatus status;
+ ma_uint32 bpf2Count;
+ ma_uint32 ibpf2;
- MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(pHeapLayout != NULL);
- pRenderedBufferList = (AudioBufferList*)pDevice->coreaudio.pAudioBufferList;
- MA_ASSERT(pRenderedBufferList);
+ MA_ZERO_OBJECT(pHeapLayout);
- /* We need to check whether or not we are outputting interleaved or non-interleaved samples. The way we do this is slightly different for each type. */
- layout = ma_stream_layout_interleaved;
- if (pRenderedBufferList->mBuffers[0].mNumberChannels != pDevice->capture.internalChannels) {
- layout = ma_stream_layout_deinterleaved;
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "INFO: Input Callback: busNumber=%d, frameCount=%d, mNumberBuffers=%d\n", busNumber, frameCount, pRenderedBufferList->mNumberBuffers);
-
- /*
- There has been a situation reported where frame count passed into this function is greater than the capacity of
- our capture buffer. There doesn't seem to be a reliable way to determine what the maximum frame count will be,
- so we need to instead resort to dynamically reallocating our buffer to ensure it's large enough to capture the
- number of frames requested by this callback.
- */
- result = ma_device_realloc_AudioBufferList__coreaudio(pDevice, frameCount, pDevice->capture.internalFormat, pDevice->capture.internalChannels, layout);
- if (result != MA_SUCCESS) {
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "Failed to allocate AudioBufferList for capture.");
- return noErr;
+ if (pConfig->order > MA_MAX_FILTER_ORDER) {
+ return MA_INVALID_ARGS;
}
- /*
- When you call AudioUnitRender(), Core Audio tries to be helpful by setting the mDataByteSize to the number of bytes
- that were actually rendered. The problem with this is that the next call can fail with -50 due to the size no longer
- being set to the capacity of the buffer, but instead the size in bytes of the previous render. This will cause a
- problem when a future call to this callback specifies a larger number of frames.
-
- To work around this we need to explicitly set the size of each buffer to their respective size in bytes.
- */
- for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; ++iBuffer) {
- pRenderedBufferList->mBuffers[iBuffer].mDataByteSize = pDevice->coreaudio.audioBufferCapInFrames * ma_get_bytes_per_sample(pDevice->capture.internalFormat) * pRenderedBufferList->mBuffers[iBuffer].mNumberChannels;
+ /* We must have an even number of order. */
+ if ((pConfig->order & 0x1) != 0) {
+ return MA_INVALID_ARGS;
}
- status = ((ma_AudioUnitRender_proc)pDevice->pContext->coreaudio.AudioUnitRender)((AudioUnit)pDevice->coreaudio.audioUnitCapture, pActionFlags, pTimeStamp, busNumber, frameCount, pRenderedBufferList);
- if (status != noErr) {
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " ERROR: AudioUnitRender() failed with %d\n", status);
- return status;
- }
+ bpf2Count = pConfig->channels / 2;
- if (layout == ma_stream_layout_interleaved) {
- for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; ++iBuffer) {
- if (pRenderedBufferList->mBuffers[iBuffer].mNumberChannels == pDevice->capture.internalChannels) {
- ma_device_handle_backend_data_callback(pDevice, NULL, pRenderedBufferList->mBuffers[iBuffer].mData, frameCount);
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " mDataByteSize=%d\n", pRenderedBufferList->mBuffers[iBuffer].mDataByteSize);
- } else {
- /*
- This case is where the number of channels in the output buffer do not match our internal channels. It could mean that it's
- not interleaved, in which case we can't handle right now since miniaudio does not yet support non-interleaved streams.
- */
- ma_uint8 silentBuffer[4096];
- ma_uint32 framesRemaining;
+ pHeapLayout->sizeInBytes = 0;
- MA_ZERO_MEMORY(silentBuffer, sizeof(silentBuffer));
+ /* BPF 2 */
+ pHeapLayout->bpf2Offset = pHeapLayout->sizeInBytes;
+ for (ibpf2 = 0; ibpf2 < bpf2Count; ibpf2 += 1) {
+ size_t bpf2HeapSizeInBytes;
+ ma_bpf2_config bpf2Config = ma_bpf2_config_init(pConfig->format, pConfig->channels, pConfig->sampleRate, pConfig->cutoffFrequency, 0.707107); /* <-- The "q" parameter does not matter for the purpose of calculating the heap size. */
- framesRemaining = frameCount;
- while (framesRemaining > 0) {
- ma_uint32 framesToSend = sizeof(silentBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
- if (framesToSend > framesRemaining) {
- framesToSend = framesRemaining;
- }
+ result = ma_bpf2_get_heap_size(&bpf2Config, &bpf2HeapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- ma_device_handle_backend_data_callback(pDevice, NULL, silentBuffer, framesToSend);
+ pHeapLayout->sizeInBytes += sizeof(ma_bpf2) + bpf2HeapSizeInBytes;
+ }
- framesRemaining -= framesToSend;
- }
+ /* Make sure allocation size is aligned. */
+ pHeapLayout->sizeInBytes = ma_align_64(pHeapLayout->sizeInBytes);
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " WARNING: Outputting silence. frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", frameCount, pRenderedBufferList->mBuffers[iBuffer].mNumberChannels, pRenderedBufferList->mBuffers[iBuffer].mDataByteSize);
- }
- }
- } else {
- /* This is the deinterleaved case. We need to interleave the audio data before sending it to the client. This assumes each buffer is the same size. */
- MA_ASSERT(pDevice->capture.internalChannels <= MA_MAX_CHANNELS); /* This should have been validated at initialization time. */
+ return MA_SUCCESS;
+}
- /*
- For safety we'll check that the internal channels is a multiple of the buffer count. If it's not it means something
- very strange has happened and we're not going to support it.
- */
- if ((pRenderedBufferList->mNumberBuffers % pDevice->capture.internalChannels) == 0) {
- ma_uint8 tempBuffer[4096];
- for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; iBuffer += pDevice->capture.internalChannels) {
- ma_uint32 framesRemaining = frameCount;
- while (framesRemaining > 0) {
- void* ppDeinterleavedBuffers[MA_MAX_CHANNELS];
- ma_uint32 iChannel;
- ma_uint32 framesToSend = sizeof(tempBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
- if (framesToSend > framesRemaining) {
- framesToSend = framesRemaining;
- }
+static ma_result ma_bpf_reinit__internal(const ma_bpf_config* pConfig, void* pHeap, ma_bpf* pBPF, ma_bool32 isNew)
+{
+ ma_result result;
+ ma_uint32 bpf2Count;
+ ma_uint32 ibpf2;
+ ma_bpf_heap_layout heapLayout; /* Only used if isNew is true. */
- for (iChannel = 0; iChannel < pDevice->capture.internalChannels; ++iChannel) {
- ppDeinterleavedBuffers[iChannel] = (void*)ma_offset_ptr(pRenderedBufferList->mBuffers[iBuffer+iChannel].mData, (frameCount - framesRemaining) * ma_get_bytes_per_sample(pDevice->capture.internalFormat));
- }
+ if (pBPF == NULL || pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
- ma_interleave_pcm_frames(pDevice->capture.internalFormat, pDevice->capture.internalChannels, framesToSend, (const void**)ppDeinterleavedBuffers, tempBuffer);
- ma_device_handle_backend_data_callback(pDevice, NULL, tempBuffer, framesToSend);
+ /* Only supporting f32 and s16. */
+ if (pConfig->format != ma_format_f32 && pConfig->format != ma_format_s16) {
+ return MA_INVALID_ARGS;
+ }
- framesRemaining -= framesToSend;
- }
- }
- }
+ /* The format cannot be changed after initialization. */
+ if (pBPF->format != ma_format_unknown && pBPF->format != pConfig->format) {
+ return MA_INVALID_OPERATION;
+ }
+
+ /* The channel count cannot be changed after initialization. */
+ if (pBPF->channels != 0 && pBPF->channels != pConfig->channels) {
+ return MA_INVALID_OPERATION;
}
- (void)pActionFlags;
- (void)pTimeStamp;
- (void)busNumber;
- (void)frameCount;
- (void)pUnusedBufferList;
+ if (pConfig->order > MA_MAX_FILTER_ORDER) {
+ return MA_INVALID_ARGS;
+ }
- return noErr;
-}
+ /* We must have an even number of order. */
+ if ((pConfig->order & 0x1) != 0) {
+ return MA_INVALID_ARGS;
+ }
-static void on_start_stop__coreaudio(void* pUserData, AudioUnit audioUnit, AudioUnitPropertyID propertyID, AudioUnitScope scope, AudioUnitElement element)
-{
- ma_device* pDevice = (ma_device*)pUserData;
- MA_ASSERT(pDevice != NULL);
+ bpf2Count = pConfig->order / 2;
- /* Don't do anything if it looks like we're just reinitializing due to a device switch. */
- if (((audioUnit == pDevice->coreaudio.audioUnitPlayback) && pDevice->coreaudio.isSwitchingPlaybackDevice) ||
- ((audioUnit == pDevice->coreaudio.audioUnitCapture) && pDevice->coreaudio.isSwitchingCaptureDevice)) {
- return;
+ /* The filter order can't change between reinits. */
+ if (!isNew) {
+ if (pBPF->bpf2Count != bpf2Count) {
+ return MA_INVALID_OPERATION;
+ }
}
- /*
- There's been a report of a deadlock here when triggered by ma_device_uninit(). It looks like
- AudioUnitGetProprty (called below) and AudioComponentInstanceDispose (called in ma_device_uninit)
- can try waiting on the same lock. I'm going to try working around this by not calling any Core
- Audio APIs in the callback when the device has been stopped or uninitialized.
- */
- if (ma_device_get_state(pDevice) == MA_STATE_UNINITIALIZED || ma_device_get_state(pDevice) == MA_STATE_STOPPING || ma_device_get_state(pDevice) == MA_STATE_STOPPED) {
- ma_stop_proc onStop = pDevice->onStop;
- if (onStop) {
- onStop(pDevice);
+ if (isNew) {
+ result = ma_bpf_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
}
- ma_event_signal(&pDevice->coreaudio.stopEvent);
- } else {
- UInt32 isRunning;
- UInt32 isRunningSize = sizeof(isRunning);
- OSStatus status = ((ma_AudioUnitGetProperty_proc)pDevice->pContext->coreaudio.AudioUnitGetProperty)(audioUnit, kAudioOutputUnitProperty_IsRunning, scope, element, &isRunning, &isRunningSize);
- if (status != noErr) {
- return; /* Don't really know what to do in this case... just ignore it, I suppose... */
- }
+ pBPF->_pHeap = pHeap;
+ MA_ZERO_MEMORY(pHeap, heapLayout.sizeInBytes);
- if (!isRunning) {
- ma_stop_proc onStop;
+ pBPF->pBPF2 = (ma_bpf2*)ma_offset_ptr(pHeap, heapLayout.bpf2Offset);
+ } else {
+ MA_ZERO_OBJECT(&heapLayout);
+ }
- /*
- The stop event is a bit annoying in Core Audio because it will be called when we automatically switch the default device. Some scenarios to consider:
+ for (ibpf2 = 0; ibpf2 < bpf2Count; ibpf2 += 1) {
+ ma_bpf2_config bpf2Config;
+ double q;
- 1) When the device is unplugged, this will be called _before_ the default device change notification.
- 2) When the device is changed via the default device change notification, this will be called _after_ the switch.
+ /* TODO: Calculate Q to make this a proper Butterworth filter. */
+ q = 0.707107;
- For case #1, we just check if there's a new default device available. If so, we just ignore the stop event. For case #2 we check a flag.
- */
- if (((audioUnit == pDevice->coreaudio.audioUnitPlayback) && pDevice->coreaudio.isDefaultPlaybackDevice) ||
- ((audioUnit == pDevice->coreaudio.audioUnitCapture) && pDevice->coreaudio.isDefaultCaptureDevice)) {
- /*
- It looks like the device is switching through an external event, such as the user unplugging the device or changing the default device
- via the operating system's sound settings. If we're re-initializing the device, we just terminate because we want the stopping of the
- device to be seamless to the client (we don't want them receiving the onStop event and thinking that the device has stopped when it
- hasn't!).
- */
- if (((audioUnit == pDevice->coreaudio.audioUnitPlayback) && pDevice->coreaudio.isSwitchingPlaybackDevice) ||
- ((audioUnit == pDevice->coreaudio.audioUnitCapture) && pDevice->coreaudio.isSwitchingCaptureDevice)) {
- return;
- }
+ bpf2Config = ma_bpf2_config_init(pConfig->format, pConfig->channels, pConfig->sampleRate, pConfig->cutoffFrequency, q);
- /*
- Getting here means the device is not reinitializing which means it may have been unplugged. From what I can see, it looks like Core Audio
- will try switching to the new default device seamlessly. We need to somehow find a way to determine whether or not Core Audio will most
- likely be successful in switching to the new device.
+ if (isNew) {
+ size_t bpf2HeapSizeInBytes;
- TODO: Try to predict if Core Audio will switch devices. If not, the onStop callback needs to be posted.
- */
- return;
+ result = ma_bpf2_get_heap_size(&bpf2Config, &bpf2HeapSizeInBytes);
+ if (result == MA_SUCCESS) {
+ result = ma_bpf2_init_preallocated(&bpf2Config, ma_offset_ptr(pHeap, heapLayout.bpf2Offset + (sizeof(ma_bpf2) * bpf2Count) + (ibpf2 * bpf2HeapSizeInBytes)), &pBPF->pBPF2[ibpf2]);
}
+ } else {
+ result = ma_bpf2_reinit(&bpf2Config, &pBPF->pBPF2[ibpf2]);
+ }
- /* Getting here means we need to stop the device. */
- onStop = pDevice->onStop;
- if (onStop) {
- onStop(pDevice);
- }
+ if (result != MA_SUCCESS) {
+ return result;
}
}
- (void)propertyID; /* Unused. */
+ pBPF->bpf2Count = bpf2Count;
+ pBPF->format = pConfig->format;
+ pBPF->channels = pConfig->channels;
+
+ return MA_SUCCESS;
}
-#if defined(MA_APPLE_DESKTOP)
-static ma_spinlock g_DeviceTrackingInitLock_CoreAudio = 0; /* A spinlock for mutal exclusion of the init/uninit of the global tracking data. Initialization to 0 is what we need. */
-static ma_uint32 g_DeviceTrackingInitCounter_CoreAudio = 0;
-static ma_mutex g_DeviceTrackingMutex_CoreAudio;
-static ma_device** g_ppTrackedDevices_CoreAudio = NULL;
-static ma_uint32 g_TrackedDeviceCap_CoreAudio = 0;
-static ma_uint32 g_TrackedDeviceCount_CoreAudio = 0;
-static OSStatus ma_default_device_changed__coreaudio(AudioObjectID objectID, UInt32 addressCount, const AudioObjectPropertyAddress* pAddresses, void* pUserData)
+MA_API ma_result ma_bpf_get_heap_size(const ma_bpf_config* pConfig, size_t* pHeapSizeInBytes)
{
- ma_device_type deviceType;
-
- /* Not sure if I really need to check this, but it makes me feel better. */
- if (addressCount == 0) {
- return noErr;
- }
+ ma_result result;
+ ma_bpf_heap_layout heapLayout;
- if (pAddresses[0].mSelector == kAudioHardwarePropertyDefaultOutputDevice) {
- deviceType = ma_device_type_playback;
- } else if (pAddresses[0].mSelector == kAudioHardwarePropertyDefaultInputDevice) {
- deviceType = ma_device_type_capture;
- } else {
- return noErr; /* Should never hit this. */
+ if (pHeapSizeInBytes == NULL) {
+ return MA_INVALID_ARGS;
}
- ma_mutex_lock(&g_DeviceTrackingMutex_CoreAudio);
- {
- ma_uint32 iDevice;
- for (iDevice = 0; iDevice < g_TrackedDeviceCount_CoreAudio; iDevice += 1) {
- ma_result reinitResult;
- ma_device* pDevice;
-
- pDevice = g_ppTrackedDevices_CoreAudio[iDevice];
- if (pDevice->type == deviceType || pDevice->type == ma_device_type_duplex) {
- if (deviceType == ma_device_type_playback) {
- pDevice->coreaudio.isSwitchingPlaybackDevice = MA_TRUE;
- reinitResult = ma_device_reinit_internal__coreaudio(pDevice, deviceType, MA_TRUE);
- pDevice->coreaudio.isSwitchingPlaybackDevice = MA_FALSE;
- } else {
- pDevice->coreaudio.isSwitchingCaptureDevice = MA_TRUE;
- reinitResult = ma_device_reinit_internal__coreaudio(pDevice, deviceType, MA_TRUE);
- pDevice->coreaudio.isSwitchingCaptureDevice = MA_FALSE;
- }
-
- if (reinitResult == MA_SUCCESS) {
- ma_device__post_init_setup(pDevice, deviceType);
+ *pHeapSizeInBytes = 0;
- /* Restart the device if required. If this fails we need to stop the device entirely. */
- if (ma_device_get_state(pDevice) == MA_STATE_STARTED) {
- OSStatus status;
- if (deviceType == ma_device_type_playback) {
- status = ((ma_AudioOutputUnitStart_proc)pDevice->pContext->coreaudio.AudioOutputUnitStart)((AudioUnit)pDevice->coreaudio.audioUnitPlayback);
- if (status != noErr) {
- if (pDevice->type == ma_device_type_duplex) {
- ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitCapture);
- }
- ma_device__set_state(pDevice, MA_STATE_STOPPED);
- }
- } else if (deviceType == ma_device_type_capture) {
- status = ((ma_AudioOutputUnitStart_proc)pDevice->pContext->coreaudio.AudioOutputUnitStart)((AudioUnit)pDevice->coreaudio.audioUnitCapture);
- if (status != noErr) {
- if (pDevice->type == ma_device_type_duplex) {
- ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitPlayback);
- }
- ma_device__set_state(pDevice, MA_STATE_STOPPED);
- }
- }
- }
- }
- }
- }
+ result = ma_bpf_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
}
- ma_mutex_unlock(&g_DeviceTrackingMutex_CoreAudio);
- /* Unused parameters. */
- (void)objectID;
- (void)pUserData;
+ *pHeapSizeInBytes = heapLayout.sizeInBytes;
- return noErr;
+ return MA_SUCCESS;
}
-static ma_result ma_context__init_device_tracking__coreaudio(ma_context* pContext)
+MA_API ma_result ma_bpf_init_preallocated(const ma_bpf_config* pConfig, void* pHeap, ma_bpf* pBPF)
{
- MA_ASSERT(pContext != NULL);
+ if (pBPF == NULL) {
+ return MA_INVALID_ARGS;
+ }
- ma_spinlock_lock(&g_DeviceTrackingInitLock_CoreAudio);
- {
- /* Don't do anything if we've already initializd device tracking. */
- if (g_DeviceTrackingInitCounter_CoreAudio == 0) {
- AudioObjectPropertyAddress propAddress;
- propAddress.mScope = kAudioObjectPropertyScopeGlobal;
- propAddress.mElement = kAudioObjectPropertyElementMaster;
+ MA_ZERO_OBJECT(pBPF);
- ma_mutex_init(&g_DeviceTrackingMutex_CoreAudio);
+ return ma_bpf_reinit__internal(pConfig, pHeap, pBPF, /*isNew*/MA_TRUE);
+}
- propAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice;
- ((ma_AudioObjectAddPropertyListener_proc)pContext->coreaudio.AudioObjectAddPropertyListener)(kAudioObjectSystemObject, &propAddress, &ma_default_device_changed__coreaudio, NULL);
+MA_API ma_result ma_bpf_init(const ma_bpf_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_bpf* pBPF)
+{
+ ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
- propAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
- ((ma_AudioObjectAddPropertyListener_proc)pContext->coreaudio.AudioObjectAddPropertyListener)(kAudioObjectSystemObject, &propAddress, &ma_default_device_changed__coreaudio, NULL);
+ result = ma_bpf_get_heap_size(pConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
}
- g_DeviceTrackingInitCounter_CoreAudio += 1;
+ } else {
+ pHeap = NULL;
+ }
+
+ result = ma_bpf_init_preallocated(pConfig, pHeap, pBPF);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
}
- ma_spinlock_unlock(&g_DeviceTrackingInitLock_CoreAudio);
+ pBPF->_ownsHeap = MA_TRUE;
return MA_SUCCESS;
}
-static ma_result ma_context__uninit_device_tracking__coreaudio(ma_context* pContext)
+MA_API void ma_bpf_uninit(ma_bpf* pBPF, const ma_allocation_callbacks* pAllocationCallbacks)
{
- MA_ASSERT(pContext != NULL);
-
- ma_spinlock_lock(&g_DeviceTrackingInitLock_CoreAudio);
- {
- if (g_DeviceTrackingInitCounter_CoreAudio > 0)
- g_DeviceTrackingInitCounter_CoreAudio -= 1;
-
- if (g_DeviceTrackingInitCounter_CoreAudio == 0) {
- AudioObjectPropertyAddress propAddress;
- propAddress.mScope = kAudioObjectPropertyScopeGlobal;
- propAddress.mElement = kAudioObjectPropertyElementMaster;
-
- propAddress.mSelector = kAudioHardwarePropertyDefaultInputDevice;
- ((ma_AudioObjectRemovePropertyListener_proc)pContext->coreaudio.AudioObjectRemovePropertyListener)(kAudioObjectSystemObject, &propAddress, &ma_default_device_changed__coreaudio, NULL);
-
- propAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
- ((ma_AudioObjectRemovePropertyListener_proc)pContext->coreaudio.AudioObjectRemovePropertyListener)(kAudioObjectSystemObject, &propAddress, &ma_default_device_changed__coreaudio, NULL);
+ ma_uint32 ibpf2;
- /* At this point there should be no tracked devices. If not there's an error somewhere. */
- if (g_ppTrackedDevices_CoreAudio != NULL) {
- ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_WARNING, "You have uninitialized all contexts while an associated device is still active.", MA_INVALID_OPERATION);
- ma_spinlock_unlock(&g_DeviceTrackingInitLock_CoreAudio);
- return MA_INVALID_OPERATION;
- }
+ if (pBPF == NULL) {
+ return;
+ }
- ma_mutex_uninit(&g_DeviceTrackingMutex_CoreAudio);
- }
+ for (ibpf2 = 0; ibpf2 < pBPF->bpf2Count; ibpf2 += 1) {
+ ma_bpf2_uninit(&pBPF->pBPF2[ibpf2], pAllocationCallbacks);
}
- ma_spinlock_unlock(&g_DeviceTrackingInitLock_CoreAudio);
- return MA_SUCCESS;
+ if (pBPF->_ownsHeap) {
+ ma_free(pBPF->_pHeap, pAllocationCallbacks);
+ }
}
-static ma_result ma_device__track__coreaudio(ma_device* pDevice)
+MA_API ma_result ma_bpf_reinit(const ma_bpf_config* pConfig, ma_bpf* pBPF)
{
- MA_ASSERT(pDevice != NULL);
+ return ma_bpf_reinit__internal(pConfig, NULL, pBPF, /*isNew*/MA_FALSE);
+}
- ma_mutex_lock(&g_DeviceTrackingMutex_CoreAudio);
- {
- /* Allocate memory if required. */
- if (g_TrackedDeviceCap_CoreAudio <= g_TrackedDeviceCount_CoreAudio) {
- ma_uint32 oldCap;
- ma_uint32 newCap;
- ma_device** ppNewDevices;
+MA_API ma_result ma_bpf_process_pcm_frames(ma_bpf* pBPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+{
+ ma_result result;
+ ma_uint32 ibpf2;
- oldCap = g_TrackedDeviceCap_CoreAudio;
- newCap = g_TrackedDeviceCap_CoreAudio * 2;
- if (newCap == 0) {
- newCap = 1;
- }
+ if (pBPF == NULL) {
+ return MA_INVALID_ARGS;
+ }
- ppNewDevices = (ma_device**)ma__realloc_from_callbacks(g_ppTrackedDevices_CoreAudio, sizeof(*g_ppTrackedDevices_CoreAudio)*newCap, sizeof(*g_ppTrackedDevices_CoreAudio)*oldCap, &pDevice->pContext->allocationCallbacks);
- if (ppNewDevices == NULL) {
- ma_mutex_unlock(&g_DeviceTrackingMutex_CoreAudio);
- return MA_OUT_OF_MEMORY;
+ /* Faster path for in-place. */
+ if (pFramesOut == pFramesIn) {
+ for (ibpf2 = 0; ibpf2 < pBPF->bpf2Count; ibpf2 += 1) {
+ result = ma_bpf2_process_pcm_frames(&pBPF->pBPF2[ibpf2], pFramesOut, pFramesOut, frameCount);
+ if (result != MA_SUCCESS) {
+ return result;
}
-
- g_ppTrackedDevices_CoreAudio = ppNewDevices;
- g_TrackedDeviceCap_CoreAudio = newCap;
}
-
- g_ppTrackedDevices_CoreAudio[g_TrackedDeviceCount_CoreAudio] = pDevice;
- g_TrackedDeviceCount_CoreAudio += 1;
}
- ma_mutex_unlock(&g_DeviceTrackingMutex_CoreAudio);
- return MA_SUCCESS;
-}
+ /* Slightly slower path for copying. */
+ if (pFramesOut != pFramesIn) {
+ ma_uint32 iFrame;
-static ma_result ma_device__untrack__coreaudio(ma_device* pDevice)
-{
- MA_ASSERT(pDevice != NULL);
+ /* */ if (pBPF->format == ma_format_f32) {
+ /* */ float* pFramesOutF32 = ( float*)pFramesOut;
+ const float* pFramesInF32 = (const float*)pFramesIn;
- ma_mutex_lock(&g_DeviceTrackingMutex_CoreAudio);
- {
- ma_uint32 iDevice;
- for (iDevice = 0; iDevice < g_TrackedDeviceCount_CoreAudio; iDevice += 1) {
- if (g_ppTrackedDevices_CoreAudio[iDevice] == pDevice) {
- /* We've found the device. We now need to remove it from the list. */
- ma_uint32 jDevice;
- for (jDevice = iDevice; jDevice < g_TrackedDeviceCount_CoreAudio-1; jDevice += 1) {
- g_ppTrackedDevices_CoreAudio[jDevice] = g_ppTrackedDevices_CoreAudio[jDevice+1];
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ MA_COPY_MEMORY(pFramesOutF32, pFramesInF32, ma_get_bytes_per_frame(pBPF->format, pBPF->channels));
+
+ for (ibpf2 = 0; ibpf2 < pBPF->bpf2Count; ibpf2 += 1) {
+ ma_bpf2_process_pcm_frame_f32(&pBPF->pBPF2[ibpf2], pFramesOutF32, pFramesOutF32);
}
- g_TrackedDeviceCount_CoreAudio -= 1;
+ pFramesOutF32 += pBPF->channels;
+ pFramesInF32 += pBPF->channels;
+ }
+ } else if (pBPF->format == ma_format_s16) {
+ /* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut;
+ const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn;
- /* If there's nothing else in the list we need to free memory. */
- if (g_TrackedDeviceCount_CoreAudio == 0) {
- ma__free_from_callbacks(g_ppTrackedDevices_CoreAudio, &pDevice->pContext->allocationCallbacks);
- g_ppTrackedDevices_CoreAudio = NULL;
- g_TrackedDeviceCap_CoreAudio = 0;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ MA_COPY_MEMORY(pFramesOutS16, pFramesInS16, ma_get_bytes_per_frame(pBPF->format, pBPF->channels));
+
+ for (ibpf2 = 0; ibpf2 < pBPF->bpf2Count; ibpf2 += 1) {
+ ma_bpf2_process_pcm_frame_s16(&pBPF->pBPF2[ibpf2], pFramesOutS16, pFramesOutS16);
}
- break;
+ pFramesOutS16 += pBPF->channels;
+ pFramesInS16 += pBPF->channels;
}
+ } else {
+ MA_ASSERT(MA_FALSE);
+ return MA_INVALID_OPERATION; /* Should never hit this. */
}
}
- ma_mutex_unlock(&g_DeviceTrackingMutex_CoreAudio);
return MA_SUCCESS;
}
-#endif
-
-#if defined(MA_APPLE_MOBILE)
-@interface ma_router_change_handler:NSObject {
- ma_device* m_pDevice;
-}
-@end
-@implementation ma_router_change_handler
--(id)init:(ma_device*)pDevice
+MA_API ma_uint32 ma_bpf_get_latency(const ma_bpf* pBPF)
{
- self = [super init];
- m_pDevice = pDevice;
-
- [[NSNotificationCenter defaultCenter] addObserver:self selector:@selector(handle_route_change:) name:AVAudioSessionRouteChangeNotification object:[AVAudioSession sharedInstance]];
+ if (pBPF == NULL) {
+ return 0;
+ }
- return self;
+ return pBPF->bpf2Count*2;
}
--(void)dealloc
-{
- [self remove_handler];
-}
--(void)remove_handler
-{
- [[NSNotificationCenter defaultCenter] removeObserver:self name:AVAudioSessionRouteChangeNotification object:nil];
-}
+/**************************************************************************************************************************************************************
--(void)handle_route_change:(NSNotification*)pNotification
+Notching Filter
+
+**************************************************************************************************************************************************************/
+MA_API ma_notch2_config ma_notch2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double q, double frequency)
{
- AVAudioSession* pSession = [AVAudioSession sharedInstance];
+ ma_notch2_config config;
- NSInteger reason = [[[pNotification userInfo] objectForKey:AVAudioSessionRouteChangeReasonKey] integerValue];
- switch (reason)
- {
- case AVAudioSessionRouteChangeReasonOldDeviceUnavailable:
- {
- ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_DEBUG, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonOldDeviceUnavailable\n");
- } break;
+ MA_ZERO_OBJECT(&config);
+ config.format = format;
+ config.channels = channels;
+ config.sampleRate = sampleRate;
+ config.q = q;
+ config.frequency = frequency;
- case AVAudioSessionRouteChangeReasonNewDeviceAvailable:
- {
- ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_DEBUG, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonNewDeviceAvailable\n");
- } break;
+ if (config.q == 0) {
+ config.q = 0.707107;
+ }
- case AVAudioSessionRouteChangeReasonNoSuitableRouteForCategory:
- {
- ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_DEBUG, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonNoSuitableRouteForCategory\n");
- } break;
+ return config;
+}
- case AVAudioSessionRouteChangeReasonWakeFromSleep:
- {
- ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_DEBUG, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonWakeFromSleep\n");
- } break;
- case AVAudioSessionRouteChangeReasonOverride:
- {
- ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_DEBUG, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonOverride\n");
- } break;
+static MA_INLINE ma_biquad_config ma_notch2__get_biquad_config(const ma_notch2_config* pConfig)
+{
+ ma_biquad_config bqConfig;
+ double q;
+ double w;
+ double s;
+ double c;
+ double a;
- case AVAudioSessionRouteChangeReasonCategoryChange:
- {
- ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_DEBUG, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonCategoryChange\n");
- } break;
+ MA_ASSERT(pConfig != NULL);
- case AVAudioSessionRouteChangeReasonUnknown:
- default:
- {
- ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_DEBUG, "[Core Audio] Route Changed: AVAudioSessionRouteChangeReasonUnknown\n");
- } break;
- }
+ q = pConfig->q;
+ w = 2 * MA_PI_D * pConfig->frequency / pConfig->sampleRate;
+ s = ma_sind(w);
+ c = ma_cosd(w);
+ a = s / (2*q);
- ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_DEBUG, "[Core Audio] Changing Route. inputNumberChannels=%d; outputNumberOfChannels=%d\n", (int)pSession.inputNumberOfChannels, (int)pSession.outputNumberOfChannels);
+ bqConfig.b0 = 1;
+ bqConfig.b1 = -2 * c;
+ bqConfig.b2 = 1;
+ bqConfig.a0 = 1 + a;
+ bqConfig.a1 = -2 * c;
+ bqConfig.a2 = 1 - a;
- /* Temporarily disabling this section of code because it appears to be causing errors. */
-#if 0
- ma_uint32 previousState = ma_device_get_state(m_pDevice);
+ bqConfig.format = pConfig->format;
+ bqConfig.channels = pConfig->channels;
- if (previousState == MA_STATE_STARTED) {
- ma_device_stop(m_pDevice);
- }
+ return bqConfig;
+}
- if (m_pDevice->type == ma_device_type_capture || m_pDevice->type == ma_device_type_duplex) {
- m_pDevice->capture.internalChannels = (ma_uint32)pSession.inputNumberOfChannels;
- m_pDevice->capture.internalSampleRate = (ma_uint32)pSession.sampleRate;
- ma_device__post_init_setup(m_pDevice, ma_device_type_capture);
- }
- if (m_pDevice->type == ma_device_type_playback || m_pDevice->type == ma_device_type_duplex) {
- m_pDevice->playback.internalChannels = (ma_uint32)pSession.outputNumberOfChannels;
- m_pDevice->playback.internalSampleRate = (ma_uint32)pSession.sampleRate;
- ma_device__post_init_setup(m_pDevice, ma_device_type_playback);
- }
+MA_API ma_result ma_notch2_get_heap_size(const ma_notch2_config* pConfig, size_t* pHeapSizeInBytes)
+{
+ ma_biquad_config bqConfig;
+ bqConfig = ma_notch2__get_biquad_config(pConfig);
- if (previousState == MA_STATE_STARTED) {
- ma_device_start(m_pDevice);
- }
-#endif
+ return ma_biquad_get_heap_size(&bqConfig, pHeapSizeInBytes);
}
-@end
-#endif
-static ma_result ma_device_uninit__coreaudio(ma_device* pDevice)
+MA_API ma_result ma_notch2_init_preallocated(const ma_notch2_config* pConfig, void* pHeap, ma_notch2* pFilter)
{
- MA_ASSERT(pDevice != NULL);
- MA_ASSERT(ma_device_get_state(pDevice) == MA_STATE_UNINITIALIZED);
+ ma_result result;
+ ma_biquad_config bqConfig;
-#if defined(MA_APPLE_DESKTOP)
- /*
- Make sure we're no longer tracking the device. It doesn't matter if we call this for a non-default device because it'll
- just gracefully ignore it.
- */
- ma_device__untrack__coreaudio(pDevice);
-#endif
-#if defined(MA_APPLE_MOBILE)
- if (pDevice->coreaudio.pRouteChangeHandler != NULL) {
- ma_router_change_handler* pRouteChangeHandler = (__bridge_transfer ma_router_change_handler*)pDevice->coreaudio.pRouteChangeHandler;
- [pRouteChangeHandler remove_handler];
+ if (pFilter == NULL) {
+ return MA_INVALID_ARGS;
}
-#endif
- if (pDevice->coreaudio.audioUnitCapture != NULL) {
- ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitCapture);
- }
- if (pDevice->coreaudio.audioUnitPlayback != NULL) {
- ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitPlayback);
+ MA_ZERO_OBJECT(pFilter);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pDevice->coreaudio.pAudioBufferList) {
- ma__free_from_callbacks(pDevice->coreaudio.pAudioBufferList, &pDevice->pContext->allocationCallbacks);
+ bqConfig = ma_notch2__get_biquad_config(pConfig);
+ result = ma_biquad_init_preallocated(&bqConfig, pHeap, &pFilter->bq);
+ if (result != MA_SUCCESS) {
+ return result;
}
return MA_SUCCESS;
}
-typedef struct
-{
- ma_bool32 allowNominalSampleRateChange;
-
- /* Input. */
- ma_format formatIn;
- ma_uint32 channelsIn;
- ma_uint32 sampleRateIn;
- ma_channel channelMapIn[MA_MAX_CHANNELS];
- ma_uint32 periodSizeInFramesIn;
- ma_uint32 periodSizeInMillisecondsIn;
- ma_uint32 periodsIn;
- ma_share_mode shareMode;
- ma_performance_profile performanceProfile;
- ma_bool32 registerStopEvent;
-
- /* Output. */
-#if defined(MA_APPLE_DESKTOP)
- AudioObjectID deviceObjectID;
-#endif
- AudioComponent component;
- AudioUnit audioUnit;
- AudioBufferList* pAudioBufferList; /* Only used for input devices. */
- ma_format formatOut;
- ma_uint32 channelsOut;
- ma_uint32 sampleRateOut;
- ma_channel channelMapOut[MA_MAX_CHANNELS];
- ma_uint32 periodSizeInFramesOut;
- ma_uint32 periodsOut;
- char deviceName[256];
-} ma_device_init_internal_data__coreaudio;
-
-static ma_result ma_device_init_internal__coreaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_init_internal_data__coreaudio* pData, void* pDevice_DoNotReference) /* <-- pDevice is typed as void* intentionally so as to avoid accidentally referencing it. */
+MA_API ma_result ma_notch2_init(const ma_notch2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_notch2* pFilter)
{
ma_result result;
- OSStatus status;
- UInt32 enableIOFlag;
- AudioStreamBasicDescription bestFormat;
- UInt32 actualPeriodSizeInFrames;
- AURenderCallbackStruct callbackInfo;
-#if defined(MA_APPLE_DESKTOP)
- AudioObjectID deviceObjectID;
-#else
- UInt32 actualPeriodSizeInFramesSize = sizeof(actualPeriodSizeInFrames);
-#endif
-
- /* This API should only be used for a single device type: playback or capture. No full-duplex mode. */
- if (deviceType == ma_device_type_duplex) {
- return MA_INVALID_ARGS;
- }
-
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(deviceType == ma_device_type_playback || deviceType == ma_device_type_capture);
-
-#if defined(MA_APPLE_DESKTOP)
- pData->deviceObjectID = 0;
-#endif
- pData->component = NULL;
- pData->audioUnit = NULL;
- pData->pAudioBufferList = NULL;
+ size_t heapSizeInBytes;
+ void* pHeap;
-#if defined(MA_APPLE_DESKTOP)
- result = ma_find_AudioObjectID(pContext, deviceType, pDeviceID, &deviceObjectID);
+ result = ma_notch2_get_heap_size(pConfig, &heapSizeInBytes);
if (result != MA_SUCCESS) {
return result;
}
- pData->deviceObjectID = deviceObjectID;
-#endif
-
- /* Core audio doesn't really use the notion of a period so we can leave this unmodified, but not too over the top. */
- pData->periodsOut = pData->periodsIn;
- if (pData->periodsOut == 0) {
- pData->periodsOut = MA_DEFAULT_PERIODS;
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+ } else {
+ pHeap = NULL;
}
- if (pData->periodsOut > 16) {
- pData->periodsOut = 16;
+
+ result = ma_notch2_init_preallocated(pConfig, pHeap, pFilter);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
}
+ pFilter->bq._ownsHeap = MA_TRUE; /* <-- This will cause the biquad to take ownership of the heap and free it when it's uninitialized. */
+ return MA_SUCCESS;
+}
- /* Audio unit. */
- status = ((ma_AudioComponentInstanceNew_proc)pContext->coreaudio.AudioComponentInstanceNew)((AudioComponent)pContext->coreaudio.component, (AudioUnit*)&pData->audioUnit);
- if (status != noErr) {
- return ma_result_from_OSStatus(status);
+MA_API void ma_notch2_uninit(ma_notch2* pFilter, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pFilter == NULL) {
+ return;
}
+ ma_biquad_uninit(&pFilter->bq, pAllocationCallbacks); /* <-- This will free the heap allocation. */
+}
- /* The input/output buses need to be explicitly enabled and disabled. We set the flag based on the output unit first, then we just swap it for input. */
- enableIOFlag = 1;
- if (deviceType == ma_device_type_capture) {
- enableIOFlag = 0;
- }
+MA_API ma_result ma_notch2_reinit(const ma_notch2_config* pConfig, ma_notch2* pFilter)
+{
+ ma_result result;
+ ma_biquad_config bqConfig;
- status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, MA_COREAUDIO_OUTPUT_BUS, &enableIOFlag, sizeof(enableIOFlag));
- if (status != noErr) {
- ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
- return ma_result_from_OSStatus(status);
+ if (pFilter == NULL || pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- enableIOFlag = (enableIOFlag == 0) ? 1 : 0;
- status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, MA_COREAUDIO_INPUT_BUS, &enableIOFlag, sizeof(enableIOFlag));
- if (status != noErr) {
- ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
- return ma_result_from_OSStatus(status);
+ bqConfig = ma_notch2__get_biquad_config(pConfig);
+ result = ma_biquad_reinit(&bqConfig, &pFilter->bq);
+ if (result != MA_SUCCESS) {
+ return result;
}
+ return MA_SUCCESS;
+}
- /* Set the device to use with this audio unit. This is only used on desktop since we are using defaults on mobile. */
-#if defined(MA_APPLE_DESKTOP)
- status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &deviceObjectID, sizeof(deviceObjectID));
- if (status != noErr) {
- ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
- return ma_result_from_OSStatus(result);
- }
-#else
- /*
- For some reason it looks like Apple is only allowing selection of the input device. There does not appear to be any way to change
- the default output route. I have no idea why this is like this, but for now we'll only be able to configure capture devices.
- */
- if (pDeviceID != NULL) {
- if (deviceType == ma_device_type_capture) {
- ma_bool32 found = MA_FALSE;
- NSArray *pInputs = [[[AVAudioSession sharedInstance] currentRoute] inputs];
- for (AVAudioSessionPortDescription* pPortDesc in pInputs) {
- if (strcmp(pDeviceID->coreaudio, [pPortDesc.UID UTF8String]) == 0) {
- [[AVAudioSession sharedInstance] setPreferredInput:pPortDesc error:nil];
- found = MA_TRUE;
- break;
- }
- }
+static MA_INLINE void ma_notch2_process_pcm_frame_s16(ma_notch2* pFilter, ma_int16* pFrameOut, const ma_int16* pFrameIn)
+{
+ ma_biquad_process_pcm_frame_s16(&pFilter->bq, pFrameOut, pFrameIn);
+}
- if (found == MA_FALSE) {
- return MA_DOES_NOT_EXIST;
- }
- }
+static MA_INLINE void ma_notch2_process_pcm_frame_f32(ma_notch2* pFilter, float* pFrameOut, const float* pFrameIn)
+{
+ ma_biquad_process_pcm_frame_f32(&pFilter->bq, pFrameOut, pFrameIn);
+}
+
+MA_API ma_result ma_notch2_process_pcm_frames(ma_notch2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+{
+ if (pFilter == NULL) {
+ return MA_INVALID_ARGS;
}
-#endif
- /*
- Format. This is the hardest part of initialization because there's a few variables to take into account.
- 1) The format must be supported by the device.
- 2) The format must be supported miniaudio.
- 3) There's a priority that miniaudio prefers.
+ return ma_biquad_process_pcm_frames(&pFilter->bq, pFramesOut, pFramesIn, frameCount);
+}
- Ideally we would like to use a format that's as close to the hardware as possible so we can get as close to a passthrough as possible. The
- most important property is the sample rate. miniaudio can do format conversion for any sample rate and channel count, but cannot do the same
- for the sample data format. If the sample data format is not supported by miniaudio it must be ignored completely.
+MA_API ma_uint32 ma_notch2_get_latency(const ma_notch2* pFilter)
+{
+ if (pFilter == NULL) {
+ return 0;
+ }
- On mobile platforms this is a bit different. We just force the use of whatever the audio unit's current format is set to.
- */
- {
- AudioStreamBasicDescription origFormat;
- UInt32 origFormatSize = sizeof(origFormat);
- AudioUnitScope formatScope = (deviceType == ma_device_type_playback) ? kAudioUnitScope_Input : kAudioUnitScope_Output;
- AudioUnitElement formatElement = (deviceType == ma_device_type_playback) ? MA_COREAUDIO_OUTPUT_BUS : MA_COREAUDIO_INPUT_BUS;
+ return ma_biquad_get_latency(&pFilter->bq);
+}
- if (deviceType == ma_device_type_playback) {
- status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, MA_COREAUDIO_OUTPUT_BUS, &origFormat, &origFormatSize);
- } else {
- status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, MA_COREAUDIO_INPUT_BUS, &origFormat, &origFormatSize);
- }
- if (status != noErr) {
- ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
- return ma_result_from_OSStatus(status);
- }
- #if defined(MA_APPLE_DESKTOP)
- result = ma_find_best_format__coreaudio(pContext, deviceObjectID, deviceType, pData->formatIn, pData->channelsIn, pData->sampleRateIn, &origFormat, &bestFormat);
- if (result != MA_SUCCESS) {
- ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
- return result;
- }
- /*
- Technical Note TN2091: Device input using the HAL Output Audio Unit
- https://developer.apple.com/library/archive/technotes/tn2091/_index.html
+/**************************************************************************************************************************************************************
- This documentation says the following:
+Peaking EQ Filter
- The internal AudioConverter can handle any *simple* conversion. Typically, this means that a client can specify ANY
- variant of the PCM formats. Consequently, the device's sample rate should match the desired sample rate. If sample rate
- conversion is needed, it can be accomplished by buffering the input and converting the data on a separate thread with
- another AudioConverter.
+**************************************************************************************************************************************************************/
+MA_API ma_peak2_config ma_peak2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double q, double frequency)
+{
+ ma_peak2_config config;
- The important part here is the mention that it can handle *simple* conversions, which does *not* include sample rate. We
- therefore want to ensure the sample rate stays consistent. This document is specifically for input, but I'm going to play it
- safe and apply the same rule to output as well.
+ MA_ZERO_OBJECT(&config);
+ config.format = format;
+ config.channels = channels;
+ config.sampleRate = sampleRate;
+ config.gainDB = gainDB;
+ config.q = q;
+ config.frequency = frequency;
- I have tried going against the documentation by setting the sample rate anyway, but this just results in AudioUnitRender()
- returning a result code of -10863. I have also tried changing the format directly on the input scope on the input bus, but
- this just results in `ca_require: IsStreamFormatWritable(inScope, inElement) NotWritable` when trying to set the format.
+ if (config.q == 0) {
+ config.q = 0.707107;
+ }
- Something that does seem to work, however, has been setting the nominal sample rate on the deivce object. The problem with
- this, however, is that it actually changes the sample rate at the operating system level and not just the application. This
- could be intrusive to the user, however, so I don't think it's wise to make this the default. Instead I'm making this a
- configuration option. When the `coreaudio.allowNominalSampleRateChange` config option is set to true, changing the sample
- rate will be allowed. Otherwise it'll be fixed to the current sample rate. To check the system-defined sample rate, run
- the Audio MIDI Setup program that comes installed on macOS and observe how the sample rate changes as the sample rate is
- changed by miniaudio.
- */
- if (pData->allowNominalSampleRateChange) {
- AudioValueRange sampleRateRange;
- AudioObjectPropertyAddress propAddress;
+ return config;
+}
- sampleRateRange.mMinimum = bestFormat.mSampleRate;
- sampleRateRange.mMaximum = bestFormat.mSampleRate;
- propAddress.mSelector = kAudioDevicePropertyNominalSampleRate;
- propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput;
- propAddress.mElement = kAudioObjectPropertyElementMaster;
+static MA_INLINE ma_biquad_config ma_peak2__get_biquad_config(const ma_peak2_config* pConfig)
+{
+ ma_biquad_config bqConfig;
+ double q;
+ double w;
+ double s;
+ double c;
+ double a;
+ double A;
- status = ((ma_AudioObjectSetPropertyData_proc)pContext->coreaudio.AudioObjectSetPropertyData)(deviceObjectID, &propAddress, 0, NULL, sizeof(sampleRateRange), &sampleRateRange);
- if (status != noErr) {
- bestFormat.mSampleRate = origFormat.mSampleRate;
- }
- } else {
- bestFormat.mSampleRate = origFormat.mSampleRate;
- }
+ MA_ASSERT(pConfig != NULL);
- status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, sizeof(bestFormat));
- if (status != noErr) {
- /* We failed to set the format, so fall back to the current format of the audio unit. */
- bestFormat = origFormat;
- }
- #else
- bestFormat = origFormat;
+ q = pConfig->q;
+ w = 2 * MA_PI_D * pConfig->frequency / pConfig->sampleRate;
+ s = ma_sind(w);
+ c = ma_cosd(w);
+ a = s / (2*q);
+ A = ma_powd(10, (pConfig->gainDB / 40));
- /*
- Sample rate is a little different here because for some reason kAudioUnitProperty_StreamFormat returns 0... Oh well. We need to instead try
- setting the sample rate to what the user has requested and then just see the results of it. Need to use some Objective-C here for this since
- it depends on Apple's AVAudioSession API. To do this we just get the shared AVAudioSession instance and then set it. Note that from what I
- can tell, it looks like the sample rate is shared between playback and capture for everything.
- */
- @autoreleasepool {
- AVAudioSession* pAudioSession = [AVAudioSession sharedInstance];
- MA_ASSERT(pAudioSession != NULL);
+ bqConfig.b0 = 1 + (a * A);
+ bqConfig.b1 = -2 * c;
+ bqConfig.b2 = 1 - (a * A);
+ bqConfig.a0 = 1 + (a / A);
+ bqConfig.a1 = -2 * c;
+ bqConfig.a2 = 1 - (a / A);
- [pAudioSession setPreferredSampleRate:(double)pData->sampleRateIn error:nil];
- bestFormat.mSampleRate = pAudioSession.sampleRate;
+ bqConfig.format = pConfig->format;
+ bqConfig.channels = pConfig->channels;
- /*
- I've had a report that the channel count returned by AudioUnitGetProperty above is inconsistent with
- AVAudioSession outputNumberOfChannels. I'm going to try using the AVAudioSession values instead.
- */
- if (deviceType == ma_device_type_playback) {
- bestFormat.mChannelsPerFrame = (UInt32)pAudioSession.outputNumberOfChannels;
- }
- if (deviceType == ma_device_type_capture) {
- bestFormat.mChannelsPerFrame = (UInt32)pAudioSession.inputNumberOfChannels;
- }
- }
+ return bqConfig;
+}
- status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, sizeof(bestFormat));
- if (status != noErr) {
- ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
- return ma_result_from_OSStatus(status);
- }
- #endif
+MA_API ma_result ma_peak2_get_heap_size(const ma_peak2_config* pConfig, size_t* pHeapSizeInBytes)
+{
+ ma_biquad_config bqConfig;
+ bqConfig = ma_peak2__get_biquad_config(pConfig);
- result = ma_format_from_AudioStreamBasicDescription(&bestFormat, &pData->formatOut);
- if (result != MA_SUCCESS) {
- ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
- return result;
- }
+ return ma_biquad_get_heap_size(&bqConfig, pHeapSizeInBytes);
+}
- if (pData->formatOut == ma_format_unknown) {
- ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
- return MA_FORMAT_NOT_SUPPORTED;
- }
+MA_API ma_result ma_peak2_init_preallocated(const ma_peak2_config* pConfig, void* pHeap, ma_peak2* pFilter)
+{
+ ma_result result;
+ ma_biquad_config bqConfig;
- pData->channelsOut = bestFormat.mChannelsPerFrame;
- pData->sampleRateOut = bestFormat.mSampleRate;
+ if (pFilter == NULL) {
+ return MA_INVALID_ARGS;
}
- /* Clamp the channel count for safety. */
- if (pData->channelsOut > MA_MAX_CHANNELS) {
- pData->channelsOut = MA_MAX_CHANNELS;
+ MA_ZERO_OBJECT(pFilter);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- /*
- Internal channel map. This is weird in my testing. If I use the AudioObject to get the
- channel map, the channel descriptions are set to "Unknown" for some reason. To work around
- this it looks like retrieving it from the AudioUnit will work. However, and this is where
- it gets weird, it doesn't seem to work with capture devices, nor at all on iOS... Therefore
- I'm going to fall back to a default assumption in these cases.
- */
-#if defined(MA_APPLE_DESKTOP)
- result = ma_get_AudioUnit_channel_map(pContext, pData->audioUnit, deviceType, pData->channelMapOut, pData->channelsOut);
+ bqConfig = ma_peak2__get_biquad_config(pConfig);
+ result = ma_biquad_init_preallocated(&bqConfig, pHeap, &pFilter->bq);
if (result != MA_SUCCESS) {
- #if 0
- /* Try falling back to the channel map from the AudioObject. */
- result = ma_get_AudioObject_channel_map(pContext, deviceObjectID, deviceType, pData->channelMapOut, pData->channelsOut);
- if (result != MA_SUCCESS) {
- return result;
- }
- #else
- /* Fall back to default assumptions. */
- ma_get_standard_channel_map(ma_standard_channel_map_default, pData->channelsOut, pData->channelMapOut);
- #endif
+ return result;
}
-#else
- /* TODO: Figure out how to get the channel map using AVAudioSession. */
- ma_get_standard_channel_map(ma_standard_channel_map_default, pData->channelsOut, pData->channelMapOut);
-#endif
+ return MA_SUCCESS;
+}
- /* Buffer size. Not allowing this to be configurable on iOS. */
- if (pData->periodSizeInFramesIn == 0) {
- if (pData->periodSizeInMillisecondsIn == 0) {
- if (pData->performanceProfile == ma_performance_profile_low_latency) {
- actualPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY, pData->sampleRateOut);
- } else {
- actualPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE, pData->sampleRateOut);
- }
- } else {
- actualPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pData->periodSizeInMillisecondsIn, pData->sampleRateOut);
+MA_API ma_result ma_peak2_init(const ma_peak2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_peak2* pFilter)
+{
+ ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
+
+ result = ma_peak2_get_heap_size(pConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
}
} else {
- actualPeriodSizeInFrames = pData->periodSizeInFramesIn;
+ pHeap = NULL;
}
-#if defined(MA_APPLE_DESKTOP)
- result = ma_set_AudioObject_buffer_size_in_frames(pContext, deviceObjectID, deviceType, &actualPeriodSizeInFrames);
+ result = ma_peak2_init_preallocated(pConfig, pHeap, pFilter);
if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
return result;
}
-#else
- /*
- I don't know how to configure buffer sizes on iOS so for now we're not allowing it to be configured. Instead we're
- just going to set it to the value of kAudioUnitProperty_MaximumFramesPerSlice.
- */
- status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(pData->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Global, 0, &actualPeriodSizeInFrames, &actualPeriodSizeInFramesSize);
- if (status != noErr) {
- ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
- return ma_result_from_OSStatus(status);
+
+ pFilter->bq._ownsHeap = MA_TRUE; /* <-- This will cause the biquad to take ownership of the heap and free it when it's uninitialized. */
+ return MA_SUCCESS;
+}
+
+MA_API void ma_peak2_uninit(ma_peak2* pFilter, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pFilter == NULL) {
+ return;
}
-#endif
+ ma_biquad_uninit(&pFilter->bq, pAllocationCallbacks); /* <-- This will free the heap allocation. */
+}
- /*
- During testing I discovered that the buffer size can be too big. You'll get an error like this:
+MA_API ma_result ma_peak2_reinit(const ma_peak2_config* pConfig, ma_peak2* pFilter)
+{
+ ma_result result;
+ ma_biquad_config bqConfig;
- kAudioUnitErr_TooManyFramesToProcess : inFramesToProcess=4096, mMaxFramesPerSlice=512
+ if (pFilter == NULL || pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
- Note how inFramesToProcess is smaller than mMaxFramesPerSlice. To fix, we need to set kAudioUnitProperty_MaximumFramesPerSlice to that
- of the size of our buffer, or do it the other way around and set our buffer size to the kAudioUnitProperty_MaximumFramesPerSlice.
- */
- status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Global, 0, &actualPeriodSizeInFrames, sizeof(actualPeriodSizeInFrames));
- if (status != noErr) {
- ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
- return ma_result_from_OSStatus(status);
+ bqConfig = ma_peak2__get_biquad_config(pConfig);
+ result = ma_biquad_reinit(&bqConfig, &pFilter->bq);
+ if (result != MA_SUCCESS) {
+ return result;
}
- pData->periodSizeInFramesOut = (ma_uint32)actualPeriodSizeInFrames;
+ return MA_SUCCESS;
+}
- /* We need a buffer list if this is an input device. We render into this in the input callback. */
- if (deviceType == ma_device_type_capture) {
- ma_bool32 isInterleaved = (bestFormat.mFormatFlags & kAudioFormatFlagIsNonInterleaved) == 0;
- AudioBufferList* pBufferList;
+static MA_INLINE void ma_peak2_process_pcm_frame_s16(ma_peak2* pFilter, ma_int16* pFrameOut, const ma_int16* pFrameIn)
+{
+ ma_biquad_process_pcm_frame_s16(&pFilter->bq, pFrameOut, pFrameIn);
+}
- pBufferList = ma_allocate_AudioBufferList__coreaudio(pData->periodSizeInFramesOut, pData->formatOut, pData->channelsOut, (isInterleaved) ? ma_stream_layout_interleaved : ma_stream_layout_deinterleaved, &pContext->allocationCallbacks);
- if (pBufferList == NULL) {
- ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
- return MA_OUT_OF_MEMORY;
- }
+static MA_INLINE void ma_peak2_process_pcm_frame_f32(ma_peak2* pFilter, float* pFrameOut, const float* pFrameIn)
+{
+ ma_biquad_process_pcm_frame_f32(&pFilter->bq, pFrameOut, pFrameIn);
+}
- pData->pAudioBufferList = pBufferList;
+MA_API ma_result ma_peak2_process_pcm_frames(ma_peak2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+{
+ if (pFilter == NULL) {
+ return MA_INVALID_ARGS;
}
- /* Callbacks. */
- callbackInfo.inputProcRefCon = pDevice_DoNotReference;
- if (deviceType == ma_device_type_playback) {
- callbackInfo.inputProc = ma_on_output__coreaudio;
- status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Global, 0, &callbackInfo, sizeof(callbackInfo));
- if (status != noErr) {
- ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
- return ma_result_from_OSStatus(status);
- }
- } else {
- callbackInfo.inputProc = ma_on_input__coreaudio;
- status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &callbackInfo, sizeof(callbackInfo));
- if (status != noErr) {
- ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
- return ma_result_from_OSStatus(status);
- }
- }
+ return ma_biquad_process_pcm_frames(&pFilter->bq, pFramesOut, pFramesIn, frameCount);
+}
- /* We need to listen for stop events. */
- if (pData->registerStopEvent) {
- status = ((ma_AudioUnitAddPropertyListener_proc)pContext->coreaudio.AudioUnitAddPropertyListener)(pData->audioUnit, kAudioOutputUnitProperty_IsRunning, on_start_stop__coreaudio, pDevice_DoNotReference);
- if (status != noErr) {
- ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
- return ma_result_from_OSStatus(status);
- }
+MA_API ma_uint32 ma_peak2_get_latency(const ma_peak2* pFilter)
+{
+ if (pFilter == NULL) {
+ return 0;
}
- /* Initialize the audio unit. */
- status = ((ma_AudioUnitInitialize_proc)pContext->coreaudio.AudioUnitInitialize)(pData->audioUnit);
- if (status != noErr) {
- ma__free_from_callbacks(pData->pAudioBufferList, &pContext->allocationCallbacks);
- pData->pAudioBufferList = NULL;
- ((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
- return ma_result_from_OSStatus(status);
- }
+ return ma_biquad_get_latency(&pFilter->bq);
+}
- /* Grab the name. */
-#if defined(MA_APPLE_DESKTOP)
- ma_get_AudioObject_name(pContext, deviceObjectID, sizeof(pData->deviceName), pData->deviceName);
-#else
- if (deviceType == ma_device_type_playback) {
- ma_strcpy_s(pData->deviceName, sizeof(pData->deviceName), MA_DEFAULT_PLAYBACK_DEVICE_NAME);
- } else {
- ma_strcpy_s(pData->deviceName, sizeof(pData->deviceName), MA_DEFAULT_CAPTURE_DEVICE_NAME);
- }
-#endif
- return result;
+/**************************************************************************************************************************************************************
+
+Low Shelf Filter
+
+**************************************************************************************************************************************************************/
+MA_API ma_loshelf2_config ma_loshelf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double shelfSlope, double frequency)
+{
+ ma_loshelf2_config config;
+
+ MA_ZERO_OBJECT(&config);
+ config.format = format;
+ config.channels = channels;
+ config.sampleRate = sampleRate;
+ config.gainDB = gainDB;
+ config.shelfSlope = shelfSlope;
+ config.frequency = frequency;
+
+ return config;
}
-#if defined(MA_APPLE_DESKTOP)
-static ma_result ma_device_reinit_internal__coreaudio(ma_device* pDevice, ma_device_type deviceType, ma_bool32 disposePreviousAudioUnit)
+
+static MA_INLINE ma_biquad_config ma_loshelf2__get_biquad_config(const ma_loshelf2_config* pConfig)
{
- ma_device_init_internal_data__coreaudio data;
- ma_result result;
+ ma_biquad_config bqConfig;
+ double w;
+ double s;
+ double c;
+ double A;
+ double S;
+ double a;
+ double sqrtA;
- /* This should only be called for playback or capture, not duplex. */
- if (deviceType == ma_device_type_duplex) {
- return MA_INVALID_ARGS;
- }
+ MA_ASSERT(pConfig != NULL);
- data.allowNominalSampleRateChange = MA_FALSE; /* Don't change the nominal sample rate when switching devices. */
+ w = 2 * MA_PI_D * pConfig->frequency / pConfig->sampleRate;
+ s = ma_sind(w);
+ c = ma_cosd(w);
+ A = ma_powd(10, (pConfig->gainDB / 40));
+ S = pConfig->shelfSlope;
+ a = s/2 * ma_sqrtd((A + 1/A) * (1/S - 1) + 2);
+ sqrtA = 2*ma_sqrtd(A)*a;
- if (deviceType == ma_device_type_capture) {
- data.formatIn = pDevice->capture.format;
- data.channelsIn = pDevice->capture.channels;
- data.sampleRateIn = pDevice->sampleRate;
- MA_COPY_MEMORY(data.channelMapIn, pDevice->capture.channelMap, sizeof(pDevice->capture.channelMap));
- data.shareMode = pDevice->capture.shareMode;
- data.performanceProfile = pDevice->coreaudio.originalPerformanceProfile;
- data.registerStopEvent = MA_TRUE;
+ bqConfig.b0 = A * ((A + 1) - (A - 1)*c + sqrtA);
+ bqConfig.b1 = 2 * A * ((A - 1) - (A + 1)*c);
+ bqConfig.b2 = A * ((A + 1) - (A - 1)*c - sqrtA);
+ bqConfig.a0 = (A + 1) + (A - 1)*c + sqrtA;
+ bqConfig.a1 = -2 * ((A - 1) + (A + 1)*c);
+ bqConfig.a2 = (A + 1) + (A - 1)*c - sqrtA;
- if (disposePreviousAudioUnit) {
- ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitCapture);
- ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitCapture);
- }
- if (pDevice->coreaudio.pAudioBufferList) {
- ma__free_from_callbacks(pDevice->coreaudio.pAudioBufferList, &pDevice->pContext->allocationCallbacks);
- }
- } else if (deviceType == ma_device_type_playback) {
- data.formatIn = pDevice->playback.format;
- data.channelsIn = pDevice->playback.channels;
- data.sampleRateIn = pDevice->sampleRate;
- MA_COPY_MEMORY(data.channelMapIn, pDevice->playback.channelMap, sizeof(pDevice->playback.channelMap));
- data.shareMode = pDevice->playback.shareMode;
- data.performanceProfile = pDevice->coreaudio.originalPerformanceProfile;
- data.registerStopEvent = (pDevice->type != ma_device_type_duplex);
+ bqConfig.format = pConfig->format;
+ bqConfig.channels = pConfig->channels;
- if (disposePreviousAudioUnit) {
- ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitPlayback);
- ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitPlayback);
- }
+ return bqConfig;
+}
+
+MA_API ma_result ma_loshelf2_get_heap_size(const ma_loshelf2_config* pConfig, size_t* pHeapSizeInBytes)
+{
+ ma_biquad_config bqConfig;
+ bqConfig = ma_loshelf2__get_biquad_config(pConfig);
+
+ return ma_biquad_get_heap_size(&bqConfig, pHeapSizeInBytes);
+}
+
+MA_API ma_result ma_loshelf2_init_preallocated(const ma_loshelf2_config* pConfig, void* pHeap, ma_loshelf2* pFilter)
+{
+ ma_result result;
+ ma_biquad_config bqConfig;
+
+ if (pFilter == NULL) {
+ return MA_INVALID_ARGS;
}
- data.periodSizeInFramesIn = pDevice->coreaudio.originalPeriodSizeInFrames;
- data.periodSizeInMillisecondsIn = pDevice->coreaudio.originalPeriodSizeInMilliseconds;
- data.periodsIn = pDevice->coreaudio.originalPeriods;
- /* Need at least 3 periods for duplex. */
- if (data.periodsIn < 3 && pDevice->type == ma_device_type_duplex) {
- data.periodsIn = 3;
+ MA_ZERO_OBJECT(pFilter);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- result = ma_device_init_internal__coreaudio(pDevice->pContext, deviceType, NULL, &data, (void*)pDevice);
+ bqConfig = ma_loshelf2__get_biquad_config(pConfig);
+ result = ma_biquad_init_preallocated(&bqConfig, pHeap, &pFilter->bq);
if (result != MA_SUCCESS) {
return result;
}
- if (deviceType == ma_device_type_capture) {
- #if defined(MA_APPLE_DESKTOP)
- pDevice->coreaudio.deviceObjectIDCapture = (ma_uint32)data.deviceObjectID;
- #endif
- pDevice->coreaudio.audioUnitCapture = (ma_ptr)data.audioUnit;
- pDevice->coreaudio.pAudioBufferList = (ma_ptr)data.pAudioBufferList;
- pDevice->coreaudio.audioBufferCapInFrames = data.periodSizeInFramesOut;
+ return MA_SUCCESS;
+}
- pDevice->capture.internalFormat = data.formatOut;
- pDevice->capture.internalChannels = data.channelsOut;
- pDevice->capture.internalSampleRate = data.sampleRateOut;
- MA_COPY_MEMORY(pDevice->capture.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut));
- pDevice->capture.internalPeriodSizeInFrames = data.periodSizeInFramesOut;
- pDevice->capture.internalPeriods = data.periodsOut;
- } else if (deviceType == ma_device_type_playback) {
- #if defined(MA_APPLE_DESKTOP)
- pDevice->coreaudio.deviceObjectIDPlayback = (ma_uint32)data.deviceObjectID;
- #endif
- pDevice->coreaudio.audioUnitPlayback = (ma_ptr)data.audioUnit;
+MA_API ma_result ma_loshelf2_init(const ma_loshelf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_loshelf2* pFilter)
+{
+ ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
- pDevice->playback.internalFormat = data.formatOut;
- pDevice->playback.internalChannels = data.channelsOut;
- pDevice->playback.internalSampleRate = data.sampleRateOut;
- MA_COPY_MEMORY(pDevice->playback.internalChannelMap, data.channelMapOut, sizeof(data.channelMapOut));
- pDevice->playback.internalPeriodSizeInFrames = data.periodSizeInFramesOut;
- pDevice->playback.internalPeriods = data.periodsOut;
+ result = ma_loshelf2_get_heap_size(pConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+ } else {
+ pHeap = NULL;
}
+ result = ma_loshelf2_init_preallocated(pConfig, pHeap, pFilter);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
+ }
+
+ pFilter->bq._ownsHeap = MA_TRUE; /* <-- This will cause the biquad to take ownership of the heap and free it when it's uninitialized. */
return MA_SUCCESS;
}
-#endif /* MA_APPLE_DESKTOP */
-static ma_result ma_device_init__coreaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+MA_API void ma_loshelf2_uninit(ma_loshelf2* pFilter, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_result result;
+ if (pFilter == NULL) {
+ return;
+ }
- MA_ASSERT(pDevice != NULL);
- MA_ASSERT(pConfig != NULL);
+ ma_biquad_uninit(&pFilter->bq, pAllocationCallbacks); /* <-- This will free the heap allocation. */
+}
- if (pConfig->deviceType == ma_device_type_loopback) {
- return MA_DEVICE_TYPE_NOT_SUPPORTED;
+MA_API ma_result ma_loshelf2_reinit(const ma_loshelf2_config* pConfig, ma_loshelf2* pFilter)
+{
+ ma_result result;
+ ma_biquad_config bqConfig;
+
+ if (pFilter == NULL || pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- /* No exclusive mode with the Core Audio backend for now. */
- if (((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive) ||
- ((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive)) {
- return MA_SHARE_MODE_NOT_SUPPORTED;
+ bqConfig = ma_loshelf2__get_biquad_config(pConfig);
+ result = ma_biquad_reinit(&bqConfig, &pFilter->bq);
+ if (result != MA_SUCCESS) {
+ return result;
}
- /* Capture needs to be initialized first. */
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
- ma_device_init_internal_data__coreaudio data;
- data.allowNominalSampleRateChange = pConfig->coreaudio.allowNominalSampleRateChange;
- data.formatIn = pDescriptorCapture->format;
- data.channelsIn = pDescriptorCapture->channels;
- data.sampleRateIn = pDescriptorCapture->sampleRate;
- MA_COPY_MEMORY(data.channelMapIn, pDescriptorCapture->channelMap, sizeof(pDescriptorCapture->channelMap));
- data.periodSizeInFramesIn = pDescriptorCapture->periodSizeInFrames;
- data.periodSizeInMillisecondsIn = pDescriptorCapture->periodSizeInMilliseconds;
- data.periodsIn = pDescriptorCapture->periodCount;
- data.shareMode = pDescriptorCapture->shareMode;
- data.performanceProfile = pConfig->performanceProfile;
- data.registerStopEvent = MA_TRUE;
+ return MA_SUCCESS;
+}
- /* Need at least 3 periods for duplex. */
- if (data.periodsIn < 3 && pConfig->deviceType == ma_device_type_duplex) {
- data.periodsIn = 3;
- }
+static MA_INLINE void ma_loshelf2_process_pcm_frame_s16(ma_loshelf2* pFilter, ma_int16* pFrameOut, const ma_int16* pFrameIn)
+{
+ ma_biquad_process_pcm_frame_s16(&pFilter->bq, pFrameOut, pFrameIn);
+}
- result = ma_device_init_internal__coreaudio(pDevice->pContext, ma_device_type_capture, pDescriptorCapture->pDeviceID, &data, (void*)pDevice);
- if (result != MA_SUCCESS) {
- return result;
- }
+static MA_INLINE void ma_loshelf2_process_pcm_frame_f32(ma_loshelf2* pFilter, float* pFrameOut, const float* pFrameIn)
+{
+ ma_biquad_process_pcm_frame_f32(&pFilter->bq, pFrameOut, pFrameIn);
+}
- pDevice->coreaudio.isDefaultCaptureDevice = (pConfig->capture.pDeviceID == NULL);
- #if defined(MA_APPLE_DESKTOP)
- pDevice->coreaudio.deviceObjectIDCapture = (ma_uint32)data.deviceObjectID;
- #endif
- pDevice->coreaudio.audioUnitCapture = (ma_ptr)data.audioUnit;
- pDevice->coreaudio.pAudioBufferList = (ma_ptr)data.pAudioBufferList;
- pDevice->coreaudio.audioBufferCapInFrames = data.periodSizeInFramesOut;
- pDevice->coreaudio.originalPeriodSizeInFrames = pDescriptorCapture->periodSizeInFrames;
- pDevice->coreaudio.originalPeriodSizeInMilliseconds = pDescriptorCapture->periodSizeInMilliseconds;
- pDevice->coreaudio.originalPeriods = pDescriptorCapture->periodCount;
- pDevice->coreaudio.originalPerformanceProfile = pConfig->performanceProfile;
+MA_API ma_result ma_loshelf2_process_pcm_frames(ma_loshelf2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+{
+ if (pFilter == NULL) {
+ return MA_INVALID_ARGS;
+ }
- pDescriptorCapture->format = data.formatOut;
- pDescriptorCapture->channels = data.channelsOut;
- pDescriptorCapture->sampleRate = data.sampleRateOut;
- MA_COPY_MEMORY(pDescriptorCapture->channelMap, data.channelMapOut, sizeof(data.channelMapOut));
- pDescriptorCapture->periodSizeInFrames = data.periodSizeInFramesOut;
- pDescriptorCapture->periodCount = data.periodsOut;
+ return ma_biquad_process_pcm_frames(&pFilter->bq, pFramesOut, pFramesIn, frameCount);
+}
- #if defined(MA_APPLE_DESKTOP)
- /*
- If we are using the default device we'll need to listen for changes to the system's default device so we can seemlessly
- switch the device in the background.
- */
- if (pConfig->capture.pDeviceID == NULL) {
- ma_device__track__coreaudio(pDevice);
- }
- #endif
+MA_API ma_uint32 ma_loshelf2_get_latency(const ma_loshelf2* pFilter)
+{
+ if (pFilter == NULL) {
+ return 0;
}
- /* Playback. */
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- ma_device_init_internal_data__coreaudio data;
- data.allowNominalSampleRateChange = pConfig->coreaudio.allowNominalSampleRateChange;
- data.formatIn = pDescriptorPlayback->format;
- data.channelsIn = pDescriptorPlayback->channels;
- data.sampleRateIn = pDescriptorPlayback->sampleRate;
- MA_COPY_MEMORY(data.channelMapIn, pDescriptorPlayback->channelMap, sizeof(pDescriptorPlayback->channelMap));
- data.shareMode = pDescriptorPlayback->shareMode;
- data.performanceProfile = pConfig->performanceProfile;
+ return ma_biquad_get_latency(&pFilter->bq);
+}
- /* In full-duplex mode we want the playback buffer to be the same size as the capture buffer. */
- if (pConfig->deviceType == ma_device_type_duplex) {
- data.periodSizeInFramesIn = pDescriptorCapture->periodSizeInFrames;
- data.periodsIn = pDescriptorCapture->periodCount;
- data.registerStopEvent = MA_FALSE;
- } else {
- data.periodSizeInFramesIn = pDescriptorPlayback->periodSizeInFrames;
- data.periodSizeInMillisecondsIn = pDescriptorPlayback->periodSizeInMilliseconds;
- data.periodsIn = pDescriptorPlayback->periodCount;
- data.registerStopEvent = MA_TRUE;
- }
- result = ma_device_init_internal__coreaudio(pDevice->pContext, ma_device_type_playback, pDescriptorPlayback->pDeviceID, &data, (void*)pDevice);
- if (result != MA_SUCCESS) {
- if (pConfig->deviceType == ma_device_type_duplex) {
- ((ma_AudioComponentInstanceDispose_proc)pDevice->pContext->coreaudio.AudioComponentInstanceDispose)((AudioUnit)pDevice->coreaudio.audioUnitCapture);
- if (pDevice->coreaudio.pAudioBufferList) {
- ma__free_from_callbacks(pDevice->coreaudio.pAudioBufferList, &pDevice->pContext->allocationCallbacks);
- }
- }
- return result;
- }
+/**************************************************************************************************************************************************************
- pDevice->coreaudio.isDefaultPlaybackDevice = (pConfig->playback.pDeviceID == NULL);
- #if defined(MA_APPLE_DESKTOP)
- pDevice->coreaudio.deviceObjectIDPlayback = (ma_uint32)data.deviceObjectID;
- #endif
- pDevice->coreaudio.audioUnitPlayback = (ma_ptr)data.audioUnit;
- pDevice->coreaudio.originalPeriodSizeInFrames = pDescriptorPlayback->periodSizeInFrames;
- pDevice->coreaudio.originalPeriodSizeInMilliseconds = pDescriptorPlayback->periodSizeInMilliseconds;
- pDevice->coreaudio.originalPeriods = pDescriptorPlayback->periodCount;
- pDevice->coreaudio.originalPerformanceProfile = pConfig->performanceProfile;
+High Shelf Filter
- pDescriptorPlayback->format = data.formatOut;
- pDescriptorPlayback->channels = data.channelsOut;
- pDescriptorPlayback->sampleRate = data.sampleRateOut;
- MA_COPY_MEMORY(pDescriptorPlayback->channelMap, data.channelMapOut, sizeof(data.channelMapOut));
- pDescriptorPlayback->periodSizeInFrames = data.periodSizeInFramesOut;
- pDescriptorPlayback->periodCount = data.periodsOut;
+**************************************************************************************************************************************************************/
+MA_API ma_hishelf2_config ma_hishelf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double shelfSlope, double frequency)
+{
+ ma_hishelf2_config config;
- #if defined(MA_APPLE_DESKTOP)
- /*
- If we are using the default device we'll need to listen for changes to the system's default device so we can seemlessly
- switch the device in the background.
- */
- if (pDescriptorPlayback->pDeviceID == NULL && (pConfig->deviceType != ma_device_type_duplex || pDescriptorCapture->pDeviceID != NULL)) {
- ma_device__track__coreaudio(pDevice);
- }
- #endif
- }
+ MA_ZERO_OBJECT(&config);
+ config.format = format;
+ config.channels = channels;
+ config.sampleRate = sampleRate;
+ config.gainDB = gainDB;
+ config.shelfSlope = shelfSlope;
+ config.frequency = frequency;
+ return config;
+}
- /*
- When stopping the device, a callback is called on another thread. We need to wait for this callback
- before returning from ma_device_stop(). This event is used for this.
- */
- ma_event_init(&pDevice->coreaudio.stopEvent);
+static MA_INLINE ma_biquad_config ma_hishelf2__get_biquad_config(const ma_hishelf2_config* pConfig)
+{
+ ma_biquad_config bqConfig;
+ double w;
+ double s;
+ double c;
+ double A;
+ double S;
+ double a;
+ double sqrtA;
- /*
- We need to detect when a route has changed so we can update the data conversion pipeline accordingly. This is done
- differently on non-Desktop Apple platforms.
- */
-#if defined(MA_APPLE_MOBILE)
- pDevice->coreaudio.pRouteChangeHandler = (__bridge_retained void*)[[ma_router_change_handler alloc] init:pDevice];
-#endif
+ MA_ASSERT(pConfig != NULL);
- return MA_SUCCESS;
+ w = 2 * MA_PI_D * pConfig->frequency / pConfig->sampleRate;
+ s = ma_sind(w);
+ c = ma_cosd(w);
+ A = ma_powd(10, (pConfig->gainDB / 40));
+ S = pConfig->shelfSlope;
+ a = s/2 * ma_sqrtd((A + 1/A) * (1/S - 1) + 2);
+ sqrtA = 2*ma_sqrtd(A)*a;
+
+ bqConfig.b0 = A * ((A + 1) + (A - 1)*c + sqrtA);
+ bqConfig.b1 = -2 * A * ((A - 1) + (A + 1)*c);
+ bqConfig.b2 = A * ((A + 1) + (A - 1)*c - sqrtA);
+ bqConfig.a0 = (A + 1) - (A - 1)*c + sqrtA;
+ bqConfig.a1 = 2 * ((A - 1) - (A + 1)*c);
+ bqConfig.a2 = (A + 1) - (A - 1)*c - sqrtA;
+
+ bqConfig.format = pConfig->format;
+ bqConfig.channels = pConfig->channels;
+
+ return bqConfig;
}
+MA_API ma_result ma_hishelf2_get_heap_size(const ma_hishelf2_config* pConfig, size_t* pHeapSizeInBytes)
+{
+ ma_biquad_config bqConfig;
+ bqConfig = ma_hishelf2__get_biquad_config(pConfig);
+
+ return ma_biquad_get_heap_size(&bqConfig, pHeapSizeInBytes);
+}
-static ma_result ma_device_start__coreaudio(ma_device* pDevice)
+MA_API ma_result ma_hishelf2_init_preallocated(const ma_hishelf2_config* pConfig, void* pHeap, ma_hishelf2* pFilter)
{
- MA_ASSERT(pDevice != NULL);
+ ma_result result;
+ ma_biquad_config bqConfig;
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- OSStatus status = ((ma_AudioOutputUnitStart_proc)pDevice->pContext->coreaudio.AudioOutputUnitStart)((AudioUnit)pDevice->coreaudio.audioUnitCapture);
- if (status != noErr) {
- return ma_result_from_OSStatus(status);
- }
+ if (pFilter == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- OSStatus status = ((ma_AudioOutputUnitStart_proc)pDevice->pContext->coreaudio.AudioOutputUnitStart)((AudioUnit)pDevice->coreaudio.audioUnitPlayback);
- if (status != noErr) {
- if (pDevice->type == ma_device_type_duplex) {
- ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitCapture);
- }
- return ma_result_from_OSStatus(status);
- }
+ MA_ZERO_OBJECT(pFilter);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ bqConfig = ma_hishelf2__get_biquad_config(pConfig);
+ result = ma_biquad_init_preallocated(&bqConfig, pHeap, &pFilter->bq);
+ if (result != MA_SUCCESS) {
+ return result;
}
return MA_SUCCESS;
}
-static ma_result ma_device_stop__coreaudio(ma_device* pDevice)
+MA_API ma_result ma_hishelf2_init(const ma_hishelf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hishelf2* pFilter)
{
- MA_ASSERT(pDevice != NULL);
+ ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
- /* It's not clear from the documentation whether or not AudioOutputUnitStop() actually drains the device or not. */
+ result = ma_hishelf2_get_heap_size(pConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- OSStatus status = ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitCapture);
- if (status != noErr) {
- return ma_result_from_OSStatus(status);
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
}
+ } else {
+ pHeap = NULL;
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- OSStatus status = ((ma_AudioOutputUnitStop_proc)pDevice->pContext->coreaudio.AudioOutputUnitStop)((AudioUnit)pDevice->coreaudio.audioUnitPlayback);
- if (status != noErr) {
- return ma_result_from_OSStatus(status);
- }
+ result = ma_hishelf2_init_preallocated(pConfig, pHeap, pFilter);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
}
- /* We need to wait for the callback to finish before returning. */
- ma_event_wait(&pDevice->coreaudio.stopEvent);
+ pFilter->bq._ownsHeap = MA_TRUE; /* <-- This will cause the biquad to take ownership of the heap and free it when it's uninitialized. */
return MA_SUCCESS;
}
+MA_API void ma_hishelf2_uninit(ma_hishelf2* pFilter, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pFilter == NULL) {
+ return;
+ }
+
+ ma_biquad_uninit(&pFilter->bq, pAllocationCallbacks); /* <-- This will free the heap allocation. */
+}
-static ma_result ma_context_uninit__coreaudio(ma_context* pContext)
+MA_API ma_result ma_hishelf2_reinit(const ma_hishelf2_config* pConfig, ma_hishelf2* pFilter)
{
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pContext->backend == ma_backend_coreaudio);
+ ma_result result;
+ ma_biquad_config bqConfig;
-#if defined(MA_APPLE_MOBILE)
- if (!pContext->coreaudio.noAudioSessionDeactivate) {
- if (![[AVAudioSession sharedInstance] setActive:false error:nil]) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "Failed to deactivate audio session.", MA_FAILED_TO_INIT_BACKEND);
- }
+ if (pFilter == NULL || pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
-#endif
-
-#if !defined(MA_NO_RUNTIME_LINKING) && !defined(MA_APPLE_MOBILE)
- ma_dlclose(pContext, pContext->coreaudio.hAudioUnit);
- ma_dlclose(pContext, pContext->coreaudio.hCoreAudio);
- ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation);
-#endif
-#if !defined(MA_APPLE_MOBILE)
- ma_context__uninit_device_tracking__coreaudio(pContext);
-#endif
+ bqConfig = ma_hishelf2__get_biquad_config(pConfig);
+ result = ma_biquad_reinit(&bqConfig, &pFilter->bq);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- (void)pContext;
return MA_SUCCESS;
}
-#if defined(MA_APPLE_MOBILE)
-static AVAudioSessionCategory ma_to_AVAudioSessionCategory(ma_ios_session_category category)
+static MA_INLINE void ma_hishelf2_process_pcm_frame_s16(ma_hishelf2* pFilter, ma_int16* pFrameOut, const ma_int16* pFrameIn)
{
- /* The "default" and "none" categories are treated different and should not be used as an input into this function. */
- MA_ASSERT(category != ma_ios_session_category_default);
- MA_ASSERT(category != ma_ios_session_category_none);
+ ma_biquad_process_pcm_frame_s16(&pFilter->bq, pFrameOut, pFrameIn);
+}
- switch (category) {
- case ma_ios_session_category_ambient: return AVAudioSessionCategoryAmbient;
- case ma_ios_session_category_solo_ambient: return AVAudioSessionCategorySoloAmbient;
- case ma_ios_session_category_playback: return AVAudioSessionCategoryPlayback;
- case ma_ios_session_category_record: return AVAudioSessionCategoryRecord;
- case ma_ios_session_category_play_and_record: return AVAudioSessionCategoryPlayAndRecord;
- case ma_ios_session_category_multi_route: return AVAudioSessionCategoryMultiRoute;
- case ma_ios_session_category_none: return AVAudioSessionCategoryAmbient;
- case ma_ios_session_category_default: return AVAudioSessionCategoryAmbient;
- default: return AVAudioSessionCategoryAmbient;
+static MA_INLINE void ma_hishelf2_process_pcm_frame_f32(ma_hishelf2* pFilter, float* pFrameOut, const float* pFrameIn)
+{
+ ma_biquad_process_pcm_frame_f32(&pFilter->bq, pFrameOut, pFrameIn);
+}
+
+MA_API ma_result ma_hishelf2_process_pcm_frames(ma_hishelf2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+{
+ if (pFilter == NULL) {
+ return MA_INVALID_ARGS;
}
+
+ return ma_biquad_process_pcm_frames(&pFilter->bq, pFramesOut, pFramesIn, frameCount);
}
-#endif
-static ma_result ma_context_init__coreaudio(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+MA_API ma_uint32 ma_hishelf2_get_latency(const ma_hishelf2* pFilter)
{
-#if !defined(MA_APPLE_MOBILE)
- ma_result result;
-#endif
+ if (pFilter == NULL) {
+ return 0;
+ }
- MA_ASSERT(pConfig != NULL);
- MA_ASSERT(pContext != NULL);
+ return ma_biquad_get_latency(&pFilter->bq);
+}
-#if defined(MA_APPLE_MOBILE)
- @autoreleasepool {
- AVAudioSession* pAudioSession = [AVAudioSession sharedInstance];
- AVAudioSessionCategoryOptions options = pConfig->coreaudio.sessionCategoryOptions;
- MA_ASSERT(pAudioSession != NULL);
- if (pConfig->coreaudio.sessionCategory == ma_ios_session_category_default) {
- /*
- I'm going to use trial and error to determine our default session category. First we'll try PlayAndRecord. If that fails
- we'll try Playback and if that fails we'll try record. If all of these fail we'll just not set the category.
- */
- #if !defined(MA_APPLE_TV) && !defined(MA_APPLE_WATCH)
- options |= AVAudioSessionCategoryOptionDefaultToSpeaker;
- #endif
+/*
+Delay
+*/
+MA_API ma_delay_config ma_delay_config_init(ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 delayInFrames, float decay)
+{
+ ma_delay_config config;
- if ([pAudioSession setCategory: AVAudioSessionCategoryPlayAndRecord withOptions:options error:nil]) {
- /* Using PlayAndRecord */
- } else if ([pAudioSession setCategory: AVAudioSessionCategoryPlayback withOptions:options error:nil]) {
- /* Using Playback */
- } else if ([pAudioSession setCategory: AVAudioSessionCategoryRecord withOptions:options error:nil]) {
- /* Using Record */
- } else {
- /* Leave as default? */
- }
- } else {
- if (pConfig->coreaudio.sessionCategory != ma_ios_session_category_none) {
- if (![pAudioSession setCategory: ma_to_AVAudioSessionCategory(pConfig->coreaudio.sessionCategory) withOptions:options error:nil]) {
- return MA_INVALID_OPERATION; /* Failed to set session category. */
- }
- }
- }
+ MA_ZERO_OBJECT(&config);
+ config.channels = channels;
+ config.sampleRate = sampleRate;
+ config.delayInFrames = delayInFrames;
+ config.delayStart = (decay == 0) ? MA_TRUE : MA_FALSE; /* Delay the start if it looks like we're not configuring an echo. */
+ config.wet = 1;
+ config.dry = 1;
+ config.decay = decay;
- if (!pConfig->coreaudio.noAudioSessionActivate) {
- if (![pAudioSession setActive:true error:nil]) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "Failed to activate audio session.", MA_FAILED_TO_INIT_BACKEND);
- }
- }
+ return config;
+}
+
+
+MA_API ma_result ma_delay_init(const ma_delay_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_delay* pDelay)
+{
+ if (pDelay == NULL) {
+ return MA_INVALID_ARGS;
}
-#endif
-#if !defined(MA_NO_RUNTIME_LINKING) && !defined(MA_APPLE_MOBILE)
- pContext->coreaudio.hCoreFoundation = ma_dlopen(pContext, "CoreFoundation.framework/CoreFoundation");
- if (pContext->coreaudio.hCoreFoundation == NULL) {
- return MA_API_NOT_FOUND;
+ MA_ZERO_OBJECT(pDelay);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- pContext->coreaudio.CFStringGetCString = ma_dlsym(pContext, pContext->coreaudio.hCoreFoundation, "CFStringGetCString");
- pContext->coreaudio.CFRelease = ma_dlsym(pContext, pContext->coreaudio.hCoreFoundation, "CFRelease");
+ if (pConfig->decay < 0 || pConfig->decay > 1) {
+ return MA_INVALID_ARGS;
+ }
+ pDelay->config = *pConfig;
+ pDelay->bufferSizeInFrames = pConfig->delayInFrames;
+ pDelay->cursor = 0;
- pContext->coreaudio.hCoreAudio = ma_dlopen(pContext, "CoreAudio.framework/CoreAudio");
- if (pContext->coreaudio.hCoreAudio == NULL) {
- ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation);
- return MA_API_NOT_FOUND;
+ pDelay->pBuffer = (float*)ma_malloc((size_t)(pDelay->bufferSizeInFrames * ma_get_bytes_per_frame(ma_format_f32, pConfig->channels)), pAllocationCallbacks);
+ if (pDelay->pBuffer == NULL) {
+ return MA_OUT_OF_MEMORY;
}
- pContext->coreaudio.AudioObjectGetPropertyData = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectGetPropertyData");
- pContext->coreaudio.AudioObjectGetPropertyDataSize = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectGetPropertyDataSize");
- pContext->coreaudio.AudioObjectSetPropertyData = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectSetPropertyData");
- pContext->coreaudio.AudioObjectAddPropertyListener = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectAddPropertyListener");
- pContext->coreaudio.AudioObjectRemovePropertyListener = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectRemovePropertyListener");
+ ma_silence_pcm_frames(pDelay->pBuffer, pDelay->bufferSizeInFrames, ma_format_f32, pConfig->channels);
- /*
- It looks like Apple has moved some APIs from AudioUnit into AudioToolbox on more recent versions of macOS. They are still
- defined in AudioUnit, but just in case they decide to remove them from there entirely I'm going to implement a fallback.
- The way it'll work is that it'll first try AudioUnit, and if the required symbols are not present there we'll fall back to
- AudioToolbox.
- */
- pContext->coreaudio.hAudioUnit = ma_dlopen(pContext, "AudioUnit.framework/AudioUnit");
- if (pContext->coreaudio.hAudioUnit == NULL) {
- ma_dlclose(pContext, pContext->coreaudio.hCoreAudio);
- ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation);
- return MA_API_NOT_FOUND;
+ return MA_SUCCESS;
+}
+
+MA_API void ma_delay_uninit(ma_delay* pDelay, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pDelay == NULL) {
+ return;
}
- if (ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioComponentFindNext") == NULL) {
- /* Couldn't find the required symbols in AudioUnit, so fall back to AudioToolbox. */
- ma_dlclose(pContext, pContext->coreaudio.hAudioUnit);
- pContext->coreaudio.hAudioUnit = ma_dlopen(pContext, "AudioToolbox.framework/AudioToolbox");
- if (pContext->coreaudio.hAudioUnit == NULL) {
- ma_dlclose(pContext, pContext->coreaudio.hCoreAudio);
- ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation);
- return MA_API_NOT_FOUND;
- }
+ ma_free(pDelay->pBuffer, pAllocationCallbacks);
+}
+
+MA_API ma_result ma_delay_process_pcm_frames(ma_delay* pDelay, void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount)
+{
+ ma_uint32 iFrame;
+ ma_uint32 iChannel;
+ float* pFramesOutF32 = (float*)pFramesOut;
+ const float* pFramesInF32 = (const float*)pFramesIn;
+
+ if (pDelay == NULL || pFramesOut == NULL || pFramesIn == NULL) {
+ return MA_INVALID_ARGS;
}
- pContext->coreaudio.AudioComponentFindNext = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioComponentFindNext");
- pContext->coreaudio.AudioComponentInstanceDispose = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioComponentInstanceDispose");
- pContext->coreaudio.AudioComponentInstanceNew = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioComponentInstanceNew");
- pContext->coreaudio.AudioOutputUnitStart = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioOutputUnitStart");
- pContext->coreaudio.AudioOutputUnitStop = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioOutputUnitStop");
- pContext->coreaudio.AudioUnitAddPropertyListener = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitAddPropertyListener");
- pContext->coreaudio.AudioUnitGetPropertyInfo = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitGetPropertyInfo");
- pContext->coreaudio.AudioUnitGetProperty = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitGetProperty");
- pContext->coreaudio.AudioUnitSetProperty = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitSetProperty");
- pContext->coreaudio.AudioUnitInitialize = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitInitialize");
- pContext->coreaudio.AudioUnitRender = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitRender");
-#else
- pContext->coreaudio.CFStringGetCString = (ma_proc)CFStringGetCString;
- pContext->coreaudio.CFRelease = (ma_proc)CFRelease;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannel = 0; iChannel < pDelay->config.channels; iChannel += 1) {
+ ma_uint32 iBuffer = (pDelay->cursor * pDelay->config.channels) + iChannel;
- #if defined(MA_APPLE_DESKTOP)
- pContext->coreaudio.AudioObjectGetPropertyData = (ma_proc)AudioObjectGetPropertyData;
- pContext->coreaudio.AudioObjectGetPropertyDataSize = (ma_proc)AudioObjectGetPropertyDataSize;
- pContext->coreaudio.AudioObjectSetPropertyData = (ma_proc)AudioObjectSetPropertyData;
- pContext->coreaudio.AudioObjectAddPropertyListener = (ma_proc)AudioObjectAddPropertyListener;
- pContext->coreaudio.AudioObjectRemovePropertyListener = (ma_proc)AudioObjectRemovePropertyListener;
- #endif
+ if (pDelay->config.delayStart) {
+ /* Delayed start. */
+
+ /* Read */
+ pFramesOutF32[iChannel] = pDelay->pBuffer[iBuffer] * pDelay->config.wet;
- pContext->coreaudio.AudioComponentFindNext = (ma_proc)AudioComponentFindNext;
- pContext->coreaudio.AudioComponentInstanceDispose = (ma_proc)AudioComponentInstanceDispose;
- pContext->coreaudio.AudioComponentInstanceNew = (ma_proc)AudioComponentInstanceNew;
- pContext->coreaudio.AudioOutputUnitStart = (ma_proc)AudioOutputUnitStart;
- pContext->coreaudio.AudioOutputUnitStop = (ma_proc)AudioOutputUnitStop;
- pContext->coreaudio.AudioUnitAddPropertyListener = (ma_proc)AudioUnitAddPropertyListener;
- pContext->coreaudio.AudioUnitGetPropertyInfo = (ma_proc)AudioUnitGetPropertyInfo;
- pContext->coreaudio.AudioUnitGetProperty = (ma_proc)AudioUnitGetProperty;
- pContext->coreaudio.AudioUnitSetProperty = (ma_proc)AudioUnitSetProperty;
- pContext->coreaudio.AudioUnitInitialize = (ma_proc)AudioUnitInitialize;
- pContext->coreaudio.AudioUnitRender = (ma_proc)AudioUnitRender;
-#endif
+ /* Feedback */
+ pDelay->pBuffer[iBuffer] = (pDelay->pBuffer[iBuffer] * pDelay->config.decay) + (pFramesInF32[iChannel] * pDelay->config.dry);
+ } else {
+ /* Immediate start */
- /* Audio component. */
- {
- AudioComponentDescription desc;
- desc.componentType = kAudioUnitType_Output;
- #if defined(MA_APPLE_DESKTOP)
- desc.componentSubType = kAudioUnitSubType_HALOutput;
- #else
- desc.componentSubType = kAudioUnitSubType_RemoteIO;
- #endif
- desc.componentManufacturer = kAudioUnitManufacturer_Apple;
- desc.componentFlags = 0;
- desc.componentFlagsMask = 0;
+ /* Feedback */
+ pDelay->pBuffer[iBuffer] = (pDelay->pBuffer[iBuffer] * pDelay->config.decay) + (pFramesInF32[iChannel] * pDelay->config.dry);
- pContext->coreaudio.component = ((ma_AudioComponentFindNext_proc)pContext->coreaudio.AudioComponentFindNext)(NULL, &desc);
- if (pContext->coreaudio.component == NULL) {
- #if !defined(MA_NO_RUNTIME_LINKING) && !defined(MA_APPLE_MOBILE)
- ma_dlclose(pContext, pContext->coreaudio.hAudioUnit);
- ma_dlclose(pContext, pContext->coreaudio.hCoreAudio);
- ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation);
- #endif
- return MA_FAILED_TO_INIT_BACKEND;
+ /* Read */
+ pFramesOutF32[iChannel] = pDelay->pBuffer[iBuffer] * pDelay->config.wet;
+ }
}
- }
-#if !defined(MA_APPLE_MOBILE)
- result = ma_context__init_device_tracking__coreaudio(pContext);
- if (result != MA_SUCCESS) {
- #if !defined(MA_NO_RUNTIME_LINKING) && !defined(MA_APPLE_MOBILE)
- ma_dlclose(pContext, pContext->coreaudio.hAudioUnit);
- ma_dlclose(pContext, pContext->coreaudio.hCoreAudio);
- ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation);
- #endif
- return result;
+ pDelay->cursor = (pDelay->cursor + 1) % pDelay->bufferSizeInFrames;
+
+ pFramesOutF32 += pDelay->config.channels;
+ pFramesInF32 += pDelay->config.channels;
}
-#endif
- pContext->coreaudio.noAudioSessionDeactivate = pConfig->coreaudio.noAudioSessionDeactivate;
+ return MA_SUCCESS;
+}
- pCallbacks->onContextInit = ma_context_init__coreaudio;
- pCallbacks->onContextUninit = ma_context_uninit__coreaudio;
- pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__coreaudio;
- pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__coreaudio;
- pCallbacks->onDeviceInit = ma_device_init__coreaudio;
- pCallbacks->onDeviceUninit = ma_device_uninit__coreaudio;
- pCallbacks->onDeviceStart = ma_device_start__coreaudio;
- pCallbacks->onDeviceStop = ma_device_stop__coreaudio;
- pCallbacks->onDeviceRead = NULL;
- pCallbacks->onDeviceWrite = NULL;
- pCallbacks->onDeviceDataLoop = NULL;
+MA_API void ma_delay_set_wet(ma_delay* pDelay, float value)
+{
+ if (pDelay == NULL) {
+ return;
+ }
- return MA_SUCCESS;
+ pDelay->config.wet = value;
}
-#endif /* Core Audio */
+MA_API float ma_delay_get_wet(const ma_delay* pDelay)
+{
+ if (pDelay == NULL) {
+ return 0;
+ }
+ return pDelay->config.wet;
+}
-/******************************************************************************
+MA_API void ma_delay_set_dry(ma_delay* pDelay, float value)
+{
+ if (pDelay == NULL) {
+ return;
+ }
-sndio Backend
+ pDelay->config.dry = value;
+}
-******************************************************************************/
-#ifdef MA_HAS_SNDIO
-#include
+MA_API float ma_delay_get_dry(const ma_delay* pDelay)
+{
+ if (pDelay == NULL) {
+ return 0;
+ }
-/*
-Only supporting OpenBSD. This did not work very well at all on FreeBSD when I tried it. Not sure if this is due
-to miniaudio's implementation or if it's some kind of system configuration issue, but basically the default device
-just doesn't emit any sound, or at times you'll hear tiny pieces. I will consider enabling this when there's
-demand for it or if I can get it tested and debugged more thoroughly.
-*/
-#if 0
-#if defined(__NetBSD__) || defined(__OpenBSD__)
-#include
-#endif
-#if defined(__FreeBSD__) || defined(__DragonFly__)
-#include
-#endif
-#endif
+ return pDelay->config.dry;
+}
-#define MA_SIO_DEVANY "default"
-#define MA_SIO_PLAY 1
-#define MA_SIO_REC 2
-#define MA_SIO_NENC 8
-#define MA_SIO_NCHAN 8
-#define MA_SIO_NRATE 16
-#define MA_SIO_NCONF 4
+MA_API void ma_delay_set_decay(ma_delay* pDelay, float value)
+{
+ if (pDelay == NULL) {
+ return;
+ }
-struct ma_sio_hdl; /* <-- Opaque */
+ pDelay->config.decay = value;
+}
-struct ma_sio_par
+MA_API float ma_delay_get_decay(const ma_delay* pDelay)
{
- unsigned int bits;
- unsigned int bps;
- unsigned int sig;
- unsigned int le;
- unsigned int msb;
- unsigned int rchan;
- unsigned int pchan;
- unsigned int rate;
- unsigned int bufsz;
- unsigned int xrun;
- unsigned int round;
- unsigned int appbufsz;
- int __pad[3];
- unsigned int __magic;
-};
+ if (pDelay == NULL) {
+ return 0;
+ }
-struct ma_sio_enc
+ return pDelay->config.decay;
+}
+
+
+MA_API ma_gainer_config ma_gainer_config_init(ma_uint32 channels, ma_uint32 smoothTimeInFrames)
{
- unsigned int bits;
- unsigned int bps;
- unsigned int sig;
- unsigned int le;
- unsigned int msb;
-};
+ ma_gainer_config config;
-struct ma_sio_conf
+ MA_ZERO_OBJECT(&config);
+ config.channels = channels;
+ config.smoothTimeInFrames = smoothTimeInFrames;
+
+ return config;
+}
+
+
+typedef struct
{
- unsigned int enc;
- unsigned int rchan;
- unsigned int pchan;
- unsigned int rate;
-};
+ size_t sizeInBytes;
+ size_t oldGainsOffset;
+ size_t newGainsOffset;
+} ma_gainer_heap_layout;
-struct ma_sio_cap
+static ma_result ma_gainer_get_heap_layout(const ma_gainer_config* pConfig, ma_gainer_heap_layout* pHeapLayout)
{
- struct ma_sio_enc enc[MA_SIO_NENC];
- unsigned int rchan[MA_SIO_NCHAN];
- unsigned int pchan[MA_SIO_NCHAN];
- unsigned int rate[MA_SIO_NRATE];
- int __pad[7];
- unsigned int nconf;
- struct ma_sio_conf confs[MA_SIO_NCONF];
-};
+ MA_ASSERT(pHeapLayout != NULL);
-typedef struct ma_sio_hdl* (* ma_sio_open_proc) (const char*, unsigned int, int);
-typedef void (* ma_sio_close_proc) (struct ma_sio_hdl*);
-typedef int (* ma_sio_setpar_proc) (struct ma_sio_hdl*, struct ma_sio_par*);
-typedef int (* ma_sio_getpar_proc) (struct ma_sio_hdl*, struct ma_sio_par*);
-typedef int (* ma_sio_getcap_proc) (struct ma_sio_hdl*, struct ma_sio_cap*);
-typedef size_t (* ma_sio_write_proc) (struct ma_sio_hdl*, const void*, size_t);
-typedef size_t (* ma_sio_read_proc) (struct ma_sio_hdl*, void*, size_t);
-typedef int (* ma_sio_start_proc) (struct ma_sio_hdl*);
-typedef int (* ma_sio_stop_proc) (struct ma_sio_hdl*);
-typedef int (* ma_sio_initpar_proc)(struct ma_sio_par*);
+ MA_ZERO_OBJECT(pHeapLayout);
-static ma_uint32 ma_get_standard_sample_rate_priority_index__sndio(ma_uint32 sampleRate) /* Lower = higher priority */
-{
- ma_uint32 i;
- for (i = 0; i < ma_countof(g_maStandardSampleRatePriorities); ++i) {
- if (g_maStandardSampleRatePriorities[i] == sampleRate) {
- return i;
- }
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- return (ma_uint32)-1;
+ if (pConfig->channels == 0) {
+ return MA_INVALID_ARGS;
+ }
+
+ pHeapLayout->sizeInBytes = 0;
+
+ /* Old gains. */
+ pHeapLayout->oldGainsOffset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += sizeof(float) * pConfig->channels;
+
+ /* New gains. */
+ pHeapLayout->newGainsOffset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += sizeof(float) * pConfig->channels;
+
+ /* Alignment. */
+ pHeapLayout->sizeInBytes = ma_align_64(pHeapLayout->sizeInBytes);
+
+ return MA_SUCCESS;
}
-static ma_format ma_format_from_sio_enc__sndio(unsigned int bits, unsigned int bps, unsigned int sig, unsigned int le, unsigned int msb)
+
+MA_API ma_result ma_gainer_get_heap_size(const ma_gainer_config* pConfig, size_t* pHeapSizeInBytes)
{
- /* We only support native-endian right now. */
- if ((ma_is_little_endian() && le == 0) || (ma_is_big_endian() && le == 1)) {
- return ma_format_unknown;
+ ma_result result;
+ ma_gainer_heap_layout heapLayout;
+
+ if (pHeapSizeInBytes == NULL) {
+ return MA_INVALID_ARGS;
}
- if (bits == 8 && bps == 1 && sig == 0) {
- return ma_format_u8;
+ *pHeapSizeInBytes = 0;
+
+ result = ma_gainer_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return MA_INVALID_ARGS;
}
- if (bits == 16 && bps == 2 && sig == 1) {
- return ma_format_s16;
+
+ *pHeapSizeInBytes = heapLayout.sizeInBytes;
+
+ return MA_SUCCESS;
+}
+
+
+MA_API ma_result ma_gainer_init_preallocated(const ma_gainer_config* pConfig, void* pHeap, ma_gainer* pGainer)
+{
+ ma_result result;
+ ma_gainer_heap_layout heapLayout;
+ ma_uint32 iChannel;
+
+ if (pGainer == NULL) {
+ return MA_INVALID_ARGS;
}
- if (bits == 24 && bps == 3 && sig == 1) {
- return ma_format_s24;
+
+ MA_ZERO_OBJECT(pGainer);
+
+ if (pConfig == NULL || pHeap == NULL) {
+ return MA_INVALID_ARGS;
}
- if (bits == 24 && bps == 4 && sig == 1 && msb == 0) {
- /*return ma_format_s24_32;*/
+
+ result = ma_gainer_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
}
- if (bits == 32 && bps == 4 && sig == 1) {
- return ma_format_s32;
+
+ pGainer->_pHeap = pHeap;
+ MA_ZERO_MEMORY(pHeap, heapLayout.sizeInBytes);
+
+ pGainer->pOldGains = (float*)ma_offset_ptr(pHeap, heapLayout.oldGainsOffset);
+ pGainer->pNewGains = (float*)ma_offset_ptr(pHeap, heapLayout.newGainsOffset);
+
+ pGainer->config = *pConfig;
+ pGainer->t = (ma_uint32)-1; /* No interpolation by default. */
+
+ for (iChannel = 0; iChannel < pConfig->channels; iChannel += 1) {
+ pGainer->pOldGains[iChannel] = 1;
+ pGainer->pNewGains[iChannel] = 1;
}
- return ma_format_unknown;
+ return MA_SUCCESS;
}
-static ma_format ma_find_best_format_from_sio_cap__sndio(struct ma_sio_cap* caps)
+MA_API ma_result ma_gainer_init(const ma_gainer_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_gainer* pGainer)
{
- ma_format bestFormat;
- unsigned int iConfig;
+ ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
- MA_ASSERT(caps != NULL);
+ result = ma_gainer_get_heap_size(pConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to retrieve the size of the heap allocation. */
+ }
- bestFormat = ma_format_unknown;
- for (iConfig = 0; iConfig < caps->nconf; iConfig += 1) {
- unsigned int iEncoding;
- for (iEncoding = 0; iEncoding < MA_SIO_NENC; iEncoding += 1) {
- unsigned int bits;
- unsigned int bps;
- unsigned int sig;
- unsigned int le;
- unsigned int msb;
- ma_format format;
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+ } else {
+ pHeap = NULL;
+ }
- if ((caps->confs[iConfig].enc & (1UL << iEncoding)) == 0) {
- continue;
- }
+ result = ma_gainer_init_preallocated(pConfig, pHeap, pGainer);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
+ }
- bits = caps->enc[iEncoding].bits;
- bps = caps->enc[iEncoding].bps;
- sig = caps->enc[iEncoding].sig;
- le = caps->enc[iEncoding].le;
- msb = caps->enc[iEncoding].msb;
- format = ma_format_from_sio_enc__sndio(bits, bps, sig, le, msb);
- if (format == ma_format_unknown) {
- continue; /* Format not supported. */
- }
+ pGainer->_ownsHeap = MA_TRUE;
+ return MA_SUCCESS;
+}
- if (bestFormat == ma_format_unknown) {
- bestFormat = format;
- } else {
- if (ma_get_format_priority_index(bestFormat) > ma_get_format_priority_index(format)) { /* <-- Lower = better. */
- bestFormat = format;
- }
- }
- }
+MA_API void ma_gainer_uninit(ma_gainer* pGainer, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pGainer == NULL) {
+ return;
}
- return bestFormat;
+ if (pGainer->_ownsHeap) {
+ ma_free(pGainer->_pHeap, pAllocationCallbacks);
+ }
}
-static ma_uint32 ma_find_best_channels_from_sio_cap__sndio(struct ma_sio_cap* caps, ma_device_type deviceType, ma_format requiredFormat)
+static float ma_gainer_calculate_current_gain(const ma_gainer* pGainer, ma_uint32 channel)
{
- ma_uint32 maxChannels;
- unsigned int iConfig;
+ float a = (float)pGainer->t / pGainer->config.smoothTimeInFrames;
+ return ma_mix_f32_fast(pGainer->pOldGains[channel], pGainer->pNewGains[channel], a);
+}
- MA_ASSERT(caps != NULL);
- MA_ASSERT(requiredFormat != ma_format_unknown);
+MA_API ma_result ma_gainer_process_pcm_frames(ma_gainer* pGainer, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+{
+ ma_uint64 iFrame;
+ ma_uint32 iChannel;
+ float* pFramesOutF32 = (float*)pFramesOut;
+ const float* pFramesInF32 = (const float*)pFramesIn;
- /* Just pick whatever configuration has the most channels. */
- maxChannels = 0;
- for (iConfig = 0; iConfig < caps->nconf; iConfig += 1) {
- /* The encoding should be of requiredFormat. */
- unsigned int iEncoding;
- for (iEncoding = 0; iEncoding < MA_SIO_NENC; iEncoding += 1) {
- unsigned int iChannel;
- unsigned int bits;
- unsigned int bps;
- unsigned int sig;
- unsigned int le;
- unsigned int msb;
- ma_format format;
+ if (pGainer == NULL) {
+ return MA_INVALID_ARGS;
+ }
- if ((caps->confs[iConfig].enc & (1UL << iEncoding)) == 0) {
- continue;
- }
+ if (pGainer->t >= pGainer->config.smoothTimeInFrames) {
+ /* Fast path. No gain calculation required. */
+ ma_copy_and_apply_volume_factor_per_channel_f32(pFramesOutF32, pFramesInF32, frameCount, pGainer->config.channels, pGainer->pNewGains);
- bits = caps->enc[iEncoding].bits;
- bps = caps->enc[iEncoding].bps;
- sig = caps->enc[iEncoding].sig;
- le = caps->enc[iEncoding].le;
- msb = caps->enc[iEncoding].msb;
- format = ma_format_from_sio_enc__sndio(bits, bps, sig, le, msb);
- if (format != requiredFormat) {
- continue;
- }
+ /* Now that some frames have been processed we need to make sure future changes to the gain are interpolated. */
+ if (pGainer->t == (ma_uint32)-1) {
+ pGainer->t = pGainer->config.smoothTimeInFrames;
+ }
+ } else {
+ /* Slow path. Need to interpolate the gain for each channel individually. */
- /* Getting here means the format is supported. Iterate over each channel count and grab the biggest one. */
- for (iChannel = 0; iChannel < MA_SIO_NCHAN; iChannel += 1) {
- unsigned int chan = 0;
- unsigned int channels;
+ /* We can allow the input and output buffers to be null in which case we'll just update the internal timer. */
+ if (pFramesOut != NULL && pFramesIn != NULL) {
+ float a = (float)pGainer->t / pGainer->config.smoothTimeInFrames;
+ float d = 1.0f / pGainer->config.smoothTimeInFrames;
+ ma_uint32 channelCount = pGainer->config.channels;
- if (deviceType == ma_device_type_playback) {
- chan = caps->confs[iConfig].pchan;
- } else {
- chan = caps->confs[iConfig].rchan;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannel = 0; iChannel < channelCount; iChannel += 1) {
+ pFramesOutF32[iChannel] = pFramesInF32[iChannel] * ma_mix_f32_fast(pGainer->pOldGains[iChannel], pGainer->pNewGains[iChannel], a);
}
- if ((chan & (1UL << iChannel)) == 0) {
- continue;
- }
+ pFramesOutF32 += channelCount;
+ pFramesInF32 += channelCount;
- if (deviceType == ma_device_type_playback) {
- channels = caps->pchan[iChannel];
- } else {
- channels = caps->rchan[iChannel];
+ a += d;
+ if (a > 1) {
+ a = 1;
}
+ }
+ }
- if (maxChannels < channels) {
- maxChannels = channels;
+ pGainer->t = (ma_uint32)ma_min(pGainer->t + frameCount, pGainer->config.smoothTimeInFrames);
+
+ #if 0 /* Reference implementation. */
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ /* We can allow the input and output buffers to be null in which case we'll just update the internal timer. */
+ if (pFramesOut != NULL && pFramesIn != NULL) {
+ for (iChannel = 0; iChannel < pGainer->config.channels; iChannel += 1) {
+ pFramesOutF32[iFrame*pGainer->config.channels + iChannel] = pFramesInF32[iFrame*pGainer->config.channels + iChannel] * ma_gainer_calculate_current_gain(pGainer, iChannel);
}
}
+
+ /* Move interpolation time forward, but don't go beyond our smoothing time. */
+ pGainer->t = ma_min(pGainer->t + 1, pGainer->config.smoothTimeInFrames);
}
+ #endif
}
- return maxChannels;
+ return MA_SUCCESS;
}
-static ma_uint32 ma_find_best_sample_rate_from_sio_cap__sndio(struct ma_sio_cap* caps, ma_device_type deviceType, ma_format requiredFormat, ma_uint32 requiredChannels)
+static void ma_gainer_set_gain_by_index(ma_gainer* pGainer, float newGain, ma_uint32 iChannel)
{
- ma_uint32 firstSampleRate;
- ma_uint32 bestSampleRate;
- unsigned int iConfig;
+ pGainer->pOldGains[iChannel] = ma_gainer_calculate_current_gain(pGainer, iChannel);
+ pGainer->pNewGains[iChannel] = newGain;
+}
- MA_ASSERT(caps != NULL);
- MA_ASSERT(requiredFormat != ma_format_unknown);
- MA_ASSERT(requiredChannels > 0);
- MA_ASSERT(requiredChannels <= MA_MAX_CHANNELS);
+static void ma_gainer_reset_smoothing_time(ma_gainer* pGainer)
+{
+ if (pGainer->t == (ma_uint32)-1) {
+ pGainer->t = pGainer->config.smoothTimeInFrames; /* No smoothing required for initial gains setting. */
+ } else {
+ pGainer->t = 0;
+ }
+}
- firstSampleRate = 0; /* <-- If the device does not support a standard rate we'll fall back to the first one that's found. */
- bestSampleRate = 0;
+MA_API ma_result ma_gainer_set_gain(ma_gainer* pGainer, float newGain)
+{
+ ma_uint32 iChannel;
- for (iConfig = 0; iConfig < caps->nconf; iConfig += 1) {
- /* The encoding should be of requiredFormat. */
- unsigned int iEncoding;
- for (iEncoding = 0; iEncoding < MA_SIO_NENC; iEncoding += 1) {
- unsigned int iChannel;
- unsigned int bits;
- unsigned int bps;
- unsigned int sig;
- unsigned int le;
- unsigned int msb;
- ma_format format;
+ if (pGainer == NULL) {
+ return MA_INVALID_ARGS;
+ }
- if ((caps->confs[iConfig].enc & (1UL << iEncoding)) == 0) {
- continue;
- }
+ for (iChannel = 0; iChannel < pGainer->config.channels; iChannel += 1) {
+ ma_gainer_set_gain_by_index(pGainer, newGain, iChannel);
+ }
- bits = caps->enc[iEncoding].bits;
- bps = caps->enc[iEncoding].bps;
- sig = caps->enc[iEncoding].sig;
- le = caps->enc[iEncoding].le;
- msb = caps->enc[iEncoding].msb;
- format = ma_format_from_sio_enc__sndio(bits, bps, sig, le, msb);
- if (format != requiredFormat) {
- continue;
- }
+ /* The smoothing time needs to be reset to ensure we always interpolate by the configured smoothing time, but only if it's not the first setting. */
+ ma_gainer_reset_smoothing_time(pGainer);
- /* Getting here means the format is supported. Iterate over each channel count and grab the biggest one. */
- for (iChannel = 0; iChannel < MA_SIO_NCHAN; iChannel += 1) {
- unsigned int chan = 0;
- unsigned int channels;
- unsigned int iRate;
+ return MA_SUCCESS;
+}
- if (deviceType == ma_device_type_playback) {
- chan = caps->confs[iConfig].pchan;
- } else {
- chan = caps->confs[iConfig].rchan;
- }
+MA_API ma_result ma_gainer_set_gains(ma_gainer* pGainer, float* pNewGains)
+{
+ ma_uint32 iChannel;
- if ((chan & (1UL << iChannel)) == 0) {
- continue;
- }
+ if (pGainer == NULL || pNewGains == NULL) {
+ return MA_INVALID_ARGS;
+ }
- if (deviceType == ma_device_type_playback) {
- channels = caps->pchan[iChannel];
- } else {
- channels = caps->rchan[iChannel];
- }
+ for (iChannel = 0; iChannel < pGainer->config.channels; iChannel += 1) {
+ ma_gainer_set_gain_by_index(pGainer, pNewGains[iChannel], iChannel);
+ }
- if (channels != requiredChannels) {
- continue;
- }
+ /* The smoothing time needs to be reset to ensure we always interpolate by the configured smoothing time, but only if it's not the first setting. */
+ ma_gainer_reset_smoothing_time(pGainer);
- /* Getting here means we have found a compatible encoding/channel pair. */
- for (iRate = 0; iRate < MA_SIO_NRATE; iRate += 1) {
- ma_uint32 rate = (ma_uint32)caps->rate[iRate];
- ma_uint32 ratePriority;
+ return MA_SUCCESS;
+}
- if (firstSampleRate == 0) {
- firstSampleRate = rate;
- }
- /* Disregard this rate if it's not a standard one. */
- ratePriority = ma_get_standard_sample_rate_priority_index__sndio(rate);
- if (ratePriority == (ma_uint32)-1) {
- continue;
- }
+MA_API ma_panner_config ma_panner_config_init(ma_format format, ma_uint32 channels)
+{
+ ma_panner_config config;
- if (ma_get_standard_sample_rate_priority_index__sndio(bestSampleRate) > ratePriority) { /* Lower = better. */
- bestSampleRate = rate;
- }
- }
+ MA_ZERO_OBJECT(&config);
+ config.format = format;
+ config.channels = channels;
+ config.mode = ma_pan_mode_balance; /* Set to balancing mode by default because it's consistent with other audio engines and most likely what the caller is expecting. */
+ config.pan = 0;
+
+ return config;
+}
+
+
+MA_API ma_result ma_panner_init(const ma_panner_config* pConfig, ma_panner* pPanner)
+{
+ if (pPanner == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ MA_ZERO_OBJECT(pPanner);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ pPanner->format = pConfig->format;
+ pPanner->channels = pConfig->channels;
+ pPanner->mode = pConfig->mode;
+ pPanner->pan = pConfig->pan;
+
+ return MA_SUCCESS;
+}
+
+static void ma_stereo_balance_pcm_frames_f32(float* pFramesOut, const float* pFramesIn, ma_uint64 frameCount, float pan)
+{
+ ma_uint64 iFrame;
+
+ if (pan > 0) {
+ float factor = 1.0f - pan;
+ if (pFramesOut == pFramesIn) {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ pFramesOut[iFrame*2 + 0] = pFramesIn[iFrame*2 + 0] * factor;
+ }
+ } else {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ pFramesOut[iFrame*2 + 0] = pFramesIn[iFrame*2 + 0] * factor;
+ pFramesOut[iFrame*2 + 1] = pFramesIn[iFrame*2 + 1];
+ }
+ }
+ } else {
+ float factor = 1.0f + pan;
+ if (pFramesOut == pFramesIn) {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ pFramesOut[iFrame*2 + 1] = pFramesIn[iFrame*2 + 1] * factor;
+ }
+ } else {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ pFramesOut[iFrame*2 + 0] = pFramesIn[iFrame*2 + 0];
+ pFramesOut[iFrame*2 + 1] = pFramesIn[iFrame*2 + 1] * factor;
}
}
}
+}
- /* If a standard sample rate was not found just fall back to the first one that was iterated. */
- if (bestSampleRate == 0) {
- bestSampleRate = firstSampleRate;
+static void ma_stereo_balance_pcm_frames(void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount, ma_format format, float pan)
+{
+ if (pan == 0) {
+ /* Fast path. No panning required. */
+ if (pFramesOut == pFramesIn) {
+ /* No-op */
+ } else {
+ ma_copy_pcm_frames(pFramesOut, pFramesIn, frameCount, format, 2);
+ }
+
+ return;
}
- return bestSampleRate;
+ switch (format) {
+ case ma_format_f32: ma_stereo_balance_pcm_frames_f32((float*)pFramesOut, (float*)pFramesIn, frameCount, pan); break;
+
+ /* Unknown format. Just copy. */
+ default:
+ {
+ ma_copy_pcm_frames(pFramesOut, pFramesIn, frameCount, format, 2);
+ } break;
+ }
}
-static ma_result ma_context_enumerate_devices__sndio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+static void ma_stereo_pan_pcm_frames_f32(float* pFramesOut, const float* pFramesIn, ma_uint64 frameCount, float pan)
{
- ma_bool32 isTerminating = MA_FALSE;
- struct ma_sio_hdl* handle;
+ ma_uint64 iFrame;
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(callback != NULL);
+ if (pan > 0) {
+ float factorL0 = 1.0f - pan;
+ float factorL1 = 0.0f + pan;
- /* sndio doesn't seem to have a good device enumeration API, so I'm therefore only enumerating over default devices for now. */
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ float sample0 = (pFramesIn[iFrame*2 + 0] * factorL0);
+ float sample1 = (pFramesIn[iFrame*2 + 0] * factorL1) + pFramesIn[iFrame*2 + 1];
- /* Playback. */
- if (!isTerminating) {
- handle = ((ma_sio_open_proc)pContext->sndio.sio_open)(MA_SIO_DEVANY, MA_SIO_PLAY, 0);
- if (handle != NULL) {
- /* Supports playback. */
- ma_device_info deviceInfo;
- MA_ZERO_OBJECT(&deviceInfo);
- ma_strcpy_s(deviceInfo.id.sndio, sizeof(deviceInfo.id.sndio), MA_SIO_DEVANY);
- ma_strcpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME);
+ pFramesOut[iFrame*2 + 0] = sample0;
+ pFramesOut[iFrame*2 + 1] = sample1;
+ }
+ } else {
+ float factorR0 = 0.0f - pan;
+ float factorR1 = 1.0f + pan;
- isTerminating = !callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ float sample0 = pFramesIn[iFrame*2 + 0] + (pFramesIn[iFrame*2 + 1] * factorR0);
+ float sample1 = (pFramesIn[iFrame*2 + 1] * factorR1);
- ((ma_sio_close_proc)pContext->sndio.sio_close)(handle);
+ pFramesOut[iFrame*2 + 0] = sample0;
+ pFramesOut[iFrame*2 + 1] = sample1;
+ }
+ }
+}
+
+static void ma_stereo_pan_pcm_frames(void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount, ma_format format, float pan)
+{
+ if (pan == 0) {
+ /* Fast path. No panning required. */
+ if (pFramesOut == pFramesIn) {
+ /* No-op */
+ } else {
+ ma_copy_pcm_frames(pFramesOut, pFramesIn, frameCount, format, 2);
}
+
+ return;
}
- /* Capture. */
- if (!isTerminating) {
- handle = ((ma_sio_open_proc)pContext->sndio.sio_open)(MA_SIO_DEVANY, MA_SIO_REC, 0);
- if (handle != NULL) {
- /* Supports capture. */
- ma_device_info deviceInfo;
- MA_ZERO_OBJECT(&deviceInfo);
- ma_strcpy_s(deviceInfo.id.sndio, sizeof(deviceInfo.id.sndio), "default");
- ma_strcpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME);
+ switch (format) {
+ case ma_format_f32: ma_stereo_pan_pcm_frames_f32((float*)pFramesOut, (float*)pFramesIn, frameCount, pan); break;
- isTerminating = !callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
+ /* Unknown format. Just copy. */
+ default:
+ {
+ ma_copy_pcm_frames(pFramesOut, pFramesIn, frameCount, format, 2);
+ } break;
+ }
+}
- ((ma_sio_close_proc)pContext->sndio.sio_close)(handle);
+MA_API ma_result ma_panner_process_pcm_frames(ma_panner* pPanner, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+{
+ if (pPanner == NULL || pFramesOut == NULL || pFramesIn == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (pPanner->channels == 2) {
+ /* Stereo case. For now assume channel 0 is left and channel right is 1, but should probably add support for a channel map. */
+ if (pPanner->mode == ma_pan_mode_balance) {
+ ma_stereo_balance_pcm_frames(pFramesOut, pFramesIn, frameCount, pPanner->format, pPanner->pan);
+ } else {
+ ma_stereo_pan_pcm_frames(pFramesOut, pFramesIn, frameCount, pPanner->format, pPanner->pan);
+ }
+ } else {
+ if (pPanner->channels == 1) {
+ /* Panning has no effect on mono streams. */
+ ma_copy_pcm_frames(pFramesOut, pFramesIn, frameCount, pPanner->format, pPanner->channels);
+ } else {
+ /* For now we're not going to support non-stereo set ups. Not sure how I want to handle this case just yet. */
+ ma_copy_pcm_frames(pFramesOut, pFramesIn, frameCount, pPanner->format, pPanner->channels);
}
}
return MA_SUCCESS;
}
-static ma_result ma_context_get_device_info__sndio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+MA_API void ma_panner_set_mode(ma_panner* pPanner, ma_pan_mode mode)
{
- char devid[256];
- struct ma_sio_hdl* handle;
- struct ma_sio_cap caps;
- unsigned int iConfig;
+ if (pPanner == NULL) {
+ return;
+ }
- MA_ASSERT(pContext != NULL);
+ pPanner->mode = mode;
+}
- /* We need to open the device before we can get information about it. */
- if (pDeviceID == NULL) {
- ma_strcpy_s(devid, sizeof(devid), MA_SIO_DEVANY);
- ma_strcpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), (deviceType == ma_device_type_playback) ? MA_DEFAULT_PLAYBACK_DEVICE_NAME : MA_DEFAULT_CAPTURE_DEVICE_NAME);
- } else {
- ma_strcpy_s(devid, sizeof(devid), pDeviceID->sndio);
- ma_strcpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), devid);
+MA_API ma_pan_mode ma_panner_get_mode(const ma_panner* pPanner)
+{
+ if (pPanner == NULL) {
+ return ma_pan_mode_balance;
}
- handle = ((ma_sio_open_proc)pContext->sndio.sio_open)(devid, (deviceType == ma_device_type_playback) ? MA_SIO_PLAY : MA_SIO_REC, 0);
- if (handle == NULL) {
- return MA_NO_DEVICE;
+ return pPanner->mode;
+}
+
+MA_API void ma_panner_set_pan(ma_panner* pPanner, float pan)
+{
+ if (pPanner == NULL) {
+ return;
}
- if (((ma_sio_getcap_proc)pContext->sndio.sio_getcap)(handle, &caps) == 0) {
- return MA_ERROR;
+ pPanner->pan = ma_clamp(pan, -1.0f, 1.0f);
+}
+
+MA_API float ma_panner_get_pan(const ma_panner* pPanner)
+{
+ if (pPanner == NULL) {
+ return 0;
}
- pDeviceInfo->nativeDataFormatCount = 0;
+ return pPanner->pan;
+}
- for (iConfig = 0; iConfig < caps.nconf; iConfig += 1) {
- /*
- The main thing we care about is that the encoding is supported by miniaudio. If it is, we want to give
- preference to some formats over others.
- */
- unsigned int iEncoding;
- unsigned int iChannel;
- unsigned int iRate;
- for (iEncoding = 0; iEncoding < MA_SIO_NENC; iEncoding += 1) {
- unsigned int bits;
- unsigned int bps;
- unsigned int sig;
- unsigned int le;
- unsigned int msb;
- ma_format format;
- if ((caps.confs[iConfig].enc & (1UL << iEncoding)) == 0) {
- continue;
- }
- bits = caps.enc[iEncoding].bits;
- bps = caps.enc[iEncoding].bps;
- sig = caps.enc[iEncoding].sig;
- le = caps.enc[iEncoding].le;
- msb = caps.enc[iEncoding].msb;
- format = ma_format_from_sio_enc__sndio(bits, bps, sig, le, msb);
- if (format == ma_format_unknown) {
- continue; /* Format not supported. */
- }
+MA_API ma_fader_config ma_fader_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate)
+{
+ ma_fader_config config;
+ MA_ZERO_OBJECT(&config);
+ config.format = format;
+ config.channels = channels;
+ config.sampleRate = sampleRate;
- /* Channels. */
- for (iChannel = 0; iChannel < MA_SIO_NCHAN; iChannel += 1) {
- unsigned int chan = 0;
- unsigned int channels;
+ return config;
+}
- if (deviceType == ma_device_type_playback) {
- chan = caps.confs[iConfig].pchan;
- } else {
- chan = caps.confs[iConfig].rchan;
- }
- if ((chan & (1UL << iChannel)) == 0) {
- continue;
- }
+MA_API ma_result ma_fader_init(const ma_fader_config* pConfig, ma_fader* pFader)
+{
+ if (pFader == NULL) {
+ return MA_INVALID_ARGS;
+ }
- if (deviceType == ma_device_type_playback) {
- channels = caps.pchan[iChannel];
- } else {
- channels = caps.rchan[iChannel];
- }
+ MA_ZERO_OBJECT(pFader);
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
- /* Sample Rates. */
- for (iRate = 0; iRate < MA_SIO_NRATE; iRate += 1) {
- if ((caps.confs[iConfig].rate & (1UL << iRate)) != 0) {
- ma_device_info_add_native_data_format(pDeviceInfo, format, channels, caps.rate[iRate], 0);
- }
- }
- }
- }
+ /* Only f32 is supported for now. */
+ if (pConfig->format != ma_format_f32) {
+ return MA_INVALID_ARGS;
}
- ((ma_sio_close_proc)pContext->sndio.sio_close)(handle);
+ pFader->config = *pConfig;
+ pFader->volumeBeg = 1;
+ pFader->volumeEnd = 1;
+ pFader->lengthInFrames = 0;
+ pFader->cursorInFrames = 0;
+
return MA_SUCCESS;
}
-static ma_result ma_device_uninit__sndio(ma_device* pDevice)
+MA_API ma_result ma_fader_process_pcm_frames(ma_fader* pFader, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
- MA_ASSERT(pDevice != NULL);
+ if (pFader == NULL) {
+ return MA_INVALID_ARGS;
+ }
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- ((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)pDevice->sndio.handleCapture);
+ /*
+ For now we need to clamp frameCount so that the cursor never overflows 32-bits. This is required for
+ the conversion to a float which we use for the linear interpolation. This might be changed later.
+ */
+ if (frameCount + pFader->cursorInFrames > UINT_MAX) {
+ frameCount = UINT_MAX - pFader->cursorInFrames;
}
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- ((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback);
+ /* Optimized path if volumeBeg and volumeEnd are equal. */
+ if (pFader->volumeBeg == pFader->volumeEnd) {
+ if (pFader->volumeBeg == 1) {
+ /* Straight copy. */
+ ma_copy_pcm_frames(pFramesOut, pFramesIn, frameCount, pFader->config.format, pFader->config.channels);
+ } else {
+ /* Copy with volume. */
+ ma_copy_and_apply_volume_and_clip_pcm_frames(pFramesOut, pFramesIn, frameCount, pFader->config.format, pFader->config.channels, pFader->volumeEnd);
+ }
+ } else {
+ /* Slower path. Volumes are different, so may need to do an interpolation. */
+ if (pFader->cursorInFrames >= pFader->lengthInFrames) {
+ /* Fast path. We've gone past the end of the fade period so just apply the end volume to all samples. */
+ ma_copy_and_apply_volume_and_clip_pcm_frames(pFramesOut, pFramesIn, frameCount, pFader->config.format, pFader->config.channels, pFader->volumeEnd);
+ } else {
+ /* Slow path. This is where we do the actual fading. */
+ ma_uint64 iFrame;
+ ma_uint32 iChannel;
+
+ /* For now we only support f32. Support for other formats will be added later. */
+ if (pFader->config.format == ma_format_f32) {
+ const float* pFramesInF32 = (const float*)pFramesIn;
+ /* */ float* pFramesOutF32 = ( float*)pFramesOut;
+
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ float a = (ma_uint32)ma_min(pFader->cursorInFrames + iFrame, pFader->lengthInFrames) / (float)((ma_uint32)pFader->lengthInFrames); /* Safe cast due to the frameCount clamp at the top of this function. */
+ float volume = ma_mix_f32_fast(pFader->volumeBeg, pFader->volumeEnd, a);
+
+ for (iChannel = 0; iChannel < pFader->config.channels; iChannel += 1) {
+ pFramesOutF32[iFrame*pFader->config.channels + iChannel] = pFramesInF32[iFrame*pFader->config.channels + iChannel] * volume;
+ }
+ }
+ } else {
+ return MA_NOT_IMPLEMENTED;
+ }
+ }
}
+ pFader->cursorInFrames += frameCount;
+
return MA_SUCCESS;
}
-static ma_result ma_device_init_handle__sndio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptor, ma_device_type deviceType)
+MA_API void ma_fader_get_data_format(const ma_fader* pFader, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate)
{
- const char* pDeviceName;
- ma_ptr handle;
- int openFlags = 0;
- struct ma_sio_cap caps;
- struct ma_sio_par par;
- const ma_device_id* pDeviceID;
- ma_format format;
- ma_uint32 channels;
- ma_uint32 sampleRate;
- ma_format internalFormat;
- ma_uint32 internalChannels;
- ma_uint32 internalSampleRate;
- ma_uint32 internalPeriodSizeInFrames;
- ma_uint32 internalPeriods;
-
- MA_ASSERT(pConfig != NULL);
- MA_ASSERT(deviceType != ma_device_type_duplex);
- MA_ASSERT(pDevice != NULL);
+ if (pFader == NULL) {
+ return;
+ }
- if (deviceType == ma_device_type_capture) {
- openFlags = MA_SIO_REC;
- } else {
- openFlags = MA_SIO_PLAY;
+ if (pFormat != NULL) {
+ *pFormat = pFader->config.format;
}
- pDeviceID = pDescriptor->pDeviceID;
- format = pDescriptor->format;
- channels = pDescriptor->channels;
- sampleRate = pDescriptor->sampleRate;
+ if (pChannels != NULL) {
+ *pChannels = pFader->config.channels;
+ }
- pDeviceName = MA_SIO_DEVANY;
- if (pDeviceID != NULL) {
- pDeviceName = pDeviceID->sndio;
+ if (pSampleRate != NULL) {
+ *pSampleRate = pFader->config.sampleRate;
}
+}
- handle = (ma_ptr)((ma_sio_open_proc)pDevice->pContext->sndio.sio_open)(pDeviceName, openFlags, 0);
- if (handle == NULL) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[sndio] Failed to open device.", MA_FAILED_TO_OPEN_BACKEND_DEVICE);
+MA_API void ma_fader_set_fade(ma_fader* pFader, float volumeBeg, float volumeEnd, ma_uint64 lengthInFrames)
+{
+ if (pFader == NULL) {
+ return;
}
- /* We need to retrieve the device caps to determine the most appropriate format to use. */
- if (((ma_sio_getcap_proc)pDevice->pContext->sndio.sio_getcap)((struct ma_sio_hdl*)handle, &caps) == 0) {
- ((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)handle);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[sndio] Failed to retrieve device caps.", MA_ERROR);
+ /* If the volume is negative, use current volume. */
+ if (volumeBeg < 0) {
+ volumeBeg = ma_fader_get_current_volume(pFader);
}
/*
- Note: sndio reports a huge range of available channels. This is inconvenient for us because there's no real
- way, as far as I can tell, to get the _actual_ channel count of the device. I'm therefore restricting this
- to the requested channels, regardless of whether or not the default channel count is requested.
-
- For hardware devices, I'm suspecting only a single channel count will be reported and we can safely use the
- value returned by ma_find_best_channels_from_sio_cap__sndio().
+ The length needs to be clamped to 32-bits due to how we convert it to a float for linear
+ interpolation reasons. I might change this requirement later, but for now it's not important.
*/
- if (deviceType == ma_device_type_capture) {
- if (format == ma_format_unknown) {
- format = ma_find_best_format_from_sio_cap__sndio(&caps);
- }
+ if (lengthInFrames > UINT_MAX) {
+ lengthInFrames = UINT_MAX;
+ }
- if (channels == 0) {
- if (strlen(pDeviceName) > strlen("rsnd/") && strncmp(pDeviceName, "rsnd/", strlen("rsnd/")) == 0) {
- channels = ma_find_best_channels_from_sio_cap__sndio(&caps, deviceType, format);
- } else {
- channels = MA_DEFAULT_CHANNELS;
- }
- }
- } else {
- if (format == ma_format_unknown) {
- format = ma_find_best_format_from_sio_cap__sndio(&caps);
- }
+ pFader->volumeBeg = volumeBeg;
+ pFader->volumeEnd = volumeEnd;
+ pFader->lengthInFrames = lengthInFrames;
+ pFader->cursorInFrames = 0; /* Reset cursor. */
+}
- if (channels == 0) {
- if (strlen(pDeviceName) > strlen("rsnd/") && strncmp(pDeviceName, "rsnd/", strlen("rsnd/")) == 0) {
- channels = ma_find_best_channels_from_sio_cap__sndio(&caps, deviceType, format);
- } else {
- channels = MA_DEFAULT_CHANNELS;
- }
- }
+MA_API float ma_fader_get_current_volume(ma_fader* pFader)
+{
+ if (pFader == NULL) {
+ return 0.0f;
}
- if (sampleRate == 0) {
- sampleRate = ma_find_best_sample_rate_from_sio_cap__sndio(&caps, pConfig->deviceType, format, channels);
+ /* The current volume depends on the position of the cursor. */
+ if (pFader->cursorInFrames == 0) {
+ return pFader->volumeBeg;
+ } else if (pFader->cursorInFrames >= pFader->lengthInFrames) {
+ return pFader->volumeEnd;
+ } else {
+ /* The cursor is somewhere inside the fading period. We can figure this out with a simple linear interpoluation between volumeBeg and volumeEnd based on our cursor position. */
+ return ma_mix_f32_fast(pFader->volumeBeg, pFader->volumeEnd, (ma_uint32)pFader->cursorInFrames / (float)((ma_uint32)pFader->lengthInFrames)); /* Safe cast to uint32 because we clamp it in ma_fader_process_pcm_frames(). */
}
+}
- ((ma_sio_initpar_proc)pDevice->pContext->sndio.sio_initpar)(&par);
- par.msb = 0;
- par.le = ma_is_little_endian();
- switch (format) {
- case ma_format_u8:
- {
- par.bits = 8;
- par.bps = 1;
- par.sig = 0;
- } break;
- case ma_format_s24:
- {
- par.bits = 24;
- par.bps = 3;
- par.sig = 1;
- } break;
- case ma_format_s32:
- {
- par.bits = 32;
- par.bps = 4;
- par.sig = 1;
- } break;
+MA_API ma_vec3f ma_vec3f_init_3f(float x, float y, float z)
+{
+ ma_vec3f v;
- case ma_format_s16:
- case ma_format_f32:
- case ma_format_unknown:
- default:
- {
- par.bits = 16;
- par.bps = 2;
- par.sig = 1;
- } break;
- }
+ v.x = x;
+ v.y = y;
+ v.z = z;
- if (deviceType == ma_device_type_capture) {
- par.rchan = channels;
- } else {
- par.pchan = channels;
- }
+ return v;
+}
- par.rate = sampleRate;
+MA_API ma_vec3f ma_vec3f_sub(ma_vec3f a, ma_vec3f b)
+{
+ return ma_vec3f_init_3f(
+ a.x - b.x,
+ a.y - b.y,
+ a.z - b.z
+ );
+}
- internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, par.rate, pConfig->performanceProfile);
+MA_API ma_vec3f ma_vec3f_neg(ma_vec3f a)
+{
+ return ma_vec3f_init_3f(
+ -a.x,
+ -a.y,
+ -a.z
+ );
+}
- par.round = internalPeriodSizeInFrames;
- par.appbufsz = par.round * pDescriptor->periodCount;
+MA_API float ma_vec3f_dot(ma_vec3f a, ma_vec3f b)
+{
+ return a.x*b.x + a.y*b.y + a.z*b.z;
+}
- if (((ma_sio_setpar_proc)pDevice->pContext->sndio.sio_setpar)((struct ma_sio_hdl*)handle, &par) == 0) {
- ((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)handle);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[sndio] Failed to set buffer size.", MA_FORMAT_NOT_SUPPORTED);
- }
+MA_API float ma_vec3f_len2(ma_vec3f v)
+{
+ return ma_vec3f_dot(v, v);
+}
- if (((ma_sio_getpar_proc)pDevice->pContext->sndio.sio_getpar)((struct ma_sio_hdl*)handle, &par) == 0) {
- ((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)handle);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[sndio] Failed to retrieve buffer size.", MA_FORMAT_NOT_SUPPORTED);
- }
+MA_API float ma_vec3f_len(ma_vec3f v)
+{
+ return (float)ma_sqrtd(ma_vec3f_len2(v));
+}
- internalFormat = ma_format_from_sio_enc__sndio(par.bits, par.bps, par.sig, par.le, par.msb);
- internalChannels = (deviceType == ma_device_type_capture) ? par.rchan : par.pchan;
- internalSampleRate = par.rate;
- internalPeriods = par.appbufsz / par.round;
- internalPeriodSizeInFrames = par.round;
+MA_API float ma_vec3f_dist(ma_vec3f a, ma_vec3f b)
+{
+ return ma_vec3f_len(ma_vec3f_sub(a, b));
+}
- if (deviceType == ma_device_type_capture) {
- pDevice->sndio.handleCapture = handle;
- } else {
- pDevice->sndio.handlePlayback = handle;
+MA_API ma_vec3f ma_vec3f_normalize(ma_vec3f v)
+{
+ float f;
+ float l = ma_vec3f_len(v);
+ if (l == 0) {
+ return ma_vec3f_init_3f(0, 0, 0);
}
- pDescriptor->format = internalFormat;
- pDescriptor->channels = internalChannels;
- pDescriptor->sampleRate = internalSampleRate;
- ma_get_standard_channel_map(ma_standard_channel_map_sndio, pDevice->playback.internalChannels, pDevice->playback.internalChannelMap);
- pDescriptor->periodSizeInFrames = internalPeriodSizeInFrames;
- pDescriptor->periodCount = internalPeriods;
-
- #ifdef MA_DEBUG_OUTPUT
- {
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "DEVICE INFO\n");
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " Format: %s\n", ma_get_format_name(internalFormat));
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " Channels: %d\n", internalChannels);
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " Sample Rate: %d\n", internalSampleRate);
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " Period Size: %d\n", internalPeriodSizeInFrames);
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " Periods: %d\n", internalPeriods);
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " appbufsz: %d\n", par.appbufsz);
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " round: %d\n", par.round);
- }
- #endif
+ f = 1 / l;
+ v.x *= f;
+ v.y *= f;
+ v.z *= f;
- return MA_SUCCESS;
+ return v;
}
-static ma_result ma_device_init__sndio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+MA_API ma_vec3f ma_vec3f_cross(ma_vec3f a, ma_vec3f b)
{
- MA_ASSERT(pDevice != NULL);
+ return ma_vec3f_init_3f(
+ a.y*b.z - a.z*b.y,
+ a.z*b.x - a.x*b.z,
+ a.x*b.y - a.y*b.x
+ );
+}
- MA_ZERO_OBJECT(&pDevice->sndio);
- if (pConfig->deviceType == ma_device_type_loopback) {
- return MA_DEVICE_TYPE_NOT_SUPPORTED;
- }
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
- ma_result result = ma_device_init_handle__sndio(pDevice, pConfig, pDescriptorCapture, ma_device_type_capture);
- if (result != MA_SUCCESS) {
- return result;
- }
+static void ma_channel_map_apply_f32(float* pFramesOut, const ma_channel* pChannelMapOut, ma_uint32 channelsOut, const float* pFramesIn, const ma_channel* pChannelMapIn, ma_uint32 channelsIn, ma_uint64 frameCount, ma_channel_mix_mode mode, ma_mono_expansion_mode monoExpansionMode);
+static ma_bool32 ma_is_spatial_channel_position(ma_channel channelPosition);
+
+
+#ifndef MA_DEFAULT_SPEED_OF_SOUND
+#define MA_DEFAULT_SPEED_OF_SOUND 343.3f
+#endif
+
+/*
+These vectors represent the direction that speakers are facing from the center point. They're used
+for panning in the spatializer. Must be normalized.
+*/
+static ma_vec3f g_maChannelDirections[MA_CHANNEL_POSITION_COUNT] = {
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_NONE */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_MONO */
+ {-0.7071f, 0.0f, -0.7071f }, /* MA_CHANNEL_FRONT_LEFT */
+ {+0.7071f, 0.0f, -0.7071f }, /* MA_CHANNEL_FRONT_RIGHT */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_FRONT_CENTER */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_LFE */
+ {-0.7071f, 0.0f, +0.7071f }, /* MA_CHANNEL_BACK_LEFT */
+ {+0.7071f, 0.0f, +0.7071f }, /* MA_CHANNEL_BACK_RIGHT */
+ {-0.3162f, 0.0f, -0.9487f }, /* MA_CHANNEL_FRONT_LEFT_CENTER */
+ {+0.3162f, 0.0f, -0.9487f }, /* MA_CHANNEL_FRONT_RIGHT_CENTER */
+ { 0.0f, 0.0f, +1.0f }, /* MA_CHANNEL_BACK_CENTER */
+ {-1.0f, 0.0f, 0.0f }, /* MA_CHANNEL_SIDE_LEFT */
+ {+1.0f, 0.0f, 0.0f }, /* MA_CHANNEL_SIDE_RIGHT */
+ { 0.0f, +1.0f, 0.0f }, /* MA_CHANNEL_TOP_CENTER */
+ {-0.5774f, +0.5774f, -0.5774f }, /* MA_CHANNEL_TOP_FRONT_LEFT */
+ { 0.0f, +0.7071f, -0.7071f }, /* MA_CHANNEL_TOP_FRONT_CENTER */
+ {+0.5774f, +0.5774f, -0.5774f }, /* MA_CHANNEL_TOP_FRONT_RIGHT */
+ {-0.5774f, +0.5774f, +0.5774f }, /* MA_CHANNEL_TOP_BACK_LEFT */
+ { 0.0f, +0.7071f, +0.7071f }, /* MA_CHANNEL_TOP_BACK_CENTER */
+ {+0.5774f, +0.5774f, +0.5774f }, /* MA_CHANNEL_TOP_BACK_RIGHT */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_0 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_1 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_2 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_3 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_4 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_5 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_6 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_7 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_8 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_9 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_10 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_11 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_12 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_13 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_14 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_15 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_16 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_17 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_18 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_19 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_20 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_21 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_22 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_23 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_24 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_25 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_26 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_27 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_28 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_29 */
+ { 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_30 */
+ { 0.0f, 0.0f, -1.0f } /* MA_CHANNEL_AUX_31 */
+};
+
+static ma_vec3f ma_get_channel_direction(ma_channel channel)
+{
+ if (channel >= MA_CHANNEL_POSITION_COUNT) {
+ return ma_vec3f_init_3f(0, 0, -1);
+ } else {
+ return g_maChannelDirections[channel];
}
+}
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- ma_result result = ma_device_init_handle__sndio(pDevice, pConfig, pDescriptorPlayback, ma_device_type_playback);
- if (result != MA_SUCCESS) {
- return result;
- }
+
+
+static float ma_attenuation_inverse(float distance, float minDistance, float maxDistance, float rolloff)
+{
+ if (minDistance >= maxDistance) {
+ return 1; /* To avoid division by zero. Do not attenuate. */
}
- return MA_SUCCESS;
+ return minDistance / (minDistance + rolloff * (ma_clamp(distance, minDistance, maxDistance) - minDistance));
}
-static ma_result ma_device_start__sndio(ma_device* pDevice)
+static float ma_attenuation_linear(float distance, float minDistance, float maxDistance, float rolloff)
{
- MA_ASSERT(pDevice != NULL);
-
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- ((ma_sio_start_proc)pDevice->pContext->sndio.sio_start)((struct ma_sio_hdl*)pDevice->sndio.handleCapture);
+ if (minDistance >= maxDistance) {
+ return 1; /* To avoid division by zero. Do not attenuate. */
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- ((ma_sio_start_proc)pDevice->pContext->sndio.sio_start)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback); /* <-- Doesn't actually playback until data is written. */
+ return 1 - rolloff * (ma_clamp(distance, minDistance, maxDistance) - minDistance) / (maxDistance - minDistance);
+}
+
+static float ma_attenuation_exponential(float distance, float minDistance, float maxDistance, float rolloff)
+{
+ if (minDistance >= maxDistance) {
+ return 1; /* To avoid division by zero. Do not attenuate. */
}
- return MA_SUCCESS;
+ return (float)ma_powd(ma_clamp(distance, minDistance, maxDistance) / minDistance, -rolloff);
}
-static ma_result ma_device_stop__sndio(ma_device* pDevice)
-{
- MA_ASSERT(pDevice != NULL);
- /*
- From the documentation:
+/*
+Dopper Effect calculation taken from the OpenAL spec, with two main differences:
- The sio_stop() function puts the audio subsystem in the same state as before sio_start() is called. It stops recording, drains the play buffer and then
- stops playback. If samples to play are queued but playback hasn't started yet then playback is forced immediately; playback will actually stop once the
- buffer is drained. In no case are samples in the play buffer discarded.
+ 1) The source to listener vector will have already been calcualted at an earlier step so we can
+ just use that directly. We need only the position of the source relative to the origin.
- Therefore, sio_stop() performs all of the necessary draining for us.
- */
+ 2) We don't scale by a frequency because we actually just want the ratio which we'll plug straight
+ into the resampler directly.
+*/
+static float ma_doppler_pitch(ma_vec3f relativePosition, ma_vec3f sourceVelocity, ma_vec3f listenVelocity, float speedOfSound, float dopplerFactor)
+{
+ float len;
+ float vls;
+ float vss;
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- ((ma_sio_stop_proc)pDevice->pContext->sndio.sio_stop)((struct ma_sio_hdl*)pDevice->sndio.handleCapture);
- }
+ len = ma_vec3f_len(relativePosition);
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- ((ma_sio_stop_proc)pDevice->pContext->sndio.sio_stop)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback);
+ /*
+ There's a case where the position of the source will be right on top of the listener in which
+ case the length will be 0 and we'll end up with a division by zero. We can just return a ratio
+ of 1.0 in this case. This is not considered in the OpenAL spec, but is necessary.
+ */
+ if (len == 0) {
+ return 1.0;
}
- return MA_SUCCESS;
+ vls = ma_vec3f_dot(relativePosition, listenVelocity) / len;
+ vss = ma_vec3f_dot(relativePosition, sourceVelocity) / len;
+
+ vls = ma_min(vls, speedOfSound / dopplerFactor);
+ vss = ma_min(vss, speedOfSound / dopplerFactor);
+
+ return (speedOfSound - dopplerFactor*vls) / (speedOfSound - dopplerFactor*vss);
}
-static ma_result ma_device_write__sndio(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten)
-{
- int result;
- if (pFramesWritten != NULL) {
- *pFramesWritten = 0;
+static void ma_get_default_channel_map_for_spatializer(ma_channel* pChannelMap, size_t channelMapCap, ma_uint32 channelCount)
+{
+ /*
+ Special case for stereo. Want to default the left and right speakers to side left and side
+ right so that they're facing directly down the X axis rather than slightly forward. Not
+ doing this will result in sounds being quieter when behind the listener. This might
+ actually be good for some scenerios, but I don't think it's an appropriate default because
+ it can be a bit unexpected.
+ */
+ if (channelCount == 2) {
+ pChannelMap[0] = MA_CHANNEL_SIDE_LEFT;
+ pChannelMap[1] = MA_CHANNEL_SIDE_RIGHT;
+ } else {
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, channelCount);
}
+}
- result = ((ma_sio_write_proc)pDevice->pContext->sndio.sio_write)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels));
- if (result == 0) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[sndio] Failed to send data from the client to the device.", MA_IO_ERROR);
- }
- if (pFramesWritten != NULL) {
- *pFramesWritten = frameCount;
- }
+MA_API ma_spatializer_listener_config ma_spatializer_listener_config_init(ma_uint32 channelsOut)
+{
+ ma_spatializer_listener_config config;
- return MA_SUCCESS;
+ MA_ZERO_OBJECT(&config);
+ config.channelsOut = channelsOut;
+ config.pChannelMapOut = NULL;
+ config.handedness = ma_handedness_right;
+ config.worldUp = ma_vec3f_init_3f(0, 1, 0);
+ config.coneInnerAngleInRadians = 6.283185f; /* 360 degrees. */
+ config.coneOuterAngleInRadians = 6.283185f; /* 360 degrees. */
+ config.coneOuterGain = 0;
+ config.speedOfSound = 343.3f; /* Same as OpenAL. Used for doppler effect. */
+
+ return config;
}
-static ma_result ma_device_read__sndio(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead)
+
+typedef struct
{
- int result;
+ size_t sizeInBytes;
+ size_t channelMapOutOffset;
+} ma_spatializer_listener_heap_layout;
- if (pFramesRead != NULL) {
- *pFramesRead = 0;
- }
+static ma_result ma_spatializer_listener_get_heap_layout(const ma_spatializer_listener_config* pConfig, ma_spatializer_listener_heap_layout* pHeapLayout)
+{
+ MA_ASSERT(pHeapLayout != NULL);
- result = ((ma_sio_read_proc)pDevice->pContext->sndio.sio_read)((struct ma_sio_hdl*)pDevice->sndio.handleCapture, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels));
- if (result == 0) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[sndio] Failed to read data from the device to be sent to the device.", MA_IO_ERROR);
+ MA_ZERO_OBJECT(pHeapLayout);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pFramesRead != NULL) {
- *pFramesRead = frameCount;
+ if (pConfig->channelsOut == 0) {
+ return MA_INVALID_ARGS;
}
- return MA_SUCCESS;
-}
+ pHeapLayout->sizeInBytes = 0;
-static ma_result ma_context_uninit__sndio(ma_context* pContext)
-{
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pContext->backend == ma_backend_sndio);
+ /* Channel map. We always need this, even for passthroughs. */
+ pHeapLayout->channelMapOutOffset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += ma_align_64(sizeof(*pConfig->pChannelMapOut) * pConfig->channelsOut);
- (void)pContext;
return MA_SUCCESS;
}
-static ma_result ma_context_init__sndio(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+
+MA_API ma_result ma_spatializer_listener_get_heap_size(const ma_spatializer_listener_config* pConfig, size_t* pHeapSizeInBytes)
{
-#ifndef MA_NO_RUNTIME_LINKING
- const char* libsndioNames[] = {
- "libsndio.so"
- };
- size_t i;
+ ma_result result;
+ ma_spatializer_listener_heap_layout heapLayout;
- for (i = 0; i < ma_countof(libsndioNames); ++i) {
- pContext->sndio.sndioSO = ma_dlopen(pContext, libsndioNames[i]);
- if (pContext->sndio.sndioSO != NULL) {
- break;
- }
+ if (pHeapSizeInBytes == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pContext->sndio.sndioSO == NULL) {
- return MA_NO_BACKEND;
- }
+ *pHeapSizeInBytes = 0;
- pContext->sndio.sio_open = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_open");
- pContext->sndio.sio_close = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_close");
- pContext->sndio.sio_setpar = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_setpar");
- pContext->sndio.sio_getpar = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_getpar");
- pContext->sndio.sio_getcap = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_getcap");
- pContext->sndio.sio_write = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_write");
- pContext->sndio.sio_read = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_read");
- pContext->sndio.sio_start = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_start");
- pContext->sndio.sio_stop = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_stop");
- pContext->sndio.sio_initpar = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_initpar");
-#else
- pContext->sndio.sio_open = sio_open;
- pContext->sndio.sio_close = sio_close;
- pContext->sndio.sio_setpar = sio_setpar;
- pContext->sndio.sio_getpar = sio_getpar;
- pContext->sndio.sio_getcap = sio_getcap;
- pContext->sndio.sio_write = sio_write;
- pContext->sndio.sio_read = sio_read;
- pContext->sndio.sio_start = sio_start;
- pContext->sndio.sio_stop = sio_stop;
- pContext->sndio.sio_initpar = sio_initpar;
-#endif
+ result = ma_spatializer_listener_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- pCallbacks->onContextInit = ma_context_init__sndio;
- pCallbacks->onContextUninit = ma_context_uninit__sndio;
- pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__sndio;
- pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__sndio;
- pCallbacks->onDeviceInit = ma_device_init__sndio;
- pCallbacks->onDeviceUninit = ma_device_uninit__sndio;
- pCallbacks->onDeviceStart = ma_device_start__sndio;
- pCallbacks->onDeviceStop = ma_device_stop__sndio;
- pCallbacks->onDeviceRead = ma_device_read__sndio;
- pCallbacks->onDeviceWrite = ma_device_write__sndio;
- pCallbacks->onDeviceDataLoop = NULL;
+ *pHeapSizeInBytes = heapLayout.sizeInBytes;
- (void)pConfig;
return MA_SUCCESS;
}
-#endif /* sndio */
-
-
-/******************************************************************************
-
-audio(4) Backend
+MA_API ma_result ma_spatializer_listener_init_preallocated(const ma_spatializer_listener_config* pConfig, void* pHeap, ma_spatializer_listener* pListener)
+{
+ ma_result result;
+ ma_spatializer_listener_heap_layout heapLayout;
-******************************************************************************/
-#ifdef MA_HAS_AUDIO4
-#include
-#include
-#include
-#include
-#include
-#include
-#include
+ if (pListener == NULL) {
+ return MA_INVALID_ARGS;
+ }
-#if defined(__OpenBSD__)
- #include
- #if defined(OpenBSD) && OpenBSD >= 201709
- #define MA_AUDIO4_USE_NEW_API
- #endif
-#endif
+ MA_ZERO_OBJECT(pListener);
-static void ma_construct_device_id__audio4(char* id, size_t idSize, const char* base, int deviceIndex)
-{
- size_t baseLen;
+ result = ma_spatializer_listener_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- MA_ASSERT(id != NULL);
- MA_ASSERT(idSize > 0);
- MA_ASSERT(deviceIndex >= 0);
+ pListener->_pHeap = pHeap;
+ MA_ZERO_MEMORY(pHeap, heapLayout.sizeInBytes);
- baseLen = strlen(base);
- MA_ASSERT(idSize > baseLen);
+ pListener->config = *pConfig;
+ pListener->position = ma_vec3f_init_3f(0, 0, 0);
+ pListener->direction = ma_vec3f_init_3f(0, 0, -1);
+ pListener->velocity = ma_vec3f_init_3f(0, 0, 0);
+ pListener->isEnabled = MA_TRUE;
- ma_strcpy_s(id, idSize, base);
- ma_itoa_s(deviceIndex, id+baseLen, idSize-baseLen, 10);
-}
+ /* Swap the forward direction if we're left handed (it was initialized based on right handed). */
+ if (pListener->config.handedness == ma_handedness_left) {
+ pListener->direction = ma_vec3f_neg(pListener->direction);
+ }
-static ma_result ma_extract_device_index_from_id__audio4(const char* id, const char* base, int* pIndexOut)
-{
- size_t idLen;
- size_t baseLen;
- const char* deviceIndexStr;
- MA_ASSERT(id != NULL);
- MA_ASSERT(base != NULL);
- MA_ASSERT(pIndexOut != NULL);
+ /* We must always have a valid channel map. */
+ pListener->config.pChannelMapOut = (ma_channel*)ma_offset_ptr(pHeap, heapLayout.channelMapOutOffset);
- idLen = strlen(id);
- baseLen = strlen(base);
- if (idLen <= baseLen) {
- return MA_ERROR; /* Doesn't look like the id starts with the base. */
+ /* Use a slightly different default channel map for stereo. */
+ if (pConfig->pChannelMapOut == NULL) {
+ ma_get_default_channel_map_for_spatializer(pListener->config.pChannelMapOut, pConfig->channelsOut, pConfig->channelsOut);
+ } else {
+ ma_channel_map_copy_or_default(pListener->config.pChannelMapOut, pConfig->channelsOut, pConfig->pChannelMapOut, pConfig->channelsOut);
}
- if (strncmp(id, base, baseLen) != 0) {
- return MA_ERROR; /* ID does not begin with base. */
+ return MA_SUCCESS;
+}
+
+MA_API ma_result ma_spatializer_listener_init(const ma_spatializer_listener_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_spatializer_listener* pListener)
+{
+ ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
+
+ result = ma_spatializer_listener_get_heap_size(pConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
}
- deviceIndexStr = id + baseLen;
- if (deviceIndexStr[0] == '\0') {
- return MA_ERROR; /* No index specified in the ID. */
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+ } else {
+ pHeap = NULL;
}
- if (pIndexOut) {
- *pIndexOut = atoi(deviceIndexStr);
+ result = ma_spatializer_listener_init_preallocated(pConfig, pHeap, pListener);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
}
+ pListener->_ownsHeap = MA_TRUE;
return MA_SUCCESS;
}
+MA_API void ma_spatializer_listener_uninit(ma_spatializer_listener* pListener, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pListener == NULL) {
+ return;
+ }
-#if !defined(MA_AUDIO4_USE_NEW_API) /* Old API */
-static ma_format ma_format_from_encoding__audio4(unsigned int encoding, unsigned int precision)
+ if (pListener->_ownsHeap) {
+ ma_free(pListener->_pHeap, pAllocationCallbacks);
+ }
+}
+
+MA_API ma_channel* ma_spatializer_listener_get_channel_map(ma_spatializer_listener* pListener)
{
- if (precision == 8 && (encoding == AUDIO_ENCODING_ULINEAR || encoding == AUDIO_ENCODING_ULINEAR || encoding == AUDIO_ENCODING_ULINEAR_LE || encoding == AUDIO_ENCODING_ULINEAR_BE)) {
- return ma_format_u8;
- } else {
- if (ma_is_little_endian() && encoding == AUDIO_ENCODING_SLINEAR_LE) {
- if (precision == 16) {
- return ma_format_s16;
- } else if (precision == 24) {
- return ma_format_s24;
- } else if (precision == 32) {
- return ma_format_s32;
- }
- } else if (ma_is_big_endian() && encoding == AUDIO_ENCODING_SLINEAR_BE) {
- if (precision == 16) {
- return ma_format_s16;
- } else if (precision == 24) {
- return ma_format_s24;
- } else if (precision == 32) {
- return ma_format_s32;
- }
- }
+ if (pListener == NULL) {
+ return NULL;
}
- return ma_format_unknown; /* Encoding not supported. */
+ return pListener->config.pChannelMapOut;
}
-static void ma_encoding_from_format__audio4(ma_format format, unsigned int* pEncoding, unsigned int* pPrecision)
+MA_API void ma_spatializer_listener_set_cone(ma_spatializer_listener* pListener, float innerAngleInRadians, float outerAngleInRadians, float outerGain)
{
- MA_ASSERT(pEncoding != NULL);
- MA_ASSERT(pPrecision != NULL);
+ if (pListener == NULL) {
+ return;
+ }
- switch (format)
- {
- case ma_format_u8:
- {
- *pEncoding = AUDIO_ENCODING_ULINEAR;
- *pPrecision = 8;
- } break;
+ pListener->config.coneInnerAngleInRadians = innerAngleInRadians;
+ pListener->config.coneOuterAngleInRadians = outerAngleInRadians;
+ pListener->config.coneOuterGain = outerGain;
+}
- case ma_format_s24:
- {
- *pEncoding = (ma_is_little_endian()) ? AUDIO_ENCODING_SLINEAR_LE : AUDIO_ENCODING_SLINEAR_BE;
- *pPrecision = 24;
- } break;
+MA_API void ma_spatializer_listener_get_cone(const ma_spatializer_listener* pListener, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain)
+{
+ if (pListener == NULL) {
+ return;
+ }
- case ma_format_s32:
- {
- *pEncoding = (ma_is_little_endian()) ? AUDIO_ENCODING_SLINEAR_LE : AUDIO_ENCODING_SLINEAR_BE;
- *pPrecision = 32;
- } break;
+ if (pInnerAngleInRadians != NULL) {
+ *pInnerAngleInRadians = pListener->config.coneInnerAngleInRadians;
+ }
- case ma_format_s16:
- case ma_format_f32:
- case ma_format_unknown:
- default:
- {
- *pEncoding = (ma_is_little_endian()) ? AUDIO_ENCODING_SLINEAR_LE : AUDIO_ENCODING_SLINEAR_BE;
- *pPrecision = 16;
- } break;
+ if (pOuterAngleInRadians != NULL) {
+ *pOuterAngleInRadians = pListener->config.coneOuterAngleInRadians;
+ }
+
+ if (pOuterGain != NULL) {
+ *pOuterGain = pListener->config.coneOuterGain;
}
}
-static ma_format ma_format_from_prinfo__audio4(struct audio_prinfo* prinfo)
+MA_API void ma_spatializer_listener_set_position(ma_spatializer_listener* pListener, float x, float y, float z)
{
- return ma_format_from_encoding__audio4(prinfo->encoding, prinfo->precision);
+ if (pListener == NULL) {
+ return;
+ }
+
+ pListener->position = ma_vec3f_init_3f(x, y, z);
}
-static ma_format ma_best_format_from_fd__audio4(int fd, ma_format preferredFormat)
+MA_API ma_vec3f ma_spatializer_listener_get_position(const ma_spatializer_listener* pListener)
{
- audio_encoding_t encoding;
- ma_uint32 iFormat;
- int counter = 0;
-
- /* First check to see if the preferred format is supported. */
- if (preferredFormat != ma_format_unknown) {
- counter = 0;
- for (;;) {
- MA_ZERO_OBJECT(&encoding);
- encoding.index = counter;
- if (ioctl(fd, AUDIO_GETENC, &encoding) < 0) {
- break;
- }
+ if (pListener == NULL) {
+ return ma_vec3f_init_3f(0, 0, 0);
+ }
- if (preferredFormat == ma_format_from_encoding__audio4(encoding.encoding, encoding.precision)) {
- return preferredFormat; /* Found the preferred format. */
- }
+ return pListener->position;
+}
- /* Getting here means this encoding does not match our preferred format so we need to more on to the next encoding. */
- counter += 1;
- }
+MA_API void ma_spatializer_listener_set_direction(ma_spatializer_listener* pListener, float x, float y, float z)
+{
+ if (pListener == NULL) {
+ return;
}
- /* Getting here means our preferred format is not supported, so fall back to our standard priorities. */
- for (iFormat = 0; iFormat < ma_countof(g_maFormatPriorities); iFormat += 1) {
- ma_format format = g_maFormatPriorities[iFormat];
+ pListener->direction = ma_vec3f_init_3f(x, y, z);
+}
- counter = 0;
- for (;;) {
- MA_ZERO_OBJECT(&encoding);
- encoding.index = counter;
- if (ioctl(fd, AUDIO_GETENC, &encoding) < 0) {
- break;
- }
+MA_API ma_vec3f ma_spatializer_listener_get_direction(const ma_spatializer_listener* pListener)
+{
+ if (pListener == NULL) {
+ return ma_vec3f_init_3f(0, 0, -1);
+ }
- if (format == ma_format_from_encoding__audio4(encoding.encoding, encoding.precision)) {
- return format; /* Found a workable format. */
- }
+ return pListener->direction;
+}
- /* Getting here means this encoding does not match our preferred format so we need to more on to the next encoding. */
- counter += 1;
- }
+MA_API void ma_spatializer_listener_set_velocity(ma_spatializer_listener* pListener, float x, float y, float z)
+{
+ if (pListener == NULL) {
+ return;
}
- /* Getting here means not appropriate format was found. */
- return ma_format_unknown;
+ pListener->velocity = ma_vec3f_init_3f(x, y, z);
}
-#else
-static ma_format ma_format_from_swpar__audio4(struct audio_swpar* par)
+
+MA_API ma_vec3f ma_spatializer_listener_get_velocity(const ma_spatializer_listener* pListener)
{
- if (par->bits == 8 && par->bps == 1 && par->sig == 0) {
- return ma_format_u8;
- }
- if (par->bits == 16 && par->bps == 2 && par->sig == 1 && par->le == ma_is_little_endian()) {
- return ma_format_s16;
+ if (pListener == NULL) {
+ return ma_vec3f_init_3f(0, 0, 0);
}
- if (par->bits == 24 && par->bps == 3 && par->sig == 1 && par->le == ma_is_little_endian()) {
- return ma_format_s24;
+
+ return pListener->velocity;
+}
+
+MA_API void ma_spatializer_listener_set_speed_of_sound(ma_spatializer_listener* pListener, float speedOfSound)
+{
+ if (pListener == NULL) {
+ return;
}
- if (par->bits == 32 && par->bps == 4 && par->sig == 1 && par->le == ma_is_little_endian()) {
- return ma_format_f32;
+
+ pListener->config.speedOfSound = speedOfSound;
+}
+
+MA_API float ma_spatializer_listener_get_speed_of_sound(const ma_spatializer_listener* pListener)
+{
+ if (pListener == NULL) {
+ return 0;
}
- /* Format not supported. */
- return ma_format_unknown;
+ return pListener->config.speedOfSound;
}
-#endif
-static ma_result ma_context_get_device_info_from_fd__audio4(ma_context* pContext, ma_device_type deviceType, int fd, ma_device_info* pDeviceInfo)
+MA_API void ma_spatializer_listener_set_world_up(ma_spatializer_listener* pListener, float x, float y, float z)
{
- audio_device_t fdDevice;
+ if (pListener == NULL) {
+ return;
+ }
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(fd >= 0);
- MA_ASSERT(pDeviceInfo != NULL);
+ pListener->config.worldUp = ma_vec3f_init_3f(x, y, z);
+}
- (void)pContext;
- (void)deviceType;
+MA_API ma_vec3f ma_spatializer_listener_get_world_up(const ma_spatializer_listener* pListener)
+{
+ if (pListener == NULL) {
+ return ma_vec3f_init_3f(0, 1, 0);
+ }
- if (ioctl(fd, AUDIO_GETDEV, &fdDevice) < 0) {
- return MA_ERROR; /* Failed to retrieve device info. */
+ return pListener->config.worldUp;
+}
+
+MA_API void ma_spatializer_listener_set_enabled(ma_spatializer_listener* pListener, ma_bool32 isEnabled)
+{
+ if (pListener == NULL) {
+ return;
}
- /* Name. */
- ma_strcpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), fdDevice.name);
+ pListener->isEnabled = isEnabled;
+}
- #if !defined(MA_AUDIO4_USE_NEW_API)
- {
- audio_info_t fdInfo;
- int counter = 0;
- ma_uint32 channels;
- ma_uint32 sampleRate;
+MA_API ma_bool32 ma_spatializer_listener_is_enabled(const ma_spatializer_listener* pListener)
+{
+ if (pListener == NULL) {
+ return MA_FALSE;
+ }
- if (ioctl(fd, AUDIO_GETINFO, &fdInfo) < 0) {
- return MA_ERROR;
- }
+ return pListener->isEnabled;
+}
- if (deviceType == ma_device_type_playback) {
- channels = fdInfo.play.channels;
- sampleRate = fdInfo.play.sample_rate;
- } else {
- channels = fdInfo.record.channels;
- sampleRate = fdInfo.record.sample_rate;
- }
- /* Supported formats. We get this by looking at the encodings. */
- pDeviceInfo->nativeDataFormatCount = 0;
- for (;;) {
- audio_encoding_t encoding;
- ma_format format;
- MA_ZERO_OBJECT(&encoding);
- encoding.index = counter;
- if (ioctl(fd, AUDIO_GETENC, &encoding) < 0) {
- break;
- }
- format = ma_format_from_encoding__audio4(encoding.encoding, encoding.precision);
- if (format != ma_format_unknown) {
- ma_device_info_add_native_data_format(pDeviceInfo, format, channels, sampleRate, 0);
- }
+MA_API ma_spatializer_config ma_spatializer_config_init(ma_uint32 channelsIn, ma_uint32 channelsOut)
+{
+ ma_spatializer_config config;
- counter += 1;
- }
- }
- #else
- {
- struct audio_swpar fdPar;
- ma_format format;
- ma_uint32 channels;
- ma_uint32 sampleRate;
+ MA_ZERO_OBJECT(&config);
+ config.channelsIn = channelsIn;
+ config.channelsOut = channelsOut;
+ config.pChannelMapIn = NULL;
+ config.attenuationModel = ma_attenuation_model_inverse;
+ config.positioning = ma_positioning_absolute;
+ config.handedness = ma_handedness_right;
+ config.minGain = 0;
+ config.maxGain = 1;
+ config.minDistance = 1;
+ config.maxDistance = MA_FLT_MAX;
+ config.rolloff = 1;
+ config.coneInnerAngleInRadians = 6.283185f; /* 360 degrees. */
+ config.coneOuterAngleInRadians = 6.283185f; /* 360 degress. */
+ config.coneOuterGain = 0.0f;
+ config.dopplerFactor = 1;
+ config.directionalAttenuationFactor = 1;
+ config.gainSmoothTimeInFrames = 360; /* 7.5ms @ 48K. */
- if (ioctl(fd, AUDIO_GETPAR, &fdPar) < 0) {
- return MA_ERROR;
- }
+ return config;
+}
- format = ma_format_from_swpar__audio4(&fdPar);
- if (format == ma_format_unknown) {
- return MA_FORMAT_NOT_SUPPORTED;
- }
- if (deviceType == ma_device_type_playback) {
- channels = fdPar.pchan;
- } else {
- channels = fdPar.rchan;
- }
+static ma_gainer_config ma_spatializer_gainer_config_init(const ma_spatializer_config* pConfig)
+{
+ MA_ASSERT(pConfig != NULL);
+ return ma_gainer_config_init(pConfig->channelsOut, pConfig->gainSmoothTimeInFrames);
+}
- sampleRate = fdPar.rate;
+static ma_result ma_spatializer_validate_config(const ma_spatializer_config* pConfig)
+{
+ MA_ASSERT(pConfig != NULL);
- pDeviceInfo->nativeDataFormatCount = 0;
- ma_device_info_add_native_data_format(pDeviceInfo, format, channels, sampleRate, 0);
+ if (pConfig->channelsIn == 0 || pConfig->channelsOut == 0) {
+ return MA_INVALID_ARGS;
}
- #endif
return MA_SUCCESS;
}
-static ma_result ma_context_enumerate_devices__audio4(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+typedef struct
{
- const int maxDevices = 64;
- char devpath[256];
- int iDevice;
+ size_t sizeInBytes;
+ size_t channelMapInOffset;
+ size_t newChannelGainsOffset;
+ size_t gainerOffset;
+} ma_spatializer_heap_layout;
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(callback != NULL);
+static ma_result ma_spatializer_get_heap_layout(const ma_spatializer_config* pConfig, ma_spatializer_heap_layout* pHeapLayout)
+{
+ ma_result result;
- /*
- Every device will be named "/dev/audioN", with a "/dev/audioctlN" equivalent. We use the "/dev/audioctlN"
- version here since we can open it even when another process has control of the "/dev/audioN" device.
- */
- for (iDevice = 0; iDevice < maxDevices; ++iDevice) {
- struct stat st;
- int fd;
- ma_bool32 isTerminating = MA_FALSE;
+ MA_ASSERT(pHeapLayout != NULL);
- ma_strcpy_s(devpath, sizeof(devpath), "/dev/audioctl");
- ma_itoa_s(iDevice, devpath+strlen(devpath), sizeof(devpath)-strlen(devpath), 10);
+ MA_ZERO_OBJECT(pHeapLayout);
- if (stat(devpath, &st) < 0) {
- break;
- }
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
- /* The device exists, but we need to check if it's usable as playback and/or capture. */
+ result = ma_spatializer_validate_config(pConfig);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- /* Playback. */
- if (!isTerminating) {
- fd = open(devpath, O_RDONLY, 0);
- if (fd >= 0) {
- /* Supports playback. */
- ma_device_info deviceInfo;
- MA_ZERO_OBJECT(&deviceInfo);
- ma_construct_device_id__audio4(deviceInfo.id.audio4, sizeof(deviceInfo.id.audio4), "/dev/audio", iDevice);
- if (ma_context_get_device_info_from_fd__audio4(pContext, ma_device_type_playback, fd, &deviceInfo) == MA_SUCCESS) {
- isTerminating = !callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
- }
+ pHeapLayout->sizeInBytes = 0;
- close(fd);
- }
- }
+ /* Channel map. */
+ pHeapLayout->channelMapInOffset = MA_SIZE_MAX; /* <-- MA_SIZE_MAX indicates no allocation necessary. */
+ if (pConfig->pChannelMapIn != NULL) {
+ pHeapLayout->channelMapInOffset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += ma_align_64(sizeof(*pConfig->pChannelMapIn) * pConfig->channelsIn);
+ }
- /* Capture. */
- if (!isTerminating) {
- fd = open(devpath, O_WRONLY, 0);
- if (fd >= 0) {
- /* Supports capture. */
- ma_device_info deviceInfo;
- MA_ZERO_OBJECT(&deviceInfo);
- ma_construct_device_id__audio4(deviceInfo.id.audio4, sizeof(deviceInfo.id.audio4), "/dev/audio", iDevice);
- if (ma_context_get_device_info_from_fd__audio4(pContext, ma_device_type_capture, fd, &deviceInfo) == MA_SUCCESS) {
- isTerminating = !callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
- }
+ /* New channel gains for output. */
+ pHeapLayout->newChannelGainsOffset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += ma_align_64(sizeof(float) * pConfig->channelsOut);
- close(fd);
- }
- }
+ /* Gainer. */
+ {
+ size_t gainerHeapSizeInBytes;
+ ma_gainer_config gainerConfig;
- if (isTerminating) {
- break;
+ gainerConfig = ma_spatializer_gainer_config_init(pConfig);
+
+ result = ma_gainer_get_heap_size(&gainerConfig, &gainerHeapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
}
+
+ pHeapLayout->gainerOffset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += ma_align_64(gainerHeapSizeInBytes);
}
return MA_SUCCESS;
}
-static ma_result ma_context_get_device_info__audio4(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+MA_API ma_result ma_spatializer_get_heap_size(const ma_spatializer_config* pConfig, size_t* pHeapSizeInBytes)
{
- int fd = -1;
- int deviceIndex = -1;
- char ctlid[256];
ma_result result;
+ ma_spatializer_heap_layout heapLayout;
- MA_ASSERT(pContext != NULL);
-
- /*
- We need to open the "/dev/audioctlN" device to get the info. To do this we need to extract the number
- from the device ID which will be in "/dev/audioN" format.
- */
- if (pDeviceID == NULL) {
- /* Default device. */
- ma_strcpy_s(ctlid, sizeof(ctlid), "/dev/audioctl");
- } else {
- /* Specific device. We need to convert from "/dev/audioN" to "/dev/audioctlN". */
- result = ma_extract_device_index_from_id__audio4(pDeviceID->audio4, "/dev/audio", &deviceIndex);
- if (result != MA_SUCCESS) {
- return result;
- }
-
- ma_construct_device_id__audio4(ctlid, sizeof(ctlid), "/dev/audioctl", deviceIndex);
+ if (pHeapSizeInBytes == NULL) {
+ return MA_INVALID_ARGS;
}
- fd = open(ctlid, (deviceType == ma_device_type_playback) ? O_WRONLY : O_RDONLY, 0);
- if (fd == -1) {
- return MA_NO_DEVICE;
- }
+ *pHeapSizeInBytes = 0; /* Safety. */
- if (deviceIndex == -1) {
- ma_strcpy_s(pDeviceInfo->id.audio4, sizeof(pDeviceInfo->id.audio4), "/dev/audio");
- } else {
- ma_construct_device_id__audio4(pDeviceInfo->id.audio4, sizeof(pDeviceInfo->id.audio4), "/dev/audio", deviceIndex);
+ result = ma_spatializer_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
}
- result = ma_context_get_device_info_from_fd__audio4(pContext, deviceType, fd, pDeviceInfo);
+ *pHeapSizeInBytes = heapLayout.sizeInBytes;
- close(fd);
- return result;
+ return MA_SUCCESS;
}
-static ma_result ma_device_uninit__audio4(ma_device* pDevice)
+
+MA_API ma_result ma_spatializer_init_preallocated(const ma_spatializer_config* pConfig, void* pHeap, ma_spatializer* pSpatializer)
{
- MA_ASSERT(pDevice != NULL);
+ ma_result result;
+ ma_spatializer_heap_layout heapLayout;
+ ma_gainer_config gainerConfig;
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- close(pDevice->audio4.fdCapture);
+ if (pSpatializer == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- close(pDevice->audio4.fdPlayback);
+ MA_ZERO_OBJECT(pSpatializer);
+
+ if (pConfig == NULL || pHeap == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ result = ma_spatializer_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ pSpatializer->_pHeap = pHeap;
+ MA_ZERO_MEMORY(pHeap, heapLayout.sizeInBytes);
+
+ pSpatializer->channelsIn = pConfig->channelsIn;
+ pSpatializer->channelsOut = pConfig->channelsOut;
+ pSpatializer->attenuationModel = pConfig->attenuationModel;
+ pSpatializer->positioning = pConfig->positioning;
+ pSpatializer->handedness = pConfig->handedness;
+ pSpatializer->minGain = pConfig->minGain;
+ pSpatializer->maxGain = pConfig->maxGain;
+ pSpatializer->minDistance = pConfig->minDistance;
+ pSpatializer->maxDistance = pConfig->maxDistance;
+ pSpatializer->rolloff = pConfig->rolloff;
+ pSpatializer->coneInnerAngleInRadians = pConfig->coneInnerAngleInRadians;
+ pSpatializer->coneOuterAngleInRadians = pConfig->coneOuterAngleInRadians;
+ pSpatializer->coneOuterGain = pConfig->coneOuterGain;
+ pSpatializer->dopplerFactor = pConfig->dopplerFactor;
+ pSpatializer->directionalAttenuationFactor = pConfig->directionalAttenuationFactor;
+ pSpatializer->gainSmoothTimeInFrames = pConfig->gainSmoothTimeInFrames;
+ pSpatializer->position = ma_vec3f_init_3f(0, 0, 0);
+ pSpatializer->direction = ma_vec3f_init_3f(0, 0, -1);
+ pSpatializer->velocity = ma_vec3f_init_3f(0, 0, 0);
+ pSpatializer->dopplerPitch = 1;
+
+ /* Swap the forward direction if we're left handed (it was initialized based on right handed). */
+ if (pSpatializer->handedness == ma_handedness_left) {
+ pSpatializer->direction = ma_vec3f_neg(pSpatializer->direction);
+ }
+
+ /* Channel map. This will be on the heap. */
+ if (pConfig->pChannelMapIn != NULL) {
+ pSpatializer->pChannelMapIn = (ma_channel*)ma_offset_ptr(pHeap, heapLayout.channelMapInOffset);
+ ma_channel_map_copy_or_default(pSpatializer->pChannelMapIn, pSpatializer->channelsIn, pConfig->pChannelMapIn, pSpatializer->channelsIn);
+ }
+
+ /* New channel gains for output channels. */
+ pSpatializer->pNewChannelGainsOut = (float*)ma_offset_ptr(pHeap, heapLayout.newChannelGainsOffset);
+
+ /* Gainer. */
+ gainerConfig = ma_spatializer_gainer_config_init(pConfig);
+
+ result = ma_gainer_init_preallocated(&gainerConfig, ma_offset_ptr(pHeap, heapLayout.gainerOffset), &pSpatializer->gainer);
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to initialize the gainer. */
}
return MA_SUCCESS;
}
-static ma_result ma_device_init_fd__audio4(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptor, ma_device_type deviceType)
+MA_API ma_result ma_spatializer_init(const ma_spatializer_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_spatializer* pSpatializer)
{
- const char* pDefaultDeviceNames[] = {
- "/dev/audio",
- "/dev/audio0"
- };
- int fd;
- int fdFlags = 0;
- ma_format internalFormat;
- ma_uint32 internalChannels;
- ma_uint32 internalSampleRate;
- ma_uint32 internalPeriodSizeInFrames;
- ma_uint32 internalPeriods;
-
- MA_ASSERT(pConfig != NULL);
- MA_ASSERT(deviceType != ma_device_type_duplex);
- MA_ASSERT(pDevice != NULL);
+ ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
- /* The first thing to do is open the file. */
- if (deviceType == ma_device_type_capture) {
- fdFlags = O_RDONLY;
- } else {
- fdFlags = O_WRONLY;
+ /* We'll need a heap allocation to retrieve the size. */
+ result = ma_spatializer_get_heap_size(pConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
}
- /*fdFlags |= O_NONBLOCK;*/
- if (pDescriptor->pDeviceID == NULL) {
- /* Default device. */
- size_t iDevice;
- for (iDevice = 0; iDevice < ma_countof(pDefaultDeviceNames); ++iDevice) {
- fd = open(pDefaultDeviceNames[iDevice], fdFlags, 0);
- if (fd != -1) {
- break;
- }
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
}
} else {
- /* Specific device. */
- fd = open(pDescriptor->pDeviceID->audio4, fdFlags, 0);
+ pHeap = NULL;
}
- if (fd == -1) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to open device.", ma_result_from_errno(errno));
+ result = ma_spatializer_init_preallocated(pConfig, pHeap, pSpatializer);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
}
- #if !defined(MA_AUDIO4_USE_NEW_API) /* Old API */
- {
- audio_info_t fdInfo;
+ pSpatializer->_ownsHeap = MA_TRUE;
+ return MA_SUCCESS;
+}
- /*
- The documentation is a little bit unclear to me as to how it handles formats. It says the
- following:
+MA_API void ma_spatializer_uninit(ma_spatializer* pSpatializer, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pSpatializer == NULL) {
+ return;
+ }
- Regardless of formats supported by underlying driver, the audio driver accepts the
- following formats.
+ ma_gainer_uninit(&pSpatializer->gainer, pAllocationCallbacks);
- By then the next sentence says this:
+ if (pSpatializer->_ownsHeap) {
+ ma_free(pSpatializer->_pHeap, pAllocationCallbacks);
+ }
+}
- `encoding` and `precision` are one of the values obtained by AUDIO_GETENC.
+static float ma_calculate_angular_gain(ma_vec3f dirA, ma_vec3f dirB, float coneInnerAngleInRadians, float coneOuterAngleInRadians, float coneOuterGain)
+{
+ /*
+ Angular attenuation.
- It sounds like a direct contradiction to me. I'm going to play this safe any only use the
- best sample format returned by AUDIO_GETENC. If the requested format is supported we'll
- use that, but otherwise we'll just use our standard format priorities to pick an
- appropriate one.
- */
- AUDIO_INITINFO(&fdInfo);
+ Unlike distance gain, the math for this is not specified by the OpenAL spec so we'll just go ahead and figure
+ this out for ourselves at the expense of possibly being inconsistent with other implementations.
- /* We get the driver to do as much of the data conversion as possible. */
- if (deviceType == ma_device_type_capture) {
- fdInfo.mode = AUMODE_RECORD;
- ma_encoding_from_format__audio4(ma_best_format_from_fd__audio4(fd, pDescriptor->format), &fdInfo.record.encoding, &fdInfo.record.precision);
+ To do cone attenuation, I'm just using the same math that we'd use to implement a basic spotlight in OpenGL. We
+ just need to get the direction from the source to the listener and then do a dot product against that and the
+ direction of the spotlight. Then we just compare that dot product against the cosine of the inner and outer
+ angles. If the dot product is greater than the the outer angle, we just use coneOuterGain. If it's less than
+ the inner angle, we just use a gain of 1. Otherwise we linearly interpolate between 1 and coneOuterGain.
+ */
+ if (coneInnerAngleInRadians < 6.283185f) {
+ float angularGain = 1;
+ float cutoffInner = (float)ma_cosd(coneInnerAngleInRadians*0.5f);
+ float cutoffOuter = (float)ma_cosd(coneOuterAngleInRadians*0.5f);
+ float d;
- if (pDescriptor->channels != 0) {
- fdInfo.record.channels = ma_clamp(pDescriptor->channels, 1, 12); /* From the documentation: `channels` ranges from 1 to 12. */
+ d = ma_vec3f_dot(dirA, dirB);
+
+ if (d > cutoffInner) {
+ /* It's inside the inner angle. */
+ angularGain = 1;
+ } else {
+ /* It's outside the inner angle. */
+ if (d > cutoffOuter) {
+ /* It's between the inner and outer angle. We need to linearly interpolate between 1 and coneOuterGain. */
+ angularGain = ma_mix_f32(coneOuterGain, 1, (d - cutoffOuter) / (cutoffInner - cutoffOuter));
+ } else {
+ /* It's outside the outer angle. */
+ angularGain = coneOuterGain;
}
+ }
- if (pDescriptor->sampleRate != 0) {
- fdInfo.record.sample_rate = ma_clamp(pDescriptor->sampleRate, 1000, 192000); /* From the documentation: `frequency` ranges from 1000Hz to 192000Hz. (They mean `sample_rate` instead of `frequency`.) */
+ /*printf("d = %f; cutoffInner = %f; cutoffOuter = %f; angularGain = %f\n", d, cutoffInner, cutoffOuter, angularGain);*/
+ return angularGain;
+ } else {
+ /* Inner angle is 360 degrees so no need to do any attenuation. */
+ return 1;
+ }
+}
+
+MA_API ma_result ma_spatializer_process_pcm_frames(ma_spatializer* pSpatializer, ma_spatializer_listener* pListener, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+{
+ ma_channel* pChannelMapIn = pSpatializer->pChannelMapIn;
+ ma_channel* pChannelMapOut = pListener->config.pChannelMapOut;
+
+ if (pSpatializer == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ /* If we're not spatializing we need to run an optimized path. */
+ if (c89atomic_load_i32(&pSpatializer->attenuationModel) == ma_attenuation_model_none) {
+ if (ma_spatializer_listener_is_enabled(pListener)) {
+ /* No attenuation is required, but we'll need to do some channel conversion. */
+ if (pSpatializer->channelsIn == pSpatializer->channelsOut) {
+ ma_copy_pcm_frames(pFramesOut, pFramesIn, frameCount, ma_format_f32, pSpatializer->channelsIn);
+ } else {
+ ma_channel_map_apply_f32((float*)pFramesOut, pChannelMapOut, pSpatializer->channelsOut, (const float*)pFramesIn, pChannelMapIn, pSpatializer->channelsIn, frameCount, ma_channel_mix_mode_rectangular, ma_mono_expansion_mode_default); /* Safe casts to float* because f32 is the only supported format. */
}
} else {
- fdInfo.mode = AUMODE_PLAY;
- ma_encoding_from_format__audio4(ma_best_format_from_fd__audio4(fd, pDescriptor->format), &fdInfo.play.encoding, &fdInfo.play.precision);
+ /* The listener is disabled. Output silence. */
+ ma_silence_pcm_frames(pFramesOut, frameCount, ma_format_f32, pSpatializer->channelsOut);
+ }
- if (pDescriptor->channels != 0) {
- fdInfo.play.channels = ma_clamp(pDescriptor->channels, 1, 12); /* From the documentation: `channels` ranges from 1 to 12. */
- }
+ /*
+ We're not doing attenuation so don't bother with doppler for now. I'm not sure if this is
+ the correct thinking so might need to review this later.
+ */
+ pSpatializer->dopplerPitch = 1;
+ } else {
+ /*
+ Let's first determine which listener the sound is closest to. Need to keep in mind that we
+ might not have a world or any listeners, in which case we just spatializer based on the
+ listener being positioned at the origin (0, 0, 0).
+ */
+ ma_vec3f relativePosNormalized;
+ ma_vec3f relativePos; /* The position relative to the listener. */
+ ma_vec3f relativeDir; /* The direction of the sound, relative to the listener. */
+ ma_vec3f listenerVel; /* The volocity of the listener. For doppler pitch calculation. */
+ float speedOfSound;
+ float distance = 0;
+ float gain = 1;
+ ma_uint32 iChannel;
+ const ma_uint32 channelsOut = pSpatializer->channelsOut;
+ const ma_uint32 channelsIn = pSpatializer->channelsIn;
+ float minDistance = ma_spatializer_get_min_distance(pSpatializer);
+ float maxDistance = ma_spatializer_get_max_distance(pSpatializer);
+ float rolloff = ma_spatializer_get_rolloff(pSpatializer);
+ float dopplerFactor = ma_spatializer_get_doppler_factor(pSpatializer);
- if (pDescriptor->sampleRate != 0) {
- fdInfo.play.sample_rate = ma_clamp(pDescriptor->sampleRate, 1000, 192000); /* From the documentation: `frequency` ranges from 1000Hz to 192000Hz. (They mean `sample_rate` instead of `frequency`.) */
- }
+ /*
+ We'll need the listener velocity for doppler pitch calculations. The speed of sound is
+ defined by the listener, so we'll grab that here too.
+ */
+ if (pListener != NULL) {
+ listenerVel = pListener->velocity;
+ speedOfSound = pListener->config.speedOfSound;
+ } else {
+ listenerVel = ma_vec3f_init_3f(0, 0, 0);
+ speedOfSound = MA_DEFAULT_SPEED_OF_SOUND;
}
- if (ioctl(fd, AUDIO_SETINFO, &fdInfo) < 0) {
- close(fd);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to set device format. AUDIO_SETINFO failed.", MA_FORMAT_NOT_SUPPORTED);
+ if (pListener == NULL || ma_spatializer_get_positioning(pSpatializer) == ma_positioning_relative) {
+ /* There's no listener or we're using relative positioning. */
+ relativePos = pSpatializer->position;
+ relativeDir = pSpatializer->direction;
+ } else {
+ /*
+ We've found a listener and we're using absolute positioning. We need to transform the
+ sound's position and direction so that it's relative to listener. Later on we'll use
+ this for determining the factors to apply to each channel to apply the panning effect.
+ */
+ ma_spatializer_get_relative_position_and_direction(pSpatializer, pListener, &relativePos, &relativeDir);
}
- if (ioctl(fd, AUDIO_GETINFO, &fdInfo) < 0) {
- close(fd);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] AUDIO_GETINFO failed.", MA_FORMAT_NOT_SUPPORTED);
+ distance = ma_vec3f_len(relativePos);
+
+ /* We've gathered the data, so now we can apply some spatialization. */
+ switch (ma_spatializer_get_attenuation_model(pSpatializer)) {
+ case ma_attenuation_model_inverse:
+ {
+ gain = ma_attenuation_inverse(distance, minDistance, maxDistance, rolloff);
+ } break;
+ case ma_attenuation_model_linear:
+ {
+ gain = ma_attenuation_linear(distance, minDistance, maxDistance, rolloff);
+ } break;
+ case ma_attenuation_model_exponential:
+ {
+ gain = ma_attenuation_exponential(distance, minDistance, maxDistance, rolloff);
+ } break;
+ case ma_attenuation_model_none:
+ default:
+ {
+ gain = 1;
+ } break;
}
- if (deviceType == ma_device_type_capture) {
- internalFormat = ma_format_from_prinfo__audio4(&fdInfo.record);
- internalChannels = fdInfo.record.channels;
- internalSampleRate = fdInfo.record.sample_rate;
+ /* Normalize the position. */
+ if (distance > 0.001f) {
+ float distanceInv = 1/distance;
+ relativePosNormalized = relativePos;
+ relativePosNormalized.x *= distanceInv;
+ relativePosNormalized.y *= distanceInv;
+ relativePosNormalized.z *= distanceInv;
} else {
- internalFormat = ma_format_from_prinfo__audio4(&fdInfo.play);
- internalChannels = fdInfo.play.channels;
- internalSampleRate = fdInfo.play.sample_rate;
+ distance = 0;
+ relativePosNormalized = ma_vec3f_init_3f(0, 0, 0);
}
- if (internalFormat == ma_format_unknown) {
- close(fd);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] The device's internal device format is not supported by miniaudio. The device is unusable.", MA_FORMAT_NOT_SUPPORTED);
- }
+ /*
+ Angular attenuation.
- /* Buffer. */
- {
- ma_uint32 internalPeriodSizeInBytes;
+ Unlike distance gain, the math for this is not specified by the OpenAL spec so we'll just go ahead and figure
+ this out for ourselves at the expense of possibly being inconsistent with other implementations.
- internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, internalSampleRate, pConfig->performanceProfile);
+ To do cone attenuation, I'm just using the same math that we'd use to implement a basic spotlight in OpenGL. We
+ just need to get the direction from the source to the listener and then do a dot product against that and the
+ direction of the spotlight. Then we just compare that dot product against the cosine of the inner and outer
+ angles. If the dot product is greater than the the outer angle, we just use coneOuterGain. If it's less than
+ the inner angle, we just use a gain of 1. Otherwise we linearly interpolate between 1 and coneOuterGain.
+ */
+ if (distance > 0) {
+ /* Source anglular gain. */
+ float spatializerConeInnerAngle;
+ float spatializerConeOuterAngle;
+ float spatializerConeOuterGain;
+ ma_spatializer_get_cone(pSpatializer, &spatializerConeInnerAngle, &spatializerConeOuterAngle, &spatializerConeOuterGain);
- internalPeriodSizeInBytes = internalPeriodSizeInFrames * ma_get_bytes_per_frame(internalFormat, internalChannels);
- if (internalPeriodSizeInBytes < 16) {
- internalPeriodSizeInBytes = 16;
- }
+ gain *= ma_calculate_angular_gain(relativeDir, ma_vec3f_neg(relativePosNormalized), spatializerConeInnerAngle, spatializerConeOuterAngle, spatializerConeOuterGain);
- internalPeriods = pDescriptor->periodCount;
- if (internalPeriods < 2) {
- internalPeriods = 2;
- }
+ /*
+ We're supporting angular gain on the listener as well for those who want to reduce the volume of sounds that
+ are positioned behind the listener. On default settings, this will have no effect.
+ */
+ if (pListener != NULL && pListener->config.coneInnerAngleInRadians < 6.283185f) {
+ ma_vec3f listenerDirection;
+ float listenerInnerAngle;
+ float listenerOuterAngle;
+ float listenerOuterGain;
+
+ if (pListener->config.handedness == ma_handedness_right) {
+ listenerDirection = ma_vec3f_init_3f(0, 0, -1);
+ } else {
+ listenerDirection = ma_vec3f_init_3f(0, 0, +1);
+ }
- /* What miniaudio calls a period, audio4 calls a block. */
- AUDIO_INITINFO(&fdInfo);
- fdInfo.hiwat = internalPeriods;
- fdInfo.lowat = internalPeriods-1;
- fdInfo.blocksize = internalPeriodSizeInBytes;
- if (ioctl(fd, AUDIO_SETINFO, &fdInfo) < 0) {
- close(fd);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to set internal buffer size. AUDIO_SETINFO failed.", MA_FORMAT_NOT_SUPPORTED);
- }
+ listenerInnerAngle = pListener->config.coneInnerAngleInRadians;
+ listenerOuterAngle = pListener->config.coneOuterAngleInRadians;
+ listenerOuterGain = pListener->config.coneOuterGain;
- internalPeriods = fdInfo.hiwat;
- internalPeriodSizeInFrames = fdInfo.blocksize / ma_get_bytes_per_frame(internalFormat, internalChannels);
+ gain *= ma_calculate_angular_gain(listenerDirection, relativePosNormalized, listenerInnerAngle, listenerOuterAngle, listenerOuterGain);
+ }
+ } else {
+ /* The sound is right on top of the listener. Don't do any angular attenuation. */
}
- }
- #else
- {
- struct audio_swpar fdPar;
- /* We need to retrieve the format of the device so we can know the channel count and sample rate. Then we can calculate the buffer size. */
- if (ioctl(fd, AUDIO_GETPAR, &fdPar) < 0) {
- close(fd);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to retrieve initial device parameters.", MA_FORMAT_NOT_SUPPORTED);
- }
- internalFormat = ma_format_from_swpar__audio4(&fdPar);
- internalChannels = (deviceType == ma_device_type_capture) ? fdPar.rchan : fdPar.pchan;
- internalSampleRate = fdPar.rate;
+ /* Clamp the gain. */
+ gain = ma_clamp(gain, ma_spatializer_get_min_gain(pSpatializer), ma_spatializer_get_max_gain(pSpatializer));
- if (internalFormat == ma_format_unknown) {
- close(fd);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] The device's internal device format is not supported by miniaudio. The device is unusable.", MA_FORMAT_NOT_SUPPORTED);
- }
+ /*
+ Panning. This is where we'll apply the gain and convert to the output channel count. We have an optimized path for
+ when we're converting to a mono stream. In that case we don't really need to do any panning - we just apply the
+ gain to the final output.
+ */
+ /*printf("distance=%f; gain=%f\n", distance, gain);*/
- /* Buffer. */
- {
- ma_uint32 internalPeriodSizeInBytes;
+ /* We must have a valid channel map here to ensure we spatialize properly. */
+ MA_ASSERT(pChannelMapOut != NULL);
- internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, internalSampleRate, pConfig->performanceProfile);
+ /*
+ We're not converting to mono so we'll want to apply some panning. This is where the feeling of something being
+ to the left, right, infront or behind the listener is calculated. I'm just using a basic model here. Note that
+ the code below is not based on any specific algorithm. I'm just implementing this off the top of my head and
+ seeing how it goes. There might be better ways to do this.
+
+ To determine the direction of the sound relative to a speaker I'm using dot products. Each speaker is given a
+ direction. For example, the left channel in a stereo system will be -1 on the X axis and the right channel will
+ be +1 on the X axis. A dot product is performed against the direction vector of the channel and the normalized
+ position of the sound.
+ */
+ for (iChannel = 0; iChannel < channelsOut; iChannel += 1) {
+ pSpatializer->pNewChannelGainsOut[iChannel] = gain;
+ }
- /* What miniaudio calls a period, audio4 calls a block. */
- internalPeriodSizeInBytes = internalPeriodSizeInFrames * ma_get_bytes_per_frame(internalFormat, internalChannels);
- if (internalPeriodSizeInBytes < 16) {
- internalPeriodSizeInBytes = 16;
- }
+ /*
+ Convert to our output channel count. If the listener is disabled we just output silence here. We cannot ignore
+ the whole section of code here because we need to update some internal spatialization state.
+ */
+ if (ma_spatializer_listener_is_enabled(pListener)) {
+ ma_channel_map_apply_f32((float*)pFramesOut, pChannelMapOut, channelsOut, (const float*)pFramesIn, pChannelMapIn, channelsIn, frameCount, ma_channel_mix_mode_rectangular, ma_mono_expansion_mode_default);
+ } else {
+ ma_silence_pcm_frames(pFramesOut, frameCount, ma_format_f32, pSpatializer->channelsOut);
+ }
- fdPar.nblks = pDescriptor->periodCount;
- fdPar.round = internalPeriodSizeInBytes;
+ /*
+ Calculate our per-channel gains. We do this based on the normalized relative position of the sound and it's
+ relation to the direction of the channel.
+ */
+ if (distance > 0) {
+ ma_vec3f unitPos = relativePos;
+ float distanceInv = 1/distance;
+ unitPos.x *= distanceInv;
+ unitPos.y *= distanceInv;
+ unitPos.z *= distanceInv;
+
+ for (iChannel = 0; iChannel < channelsOut; iChannel += 1) {
+ ma_channel channelOut;
+ float d;
+ float dMin;
+
+ channelOut = ma_channel_map_get_channel(pChannelMapOut, channelsOut, iChannel);
+ if (ma_is_spatial_channel_position(channelOut)) {
+ d = ma_mix_f32_fast(1, ma_vec3f_dot(unitPos, ma_get_channel_direction(channelOut)), ma_spatializer_get_directional_attenuation_factor(pSpatializer));
+ } else {
+ d = 1; /* It's not a spatial channel so there's no real notion of direction. */
+ }
- if (ioctl(fd, AUDIO_SETPAR, &fdPar) < 0) {
- close(fd);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to set device parameters.", MA_FORMAT_NOT_SUPPORTED);
- }
+ /*
+ In my testing, if the panning effect is too aggressive it makes spatialization feel uncomfortable.
+ The "dMin" variable below is used to control the aggressiveness of the panning effect. When set to
+ 0, panning will be most extreme and any sounds that are positioned on the opposite side of the
+ speaker will be completely silent from that speaker. Not only does this feel uncomfortable, it
+ doesn't even remotely represent the real world at all because sounds that come from your right side
+ are still clearly audible from your left side. Setting "dMin" to 1 will result in no panning at
+ all, which is also not ideal. By setting it to something greater than 0, the spatialization effect
+ becomes much less dramatic and a lot more bearable.
+
+ Summary: 0 = more extreme panning; 1 = no panning.
+ */
+ dMin = 0.2f; /* TODO: Consider making this configurable. */
- if (ioctl(fd, AUDIO_GETPAR, &fdPar) < 0) {
- close(fd);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to retrieve actual device parameters.", MA_FORMAT_NOT_SUPPORTED);
+ /*
+ At this point, "d" will be positive if the sound is on the same side as the channel and negative if
+ it's on the opposite side. It will be in the range of -1..1. There's two ways I can think of to
+ calculate a panning value. The first is to simply convert it to 0..1, however this has a problem
+ which I'm not entirely happy with. Considering a stereo system, when a sound is positioned right
+ in front of the listener it'll result in each speaker getting a gain of 0.5. I don't know if I like
+ the idea of having a scaling factor of 0.5 being applied to a sound when it's sitting right in front
+ of the listener. I would intuitively expect that to be played at full volume, or close to it.
+
+ The second idea I think of is to only apply a reduction in gain when the sound is on the opposite
+ side of the speaker. That is, reduce the gain only when the dot product is negative. The problem
+ with this is that there will not be any attenuation as the sound sweeps around the 180 degrees
+ where the dot product is positive. The idea with this option is that you leave the gain at 1 when
+ the sound is being played on the same side as the speaker and then you just reduce the volume when
+ the sound is on the other side.
+
+ The summarize, I think the first option should give a better sense of spatialization, but the second
+ option is better for preserving the sound's power.
+
+ UPDATE: In my testing, I find the first option to sound better. You can feel the sense of space a
+ bit better, but you can also hear the reduction in volume when it's right in front.
+ */
+ #if 1
+ {
+ /*
+ Scale the dot product from -1..1 to 0..1. Will result in a sound directly in front losing power
+ by being played at 0.5 gain.
+ */
+ d = (d + 1) * 0.5f; /* -1..1 to 0..1 */
+ d = ma_max(d, dMin);
+ pSpatializer->pNewChannelGainsOut[iChannel] *= d;
+ }
+ #else
+ {
+ /*
+ Only reduce the volume of the sound if it's on the opposite side. This path keeps the volume more
+ consistent, but comes at the expense of a worse sense of space and positioning.
+ */
+ if (d < 0) {
+ d += 1; /* Move into the positive range. */
+ d = ma_max(d, dMin);
+ channelGainsOut[iChannel] *= d;
+ }
+ }
+ #endif
}
+ } else {
+ /* Assume the sound is right on top of us. Don't do any panning. */
}
- internalFormat = ma_format_from_swpar__audio4(&fdPar);
- internalChannels = (deviceType == ma_device_type_capture) ? fdPar.rchan : fdPar.pchan;
- internalSampleRate = fdPar.rate;
- internalPeriods = fdPar.nblks;
- internalPeriodSizeInFrames = fdPar.round / ma_get_bytes_per_frame(internalFormat, internalChannels);
- }
- #endif
-
- if (internalFormat == ma_format_unknown) {
- close(fd);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] The device's internal device format is not supported by miniaudio. The device is unusable.", MA_FORMAT_NOT_SUPPORTED);
- }
+ /* Now we need to apply the volume to each channel. This needs to run through the gainer to ensure we get a smooth volume transition. */
+ ma_gainer_set_gains(&pSpatializer->gainer, pSpatializer->pNewChannelGainsOut);
+ ma_gainer_process_pcm_frames(&pSpatializer->gainer, pFramesOut, pFramesOut, frameCount);
- if (deviceType == ma_device_type_capture) {
- pDevice->audio4.fdCapture = fd;
- } else {
- pDevice->audio4.fdPlayback = fd;
+ /*
+ Before leaving we'll want to update our doppler pitch so that the caller can apply some
+ pitch shifting if they desire. Note that we need to negate the relative position here
+ because the doppler calculation needs to be source-to-listener, but ours is listener-to-
+ source.
+ */
+ if (dopplerFactor > 0) {
+ pSpatializer->dopplerPitch = ma_doppler_pitch(ma_vec3f_sub(pListener->position, pSpatializer->position), pSpatializer->velocity, listenerVel, speedOfSound, dopplerFactor);
+ } else {
+ pSpatializer->dopplerPitch = 1;
+ }
}
- pDescriptor->format = internalFormat;
- pDescriptor->channels = internalChannels;
- pDescriptor->sampleRate = internalSampleRate;
- ma_get_standard_channel_map(ma_standard_channel_map_sound4, internalChannels, pDescriptor->channelMap);
- pDescriptor->periodSizeInFrames = internalPeriodSizeInFrames;
- pDescriptor->periodCount = internalPeriods;
-
return MA_SUCCESS;
}
-static ma_result ma_device_init__audio4(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+MA_API ma_uint32 ma_spatializer_get_input_channels(const ma_spatializer* pSpatializer)
{
- MA_ASSERT(pDevice != NULL);
+ if (pSpatializer == NULL) {
+ return 0;
+ }
- MA_ZERO_OBJECT(&pDevice->audio4);
+ return pSpatializer->channelsIn;
+}
- if (pConfig->deviceType == ma_device_type_loopback) {
- return MA_DEVICE_TYPE_NOT_SUPPORTED;
+MA_API ma_uint32 ma_spatializer_get_output_channels(const ma_spatializer* pSpatializer)
+{
+ if (pSpatializer == NULL) {
+ return 0;
}
- pDevice->audio4.fdCapture = -1;
- pDevice->audio4.fdPlayback = -1;
+ return pSpatializer->channelsOut;
+}
- /*
- The version of the operating system dictates whether or not the device is exclusive or shared. NetBSD
- introduced in-kernel mixing which means it's shared. All other BSD flavours are exclusive as far as
- I'm aware.
- */
-#if defined(__NetBSD_Version__) && __NetBSD_Version__ >= 800000000
- /* NetBSD 8.0+ */
- if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) ||
- ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) {
- return MA_SHARE_MODE_NOT_SUPPORTED;
+MA_API void ma_spatializer_set_attenuation_model(ma_spatializer* pSpatializer, ma_attenuation_model attenuationModel)
+{
+ if (pSpatializer == NULL) {
+ return;
}
-#else
- /* All other flavors. */
-#endif
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
- ma_result result = ma_device_init_fd__audio4(pDevice, pConfig, pDescriptorCapture, ma_device_type_capture);
- if (result != MA_SUCCESS) {
- return result;
- }
- }
+ c89atomic_exchange_i32(&pSpatializer->attenuationModel, attenuationModel);
+}
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- ma_result result = ma_device_init_fd__audio4(pDevice, pConfig, pDescriptorPlayback, ma_device_type_playback);
- if (result != MA_SUCCESS) {
- if (pConfig->deviceType == ma_device_type_duplex) {
- close(pDevice->audio4.fdCapture);
- }
- return result;
- }
+MA_API ma_attenuation_model ma_spatializer_get_attenuation_model(const ma_spatializer* pSpatializer)
+{
+ if (pSpatializer == NULL) {
+ return ma_attenuation_model_none;
}
- return MA_SUCCESS;
+ return (ma_attenuation_model)c89atomic_load_i32(&pSpatializer->attenuationModel);
}
-static ma_result ma_device_start__audio4(ma_device* pDevice)
+MA_API void ma_spatializer_set_positioning(ma_spatializer* pSpatializer, ma_positioning positioning)
{
- MA_ASSERT(pDevice != NULL);
-
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- if (pDevice->audio4.fdCapture == -1) {
- return MA_INVALID_ARGS;
- }
+ if (pSpatializer == NULL) {
+ return;
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- if (pDevice->audio4.fdPlayback == -1) {
- return MA_INVALID_ARGS;
- }
+ c89atomic_exchange_i32(&pSpatializer->positioning, positioning);
+}
+
+MA_API ma_positioning ma_spatializer_get_positioning(const ma_spatializer* pSpatializer)
+{
+ if (pSpatializer == NULL) {
+ return ma_positioning_absolute;
}
- return MA_SUCCESS;
+ return (ma_positioning)c89atomic_load_i32(&pSpatializer->positioning);
}
-static ma_result ma_device_stop_fd__audio4(ma_device* pDevice, int fd)
+MA_API void ma_spatializer_set_rolloff(ma_spatializer* pSpatializer, float rolloff)
{
- if (fd == -1) {
- return MA_INVALID_ARGS;
+ if (pSpatializer == NULL) {
+ return;
}
-#if !defined(MA_AUDIO4_USE_NEW_API)
- if (ioctl(fd, AUDIO_FLUSH, 0) < 0) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to stop device. AUDIO_FLUSH failed.", ma_result_from_errno(errno));
- }
-#else
- if (ioctl(fd, AUDIO_STOP, 0) < 0) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to stop device. AUDIO_STOP failed.", ma_result_from_errno(errno));
+ c89atomic_exchange_f32(&pSpatializer->rolloff, rolloff);
+}
+
+MA_API float ma_spatializer_get_rolloff(const ma_spatializer* pSpatializer)
+{
+ if (pSpatializer == NULL) {
+ return 0;
}
-#endif
- return MA_SUCCESS;
+ return c89atomic_load_f32(&pSpatializer->rolloff);
}
-static ma_result ma_device_stop__audio4(ma_device* pDevice)
+MA_API void ma_spatializer_set_min_gain(ma_spatializer* pSpatializer, float minGain)
{
- MA_ASSERT(pDevice != NULL);
+ if (pSpatializer == NULL) {
+ return;
+ }
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- ma_result result;
+ c89atomic_exchange_f32(&pSpatializer->minGain, minGain);
+}
- result = ma_device_stop_fd__audio4(pDevice, pDevice->audio4.fdCapture);
- if (result != MA_SUCCESS) {
- return result;
- }
+MA_API float ma_spatializer_get_min_gain(const ma_spatializer* pSpatializer)
+{
+ if (pSpatializer == NULL) {
+ return 0;
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- ma_result result;
-
- /* Drain the device first. If this fails we'll just need to flush without draining. Unfortunately draining isn't available on newer version of OpenBSD. */
- #if !defined(MA_AUDIO4_USE_NEW_API)
- ioctl(pDevice->audio4.fdPlayback, AUDIO_DRAIN, 0);
- #endif
+ return c89atomic_load_f32(&pSpatializer->minGain);
+}
- /* Here is where the device is stopped immediately. */
- result = ma_device_stop_fd__audio4(pDevice, pDevice->audio4.fdPlayback);
- if (result != MA_SUCCESS) {
- return result;
- }
+MA_API void ma_spatializer_set_max_gain(ma_spatializer* pSpatializer, float maxGain)
+{
+ if (pSpatializer == NULL) {
+ return;
}
- return MA_SUCCESS;
+ c89atomic_exchange_f32(&pSpatializer->maxGain, maxGain);
}
-static ma_result ma_device_write__audio4(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten)
+MA_API float ma_spatializer_get_max_gain(const ma_spatializer* pSpatializer)
{
- int result;
-
- if (pFramesWritten != NULL) {
- *pFramesWritten = 0;
+ if (pSpatializer == NULL) {
+ return 0;
}
- result = write(pDevice->audio4.fdPlayback, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels));
- if (result < 0) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to write data to the device.", ma_result_from_errno(errno));
- }
+ return c89atomic_load_f32(&pSpatializer->maxGain);
+}
- if (pFramesWritten != NULL) {
- *pFramesWritten = (ma_uint32)result / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
+MA_API void ma_spatializer_set_min_distance(ma_spatializer* pSpatializer, float minDistance)
+{
+ if (pSpatializer == NULL) {
+ return;
}
- return MA_SUCCESS;
+ c89atomic_exchange_f32(&pSpatializer->minDistance, minDistance);
}
-static ma_result ma_device_read__audio4(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead)
+MA_API float ma_spatializer_get_min_distance(const ma_spatializer* pSpatializer)
{
- int result;
-
- if (pFramesRead != NULL) {
- *pFramesRead = 0;
+ if (pSpatializer == NULL) {
+ return 0;
}
- result = read(pDevice->audio4.fdCapture, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels));
- if (result < 0) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[audio4] Failed to read data from the device.", ma_result_from_errno(errno));
+ return c89atomic_load_f32(&pSpatializer->minDistance);
+}
+
+MA_API void ma_spatializer_set_max_distance(ma_spatializer* pSpatializer, float maxDistance)
+{
+ if (pSpatializer == NULL) {
+ return;
}
- if (pFramesRead != NULL) {
- *pFramesRead = (ma_uint32)result / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
+ c89atomic_exchange_f32(&pSpatializer->maxDistance, maxDistance);
+}
+
+MA_API float ma_spatializer_get_max_distance(const ma_spatializer* pSpatializer)
+{
+ if (pSpatializer == NULL) {
+ return 0;
}
- return MA_SUCCESS;
+ return c89atomic_load_f32(&pSpatializer->maxDistance);
}
-static ma_result ma_context_uninit__audio4(ma_context* pContext)
+MA_API void ma_spatializer_set_cone(ma_spatializer* pSpatializer, float innerAngleInRadians, float outerAngleInRadians, float outerGain)
{
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pContext->backend == ma_backend_audio4);
+ if (pSpatializer == NULL) {
+ return;
+ }
- (void)pContext;
- return MA_SUCCESS;
+ c89atomic_exchange_f32(&pSpatializer->coneInnerAngleInRadians, innerAngleInRadians);
+ c89atomic_exchange_f32(&pSpatializer->coneOuterAngleInRadians, outerAngleInRadians);
+ c89atomic_exchange_f32(&pSpatializer->coneOuterGain, outerGain);
}
-static ma_result ma_context_init__audio4(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+MA_API void ma_spatializer_get_cone(const ma_spatializer* pSpatializer, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain)
{
- MA_ASSERT(pContext != NULL);
+ if (pSpatializer == NULL) {
+ return;
+ }
- (void)pConfig;
+ if (pInnerAngleInRadians != NULL) {
+ *pInnerAngleInRadians = c89atomic_load_f32(&pSpatializer->coneInnerAngleInRadians);
+ }
- pCallbacks->onContextInit = ma_context_init__audio4;
- pCallbacks->onContextUninit = ma_context_uninit__audio4;
- pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__audio4;
- pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__audio4;
- pCallbacks->onDeviceInit = ma_device_init__audio4;
- pCallbacks->onDeviceUninit = ma_device_uninit__audio4;
- pCallbacks->onDeviceStart = ma_device_start__audio4;
- pCallbacks->onDeviceStop = ma_device_stop__audio4;
- pCallbacks->onDeviceRead = ma_device_read__audio4;
- pCallbacks->onDeviceWrite = ma_device_write__audio4;
- pCallbacks->onDeviceDataLoop = NULL;
+ if (pOuterAngleInRadians != NULL) {
+ *pOuterAngleInRadians = c89atomic_load_f32(&pSpatializer->coneOuterAngleInRadians);
+ }
- return MA_SUCCESS;
+ if (pOuterGain != NULL) {
+ *pOuterGain = c89atomic_load_f32(&pSpatializer->coneOuterGain);
+ }
}
-#endif /* audio4 */
+MA_API void ma_spatializer_set_doppler_factor(ma_spatializer* pSpatializer, float dopplerFactor)
+{
+ if (pSpatializer == NULL) {
+ return;
+ }
-/******************************************************************************
+ c89atomic_exchange_f32(&pSpatializer->dopplerFactor, dopplerFactor);
+}
-OSS Backend
+MA_API float ma_spatializer_get_doppler_factor(const ma_spatializer* pSpatializer)
+{
+ if (pSpatializer == NULL) {
+ return 1;
+ }
-******************************************************************************/
-#ifdef MA_HAS_OSS
-#include
-#include
-#include
-#include
+ return c89atomic_load_f32(&pSpatializer->dopplerFactor);
+}
-#ifndef SNDCTL_DSP_HALT
-#define SNDCTL_DSP_HALT SNDCTL_DSP_RESET
-#endif
+MA_API void ma_spatializer_set_directional_attenuation_factor(ma_spatializer* pSpatializer, float directionalAttenuationFactor)
+{
+ if (pSpatializer == NULL) {
+ return;
+ }
-#define MA_OSS_DEFAULT_DEVICE_NAME "/dev/dsp"
+ c89atomic_exchange_f32(&pSpatializer->directionalAttenuationFactor, directionalAttenuationFactor);
+}
-static int ma_open_temp_device__oss()
+MA_API float ma_spatializer_get_directional_attenuation_factor(const ma_spatializer* pSpatializer)
{
- /* The OSS sample code uses "/dev/mixer" as the device for getting system properties so I'm going to do the same. */
- int fd = open("/dev/mixer", O_RDONLY, 0);
- if (fd >= 0) {
- return fd;
+ if (pSpatializer == NULL) {
+ return 1;
}
- return -1;
+ return c89atomic_load_f32(&pSpatializer->directionalAttenuationFactor);
}
-static ma_result ma_context_open_device__oss(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, int* pfd)
+MA_API void ma_spatializer_set_position(ma_spatializer* pSpatializer, float x, float y, float z)
{
- const char* deviceName;
- int flags;
+ if (pSpatializer == NULL) {
+ return;
+ }
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pfd != NULL);
- (void)pContext;
+ pSpatializer->position = ma_vec3f_init_3f(x, y, z);
+}
- *pfd = -1;
+MA_API ma_vec3f ma_spatializer_get_position(const ma_spatializer* pSpatializer)
+{
+ if (pSpatializer == NULL) {
+ return ma_vec3f_init_3f(0, 0, 0);
+ }
- /* This function should only be called for playback or capture, not duplex. */
- if (deviceType == ma_device_type_duplex) {
- return MA_INVALID_ARGS;
+ return pSpatializer->position;
+}
+
+MA_API void ma_spatializer_set_direction(ma_spatializer* pSpatializer, float x, float y, float z)
+{
+ if (pSpatializer == NULL) {
+ return;
}
- deviceName = MA_OSS_DEFAULT_DEVICE_NAME;
- if (pDeviceID != NULL) {
- deviceName = pDeviceID->oss;
+ pSpatializer->direction = ma_vec3f_init_3f(x, y, z);
+}
+
+MA_API ma_vec3f ma_spatializer_get_direction(const ma_spatializer* pSpatializer)
+{
+ if (pSpatializer == NULL) {
+ return ma_vec3f_init_3f(0, 0, -1);
}
- flags = (deviceType == ma_device_type_playback) ? O_WRONLY : O_RDONLY;
- if (shareMode == ma_share_mode_exclusive) {
- flags |= O_EXCL;
+ return pSpatializer->direction;
+}
+
+MA_API void ma_spatializer_set_velocity(ma_spatializer* pSpatializer, float x, float y, float z)
+{
+ if (pSpatializer == NULL) {
+ return;
}
- *pfd = open(deviceName, flags, 0);
- if (*pfd == -1) {
- return ma_result_from_errno(errno);
+ pSpatializer->velocity = ma_vec3f_init_3f(x, y, z);
+}
+
+MA_API ma_vec3f ma_spatializer_get_velocity(const ma_spatializer* pSpatializer)
+{
+ if (pSpatializer == NULL) {
+ return ma_vec3f_init_3f(0, 0, 0);
}
- return MA_SUCCESS;
+ return pSpatializer->velocity;
}
-static ma_result ma_context_enumerate_devices__oss(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+MA_API void ma_spatializer_get_relative_position_and_direction(const ma_spatializer* pSpatializer, const ma_spatializer_listener* pListener, ma_vec3f* pRelativePos, ma_vec3f* pRelativeDir)
{
- int fd;
- oss_sysinfo si;
- int result;
+ if (pRelativePos != NULL) {
+ pRelativePos->x = 0;
+ pRelativePos->y = 0;
+ pRelativePos->z = 0;
+ }
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(callback != NULL);
+ if (pRelativeDir != NULL) {
+ pRelativeDir->x = 0;
+ pRelativeDir->y = 0;
+ pRelativeDir->z = -1;
+ }
- fd = ma_open_temp_device__oss();
- if (fd == -1) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[OSS] Failed to open a temporary device for retrieving system information used for device enumeration.", MA_NO_BACKEND);
+ if (pSpatializer == NULL) {
+ return;
}
- result = ioctl(fd, SNDCTL_SYSINFO, &si);
- if (result != -1) {
- int iAudioDevice;
- for (iAudioDevice = 0; iAudioDevice < si.numaudios; ++iAudioDevice) {
- oss_audioinfo ai;
- ai.dev = iAudioDevice;
- result = ioctl(fd, SNDCTL_AUDIOINFO, &ai);
- if (result != -1) {
- if (ai.devnode[0] != '\0') { /* <-- Can be blank, according to documentation. */
- ma_device_info deviceInfo;
- ma_bool32 isTerminating = MA_FALSE;
+ if (pListener == NULL || ma_spatializer_get_positioning(pSpatializer) == ma_positioning_relative) {
+ /* There's no listener or we're using relative positioning. */
+ if (pRelativePos != NULL) {
+ *pRelativePos = pSpatializer->position;
+ }
+ if (pRelativeDir != NULL) {
+ *pRelativeDir = pSpatializer->direction;
+ }
+ } else {
+ ma_vec3f v;
+ ma_vec3f axisX;
+ ma_vec3f axisY;
+ ma_vec3f axisZ;
+ float m[4][4];
- MA_ZERO_OBJECT(&deviceInfo);
+ /*
+ We need to calcualte the right vector from our forward and up vectors. This is done with
+ a cross product.
+ */
+ axisZ = ma_vec3f_normalize(pListener->direction); /* Normalization required here because we can't trust the caller. */
+ axisX = ma_vec3f_normalize(ma_vec3f_cross(axisZ, pListener->config.worldUp)); /* Normalization required here because the world up vector may not be perpendicular with the forward vector. */
- /* ID */
- ma_strncpy_s(deviceInfo.id.oss, sizeof(deviceInfo.id.oss), ai.devnode, (size_t)-1);
+ /*
+ The calculation of axisX above can result in a zero-length vector if the listener is
+ looking straight up on the Y axis. We'll need to fall back to a +X in this case so that
+ the calculations below don't fall apart. This is where a quaternion based listener and
+ sound orientation would come in handy.
+ */
+ if (ma_vec3f_len2(axisX) == 0) {
+ axisX = ma_vec3f_init_3f(1, 0, 0);
+ }
- /*
- The human readable device name should be in the "ai.handle" variable, but it can
- sometimes be empty in which case we just fall back to "ai.name" which is less user
- friendly, but usually has a value.
- */
- if (ai.handle[0] != '\0') {
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), ai.handle, (size_t)-1);
- } else {
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), ai.name, (size_t)-1);
- }
+ axisY = ma_vec3f_cross(axisX, axisZ); /* No normalization is required here because axisX and axisZ are unit length and perpendicular. */
- /* The device can be both playback and capture. */
- if (!isTerminating && (ai.caps & PCM_CAP_OUTPUT) != 0) {
- isTerminating = !callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
- }
- if (!isTerminating && (ai.caps & PCM_CAP_INPUT) != 0) {
- isTerminating = !callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
- }
+ /*
+ We need to swap the X axis if we're left handed because otherwise the cross product above
+ will have resulted in it pointing in the wrong direction (right handed was assumed in the
+ cross products above).
+ */
+ if (pListener->config.handedness == ma_handedness_left) {
+ axisX = ma_vec3f_neg(axisX);
+ }
- if (isTerminating) {
- break;
- }
- }
- }
+ /* Lookat. */
+ m[0][0] = axisX.x; m[1][0] = axisX.y; m[2][0] = axisX.z; m[3][0] = -ma_vec3f_dot(axisX, pListener->position);
+ m[0][1] = axisY.x; m[1][1] = axisY.y; m[2][1] = axisY.z; m[3][1] = -ma_vec3f_dot(axisY, pListener->position);
+ m[0][2] = -axisZ.x; m[1][2] = -axisZ.y; m[2][2] = -axisZ.z; m[3][2] = -ma_vec3f_dot(ma_vec3f_neg(axisZ), pListener->position);
+ m[0][3] = 0; m[1][3] = 0; m[2][3] = 0; m[3][3] = 1;
+
+ /*
+ Multiply the lookat matrix by the spatializer position to transform it to listener
+ space. This allows calculations to work based on the sound being relative to the
+ origin which makes things simpler.
+ */
+ if (pRelativePos != NULL) {
+ v = pSpatializer->position;
+ pRelativePos->x = m[0][0] * v.x + m[1][0] * v.y + m[2][0] * v.z + m[3][0] * 1;
+ pRelativePos->y = m[0][1] * v.x + m[1][1] * v.y + m[2][1] * v.z + m[3][1] * 1;
+ pRelativePos->z = m[0][2] * v.x + m[1][2] * v.y + m[2][2] * v.z + m[3][2] * 1;
}
- } else {
- close(fd);
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[OSS] Failed to retrieve system information for device enumeration.", MA_NO_BACKEND);
- }
- close(fd);
- return MA_SUCCESS;
+ /*
+ The direction of the sound needs to also be transformed so that it's relative to the
+ rotation of the listener.
+ */
+ if (pRelativeDir != NULL) {
+ v = pSpatializer->direction;
+ pRelativeDir->x = m[0][0] * v.x + m[1][0] * v.y + m[2][0] * v.z;
+ pRelativeDir->y = m[0][1] * v.x + m[1][1] * v.y + m[2][1] * v.z;
+ pRelativeDir->z = m[0][2] * v.x + m[1][2] * v.y + m[2][2] * v.z;
+ }
+ }
}
-static void ma_context_add_native_data_format__oss(ma_context* pContext, oss_audioinfo* pAudioInfo, ma_format format, ma_device_info* pDeviceInfo)
+
+
+
+/**************************************************************************************************************************************************************
+
+Resampling
+
+**************************************************************************************************************************************************************/
+MA_API ma_linear_resampler_config ma_linear_resampler_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut)
{
- unsigned int minChannels;
- unsigned int maxChannels;
- unsigned int iRate;
+ ma_linear_resampler_config config;
+ MA_ZERO_OBJECT(&config);
+ config.format = format;
+ config.channels = channels;
+ config.sampleRateIn = sampleRateIn;
+ config.sampleRateOut = sampleRateOut;
+ config.lpfOrder = ma_min(MA_DEFAULT_RESAMPLER_LPF_ORDER, MA_MAX_FILTER_ORDER);
+ config.lpfNyquistFactor = 1;
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pAudioInfo != NULL);
- MA_ASSERT(pDeviceInfo != NULL);
+ return config;
+}
- /* If we support all channels we just report 0. */
- minChannels = ma_clamp(pAudioInfo->min_channels, MA_MIN_CHANNELS, MA_MAX_CHANNELS);
- maxChannels = ma_clamp(pAudioInfo->max_channels, MA_MIN_CHANNELS, MA_MAX_CHANNELS);
+typedef struct
+{
+ size_t sizeInBytes;
+ size_t x0Offset;
+ size_t x1Offset;
+ size_t lpfOffset;
+} ma_linear_resampler_heap_layout;
+
+
+static void ma_linear_resampler_adjust_timer_for_new_rate(ma_linear_resampler* pResampler, ma_uint32 oldSampleRateOut, ma_uint32 newSampleRateOut)
+{
/*
- OSS has this annoying thing where sample rates can be reported in two ways. We prefer explicitness,
- which OSS has in the form of nrates/rates, however there are times where nrates can be 0, in which
- case we'll need to use min_rate and max_rate and report only standard rates.
+ So what's happening here? Basically we need to adjust the fractional component of the time advance based on the new rate. The old time advance will
+ be based on the old sample rate, but we are needing to adjust it to that it's based on the new sample rate.
*/
- if (pAudioInfo->nrates > 0) {
- for (iRate = 0; iRate < pAudioInfo->nrates; iRate += 1) {
- unsigned int rate = pAudioInfo->rates[iRate];
-
- if (minChannels == MA_MIN_CHANNELS && maxChannels == MA_MAX_CHANNELS) {
- ma_device_info_add_native_data_format(pDeviceInfo, format, 0, rate, 0); /* Set the channel count to 0 to indicate that all channel counts are supported. */
- } else {
- unsigned int iChannel;
- for (iChannel = minChannels; iChannel <= maxChannels; iChannel += 1) {
- ma_device_info_add_native_data_format(pDeviceInfo, format, iChannel, rate, 0);
- }
- }
- }
- } else {
- for (iRate = 0; iRate < ma_countof(g_maStandardSampleRatePriorities); iRate += 1) {
- ma_uint32 standardRate = g_maStandardSampleRatePriorities[iRate];
+ ma_uint32 oldRateTimeWhole = pResampler->inTimeFrac / oldSampleRateOut; /* <-- This should almost never be anything other than 0, but leaving it here to make this more general and robust just in case. */
+ ma_uint32 oldRateTimeFract = pResampler->inTimeFrac % oldSampleRateOut;
- if (standardRate >= (ma_uint32)pAudioInfo->min_rate && standardRate <= (ma_uint32)pAudioInfo->max_rate) {
- if (minChannels == MA_MIN_CHANNELS && maxChannels == MA_MAX_CHANNELS) {
- ma_device_info_add_native_data_format(pDeviceInfo, format, 0, standardRate, 0); /* Set the channel count to 0 to indicate that all channel counts are supported. */
- } else {
- unsigned int iChannel;
- for (iChannel = minChannels; iChannel <= maxChannels; iChannel += 1) {
- ma_device_info_add_native_data_format(pDeviceInfo, format, iChannel, standardRate, 0);
- }
- }
- }
- }
- }
+ pResampler->inTimeFrac =
+ (oldRateTimeWhole * newSampleRateOut) +
+ ((oldRateTimeFract * newSampleRateOut) / oldSampleRateOut);
+
+ /* Make sure the fractional part is less than the output sample rate. */
+ pResampler->inTimeInt += pResampler->inTimeFrac / pResampler->config.sampleRateOut;
+ pResampler->inTimeFrac = pResampler->inTimeFrac % pResampler->config.sampleRateOut;
}
-static ma_result ma_context_get_device_info__oss(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+static ma_result ma_linear_resampler_set_rate_internal(ma_linear_resampler* pResampler, void* pHeap, ma_linear_resampler_heap_layout* pHeapLayout, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut, ma_bool32 isResamplerAlreadyInitialized)
{
- ma_bool32 foundDevice;
- int fdTemp;
- oss_sysinfo si;
- int result;
-
- MA_ASSERT(pContext != NULL);
-
- /* Handle the default device a little differently. */
- if (pDeviceID == NULL) {
- if (deviceType == ma_device_type_playback) {
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
- } else {
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
- }
+ ma_result result;
+ ma_uint32 gcf;
+ ma_uint32 lpfSampleRate;
+ double lpfCutoffFrequency;
+ ma_lpf_config lpfConfig;
+ ma_uint32 oldSampleRateOut; /* Required for adjusting time advance down the bottom. */
- return MA_SUCCESS;
+ if (pResampler == NULL) {
+ return MA_INVALID_ARGS;
}
-
- /* If we get here it means we are _not_ using the default device. */
- foundDevice = MA_FALSE;
-
- fdTemp = ma_open_temp_device__oss();
- if (fdTemp == -1) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[OSS] Failed to open a temporary device for retrieving system information used for device enumeration.", MA_NO_BACKEND);
+ if (sampleRateIn == 0 || sampleRateOut == 0) {
+ return MA_INVALID_ARGS;
}
- result = ioctl(fdTemp, SNDCTL_SYSINFO, &si);
- if (result != -1) {
- int iAudioDevice;
- for (iAudioDevice = 0; iAudioDevice < si.numaudios; ++iAudioDevice) {
- oss_audioinfo ai;
- ai.dev = iAudioDevice;
- result = ioctl(fdTemp, SNDCTL_AUDIOINFO, &ai);
- if (result != -1) {
- if (ma_strcmp(ai.devnode, pDeviceID->oss) == 0) {
- /* It has the same name, so now just confirm the type. */
- if ((deviceType == ma_device_type_playback && ((ai.caps & PCM_CAP_OUTPUT) != 0)) ||
- (deviceType == ma_device_type_capture && ((ai.caps & PCM_CAP_INPUT) != 0))) {
- unsigned int formatMask;
-
- /* ID */
- ma_strncpy_s(pDeviceInfo->id.oss, sizeof(pDeviceInfo->id.oss), ai.devnode, (size_t)-1);
+ oldSampleRateOut = pResampler->config.sampleRateOut;
- /*
- The human readable device name should be in the "ai.handle" variable, but it can
- sometimes be empty in which case we just fall back to "ai.name" which is less user
- friendly, but usually has a value.
- */
- if (ai.handle[0] != '\0') {
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), ai.handle, (size_t)-1);
- } else {
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), ai.name, (size_t)-1);
- }
+ pResampler->config.sampleRateIn = sampleRateIn;
+ pResampler->config.sampleRateOut = sampleRateOut;
+ /* Simplify the sample rate. */
+ gcf = ma_gcf_u32(pResampler->config.sampleRateIn, pResampler->config.sampleRateOut);
+ pResampler->config.sampleRateIn /= gcf;
+ pResampler->config.sampleRateOut /= gcf;
- pDeviceInfo->nativeDataFormatCount = 0;
+ /* Always initialize the low-pass filter, even when the order is 0. */
+ if (pResampler->config.lpfOrder > MA_MAX_FILTER_ORDER) {
+ return MA_INVALID_ARGS;
+ }
- if (deviceType == ma_device_type_playback) {
- formatMask = ai.oformats;
- } else {
- formatMask = ai.iformats;
- }
+ lpfSampleRate = (ma_uint32)(ma_max(pResampler->config.sampleRateIn, pResampler->config.sampleRateOut));
+ lpfCutoffFrequency = ( double)(ma_min(pResampler->config.sampleRateIn, pResampler->config.sampleRateOut) * 0.5 * pResampler->config.lpfNyquistFactor);
- if (((formatMask & AFMT_S16_LE) != 0 && ma_is_little_endian()) || (AFMT_S16_BE && ma_is_big_endian())) {
- ma_context_add_native_data_format__oss(pContext, &ai, ma_format_s16, pDeviceInfo);
- }
- if (((formatMask & AFMT_S32_LE) != 0 && ma_is_little_endian()) || (AFMT_S32_BE && ma_is_big_endian())) {
- ma_context_add_native_data_format__oss(pContext, &ai, ma_format_s32, pDeviceInfo);
- }
- if ((formatMask & AFMT_U8) != 0) {
- ma_context_add_native_data_format__oss(pContext, &ai, ma_format_u8, pDeviceInfo);
- }
+ lpfConfig = ma_lpf_config_init(pResampler->config.format, pResampler->config.channels, lpfSampleRate, lpfCutoffFrequency, pResampler->config.lpfOrder);
- foundDevice = MA_TRUE;
- break;
- }
- }
- }
- }
+ /*
+ If the resampler is alreay initialized we don't want to do a fresh initialization of the low-pass filter because it will result in the cached frames
+ getting cleared. Instead we re-initialize the filter which will maintain any cached frames.
+ */
+ if (isResamplerAlreadyInitialized) {
+ result = ma_lpf_reinit(&lpfConfig, &pResampler->lpf);
} else {
- close(fdTemp);
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[OSS] Failed to retrieve system information for device enumeration.", MA_NO_BACKEND);
+ result = ma_lpf_init_preallocated(&lpfConfig, ma_offset_ptr(pHeap, pHeapLayout->lpfOffset), &pResampler->lpf);
}
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- close(fdTemp);
- if (!foundDevice) {
- return MA_NO_DEVICE;
- }
+ pResampler->inAdvanceInt = pResampler->config.sampleRateIn / pResampler->config.sampleRateOut;
+ pResampler->inAdvanceFrac = pResampler->config.sampleRateIn % pResampler->config.sampleRateOut;
+
+ /* Our timer was based on the old rate. We need to adjust it so that it's based on the new rate. */
+ ma_linear_resampler_adjust_timer_for_new_rate(pResampler, oldSampleRateOut, pResampler->config.sampleRateOut);
return MA_SUCCESS;
}
-static ma_result ma_device_uninit__oss(ma_device* pDevice)
+static ma_result ma_linear_resampler_get_heap_layout(const ma_linear_resampler_config* pConfig, ma_linear_resampler_heap_layout* pHeapLayout)
{
- MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(pHeapLayout != NULL);
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- close(pDevice->oss.fdCapture);
- }
+ MA_ZERO_OBJECT(pHeapLayout);
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- close(pDevice->oss.fdPlayback);
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- return MA_SUCCESS;
-}
+ if (pConfig->format != ma_format_f32 && pConfig->format != ma_format_s16) {
+ return MA_INVALID_ARGS;
+ }
-static int ma_format_to_oss(ma_format format)
-{
- int ossFormat = AFMT_U8;
- switch (format) {
- case ma_format_s16: ossFormat = (ma_is_little_endian()) ? AFMT_S16_LE : AFMT_S16_BE; break;
- case ma_format_s24: ossFormat = (ma_is_little_endian()) ? AFMT_S32_LE : AFMT_S32_BE; break;
- case ma_format_s32: ossFormat = (ma_is_little_endian()) ? AFMT_S32_LE : AFMT_S32_BE; break;
- case ma_format_f32: ossFormat = (ma_is_little_endian()) ? AFMT_S16_LE : AFMT_S16_BE; break;
- case ma_format_u8:
- default: ossFormat = AFMT_U8; break;
+ if (pConfig->channels == 0) {
+ return MA_INVALID_ARGS;
}
- return ossFormat;
-}
+ pHeapLayout->sizeInBytes = 0;
-static ma_format ma_format_from_oss(int ossFormat)
-{
- if (ossFormat == AFMT_U8) {
- return ma_format_u8;
+ /* x0 */
+ pHeapLayout->x0Offset = pHeapLayout->sizeInBytes;
+ if (pConfig->format == ma_format_f32) {
+ pHeapLayout->sizeInBytes += sizeof(float) * pConfig->channels;
} else {
- if (ma_is_little_endian()) {
- switch (ossFormat) {
- case AFMT_S16_LE: return ma_format_s16;
- case AFMT_S32_LE: return ma_format_s32;
- default: return ma_format_unknown;
- }
- } else {
- switch (ossFormat) {
- case AFMT_S16_BE: return ma_format_s16;
- case AFMT_S32_BE: return ma_format_s32;
- default: return ma_format_unknown;
- }
+ pHeapLayout->sizeInBytes += sizeof(ma_int16) * pConfig->channels;
+ }
+
+ /* x1 */
+ pHeapLayout->x1Offset = pHeapLayout->sizeInBytes;
+ if (pConfig->format == ma_format_f32) {
+ pHeapLayout->sizeInBytes += sizeof(float) * pConfig->channels;
+ } else {
+ pHeapLayout->sizeInBytes += sizeof(ma_int16) * pConfig->channels;
+ }
+
+ /* LPF */
+ pHeapLayout->lpfOffset = pHeapLayout->sizeInBytes;
+ {
+ ma_result result;
+ size_t lpfHeapSizeInBytes;
+ ma_lpf_config lpfConfig = ma_lpf_config_init(pConfig->format, pConfig->channels, 1, 1, pConfig->lpfOrder); /* Sample rate and cutoff frequency do not matter. */
+
+ result = ma_lpf_get_heap_size(&lpfConfig, &lpfHeapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
}
+
+ pHeapLayout->sizeInBytes += lpfHeapSizeInBytes;
}
- return ma_format_unknown;
+ /* Make sure allocation size is aligned. */
+ pHeapLayout->sizeInBytes = ma_align_64(pHeapLayout->sizeInBytes);
+
+ return MA_SUCCESS;
}
-static ma_result ma_device_init_fd__oss(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptor, ma_device_type deviceType)
+MA_API ma_result ma_linear_resampler_get_heap_size(const ma_linear_resampler_config* pConfig, size_t* pHeapSizeInBytes)
{
ma_result result;
- int ossResult;
- int fd;
- const ma_device_id* pDeviceID = NULL;
- ma_share_mode shareMode;
- int ossFormat;
- int ossChannels;
- int ossSampleRate;
- int ossFragment;
+ ma_linear_resampler_heap_layout heapLayout;
- MA_ASSERT(pDevice != NULL);
- MA_ASSERT(pConfig != NULL);
- MA_ASSERT(deviceType != ma_device_type_duplex);
+ if (pHeapSizeInBytes == NULL) {
+ return MA_INVALID_ARGS;
+ }
- pDeviceID = pDescriptor->pDeviceID;
- shareMode = pDescriptor->shareMode;
- ossFormat = ma_format_to_oss((pDescriptor->format != ma_format_unknown) ? pDescriptor->format : ma_format_s16); /* Use s16 by default because OSS doesn't like floating point. */
- ossChannels = (int)(pDescriptor->channels > 0) ? pDescriptor->channels : MA_DEFAULT_CHANNELS;
- ossSampleRate = (int)(pDescriptor->sampleRate > 0) ? pDescriptor->sampleRate : MA_DEFAULT_SAMPLE_RATE;
+ *pHeapSizeInBytes = 0;
- result = ma_context_open_device__oss(pDevice->pContext, deviceType, pDeviceID, shareMode, &fd);
+ result = ma_linear_resampler_get_heap_layout(pConfig, &heapLayout);
if (result != MA_SUCCESS) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to open device.", result);
+ return result;
}
- /*
- The OSS documantation is very clear about the order we should be initializing the device's properties:
- 1) Format
- 2) Channels
- 3) Sample rate.
- */
+ *pHeapSizeInBytes = heapLayout.sizeInBytes;
- /* Format. */
- ossResult = ioctl(fd, SNDCTL_DSP_SETFMT, &ossFormat);
- if (ossResult == -1) {
- close(fd);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to set format.", MA_FORMAT_NOT_SUPPORTED);
- }
+ return MA_SUCCESS;
+}
- /* Channels. */
- ossResult = ioctl(fd, SNDCTL_DSP_CHANNELS, &ossChannels);
- if (ossResult == -1) {
- close(fd);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to set channel count.", MA_FORMAT_NOT_SUPPORTED);
- }
+MA_API ma_result ma_linear_resampler_init_preallocated(const ma_linear_resampler_config* pConfig, void* pHeap, ma_linear_resampler* pResampler)
+{
+ ma_result result;
+ ma_linear_resampler_heap_layout heapLayout;
- /* Sample Rate. */
- ossResult = ioctl(fd, SNDCTL_DSP_SPEED, &ossSampleRate);
- if (ossResult == -1) {
- close(fd);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to set sample rate.", MA_FORMAT_NOT_SUPPORTED);
+ if (pResampler == NULL) {
+ return MA_INVALID_ARGS;
}
- /*
- Buffer.
-
- The documentation says that the fragment settings should be set as soon as possible, but I'm not sure if
- it should be done before or after format/channels/rate.
-
- OSS wants the fragment size in bytes and a power of 2. When setting, we specify the power, not the actual
- value.
- */
- {
- ma_uint32 periodSizeInFrames;
- ma_uint32 periodSizeInBytes;
- ma_uint32 ossFragmentSizePower;
-
- periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, (ma_uint32)ossSampleRate, pConfig->performanceProfile);
+ MA_ZERO_OBJECT(pResampler);
- periodSizeInBytes = ma_round_to_power_of_2(periodSizeInFrames * ma_get_bytes_per_frame(ma_format_from_oss(ossFormat), ossChannels));
- if (periodSizeInBytes < 16) {
- periodSizeInBytes = 16;
- }
+ result = ma_linear_resampler_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- ossFragmentSizePower = 4;
- periodSizeInBytes >>= 4;
- while (periodSizeInBytes >>= 1) {
- ossFragmentSizePower += 1;
- }
+ pResampler->config = *pConfig;
- ossFragment = (int)((pConfig->periods << 16) | ossFragmentSizePower);
- ossResult = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &ossFragment);
- if (ossResult == -1) {
- close(fd);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to set fragment size and period count.", MA_FORMAT_NOT_SUPPORTED);
- }
- }
+ pResampler->_pHeap = pHeap;
+ MA_ZERO_MEMORY(pHeap, heapLayout.sizeInBytes);
- /* Internal settings. */
- if (deviceType == ma_device_type_capture) {
- pDevice->oss.fdCapture = fd;
+ if (pConfig->format == ma_format_f32) {
+ pResampler->x0.f32 = (float*)ma_offset_ptr(pHeap, heapLayout.x0Offset);
+ pResampler->x1.f32 = (float*)ma_offset_ptr(pHeap, heapLayout.x1Offset);
} else {
- pDevice->oss.fdPlayback = fd;
+ pResampler->x0.s16 = (ma_int16*)ma_offset_ptr(pHeap, heapLayout.x0Offset);
+ pResampler->x1.s16 = (ma_int16*)ma_offset_ptr(pHeap, heapLayout.x1Offset);
}
- pDescriptor->format = ma_format_from_oss(ossFormat);
- pDescriptor->channels = ossChannels;
- pDescriptor->sampleRate = ossSampleRate;
- ma_get_standard_channel_map(ma_standard_channel_map_sound4, pDescriptor->channels, pDescriptor->channelMap);
- pDescriptor->periodCount = (ma_uint32)(ossFragment >> 16);
- pDescriptor->periodSizeInFrames = (ma_uint32)(1 << (ossFragment & 0xFFFF)) / ma_get_bytes_per_frame(pDescriptor->format, pDescriptor->channels);
-
- if (pDescriptor->format == ma_format_unknown) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] The device's internal format is not supported by miniaudio.", MA_FORMAT_NOT_SUPPORTED);
+ /* Setting the rate will set up the filter and time advances for us. */
+ result = ma_linear_resampler_set_rate_internal(pResampler, pHeap, &heapLayout, pConfig->sampleRateIn, pConfig->sampleRateOut, /* isResamplerAlreadyInitialized = */ MA_FALSE);
+ if (result != MA_SUCCESS) {
+ return result;
}
+ pResampler->inTimeInt = 1; /* Set this to one to force an input sample to always be loaded for the first output frame. */
+ pResampler->inTimeFrac = 0;
+
return MA_SUCCESS;
}
-static ma_result ma_device_init__oss(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+MA_API ma_result ma_linear_resampler_init(const ma_linear_resampler_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_linear_resampler* pResampler)
{
- MA_ASSERT(pDevice != NULL);
- MA_ASSERT(pConfig != NULL);
-
- MA_ZERO_OBJECT(&pDevice->oss);
+ ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
- if (pConfig->deviceType == ma_device_type_loopback) {
- return MA_DEVICE_TYPE_NOT_SUPPORTED;
+ result = ma_linear_resampler_get_heap_size(pConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
}
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
- ma_result result = ma_device_init_fd__oss(pDevice, pConfig, pDescriptorCapture, ma_device_type_capture);
- if (result != MA_SUCCESS) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to open device.", result);
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
}
+ } else {
+ pHeap = NULL;
}
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- ma_result result = ma_device_init_fd__oss(pDevice, pConfig, pDescriptorPlayback, ma_device_type_playback);
- if (result != MA_SUCCESS) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to open device.", result);
- }
+ result = ma_linear_resampler_init_preallocated(pConfig, pHeap, pResampler);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
}
+ pResampler->_ownsHeap = MA_TRUE;
return MA_SUCCESS;
}
-/*
-Note on Starting and Stopping
-=============================
-In the past I was using SNDCTL_DSP_HALT to stop the device, however this results in issues when
-trying to resume the device again. If we use SNDCTL_DSP_HALT, the next write() or read() will
-fail. Instead what we need to do is just not write or read to and from the device when the
-device is not running.
+MA_API void ma_linear_resampler_uninit(ma_linear_resampler* pResampler, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pResampler == NULL) {
+ return;
+ }
-As a result, both the start and stop functions for OSS are just empty stubs. The starting and
-stopping logic is handled by ma_device_write__oss() and ma_device_read__oss(). These will check
-the device state, and if the device is stopped they will simply not do any kind of processing.
+ ma_lpf_uninit(&pResampler->lpf, pAllocationCallbacks);
-The downside to this technique is that I've noticed a fairly lengthy delay in stopping the
-device, up to a second. This is on a virtual machine, and as such might just be due to the
-virtual drivers, but I'm not fully sure. I am not sure how to work around this problem so for
-the moment that's just how it's going to have to be.
+ if (pResampler->_ownsHeap) {
+ ma_free(pResampler->_pHeap, pAllocationCallbacks);
+ }
+}
-When starting the device, OSS will automatically start it when write() or read() is called.
-*/
-static ma_result ma_device_start__oss(ma_device* pDevice)
+static MA_INLINE ma_int16 ma_linear_resampler_mix_s16(ma_int16 x, ma_int16 y, ma_int32 a, const ma_int32 shift)
{
- MA_ASSERT(pDevice != NULL);
+ ma_int32 b;
+ ma_int32 c;
+ ma_int32 r;
- /* The device is automatically started with reading and writing. */
- (void)pDevice;
+ MA_ASSERT(a <= (1<> shift);
}
-static ma_result ma_device_stop__oss(ma_device* pDevice)
+static void ma_linear_resampler_interpolate_frame_s16(ma_linear_resampler* pResampler, ma_int16* MA_RESTRICT pFrameOut)
{
- MA_ASSERT(pDevice != NULL);
+ ma_uint32 c;
+ ma_uint32 a;
+ const ma_uint32 channels = pResampler->config.channels;
+ const ma_uint32 shift = 12;
- /* See note above on why this is empty. */
- (void)pDevice;
+ MA_ASSERT(pResampler != NULL);
+ MA_ASSERT(pFrameOut != NULL);
- return MA_SUCCESS;
+ a = (pResampler->inTimeFrac << shift) / pResampler->config.sampleRateOut;
+
+ MA_ASSUME(channels > 0);
+ for (c = 0; c < channels; c += 1) {
+ ma_int16 s = ma_linear_resampler_mix_s16(pResampler->x0.s16[c], pResampler->x1.s16[c], a, shift);
+ pFrameOut[c] = s;
+ }
}
-static ma_result ma_device_write__oss(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten)
+
+static void ma_linear_resampler_interpolate_frame_f32(ma_linear_resampler* pResampler, float* MA_RESTRICT pFrameOut)
{
- int resultOSS;
- ma_uint32 deviceState;
+ ma_uint32 c;
+ float a;
+ const ma_uint32 channels = pResampler->config.channels;
- if (pFramesWritten != NULL) {
- *pFramesWritten = 0;
- }
+ MA_ASSERT(pResampler != NULL);
+ MA_ASSERT(pFrameOut != NULL);
- /* Don't do any processing if the device is stopped. */
- deviceState = ma_device_get_state(pDevice);
- if (deviceState != MA_STATE_STARTED && deviceState != MA_STATE_STARTING) {
- return MA_SUCCESS;
- }
+ a = (float)pResampler->inTimeFrac / pResampler->config.sampleRateOut;
- resultOSS = write(pDevice->oss.fdPlayback, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels));
- if (resultOSS < 0) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to send data from the client to the device.", ma_result_from_errno(errno));
+ MA_ASSUME(channels > 0);
+ for (c = 0; c < channels; c += 1) {
+ float s = ma_mix_f32_fast(pResampler->x0.f32[c], pResampler->x1.f32[c], a);
+ pFrameOut[c] = s;
}
+}
- if (pFramesWritten != NULL) {
- *pFramesWritten = (ma_uint32)resultOSS / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
+static ma_result ma_linear_resampler_process_pcm_frames_s16_downsample(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+{
+ const ma_int16* pFramesInS16;
+ /* */ ma_int16* pFramesOutS16;
+ ma_uint64 frameCountIn;
+ ma_uint64 frameCountOut;
+ ma_uint64 framesProcessedIn;
+ ma_uint64 framesProcessedOut;
+
+ MA_ASSERT(pResampler != NULL);
+ MA_ASSERT(pFrameCountIn != NULL);
+ MA_ASSERT(pFrameCountOut != NULL);
+
+ pFramesInS16 = (const ma_int16*)pFramesIn;
+ pFramesOutS16 = ( ma_int16*)pFramesOut;
+ frameCountIn = *pFrameCountIn;
+ frameCountOut = *pFrameCountOut;
+ framesProcessedIn = 0;
+ framesProcessedOut = 0;
+
+ while (framesProcessedOut < frameCountOut) {
+ /* Before interpolating we need to load the buffers. When doing this we need to ensure we run every input sample through the filter. */
+ while (pResampler->inTimeInt > 0 && frameCountIn > framesProcessedIn) {
+ ma_uint32 iChannel;
+
+ if (pFramesInS16 != NULL) {
+ for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
+ pResampler->x0.s16[iChannel] = pResampler->x1.s16[iChannel];
+ pResampler->x1.s16[iChannel] = pFramesInS16[iChannel];
+ }
+ pFramesInS16 += pResampler->config.channels;
+ } else {
+ for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
+ pResampler->x0.s16[iChannel] = pResampler->x1.s16[iChannel];
+ pResampler->x1.s16[iChannel] = 0;
+ }
+ }
+
+ /* Filter. */
+ ma_lpf_process_pcm_frame_s16(&pResampler->lpf, pResampler->x1.s16, pResampler->x1.s16);
+
+ framesProcessedIn += 1;
+ pResampler->inTimeInt -= 1;
+ }
+
+ if (pResampler->inTimeInt > 0) {
+ break; /* Ran out of input data. */
+ }
+
+ /* Getting here means the frames have been loaded and filtered and we can generate the next output frame. */
+ if (pFramesOutS16 != NULL) {
+ MA_ASSERT(pResampler->inTimeInt == 0);
+ ma_linear_resampler_interpolate_frame_s16(pResampler, pFramesOutS16);
+
+ pFramesOutS16 += pResampler->config.channels;
+ }
+
+ framesProcessedOut += 1;
+
+ /* Advance time forward. */
+ pResampler->inTimeInt += pResampler->inAdvanceInt;
+ pResampler->inTimeFrac += pResampler->inAdvanceFrac;
+ if (pResampler->inTimeFrac >= pResampler->config.sampleRateOut) {
+ pResampler->inTimeFrac -= pResampler->config.sampleRateOut;
+ pResampler->inTimeInt += 1;
+ }
}
+ *pFrameCountIn = framesProcessedIn;
+ *pFrameCountOut = framesProcessedOut;
+
return MA_SUCCESS;
}
-static ma_result ma_device_read__oss(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead)
+static ma_result ma_linear_resampler_process_pcm_frames_s16_upsample(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
- int resultOSS;
- ma_uint32 deviceState;
+ const ma_int16* pFramesInS16;
+ /* */ ma_int16* pFramesOutS16;
+ ma_uint64 frameCountIn;
+ ma_uint64 frameCountOut;
+ ma_uint64 framesProcessedIn;
+ ma_uint64 framesProcessedOut;
- if (pFramesRead != NULL) {
- *pFramesRead = 0;
- }
+ MA_ASSERT(pResampler != NULL);
+ MA_ASSERT(pFrameCountIn != NULL);
+ MA_ASSERT(pFrameCountOut != NULL);
- /* Don't do any processing if the device is stopped. */
- deviceState = ma_device_get_state(pDevice);
- if (deviceState != MA_STATE_STARTED && deviceState != MA_STATE_STARTING) {
- return MA_SUCCESS;
- }
+ pFramesInS16 = (const ma_int16*)pFramesIn;
+ pFramesOutS16 = ( ma_int16*)pFramesOut;
+ frameCountIn = *pFrameCountIn;
+ frameCountOut = *pFrameCountOut;
+ framesProcessedIn = 0;
+ framesProcessedOut = 0;
+
+ while (framesProcessedOut < frameCountOut) {
+ /* Before interpolating we need to load the buffers. */
+ while (pResampler->inTimeInt > 0 && frameCountIn > framesProcessedIn) {
+ ma_uint32 iChannel;
+
+ if (pFramesInS16 != NULL) {
+ for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
+ pResampler->x0.s16[iChannel] = pResampler->x1.s16[iChannel];
+ pResampler->x1.s16[iChannel] = pFramesInS16[iChannel];
+ }
+ pFramesInS16 += pResampler->config.channels;
+ } else {
+ for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
+ pResampler->x0.s16[iChannel] = pResampler->x1.s16[iChannel];
+ pResampler->x1.s16[iChannel] = 0;
+ }
+ }
+
+ framesProcessedIn += 1;
+ pResampler->inTimeInt -= 1;
+ }
+
+ if (pResampler->inTimeInt > 0) {
+ break; /* Ran out of input data. */
+ }
+
+ /* Getting here means the frames have been loaded and we can generate the next output frame. */
+ if (pFramesOutS16 != NULL) {
+ MA_ASSERT(pResampler->inTimeInt == 0);
+ ma_linear_resampler_interpolate_frame_s16(pResampler, pFramesOutS16);
+
+ /* Filter. */
+ ma_lpf_process_pcm_frame_s16(&pResampler->lpf, pFramesOutS16, pFramesOutS16);
+
+ pFramesOutS16 += pResampler->config.channels;
+ }
+
+ framesProcessedOut += 1;
- resultOSS = read(pDevice->oss.fdCapture, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels));
- if (resultOSS < 0) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OSS] Failed to read data from the device to be sent to the client.", ma_result_from_errno(errno));
+ /* Advance time forward. */
+ pResampler->inTimeInt += pResampler->inAdvanceInt;
+ pResampler->inTimeFrac += pResampler->inAdvanceFrac;
+ if (pResampler->inTimeFrac >= pResampler->config.sampleRateOut) {
+ pResampler->inTimeFrac -= pResampler->config.sampleRateOut;
+ pResampler->inTimeInt += 1;
+ }
}
- if (pFramesRead != NULL) {
- *pFramesRead = (ma_uint32)resultOSS / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
- }
+ *pFrameCountIn = framesProcessedIn;
+ *pFrameCountOut = framesProcessedOut;
return MA_SUCCESS;
}
-static ma_result ma_context_uninit__oss(ma_context* pContext)
+static ma_result ma_linear_resampler_process_pcm_frames_s16(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pContext->backend == ma_backend_oss);
+ MA_ASSERT(pResampler != NULL);
- (void)pContext;
- return MA_SUCCESS;
+ if (pResampler->config.sampleRateIn > pResampler->config.sampleRateOut) {
+ return ma_linear_resampler_process_pcm_frames_s16_downsample(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
+ } else {
+ return ma_linear_resampler_process_pcm_frames_s16_upsample(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
+ }
}
-static ma_result ma_context_init__oss(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
-{
- int fd;
- int ossVersion;
- int result;
- MA_ASSERT(pContext != NULL);
+static ma_result ma_linear_resampler_process_pcm_frames_f32_downsample(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+{
+ const float* pFramesInF32;
+ /* */ float* pFramesOutF32;
+ ma_uint64 frameCountIn;
+ ma_uint64 frameCountOut;
+ ma_uint64 framesProcessedIn;
+ ma_uint64 framesProcessedOut;
- (void)pConfig;
+ MA_ASSERT(pResampler != NULL);
+ MA_ASSERT(pFrameCountIn != NULL);
+ MA_ASSERT(pFrameCountOut != NULL);
- /* Try opening a temporary device first so we can get version information. This is closed at the end. */
- fd = ma_open_temp_device__oss();
- if (fd == -1) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[OSS] Failed to open temporary device for retrieving system properties.", MA_NO_BACKEND); /* Looks liks OSS isn't installed, or there are no available devices. */
- }
+ pFramesInF32 = (const float*)pFramesIn;
+ pFramesOutF32 = ( float*)pFramesOut;
+ frameCountIn = *pFrameCountIn;
+ frameCountOut = *pFrameCountOut;
+ framesProcessedIn = 0;
+ framesProcessedOut = 0;
- /* Grab the OSS version. */
- ossVersion = 0;
- result = ioctl(fd, OSS_GETVERSION, &ossVersion);
- if (result == -1) {
- close(fd);
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "[OSS] Failed to retrieve OSS version.", MA_NO_BACKEND);
- }
+ while (framesProcessedOut < frameCountOut) {
+ /* Before interpolating we need to load the buffers. When doing this we need to ensure we run every input sample through the filter. */
+ while (pResampler->inTimeInt > 0 && frameCountIn > framesProcessedIn) {
+ ma_uint32 iChannel;
- /* The file handle to temp device is no longer needed. Close ASAP. */
- close(fd);
+ if (pFramesInF32 != NULL) {
+ for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
+ pResampler->x0.f32[iChannel] = pResampler->x1.f32[iChannel];
+ pResampler->x1.f32[iChannel] = pFramesInF32[iChannel];
+ }
+ pFramesInF32 += pResampler->config.channels;
+ } else {
+ for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
+ pResampler->x0.f32[iChannel] = pResampler->x1.f32[iChannel];
+ pResampler->x1.f32[iChannel] = 0;
+ }
+ }
- pContext->oss.versionMajor = ((ossVersion & 0xFF0000) >> 16);
- pContext->oss.versionMinor = ((ossVersion & 0x00FF00) >> 8);
+ /* Filter. */
+ ma_lpf_process_pcm_frame_f32(&pResampler->lpf, pResampler->x1.f32, pResampler->x1.f32);
- pCallbacks->onContextInit = ma_context_init__oss;
- pCallbacks->onContextUninit = ma_context_uninit__oss;
- pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__oss;
- pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__oss;
- pCallbacks->onDeviceInit = ma_device_init__oss;
- pCallbacks->onDeviceUninit = ma_device_uninit__oss;
- pCallbacks->onDeviceStart = ma_device_start__oss;
- pCallbacks->onDeviceStop = ma_device_stop__oss;
- pCallbacks->onDeviceRead = ma_device_read__oss;
- pCallbacks->onDeviceWrite = ma_device_write__oss;
- pCallbacks->onDeviceDataLoop = NULL;
+ framesProcessedIn += 1;
+ pResampler->inTimeInt -= 1;
+ }
- return MA_SUCCESS;
-}
-#endif /* OSS */
+ if (pResampler->inTimeInt > 0) {
+ break; /* Ran out of input data. */
+ }
+ /* Getting here means the frames have been loaded and filtered and we can generate the next output frame. */
+ if (pFramesOutF32 != NULL) {
+ MA_ASSERT(pResampler->inTimeInt == 0);
+ ma_linear_resampler_interpolate_frame_f32(pResampler, pFramesOutF32);
-/******************************************************************************
+ pFramesOutF32 += pResampler->config.channels;
+ }
-AAudio Backend
+ framesProcessedOut += 1;
-******************************************************************************/
-#ifdef MA_HAS_AAUDIO
+ /* Advance time forward. */
+ pResampler->inTimeInt += pResampler->inAdvanceInt;
+ pResampler->inTimeFrac += pResampler->inAdvanceFrac;
+ if (pResampler->inTimeFrac >= pResampler->config.sampleRateOut) {
+ pResampler->inTimeFrac -= pResampler->config.sampleRateOut;
+ pResampler->inTimeInt += 1;
+ }
+ }
-/*#include */
+ *pFrameCountIn = framesProcessedIn;
+ *pFrameCountOut = framesProcessedOut;
-typedef int32_t ma_aaudio_result_t;
-typedef int32_t ma_aaudio_direction_t;
-typedef int32_t ma_aaudio_sharing_mode_t;
-typedef int32_t ma_aaudio_format_t;
-typedef int32_t ma_aaudio_stream_state_t;
-typedef int32_t ma_aaudio_performance_mode_t;
-typedef int32_t ma_aaudio_usage_t;
-typedef int32_t ma_aaudio_content_type_t;
-typedef int32_t ma_aaudio_input_preset_t;
-typedef int32_t ma_aaudio_data_callback_result_t;
-typedef struct ma_AAudioStreamBuilder_t* ma_AAudioStreamBuilder;
-typedef struct ma_AAudioStream_t* ma_AAudioStream;
+ return MA_SUCCESS;
+}
-#define MA_AAUDIO_UNSPECIFIED 0
+static ma_result ma_linear_resampler_process_pcm_frames_f32_upsample(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+{
+ const float* pFramesInF32;
+ /* */ float* pFramesOutF32;
+ ma_uint64 frameCountIn;
+ ma_uint64 frameCountOut;
+ ma_uint64 framesProcessedIn;
+ ma_uint64 framesProcessedOut;
-/* Result codes. miniaudio only cares about the success code. */
-#define MA_AAUDIO_OK 0
+ MA_ASSERT(pResampler != NULL);
+ MA_ASSERT(pFrameCountIn != NULL);
+ MA_ASSERT(pFrameCountOut != NULL);
-/* Directions. */
-#define MA_AAUDIO_DIRECTION_OUTPUT 0
-#define MA_AAUDIO_DIRECTION_INPUT 1
+ pFramesInF32 = (const float*)pFramesIn;
+ pFramesOutF32 = ( float*)pFramesOut;
+ frameCountIn = *pFrameCountIn;
+ frameCountOut = *pFrameCountOut;
+ framesProcessedIn = 0;
+ framesProcessedOut = 0;
-/* Sharing modes. */
-#define MA_AAUDIO_SHARING_MODE_EXCLUSIVE 0
-#define MA_AAUDIO_SHARING_MODE_SHARED 1
+ while (framesProcessedOut < frameCountOut) {
+ /* Before interpolating we need to load the buffers. */
+ while (pResampler->inTimeInt > 0 && frameCountIn > framesProcessedIn) {
+ ma_uint32 iChannel;
-/* Formats. */
-#define MA_AAUDIO_FORMAT_PCM_I16 1
-#define MA_AAUDIO_FORMAT_PCM_FLOAT 2
+ if (pFramesInF32 != NULL) {
+ for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
+ pResampler->x0.f32[iChannel] = pResampler->x1.f32[iChannel];
+ pResampler->x1.f32[iChannel] = pFramesInF32[iChannel];
+ }
+ pFramesInF32 += pResampler->config.channels;
+ } else {
+ for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
+ pResampler->x0.f32[iChannel] = pResampler->x1.f32[iChannel];
+ pResampler->x1.f32[iChannel] = 0;
+ }
+ }
-/* Stream states. */
-#define MA_AAUDIO_STREAM_STATE_UNINITIALIZED 0
-#define MA_AAUDIO_STREAM_STATE_UNKNOWN 1
-#define MA_AAUDIO_STREAM_STATE_OPEN 2
-#define MA_AAUDIO_STREAM_STATE_STARTING 3
-#define MA_AAUDIO_STREAM_STATE_STARTED 4
-#define MA_AAUDIO_STREAM_STATE_PAUSING 5
-#define MA_AAUDIO_STREAM_STATE_PAUSED 6
-#define MA_AAUDIO_STREAM_STATE_FLUSHING 7
-#define MA_AAUDIO_STREAM_STATE_FLUSHED 8
-#define MA_AAUDIO_STREAM_STATE_STOPPING 9
-#define MA_AAUDIO_STREAM_STATE_STOPPED 10
-#define MA_AAUDIO_STREAM_STATE_CLOSING 11
-#define MA_AAUDIO_STREAM_STATE_CLOSED 12
-#define MA_AAUDIO_STREAM_STATE_DISCONNECTED 13
+ framesProcessedIn += 1;
+ pResampler->inTimeInt -= 1;
+ }
-/* Performance modes. */
-#define MA_AAUDIO_PERFORMANCE_MODE_NONE 10
-#define MA_AAUDIO_PERFORMANCE_MODE_POWER_SAVING 11
-#define MA_AAUDIO_PERFORMANCE_MODE_LOW_LATENCY 12
+ if (pResampler->inTimeInt > 0) {
+ break; /* Ran out of input data. */
+ }
-/* Usage types. */
-#define MA_AAUDIO_USAGE_MEDIA 1
-#define MA_AAUDIO_USAGE_VOICE_COMMUNICATION 2
-#define MA_AAUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING 3
-#define MA_AAUDIO_USAGE_ALARM 4
-#define MA_AAUDIO_USAGE_NOTIFICATION 5
-#define MA_AAUDIO_USAGE_NOTIFICATION_RINGTONE 6
-#define MA_AAUDIO_USAGE_NOTIFICATION_EVENT 10
-#define MA_AAUDIO_USAGE_ASSISTANCE_ACCESSIBILITY 11
-#define MA_AAUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE 12
-#define MA_AAUDIO_USAGE_ASSISTANCE_SONIFICATION 13
-#define MA_AAUDIO_USAGE_GAME 14
-#define MA_AAUDIO_USAGE_ASSISTANT 16
-#define MA_AAUDIO_SYSTEM_USAGE_EMERGENCY 1000
-#define MA_AAUDIO_SYSTEM_USAGE_SAFETY 1001
-#define MA_AAUDIO_SYSTEM_USAGE_VEHICLE_STATUS 1002
-#define MA_AAUDIO_SYSTEM_USAGE_ANNOUNCEMENT 1003
+ /* Getting here means the frames have been loaded and we can generate the next output frame. */
+ if (pFramesOutF32 != NULL) {
+ MA_ASSERT(pResampler->inTimeInt == 0);
+ ma_linear_resampler_interpolate_frame_f32(pResampler, pFramesOutF32);
-/* Content types. */
-#define MA_AAUDIO_CONTENT_TYPE_SPEECH 1
-#define MA_AAUDIO_CONTENT_TYPE_MUSIC 2
-#define MA_AAUDIO_CONTENT_TYPE_MOVIE 3
-#define MA_AAUDIO_CONTENT_TYPE_SONIFICATION 4
+ /* Filter. */
+ ma_lpf_process_pcm_frame_f32(&pResampler->lpf, pFramesOutF32, pFramesOutF32);
-/* Input presets. */
-#define MA_AAUDIO_INPUT_PRESET_GENERIC 1
-#define MA_AAUDIO_INPUT_PRESET_CAMCORDER 5
-#define MA_AAUDIO_INPUT_PRESET_VOICE_RECOGNITION 6
-#define MA_AAUDIO_INPUT_PRESET_VOICE_COMMUNICATION 7
-#define MA_AAUDIO_INPUT_PRESET_UNPROCESSED 9
-#define MA_AAUDIO_INPUT_PRESET_VOICE_PERFORMANCE 10
+ pFramesOutF32 += pResampler->config.channels;
+ }
-/* Callback results. */
-#define MA_AAUDIO_CALLBACK_RESULT_CONTINUE 0
-#define MA_AAUDIO_CALLBACK_RESULT_STOP 1
+ framesProcessedOut += 1;
+ /* Advance time forward. */
+ pResampler->inTimeInt += pResampler->inAdvanceInt;
+ pResampler->inTimeFrac += pResampler->inAdvanceFrac;
+ if (pResampler->inTimeFrac >= pResampler->config.sampleRateOut) {
+ pResampler->inTimeFrac -= pResampler->config.sampleRateOut;
+ pResampler->inTimeInt += 1;
+ }
+ }
-typedef ma_aaudio_data_callback_result_t (* ma_AAudioStream_dataCallback) (ma_AAudioStream* pStream, void* pUserData, void* pAudioData, int32_t numFrames);
-typedef void (* ma_AAudioStream_errorCallback)(ma_AAudioStream *pStream, void *pUserData, ma_aaudio_result_t error);
+ *pFrameCountIn = framesProcessedIn;
+ *pFrameCountOut = framesProcessedOut;
-typedef ma_aaudio_result_t (* MA_PFN_AAudio_createStreamBuilder) (ma_AAudioStreamBuilder** ppBuilder);
-typedef ma_aaudio_result_t (* MA_PFN_AAudioStreamBuilder_delete) (ma_AAudioStreamBuilder* pBuilder);
-typedef void (* MA_PFN_AAudioStreamBuilder_setDeviceId) (ma_AAudioStreamBuilder* pBuilder, int32_t deviceId);
-typedef void (* MA_PFN_AAudioStreamBuilder_setDirection) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_direction_t direction);
-typedef void (* MA_PFN_AAudioStreamBuilder_setSharingMode) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_sharing_mode_t sharingMode);
-typedef void (* MA_PFN_AAudioStreamBuilder_setFormat) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_format_t format);
-typedef void (* MA_PFN_AAudioStreamBuilder_setChannelCount) (ma_AAudioStreamBuilder* pBuilder, int32_t channelCount);
-typedef void (* MA_PFN_AAudioStreamBuilder_setSampleRate) (ma_AAudioStreamBuilder* pBuilder, int32_t sampleRate);
-typedef void (* MA_PFN_AAudioStreamBuilder_setBufferCapacityInFrames)(ma_AAudioStreamBuilder* pBuilder, int32_t numFrames);
-typedef void (* MA_PFN_AAudioStreamBuilder_setFramesPerDataCallback) (ma_AAudioStreamBuilder* pBuilder, int32_t numFrames);
-typedef void (* MA_PFN_AAudioStreamBuilder_setDataCallback) (ma_AAudioStreamBuilder* pBuilder, ma_AAudioStream_dataCallback callback, void* pUserData);
-typedef void (* MA_PFN_AAudioStreamBuilder_setErrorCallback) (ma_AAudioStreamBuilder* pBuilder, ma_AAudioStream_errorCallback callback, void* pUserData);
-typedef void (* MA_PFN_AAudioStreamBuilder_setPerformanceMode) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_performance_mode_t mode);
-typedef void (* MA_PFN_AAudioStreamBuilder_setUsage) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_usage_t contentType);
-typedef void (* MA_PFN_AAudioStreamBuilder_setContentType) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_content_type_t contentType);
-typedef void (* MA_PFN_AAudioStreamBuilder_setInputPreset) (ma_AAudioStreamBuilder* pBuilder, ma_aaudio_input_preset_t inputPreset);
-typedef ma_aaudio_result_t (* MA_PFN_AAudioStreamBuilder_openStream) (ma_AAudioStreamBuilder* pBuilder, ma_AAudioStream** ppStream);
-typedef ma_aaudio_result_t (* MA_PFN_AAudioStream_close) (ma_AAudioStream* pStream);
-typedef ma_aaudio_stream_state_t (* MA_PFN_AAudioStream_getState) (ma_AAudioStream* pStream);
-typedef ma_aaudio_result_t (* MA_PFN_AAudioStream_waitForStateChange) (ma_AAudioStream* pStream, ma_aaudio_stream_state_t inputState, ma_aaudio_stream_state_t* pNextState, int64_t timeoutInNanoseconds);
-typedef ma_aaudio_format_t (* MA_PFN_AAudioStream_getFormat) (ma_AAudioStream* pStream);
-typedef int32_t (* MA_PFN_AAudioStream_getChannelCount) (ma_AAudioStream* pStream);
-typedef int32_t (* MA_PFN_AAudioStream_getSampleRate) (ma_AAudioStream* pStream);
-typedef int32_t (* MA_PFN_AAudioStream_getBufferCapacityInFrames) (ma_AAudioStream* pStream);
-typedef int32_t (* MA_PFN_AAudioStream_getFramesPerDataCallback) (ma_AAudioStream* pStream);
-typedef int32_t (* MA_PFN_AAudioStream_getFramesPerBurst) (ma_AAudioStream* pStream);
-typedef ma_aaudio_result_t (* MA_PFN_AAudioStream_requestStart) (ma_AAudioStream* pStream);
-typedef ma_aaudio_result_t (* MA_PFN_AAudioStream_requestStop) (ma_AAudioStream* pStream);
+ return MA_SUCCESS;
+}
-static ma_result ma_result_from_aaudio(ma_aaudio_result_t resultAA)
+static ma_result ma_linear_resampler_process_pcm_frames_f32(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
- switch (resultAA)
- {
- case MA_AAUDIO_OK: return MA_SUCCESS;
- default: break;
- }
+ MA_ASSERT(pResampler != NULL);
- return MA_ERROR;
+ if (pResampler->config.sampleRateIn > pResampler->config.sampleRateOut) {
+ return ma_linear_resampler_process_pcm_frames_f32_downsample(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
+ } else {
+ return ma_linear_resampler_process_pcm_frames_f32_upsample(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
+ }
}
-static ma_aaudio_usage_t ma_to_usage__aaudio(ma_aaudio_usage usage)
+
+MA_API ma_result ma_linear_resampler_process_pcm_frames(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
- switch (usage) {
- case ma_aaudio_usage_announcement: return MA_AAUDIO_USAGE_MEDIA;
- case ma_aaudio_usage_emergency: return MA_AAUDIO_USAGE_VOICE_COMMUNICATION;
- case ma_aaudio_usage_safety: return MA_AAUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING;
- case ma_aaudio_usage_vehicle_status: return MA_AAUDIO_USAGE_ALARM;
- case ma_aaudio_usage_alarm: return MA_AAUDIO_USAGE_NOTIFICATION;
- case ma_aaudio_usage_assistance_accessibility: return MA_AAUDIO_USAGE_NOTIFICATION_RINGTONE;
- case ma_aaudio_usage_assistance_navigation_guidance: return MA_AAUDIO_USAGE_NOTIFICATION_EVENT;
- case ma_aaudio_usage_assistance_sonification: return MA_AAUDIO_USAGE_ASSISTANCE_ACCESSIBILITY;
- case ma_aaudio_usage_assitant: return MA_AAUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE;
- case ma_aaudio_usage_game: return MA_AAUDIO_USAGE_ASSISTANCE_SONIFICATION;
- case ma_aaudio_usage_media: return MA_AAUDIO_USAGE_GAME;
- case ma_aaudio_usage_notification: return MA_AAUDIO_USAGE_ASSISTANT;
- case ma_aaudio_usage_notification_event: return MA_AAUDIO_SYSTEM_USAGE_EMERGENCY;
- case ma_aaudio_usage_notification_ringtone: return MA_AAUDIO_SYSTEM_USAGE_SAFETY;
- case ma_aaudio_usage_voice_communication: return MA_AAUDIO_SYSTEM_USAGE_VEHICLE_STATUS;
- case ma_aaudio_usage_voice_communication_signalling: return MA_AAUDIO_SYSTEM_USAGE_ANNOUNCEMENT;
- default: break;
+ if (pResampler == NULL) {
+ return MA_INVALID_ARGS;
}
- return MA_AAUDIO_USAGE_MEDIA;
+ /* */ if (pResampler->config.format == ma_format_s16) {
+ return ma_linear_resampler_process_pcm_frames_s16(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
+ } else if (pResampler->config.format == ma_format_f32) {
+ return ma_linear_resampler_process_pcm_frames_f32(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
+ } else {
+ /* Should never get here. Getting here means the format is not supported and you didn't check the return value of ma_linear_resampler_init(). */
+ MA_ASSERT(MA_FALSE);
+ return MA_INVALID_ARGS;
+ }
}
-static ma_aaudio_content_type_t ma_to_content_type__aaudio(ma_aaudio_content_type contentType)
-{
- switch (contentType) {
- case ma_aaudio_content_type_movie: return MA_AAUDIO_CONTENT_TYPE_MOVIE;
- case ma_aaudio_content_type_music: return MA_AAUDIO_CONTENT_TYPE_MUSIC;
- case ma_aaudio_content_type_sonification: return MA_AAUDIO_CONTENT_TYPE_SONIFICATION;
- case ma_aaudio_content_type_speech: return MA_AAUDIO_CONTENT_TYPE_SPEECH;
- default: break;
- }
- return MA_AAUDIO_CONTENT_TYPE_SPEECH;
+MA_API ma_result ma_linear_resampler_set_rate(ma_linear_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut)
+{
+ return ma_linear_resampler_set_rate_internal(pResampler, NULL, NULL, sampleRateIn, sampleRateOut, /* isResamplerAlreadyInitialized = */ MA_TRUE);
}
-static ma_aaudio_input_preset_t ma_to_input_preset__aaudio(ma_aaudio_input_preset inputPreset)
+MA_API ma_result ma_linear_resampler_set_rate_ratio(ma_linear_resampler* pResampler, float ratioInOut)
{
- switch (inputPreset) {
- case ma_aaudio_input_preset_generic: return MA_AAUDIO_INPUT_PRESET_GENERIC;
- case ma_aaudio_input_preset_camcorder: return MA_AAUDIO_INPUT_PRESET_CAMCORDER;
- case ma_aaudio_input_preset_unprocessed: return MA_AAUDIO_INPUT_PRESET_UNPROCESSED;
- case ma_aaudio_input_preset_voice_recognition: return MA_AAUDIO_INPUT_PRESET_VOICE_RECOGNITION;
- case ma_aaudio_input_preset_voice_communication: return MA_AAUDIO_INPUT_PRESET_VOICE_COMMUNICATION;
- case ma_aaudio_input_preset_voice_performance: return MA_AAUDIO_INPUT_PRESET_VOICE_PERFORMANCE;
- default: break;
+ ma_uint32 n;
+ ma_uint32 d;
+
+ if (pResampler == NULL) {
+ return MA_INVALID_ARGS;
}
- return MA_AAUDIO_INPUT_PRESET_GENERIC;
-}
+ if (ratioInOut <= 0) {
+ return MA_INVALID_ARGS;
+ }
-static void ma_stream_error_callback__aaudio(ma_AAudioStream* pStream, void* pUserData, ma_aaudio_result_t error)
-{
- ma_device* pDevice = (ma_device*)pUserData;
- MA_ASSERT(pDevice != NULL);
+ d = 1000;
+ n = (ma_uint32)(ratioInOut * d);
- (void)error;
+ if (n == 0) {
+ return MA_INVALID_ARGS; /* Ratio too small. */
+ }
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[AAudio] ERROR CALLBACK: error=%d, AAudioStream_getState()=%d\n", error, ((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream));
+ MA_ASSERT(n != 0);
- /*
- From the documentation for AAudio, when a device is disconnected all we can do is stop it. However, we cannot stop it from the callback - we need
- to do it from another thread. Therefore we are going to use an event thread for the AAudio backend to do this cleanly and safely.
- */
- if (((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream) == MA_AAUDIO_STREAM_STATE_DISCONNECTED) {
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[AAudio] Device Disconnected.\n");
- }
+ return ma_linear_resampler_set_rate(pResampler, n, d);
}
-static ma_aaudio_data_callback_result_t ma_stream_data_callback_capture__aaudio(ma_AAudioStream* pStream, void* pUserData, void* pAudioData, int32_t frameCount)
+MA_API ma_uint64 ma_linear_resampler_get_input_latency(const ma_linear_resampler* pResampler)
{
- ma_device* pDevice = (ma_device*)pUserData;
- MA_ASSERT(pDevice != NULL);
-
- ma_device_handle_backend_data_callback(pDevice, NULL, pAudioData, frameCount);
+ if (pResampler == NULL) {
+ return 0;
+ }
- (void)pStream;
- return MA_AAUDIO_CALLBACK_RESULT_CONTINUE;
+ return 1 + ma_lpf_get_latency(&pResampler->lpf);
}
-static ma_aaudio_data_callback_result_t ma_stream_data_callback_playback__aaudio(ma_AAudioStream* pStream, void* pUserData, void* pAudioData, int32_t frameCount)
+MA_API ma_uint64 ma_linear_resampler_get_output_latency(const ma_linear_resampler* pResampler)
{
- ma_device* pDevice = (ma_device*)pUserData;
- MA_ASSERT(pDevice != NULL);
-
- ma_device_handle_backend_data_callback(pDevice, pAudioData, NULL, frameCount);
+ if (pResampler == NULL) {
+ return 0;
+ }
- (void)pStream;
- return MA_AAUDIO_CALLBACK_RESULT_CONTINUE;
+ return ma_linear_resampler_get_input_latency(pResampler) * pResampler->config.sampleRateOut / pResampler->config.sampleRateIn;
}
-static ma_result ma_create_and_configure_AAudioStreamBuilder__aaudio(ma_context* pContext, const ma_device_id* pDeviceID, ma_device_type deviceType, ma_share_mode shareMode, const ma_device_descriptor* pDescriptor, const ma_device_config* pConfig, ma_device* pDevice, ma_AAudioStreamBuilder** ppBuilder)
+MA_API ma_result ma_linear_resampler_get_required_input_frame_count(const ma_linear_resampler* pResampler, ma_uint64 outputFrameCount, ma_uint64* pInputFrameCount)
{
- ma_AAudioStreamBuilder* pBuilder;
- ma_aaudio_result_t resultAA;
- ma_uint32 bufferCapacityInFrames;
-
- /* Safety. */
- *ppBuilder = NULL;
-
- resultAA = ((MA_PFN_AAudio_createStreamBuilder)pContext->aaudio.AAudio_createStreamBuilder)(&pBuilder);
- if (resultAA != MA_AAUDIO_OK) {
- return ma_result_from_aaudio(resultAA);
- }
+ ma_uint64 inputFrameCount;
- if (pDeviceID != NULL) {
- ((MA_PFN_AAudioStreamBuilder_setDeviceId)pContext->aaudio.AAudioStreamBuilder_setDeviceId)(pBuilder, pDeviceID->aaudio);
+ if (pInputFrameCount == NULL) {
+ return MA_INVALID_ARGS;
}
- ((MA_PFN_AAudioStreamBuilder_setDirection)pContext->aaudio.AAudioStreamBuilder_setDirection)(pBuilder, (deviceType == ma_device_type_playback) ? MA_AAUDIO_DIRECTION_OUTPUT : MA_AAUDIO_DIRECTION_INPUT);
- ((MA_PFN_AAudioStreamBuilder_setSharingMode)pContext->aaudio.AAudioStreamBuilder_setSharingMode)(pBuilder, (shareMode == ma_share_mode_shared) ? MA_AAUDIO_SHARING_MODE_SHARED : MA_AAUDIO_SHARING_MODE_EXCLUSIVE);
+ *pInputFrameCount = 0;
+ if (pResampler == NULL) {
+ return MA_INVALID_ARGS;
+ }
- /* If we have a device descriptor make sure we configure the stream builder to take our requested parameters. */
- if (pDescriptor != NULL) {
- MA_ASSERT(pConfig != NULL); /* We must have a device config if we also have a descriptor. The config is required for AAudio specific configuration options. */
+ if (outputFrameCount == 0) {
+ return MA_SUCCESS;
+ }
- if (pDescriptor->sampleRate != 0) {
- ((MA_PFN_AAudioStreamBuilder_setSampleRate)pContext->aaudio.AAudioStreamBuilder_setSampleRate)(pBuilder, pDescriptor->sampleRate);
- }
+ /* Any whole input frames are consumed before the first output frame is generated. */
+ inputFrameCount = pResampler->inTimeInt;
+ outputFrameCount -= 1;
- if (deviceType == ma_device_type_capture) {
- if (pDescriptor->channels != 0) {
- ((MA_PFN_AAudioStreamBuilder_setChannelCount)pContext->aaudio.AAudioStreamBuilder_setChannelCount)(pBuilder, pDescriptor->channels);
- }
- if (pDescriptor->format != ma_format_unknown) {
- ((MA_PFN_AAudioStreamBuilder_setFormat)pContext->aaudio.AAudioStreamBuilder_setFormat)(pBuilder, (pDescriptor->format == ma_format_s16) ? MA_AAUDIO_FORMAT_PCM_I16 : MA_AAUDIO_FORMAT_PCM_FLOAT);
- }
- } else {
- if (pDescriptor->channels != 0) {
- ((MA_PFN_AAudioStreamBuilder_setChannelCount)pContext->aaudio.AAudioStreamBuilder_setChannelCount)(pBuilder, pDescriptor->channels);
- }
- if (pDescriptor->format != ma_format_unknown) {
- ((MA_PFN_AAudioStreamBuilder_setFormat)pContext->aaudio.AAudioStreamBuilder_setFormat)(pBuilder, (pDescriptor->format == ma_format_s16) ? MA_AAUDIO_FORMAT_PCM_I16 : MA_AAUDIO_FORMAT_PCM_FLOAT);
- }
- }
+ /* The rest of the output frames can be calculated in constant time. */
+ inputFrameCount += outputFrameCount * pResampler->inAdvanceInt;
+ inputFrameCount += (pResampler->inTimeFrac + (outputFrameCount * pResampler->inAdvanceFrac)) / pResampler->config.sampleRateOut;
- /*
- AAudio is annoying when it comes to it's buffer calculation stuff because it doesn't let you
- retrieve the actual sample rate until after you've opened the stream. But you need to configure
- the buffer capacity before you open the stream... :/
+ *pInputFrameCount = inputFrameCount;
- To solve, we're just going to assume MA_DEFAULT_SAMPLE_RATE (48000) and move on.
- */
- bufferCapacityInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, pDescriptor->sampleRate, pConfig->performanceProfile) * pDescriptor->periodCount;
+ return MA_SUCCESS;
+}
- ((MA_PFN_AAudioStreamBuilder_setBufferCapacityInFrames)pContext->aaudio.AAudioStreamBuilder_setBufferCapacityInFrames)(pBuilder, bufferCapacityInFrames);
- ((MA_PFN_AAudioStreamBuilder_setFramesPerDataCallback)pContext->aaudio.AAudioStreamBuilder_setFramesPerDataCallback)(pBuilder, bufferCapacityInFrames / pDescriptor->periodCount);
+MA_API ma_result ma_linear_resampler_get_expected_output_frame_count(const ma_linear_resampler* pResampler, ma_uint64 inputFrameCount, ma_uint64* pOutputFrameCount)
+{
+ ma_uint64 outputFrameCount;
+ ma_uint64 preliminaryInputFrameCountFromFrac;
+ ma_uint64 preliminaryInputFrameCount;
- if (deviceType == ma_device_type_capture) {
- if (pConfig->aaudio.inputPreset != ma_aaudio_input_preset_default && pContext->aaudio.AAudioStreamBuilder_setInputPreset != NULL) {
- ((MA_PFN_AAudioStreamBuilder_setInputPreset)pContext->aaudio.AAudioStreamBuilder_setInputPreset)(pBuilder, ma_to_input_preset__aaudio(pConfig->aaudio.inputPreset));
- }
+ if (pOutputFrameCount == NULL) {
+ return MA_INVALID_ARGS;
+ }
- ((MA_PFN_AAudioStreamBuilder_setDataCallback)pContext->aaudio.AAudioStreamBuilder_setDataCallback)(pBuilder, ma_stream_data_callback_capture__aaudio, (void*)pDevice);
- } else {
- if (pConfig->aaudio.usage != ma_aaudio_usage_default && pContext->aaudio.AAudioStreamBuilder_setUsage != NULL) {
- ((MA_PFN_AAudioStreamBuilder_setUsage)pContext->aaudio.AAudioStreamBuilder_setUsage)(pBuilder, ma_to_usage__aaudio(pConfig->aaudio.usage));
- }
+ *pOutputFrameCount = 0;
- if (pConfig->aaudio.contentType != ma_aaudio_content_type_default && pContext->aaudio.AAudioStreamBuilder_setContentType != NULL) {
- ((MA_PFN_AAudioStreamBuilder_setContentType)pContext->aaudio.AAudioStreamBuilder_setContentType)(pBuilder, ma_to_content_type__aaudio(pConfig->aaudio.contentType));
- }
+ if (pResampler == NULL) {
+ return MA_INVALID_ARGS;
+ }
- ((MA_PFN_AAudioStreamBuilder_setDataCallback)pContext->aaudio.AAudioStreamBuilder_setDataCallback)(pBuilder, ma_stream_data_callback_playback__aaudio, (void*)pDevice);
- }
+ /*
+ The first step is to get a preliminary output frame count. This will either be exactly equal to what we need, or less by 1. We need to
+ determine how many input frames will be consumed by this value. If it's greater than our original input frame count it means we won't
+ be able to generate an extra frame because we will have run out of input data. Otherwise we will have enough input for the generation
+ of an extra output frame. This add-by-one logic is necessary due to how the data loading logic works when processing frames.
+ */
+ outputFrameCount = (inputFrameCount * pResampler->config.sampleRateOut) / pResampler->config.sampleRateIn;
- /* Not sure how this affects things, but since there's a mapping between miniaudio's performance profiles and AAudio's performance modes, let go ahead and set it. */
- ((MA_PFN_AAudioStreamBuilder_setPerformanceMode)pContext->aaudio.AAudioStreamBuilder_setPerformanceMode)(pBuilder, (pConfig->performanceProfile == ma_performance_profile_low_latency) ? MA_AAUDIO_PERFORMANCE_MODE_LOW_LATENCY : MA_AAUDIO_PERFORMANCE_MODE_NONE);
+ /*
+ We need to determine how many *whole* input frames will have been processed to generate our preliminary output frame count. This is
+ used in the logic below to determine whether or not we need to add an extra output frame.
+ */
+ preliminaryInputFrameCountFromFrac = (pResampler->inTimeFrac + outputFrameCount*pResampler->inAdvanceFrac) / pResampler->config.sampleRateOut;
+ preliminaryInputFrameCount = (pResampler->inTimeInt + outputFrameCount*pResampler->inAdvanceInt ) + preliminaryInputFrameCountFromFrac;
- /* We need to set an error callback to detect device changes. */
- if (pDevice != NULL) { /* <-- pDevice should never be null if pDescriptor is not null, which is always the case if we hit this branch. Check anyway for safety. */
- ((MA_PFN_AAudioStreamBuilder_setErrorCallback)pContext->aaudio.AAudioStreamBuilder_setErrorCallback)(pBuilder, ma_stream_error_callback__aaudio, (void*)pDevice);
- }
+ /*
+ If the total number of *whole* input frames that would be required to generate our preliminary output frame count is greather than
+ the amount of whole input frames we have available as input we need to *not* add an extra output frame as there won't be enough data
+ to actually process. Otherwise we need to add the extra output frame.
+ */
+ if (preliminaryInputFrameCount <= inputFrameCount) {
+ outputFrameCount += 1;
}
- *ppBuilder = pBuilder;
+ *pOutputFrameCount = outputFrameCount;
return MA_SUCCESS;
}
-static ma_result ma_open_stream_and_close_builder__aaudio(ma_context* pContext, ma_AAudioStreamBuilder* pBuilder, ma_AAudioStream** ppStream)
+
+/* Linear resampler backend vtable. */
+static ma_linear_resampler_config ma_resampling_backend_get_config__linear(const ma_resampler_config* pConfig)
{
- ma_result result;
+ ma_linear_resampler_config linearConfig;
- result = ma_result_from_aaudio(((MA_PFN_AAudioStreamBuilder_openStream)pContext->aaudio.AAudioStreamBuilder_openStream)(pBuilder, ppStream));
- ((MA_PFN_AAudioStreamBuilder_delete)pContext->aaudio.AAudioStreamBuilder_delete)(pBuilder);
+ linearConfig = ma_linear_resampler_config_init(pConfig->format, pConfig->channels, pConfig->sampleRateIn, pConfig->sampleRateOut);
+ linearConfig.lpfOrder = pConfig->linear.lpfOrder;
- return result;
+ return linearConfig;
}
-static ma_result ma_open_stream_basic__aaudio(ma_context* pContext, const ma_device_id* pDeviceID, ma_device_type deviceType, ma_share_mode shareMode, ma_AAudioStream** ppStream)
+static ma_result ma_resampling_backend_get_heap_size__linear(void* pUserData, const ma_resampler_config* pConfig, size_t* pHeapSizeInBytes)
{
- ma_result result;
- ma_AAudioStreamBuilder* pBuilder;
+ ma_linear_resampler_config linearConfig;
- *ppStream = NULL;
+ (void)pUserData;
- result = ma_create_and_configure_AAudioStreamBuilder__aaudio(pContext, pDeviceID, deviceType, shareMode, NULL, NULL, NULL, &pBuilder);
- if (result != MA_SUCCESS) {
- return result;
- }
+ linearConfig = ma_resampling_backend_get_config__linear(pConfig);
- return ma_open_stream_and_close_builder__aaudio(pContext, pBuilder, ppStream);
+ return ma_linear_resampler_get_heap_size(&linearConfig, pHeapSizeInBytes);
}
-static ma_result ma_open_stream__aaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_type deviceType, const ma_device_descriptor* pDescriptor, ma_AAudioStream** ppStream)
+static ma_result ma_resampling_backend_init__linear(void* pUserData, const ma_resampler_config* pConfig, void* pHeap, ma_resampling_backend** ppBackend)
{
+ ma_resampler* pResampler = (ma_resampler*)pUserData;
ma_result result;
- ma_AAudioStreamBuilder* pBuilder;
+ ma_linear_resampler_config linearConfig;
- MA_ASSERT(pConfig != NULL);
- MA_ASSERT(pConfig->deviceType != ma_device_type_duplex); /* This function should not be called for a full-duplex device type. */
+ (void)pUserData;
- *ppStream = NULL;
+ linearConfig = ma_resampling_backend_get_config__linear(pConfig);
- result = ma_create_and_configure_AAudioStreamBuilder__aaudio(pDevice->pContext, pDescriptor->pDeviceID, deviceType, pDescriptor->shareMode, pDescriptor, pConfig, pDevice, &pBuilder);
+ result = ma_linear_resampler_init_preallocated(&linearConfig, pHeap, &pResampler->state.linear);
if (result != MA_SUCCESS) {
return result;
}
- return ma_open_stream_and_close_builder__aaudio(pDevice->pContext, pBuilder, ppStream);
+ *ppBackend = &pResampler->state.linear;
+
+ return MA_SUCCESS;
}
-static ma_result ma_close_stream__aaudio(ma_context* pContext, ma_AAudioStream* pStream)
+static void ma_resampling_backend_uninit__linear(void* pUserData, ma_resampling_backend* pBackend, const ma_allocation_callbacks* pAllocationCallbacks)
{
- return ma_result_from_aaudio(((MA_PFN_AAudioStream_close)pContext->aaudio.AAudioStream_close)(pStream));
+ (void)pUserData;
+
+ ma_linear_resampler_uninit((ma_linear_resampler*)pBackend, pAllocationCallbacks);
}
-static ma_bool32 ma_has_default_device__aaudio(ma_context* pContext, ma_device_type deviceType)
+static ma_result ma_resampling_backend_process__linear(void* pUserData, ma_resampling_backend* pBackend, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
- /* The only way to know this is to try creating a stream. */
- ma_AAudioStream* pStream;
- ma_result result = ma_open_stream_basic__aaudio(pContext, NULL, deviceType, ma_share_mode_shared, &pStream);
- if (result != MA_SUCCESS) {
- return MA_FALSE;
- }
+ (void)pUserData;
- ma_close_stream__aaudio(pContext, pStream);
- return MA_TRUE;
+ return ma_linear_resampler_process_pcm_frames((ma_linear_resampler*)pBackend, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
}
-static ma_result ma_wait_for_simple_state_transition__aaudio(ma_context* pContext, ma_AAudioStream* pStream, ma_aaudio_stream_state_t oldState, ma_aaudio_stream_state_t newState)
+static ma_result ma_resampling_backend_set_rate__linear(void* pUserData, ma_resampling_backend* pBackend, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut)
{
- ma_aaudio_stream_state_t actualNewState;
- ma_aaudio_result_t resultAA = ((MA_PFN_AAudioStream_waitForStateChange)pContext->aaudio.AAudioStream_waitForStateChange)(pStream, oldState, &actualNewState, 5000000000); /* 5 second timeout. */
- if (resultAA != MA_AAUDIO_OK) {
- return ma_result_from_aaudio(resultAA);
- }
-
- if (newState != actualNewState) {
- return MA_ERROR; /* Failed to transition into the expected state. */
- }
+ (void)pUserData;
- return MA_SUCCESS;
+ return ma_linear_resampler_set_rate((ma_linear_resampler*)pBackend, sampleRateIn, sampleRateOut);
}
-
-static ma_result ma_context_enumerate_devices__aaudio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+static ma_uint64 ma_resampling_backend_get_input_latency__linear(void* pUserData, const ma_resampling_backend* pBackend)
{
- ma_bool32 cbResult = MA_TRUE;
+ (void)pUserData;
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(callback != NULL);
+ return ma_linear_resampler_get_input_latency((const ma_linear_resampler*)pBackend);
+}
- /* Unfortunately AAudio does not have an enumeration API. Therefore I'm only going to report default devices, but only if it can instantiate a stream. */
+static ma_uint64 ma_resampling_backend_get_output_latency__linear(void* pUserData, const ma_resampling_backend* pBackend)
+{
+ (void)pUserData;
- /* Playback. */
- if (cbResult) {
- ma_device_info deviceInfo;
- MA_ZERO_OBJECT(&deviceInfo);
- deviceInfo.id.aaudio = MA_AAUDIO_UNSPECIFIED;
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
+ return ma_linear_resampler_get_output_latency((const ma_linear_resampler*)pBackend);
+}
- if (ma_has_default_device__aaudio(pContext, ma_device_type_playback)) {
- cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
- }
- }
+static ma_result ma_resampling_backend_get_required_input_frame_count__linear(void* pUserData, const ma_resampling_backend* pBackend, ma_uint64 outputFrameCount, ma_uint64* pInputFrameCount)
+{
+ (void)pUserData;
- /* Capture. */
- if (cbResult) {
- ma_device_info deviceInfo;
- MA_ZERO_OBJECT(&deviceInfo);
- deviceInfo.id.aaudio = MA_AAUDIO_UNSPECIFIED;
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
+ return ma_linear_resampler_get_required_input_frame_count((const ma_linear_resampler*)pBackend, outputFrameCount, pInputFrameCount);
+}
- if (ma_has_default_device__aaudio(pContext, ma_device_type_capture)) {
- cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
- }
- }
+static ma_result ma_resampling_backend_get_expected_output_frame_count__linear(void* pUserData, const ma_resampling_backend* pBackend, ma_uint64 inputFrameCount, ma_uint64* pOutputFrameCount)
+{
+ (void)pUserData;
- return MA_SUCCESS;
+ return ma_linear_resampler_get_expected_output_frame_count((const ma_linear_resampler*)pBackend, inputFrameCount, pOutputFrameCount);
}
-static void ma_context_add_native_data_format_from_AAudioStream_ex__aaudio(ma_context* pContext, ma_AAudioStream* pStream, ma_format format, ma_uint32 flags, ma_device_info* pDeviceInfo)
+static ma_resampling_backend_vtable g_ma_linear_resampler_vtable =
{
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pStream != NULL);
- MA_ASSERT(pDeviceInfo != NULL);
+ ma_resampling_backend_get_heap_size__linear,
+ ma_resampling_backend_init__linear,
+ ma_resampling_backend_uninit__linear,
+ ma_resampling_backend_process__linear,
+ ma_resampling_backend_set_rate__linear,
+ ma_resampling_backend_get_input_latency__linear,
+ ma_resampling_backend_get_output_latency__linear,
+ ma_resampling_backend_get_required_input_frame_count__linear,
+ ma_resampling_backend_get_expected_output_frame_count__linear
+};
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = format;
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = ((MA_PFN_AAudioStream_getChannelCount)pContext->aaudio.AAudioStream_getChannelCount)(pStream);
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = ((MA_PFN_AAudioStream_getSampleRate)pContext->aaudio.AAudioStream_getSampleRate)(pStream);
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = flags;
- pDeviceInfo->nativeDataFormatCount += 1;
-}
-static void ma_context_add_native_data_format_from_AAudioStream__aaudio(ma_context* pContext, ma_AAudioStream* pStream, ma_uint32 flags, ma_device_info* pDeviceInfo)
+
+MA_API ma_resampler_config ma_resampler_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut, ma_resample_algorithm algorithm)
{
- /* AAudio supports s16 and f32. */
- ma_context_add_native_data_format_from_AAudioStream_ex__aaudio(pContext, pStream, ma_format_f32, flags, pDeviceInfo);
- ma_context_add_native_data_format_from_AAudioStream_ex__aaudio(pContext, pStream, ma_format_s16, flags, pDeviceInfo);
+ ma_resampler_config config;
+
+ MA_ZERO_OBJECT(&config);
+ config.format = format;
+ config.channels = channels;
+ config.sampleRateIn = sampleRateIn;
+ config.sampleRateOut = sampleRateOut;
+ config.algorithm = algorithm;
+
+ /* Linear. */
+ config.linear.lpfOrder = ma_min(MA_DEFAULT_RESAMPLER_LPF_ORDER, MA_MAX_FILTER_ORDER);
+
+ return config;
}
-static ma_result ma_context_get_device_info__aaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+static ma_result ma_resampler_get_vtable(const ma_resampler_config* pConfig, ma_resampler* pResampler, ma_resampling_backend_vtable** ppVTable, void** ppUserData)
{
- ma_AAudioStream* pStream;
- ma_result result;
+ MA_ASSERT(pConfig != NULL);
+ MA_ASSERT(ppVTable != NULL);
+ MA_ASSERT(ppUserData != NULL);
- MA_ASSERT(pContext != NULL);
+ /* Safety. */
+ *ppVTable = NULL;
+ *ppUserData = NULL;
- /* ID */
- if (pDeviceID != NULL) {
- pDeviceInfo->id.aaudio = pDeviceID->aaudio;
- } else {
- pDeviceInfo->id.aaudio = MA_AAUDIO_UNSPECIFIED;
+ switch (pConfig->algorithm)
+ {
+ case ma_resample_algorithm_linear:
+ {
+ *ppVTable = &g_ma_linear_resampler_vtable;
+ *ppUserData = pResampler;
+ } break;
+
+ case ma_resample_algorithm_custom:
+ {
+ *ppVTable = pConfig->pBackendVTable;
+ *ppUserData = pConfig->pBackendUserData;
+ } break;
+
+ default: return MA_INVALID_ARGS;
}
- /* Name */
- if (deviceType == ma_device_type_playback) {
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
- } else {
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
+ return MA_SUCCESS;
+}
+
+MA_API ma_result ma_resampler_get_heap_size(const ma_resampler_config* pConfig, size_t* pHeapSizeInBytes)
+{
+ ma_result result;
+ ma_resampling_backend_vtable* pVTable;
+ void* pVTableUserData;
+
+ if (pHeapSizeInBytes == NULL) {
+ return MA_INVALID_ARGS;
}
+ *pHeapSizeInBytes = 0;
- pDeviceInfo->nativeDataFormatCount = 0;
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
- /* We'll need to open the device to get accurate sample rate and channel count information. */
- result = ma_open_stream_basic__aaudio(pContext, pDeviceID, deviceType, ma_share_mode_shared, &pStream);
+ result = ma_resampler_get_vtable(pConfig, NULL, &pVTable, &pVTableUserData);
if (result != MA_SUCCESS) {
return result;
}
- ma_context_add_native_data_format_from_AAudioStream__aaudio(pContext, pStream, 0, pDeviceInfo);
+ if (pVTable == NULL || pVTable->onGetHeapSize == NULL) {
+ return MA_NOT_IMPLEMENTED;
+ }
- ma_close_stream__aaudio(pContext, pStream);
- pStream = NULL;
+ result = pVTable->onGetHeapSize(pVTableUserData, pConfig, pHeapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
return MA_SUCCESS;
}
-
-static ma_result ma_device_uninit__aaudio(ma_device* pDevice)
+MA_API ma_result ma_resampler_init_preallocated(const ma_resampler_config* pConfig, void* pHeap, ma_resampler* pResampler)
{
- MA_ASSERT(pDevice != NULL);
+ ma_result result;
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- ma_close_stream__aaudio(pDevice->pContext, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture);
- pDevice->aaudio.pStreamCapture = NULL;
+ if (pResampler == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- ma_close_stream__aaudio(pDevice->pContext, (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback);
- pDevice->aaudio.pStreamPlayback = NULL;
+ MA_ZERO_OBJECT(pResampler);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ pResampler->_pHeap = pHeap;
+ pResampler->format = pConfig->format;
+ pResampler->channels = pConfig->channels;
+ pResampler->sampleRateIn = pConfig->sampleRateIn;
+ pResampler->sampleRateOut = pConfig->sampleRateOut;
+
+ result = ma_resampler_get_vtable(pConfig, pResampler, &pResampler->pBackendVTable, &pResampler->pBackendUserData);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ if (pResampler->pBackendVTable == NULL || pResampler->pBackendVTable->onInit == NULL) {
+ return MA_NOT_IMPLEMENTED; /* onInit not implemented. */
+ }
+
+ result = pResampler->pBackendVTable->onInit(pResampler->pBackendUserData, pConfig, pHeap, &pResampler->pBackend);
+ if (result != MA_SUCCESS) {
+ return result;
}
return MA_SUCCESS;
}
-static ma_result ma_device_init_by_type__aaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_type deviceType, ma_device_descriptor* pDescriptor, ma_AAudioStream** ppStream)
+MA_API ma_result ma_resampler_init(const ma_resampler_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_resampler* pResampler)
{
ma_result result;
- int32_t bufferCapacityInFrames;
- int32_t framesPerDataCallback;
- ma_AAudioStream* pStream;
+ size_t heapSizeInBytes;
+ void* pHeap;
- MA_ASSERT(pDevice != NULL);
- MA_ASSERT(pConfig != NULL);
- MA_ASSERT(pDescriptor != NULL);
-
- *ppStream = NULL; /* Safety. */
-
- /* First step is to open the stream. From there we'll be able to extract the internal configuration. */
- result = ma_open_stream__aaudio(pDevice, pConfig, deviceType, pDescriptor, &pStream);
+ result = ma_resampler_get_heap_size(pConfig, &heapSizeInBytes);
if (result != MA_SUCCESS) {
- return result; /* Failed to open the AAudio stream. */
+ return result;
}
- /* Now extract the internal configuration. */
- pDescriptor->format = (((MA_PFN_AAudioStream_getFormat)pDevice->pContext->aaudio.AAudioStream_getFormat)(pStream) == MA_AAUDIO_FORMAT_PCM_I16) ? ma_format_s16 : ma_format_f32;
- pDescriptor->channels = ((MA_PFN_AAudioStream_getChannelCount)pDevice->pContext->aaudio.AAudioStream_getChannelCount)(pStream);
- pDescriptor->sampleRate = ((MA_PFN_AAudioStream_getSampleRate)pDevice->pContext->aaudio.AAudioStream_getSampleRate)(pStream);
-
- /* For the channel map we need to be sure we don't overflow any buffers. */
- if (pDescriptor->channels <= MA_MAX_CHANNELS) {
- ma_get_standard_channel_map(ma_standard_channel_map_default, pDescriptor->channels, pDescriptor->channelMap); /* <-- Cannot find info on channel order, so assuming a default. */
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
} else {
- ma_channel_map_init_blank(MA_MAX_CHANNELS, pDescriptor->channelMap); /* Too many channels. Use a blank channel map. */
+ pHeap = NULL;
}
- bufferCapacityInFrames = ((MA_PFN_AAudioStream_getBufferCapacityInFrames)pDevice->pContext->aaudio.AAudioStream_getBufferCapacityInFrames)(pStream);
- framesPerDataCallback = ((MA_PFN_AAudioStream_getFramesPerDataCallback)pDevice->pContext->aaudio.AAudioStream_getFramesPerDataCallback)(pStream);
-
- if (framesPerDataCallback > 0) {
- pDescriptor->periodSizeInFrames = framesPerDataCallback;
- pDescriptor->periodCount = bufferCapacityInFrames / framesPerDataCallback;
- } else {
- pDescriptor->periodSizeInFrames = bufferCapacityInFrames;
- pDescriptor->periodCount = 1;
+ result = ma_resampler_init_preallocated(pConfig, pHeap, pResampler);
+ if (result != MA_SUCCESS) {
+ return result;
}
- *ppStream = pStream;
-
+ pResampler->_ownsHeap = MA_TRUE;
return MA_SUCCESS;
}
-static ma_result ma_device_init__aaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+MA_API void ma_resampler_uninit(ma_resampler* pResampler, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_result result;
+ if (pResampler == NULL) {
+ return;
+ }
- MA_ASSERT(pDevice != NULL);
+ if (pResampler->pBackendVTable == NULL || pResampler->pBackendVTable->onUninit == NULL) {
+ return;
+ }
- if (pConfig->deviceType == ma_device_type_loopback) {
- return MA_DEVICE_TYPE_NOT_SUPPORTED;
+ pResampler->pBackendVTable->onUninit(pResampler->pBackendUserData, pResampler->pBackend, pAllocationCallbacks);
+
+ if (pResampler->_ownsHeap) {
+ ma_free(pResampler->_pHeap, pAllocationCallbacks);
}
+}
- /* No exclusive mode with AAudio. */
- if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) ||
- ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) {
- return MA_SHARE_MODE_NOT_SUPPORTED;
+MA_API ma_result ma_resampler_process_pcm_frames(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+{
+ if (pResampler == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
- result = ma_device_init_by_type__aaudio(pDevice, pConfig, ma_device_type_capture, pDescriptorCapture, (ma_AAudioStream**)&pDevice->aaudio.pStreamCapture);
- if (result != MA_SUCCESS) {
- return result;
- }
+ if (pFrameCountOut == NULL && pFrameCountIn == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- result = ma_device_init_by_type__aaudio(pDevice, pConfig, ma_device_type_playback, pDescriptorPlayback, (ma_AAudioStream**)&pDevice->aaudio.pStreamPlayback);
- if (result != MA_SUCCESS) {
- return result;
- }
+ if (pResampler->pBackendVTable == NULL || pResampler->pBackendVTable->onProcess == NULL) {
+ return MA_NOT_IMPLEMENTED;
}
- return MA_SUCCESS;
+ return pResampler->pBackendVTable->onProcess(pResampler->pBackendUserData, pResampler->pBackend, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
}
-static ma_result ma_device_start_stream__aaudio(ma_device* pDevice, ma_AAudioStream* pStream)
+MA_API ma_result ma_resampler_set_rate(ma_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut)
{
- ma_aaudio_result_t resultAA;
- ma_aaudio_stream_state_t currentState;
-
- MA_ASSERT(pDevice != NULL);
+ ma_result result;
- resultAA = ((MA_PFN_AAudioStream_requestStart)pDevice->pContext->aaudio.AAudioStream_requestStart)(pStream);
- if (resultAA != MA_AAUDIO_OK) {
- return ma_result_from_aaudio(resultAA);
+ if (pResampler == NULL) {
+ return MA_INVALID_ARGS;
}
- /* Do we actually need to wait for the device to transition into it's started state? */
-
- /* The device should be in either a starting or started state. If it's not set to started we need to wait for it to transition. It should go from starting to started. */
- currentState = ((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream);
- if (currentState != MA_AAUDIO_STREAM_STATE_STARTED) {
- ma_result result;
+ if (sampleRateIn == 0 || sampleRateOut == 0) {
+ return MA_INVALID_ARGS;
+ }
- if (currentState != MA_AAUDIO_STREAM_STATE_STARTING) {
- return MA_ERROR; /* Expecting the stream to be a starting or started state. */
- }
+ if (pResampler->pBackendVTable == NULL || pResampler->pBackendVTable->onSetRate == NULL) {
+ return MA_NOT_IMPLEMENTED;
+ }
- result = ma_wait_for_simple_state_transition__aaudio(pDevice->pContext, pStream, currentState, MA_AAUDIO_STREAM_STATE_STARTED);
- if (result != MA_SUCCESS) {
- return result;
- }
+ result = pResampler->pBackendVTable->onSetRate(pResampler->pBackendUserData, pResampler->pBackend, sampleRateIn, sampleRateOut);
+ if (result != MA_SUCCESS) {
+ return result;
}
+ pResampler->sampleRateIn = sampleRateIn;
+ pResampler->sampleRateOut = sampleRateOut;
+
return MA_SUCCESS;
}
-static ma_result ma_device_stop_stream__aaudio(ma_device* pDevice, ma_AAudioStream* pStream)
+MA_API ma_result ma_resampler_set_rate_ratio(ma_resampler* pResampler, float ratio)
{
- ma_aaudio_result_t resultAA;
- ma_aaudio_stream_state_t currentState;
-
- MA_ASSERT(pDevice != NULL);
+ ma_uint32 n;
+ ma_uint32 d;
- /*
- From the AAudio documentation:
+ if (pResampler == NULL) {
+ return MA_INVALID_ARGS;
+ }
- The stream will stop after all of the data currently buffered has been played.
+ if (ratio <= 0) {
+ return MA_INVALID_ARGS;
+ }
- This maps with miniaudio's requirement that device's be drained which means we don't need to implement any draining logic.
- */
+ d = 1000;
+ n = (ma_uint32)(ratio * d);
- resultAA = ((MA_PFN_AAudioStream_requestStop)pDevice->pContext->aaudio.AAudioStream_requestStop)(pStream);
- if (resultAA != MA_AAUDIO_OK) {
- return ma_result_from_aaudio(resultAA);
+ if (n == 0) {
+ return MA_INVALID_ARGS; /* Ratio too small. */
}
- /* The device should be in either a stopping or stopped state. If it's not set to started we need to wait for it to transition. It should go from stopping to stopped. */
- currentState = ((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream);
- if (currentState != MA_AAUDIO_STREAM_STATE_STOPPED) {
- ma_result result;
+ MA_ASSERT(n != 0);
- if (currentState != MA_AAUDIO_STREAM_STATE_STOPPING) {
- return MA_ERROR; /* Expecting the stream to be a stopping or stopped state. */
- }
+ return ma_resampler_set_rate(pResampler, n, d);
+}
- result = ma_wait_for_simple_state_transition__aaudio(pDevice->pContext, pStream, currentState, MA_AAUDIO_STREAM_STATE_STOPPED);
- if (result != MA_SUCCESS) {
- return result;
- }
+MA_API ma_uint64 ma_resampler_get_input_latency(const ma_resampler* pResampler)
+{
+ if (pResampler == NULL) {
+ return 0;
}
- return MA_SUCCESS;
+ if (pResampler->pBackendVTable == NULL || pResampler->pBackendVTable->onGetInputLatency == NULL) {
+ return 0;
+ }
+
+ return pResampler->pBackendVTable->onGetInputLatency(pResampler->pBackendUserData, pResampler->pBackend);
}
-static ma_result ma_device_start__aaudio(ma_device* pDevice)
+MA_API ma_uint64 ma_resampler_get_output_latency(const ma_resampler* pResampler)
{
- MA_ASSERT(pDevice != NULL);
-
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- ma_result result = ma_device_start_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture);
- if (result != MA_SUCCESS) {
- return result;
- }
+ if (pResampler == NULL) {
+ return 0;
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- ma_result result = ma_device_start_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback);
- if (result != MA_SUCCESS) {
- if (pDevice->type == ma_device_type_duplex) {
- ma_device_stop_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture);
- }
- return result;
- }
+ if (pResampler->pBackendVTable == NULL || pResampler->pBackendVTable->onGetOutputLatency == NULL) {
+ return 0;
}
- return MA_SUCCESS;
+ return pResampler->pBackendVTable->onGetOutputLatency(pResampler->pBackendUserData, pResampler->pBackend);
}
-static ma_result ma_device_stop__aaudio(ma_device* pDevice)
+MA_API ma_result ma_resampler_get_required_input_frame_count(const ma_resampler* pResampler, ma_uint64 outputFrameCount, ma_uint64* pInputFrameCount)
{
- ma_stop_proc onStop;
-
- MA_ASSERT(pDevice != NULL);
-
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- ma_result result = ma_device_stop_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture);
- if (result != MA_SUCCESS) {
- return result;
- }
+ if (pInputFrameCount == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- ma_result result = ma_device_stop_stream__aaudio(pDevice, (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback);
- if (result != MA_SUCCESS) {
- return result;
- }
+ *pInputFrameCount = 0;
+
+ if (pResampler == NULL) {
+ return MA_INVALID_ARGS;
}
- onStop = pDevice->onStop;
- if (onStop) {
- onStop(pDevice);
+ if (pResampler->pBackendVTable == NULL || pResampler->pBackendVTable->onGetRequiredInputFrameCount == NULL) {
+ return MA_NOT_IMPLEMENTED;
}
- return MA_SUCCESS;
+ return pResampler->pBackendVTable->onGetRequiredInputFrameCount(pResampler->pBackendUserData, pResampler->pBackend, outputFrameCount, pInputFrameCount);
}
-
-static ma_result ma_context_uninit__aaudio(ma_context* pContext)
+MA_API ma_result ma_resampler_get_expected_output_frame_count(const ma_resampler* pResampler, ma_uint64 inputFrameCount, ma_uint64* pOutputFrameCount)
{
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pContext->backend == ma_backend_aaudio);
+ if (pOutputFrameCount == NULL) {
+ return MA_INVALID_ARGS;
+ }
- ma_dlclose(pContext, pContext->aaudio.hAAudio);
- pContext->aaudio.hAAudio = NULL;
+ *pOutputFrameCount = 0;
- return MA_SUCCESS;
+ if (pResampler == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (pResampler->pBackendVTable == NULL || pResampler->pBackendVTable->onGetExpectedOutputFrameCount == NULL) {
+ return MA_NOT_IMPLEMENTED;
+ }
+
+ return pResampler->pBackendVTable->onGetExpectedOutputFrameCount(pResampler->pBackendUserData, pResampler->pBackend, inputFrameCount, pOutputFrameCount);
}
-static ma_result ma_context_init__aaudio(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
-{
- size_t i;
- const char* libNames[] = {
- "libaaudio.so"
- };
+/**************************************************************************************************************************************************************
- for (i = 0; i < ma_countof(libNames); ++i) {
- pContext->aaudio.hAAudio = ma_dlopen(pContext, libNames[i]);
- if (pContext->aaudio.hAAudio != NULL) {
- break;
- }
- }
+Channel Conversion
- if (pContext->aaudio.hAAudio == NULL) {
- return MA_FAILED_TO_INIT_BACKEND;
- }
+**************************************************************************************************************************************************************/
+#ifndef MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT
+#define MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT 12
+#endif
- pContext->aaudio.AAudio_createStreamBuilder = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudio_createStreamBuilder");
- pContext->aaudio.AAudioStreamBuilder_delete = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_delete");
- pContext->aaudio.AAudioStreamBuilder_setDeviceId = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setDeviceId");
- pContext->aaudio.AAudioStreamBuilder_setDirection = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setDirection");
- pContext->aaudio.AAudioStreamBuilder_setSharingMode = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setSharingMode");
- pContext->aaudio.AAudioStreamBuilder_setFormat = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setFormat");
- pContext->aaudio.AAudioStreamBuilder_setChannelCount = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setChannelCount");
- pContext->aaudio.AAudioStreamBuilder_setSampleRate = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setSampleRate");
- pContext->aaudio.AAudioStreamBuilder_setBufferCapacityInFrames = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setBufferCapacityInFrames");
- pContext->aaudio.AAudioStreamBuilder_setFramesPerDataCallback = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setFramesPerDataCallback");
- pContext->aaudio.AAudioStreamBuilder_setDataCallback = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setDataCallback");
- pContext->aaudio.AAudioStreamBuilder_setErrorCallback = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setErrorCallback");
- pContext->aaudio.AAudioStreamBuilder_setPerformanceMode = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setPerformanceMode");
- pContext->aaudio.AAudioStreamBuilder_setUsage = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setUsage");
- pContext->aaudio.AAudioStreamBuilder_setContentType = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setContentType");
- pContext->aaudio.AAudioStreamBuilder_setInputPreset = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_setInputPreset");
- pContext->aaudio.AAudioStreamBuilder_openStream = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStreamBuilder_openStream");
- pContext->aaudio.AAudioStream_close = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_close");
- pContext->aaudio.AAudioStream_getState = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getState");
- pContext->aaudio.AAudioStream_waitForStateChange = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_waitForStateChange");
- pContext->aaudio.AAudioStream_getFormat = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getFormat");
- pContext->aaudio.AAudioStream_getChannelCount = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getChannelCount");
- pContext->aaudio.AAudioStream_getSampleRate = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getSampleRate");
- pContext->aaudio.AAudioStream_getBufferCapacityInFrames = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getBufferCapacityInFrames");
- pContext->aaudio.AAudioStream_getFramesPerDataCallback = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getFramesPerDataCallback");
- pContext->aaudio.AAudioStream_getFramesPerBurst = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_getFramesPerBurst");
- pContext->aaudio.AAudioStream_requestStart = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_requestStart");
- pContext->aaudio.AAudioStream_requestStop = (ma_proc)ma_dlsym(pContext, pContext->aaudio.hAAudio, "AAudioStream_requestStop");
+#define MA_PLANE_LEFT 0
+#define MA_PLANE_RIGHT 1
+#define MA_PLANE_FRONT 2
+#define MA_PLANE_BACK 3
+#define MA_PLANE_BOTTOM 4
+#define MA_PLANE_TOP 5
+
+static float g_maChannelPlaneRatios[MA_CHANNEL_POSITION_COUNT][6] = {
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_NONE */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_MONO */
+ { 0.5f, 0.0f, 0.5f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_LEFT */
+ { 0.0f, 0.5f, 0.5f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_RIGHT */
+ { 0.0f, 0.0f, 1.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_CENTER */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_LFE */
+ { 0.5f, 0.0f, 0.0f, 0.5f, 0.0f, 0.0f}, /* MA_CHANNEL_BACK_LEFT */
+ { 0.0f, 0.5f, 0.0f, 0.5f, 0.0f, 0.0f}, /* MA_CHANNEL_BACK_RIGHT */
+ { 0.25f, 0.0f, 0.75f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_LEFT_CENTER */
+ { 0.0f, 0.25f, 0.75f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_RIGHT_CENTER */
+ { 0.0f, 0.0f, 0.0f, 1.0f, 0.0f, 0.0f}, /* MA_CHANNEL_BACK_CENTER */
+ { 1.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_SIDE_LEFT */
+ { 0.0f, 1.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_SIDE_RIGHT */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 1.0f}, /* MA_CHANNEL_TOP_CENTER */
+ { 0.33f, 0.0f, 0.33f, 0.0f, 0.0f, 0.34f}, /* MA_CHANNEL_TOP_FRONT_LEFT */
+ { 0.0f, 0.0f, 0.5f, 0.0f, 0.0f, 0.5f}, /* MA_CHANNEL_TOP_FRONT_CENTER */
+ { 0.0f, 0.33f, 0.33f, 0.0f, 0.0f, 0.34f}, /* MA_CHANNEL_TOP_FRONT_RIGHT */
+ { 0.33f, 0.0f, 0.0f, 0.33f, 0.0f, 0.34f}, /* MA_CHANNEL_TOP_BACK_LEFT */
+ { 0.0f, 0.0f, 0.0f, 0.5f, 0.0f, 0.5f}, /* MA_CHANNEL_TOP_BACK_CENTER */
+ { 0.0f, 0.33f, 0.0f, 0.33f, 0.0f, 0.34f}, /* MA_CHANNEL_TOP_BACK_RIGHT */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_0 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_1 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_2 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_3 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_4 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_5 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_6 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_7 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_8 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_9 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_10 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_11 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_12 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_13 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_14 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_15 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_16 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_17 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_18 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_19 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_20 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_21 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_22 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_23 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_24 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_25 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_26 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_27 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_28 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_29 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_30 */
+ { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_31 */
+};
+static float ma_calculate_channel_position_rectangular_weight(ma_channel channelPositionA, ma_channel channelPositionB)
+{
+ /*
+ Imagine the following simplified example: You have a single input speaker which is the front/left speaker which you want to convert to
+ the following output configuration:
- pCallbacks->onContextInit = ma_context_init__aaudio;
- pCallbacks->onContextUninit = ma_context_uninit__aaudio;
- pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__aaudio;
- pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__aaudio;
- pCallbacks->onDeviceInit = ma_device_init__aaudio;
- pCallbacks->onDeviceUninit = ma_device_uninit__aaudio;
- pCallbacks->onDeviceStart = ma_device_start__aaudio;
- pCallbacks->onDeviceStop = ma_device_stop__aaudio;
- pCallbacks->onDeviceRead = NULL; /* Not used because AAudio is asynchronous. */
- pCallbacks->onDeviceWrite = NULL; /* Not used because AAudio is asynchronous. */
- pCallbacks->onDeviceDataLoop = NULL; /* Not used because AAudio is asynchronous. */
+ - front/left
+ - side/left
+ - back/left
- (void)pConfig;
- return MA_SUCCESS;
-}
-#endif /* AAudio */
+ The front/left output is easy - it the same speaker position so it receives the full contribution of the front/left input. The amount
+ of contribution to apply to the side/left and back/left speakers, however, is a bit more complicated.
+ Imagine the front/left speaker as emitting audio from two planes - the front plane and the left plane. You can think of the front/left
+ speaker emitting half of it's total volume from the front, and the other half from the left. Since part of it's volume is being emitted
+ from the left side, and the side/left and back/left channels also emit audio from the left plane, one would expect that they would
+ receive some amount of contribution from front/left speaker. The amount of contribution depends on how many planes are shared between
+ the two speakers. Note that in the examples below I've added a top/front/left speaker as an example just to show how the math works
+ across 3 spatial dimensions.
-/******************************************************************************
+ The first thing to do is figure out how each speaker's volume is spread over each of plane:
+ - front/left: 2 planes (front and left) = 1/2 = half it's total volume on each plane
+ - side/left: 1 plane (left only) = 1/1 = entire volume from left plane
+ - back/left: 2 planes (back and left) = 1/2 = half it's total volume on each plane
+ - top/front/left: 3 planes (top, front and left) = 1/3 = one third it's total volume on each plane
-OpenSL|ES Backend
+ The amount of volume each channel contributes to each of it's planes is what controls how much it is willing to given and take to other
+ channels on the same plane. The volume that is willing to the given by one channel is multiplied by the volume that is willing to be
+ taken by the other to produce the final contribution.
+ */
-******************************************************************************/
-#ifdef MA_HAS_OPENSL
-#include
-#ifdef MA_ANDROID
-#include
-#endif
+ /* Contribution = Sum(Volume to Give * Volume to Take) */
+ float contribution =
+ g_maChannelPlaneRatios[channelPositionA][0] * g_maChannelPlaneRatios[channelPositionB][0] +
+ g_maChannelPlaneRatios[channelPositionA][1] * g_maChannelPlaneRatios[channelPositionB][1] +
+ g_maChannelPlaneRatios[channelPositionA][2] * g_maChannelPlaneRatios[channelPositionB][2] +
+ g_maChannelPlaneRatios[channelPositionA][3] * g_maChannelPlaneRatios[channelPositionB][3] +
+ g_maChannelPlaneRatios[channelPositionA][4] * g_maChannelPlaneRatios[channelPositionB][4] +
+ g_maChannelPlaneRatios[channelPositionA][5] * g_maChannelPlaneRatios[channelPositionB][5];
-typedef SLresult (SLAPIENTRY * ma_slCreateEngine_proc)(SLObjectItf* pEngine, SLuint32 numOptions, SLEngineOption* pEngineOptions, SLuint32 numInterfaces, SLInterfaceID* pInterfaceIds, SLboolean* pInterfaceRequired);
+ return contribution;
+}
-/* OpenSL|ES has one-per-application objects :( */
-static SLObjectItf g_maEngineObjectSL = NULL;
-static SLEngineItf g_maEngineSL = NULL;
-static ma_uint32 g_maOpenSLInitCounter = 0;
-static ma_spinlock g_maOpenSLSpinlock = 0; /* For init/uninit. */
+MA_API ma_channel_converter_config ma_channel_converter_config_init(ma_format format, ma_uint32 channelsIn, const ma_channel* pChannelMapIn, ma_uint32 channelsOut, const ma_channel* pChannelMapOut, ma_channel_mix_mode mixingMode)
+{
+ ma_channel_converter_config config;
-#define MA_OPENSL_OBJ(p) (*((SLObjectItf)(p)))
-#define MA_OPENSL_OUTPUTMIX(p) (*((SLOutputMixItf)(p)))
-#define MA_OPENSL_PLAY(p) (*((SLPlayItf)(p)))
-#define MA_OPENSL_RECORD(p) (*((SLRecordItf)(p)))
+ MA_ZERO_OBJECT(&config);
+ config.format = format;
+ config.channelsIn = channelsIn;
+ config.channelsOut = channelsOut;
+ config.pChannelMapIn = pChannelMapIn;
+ config.pChannelMapOut = pChannelMapOut;
+ config.mixingMode = mixingMode;
-#ifdef MA_ANDROID
-#define MA_OPENSL_BUFFERQUEUE(p) (*((SLAndroidSimpleBufferQueueItf)(p)))
-#else
-#define MA_OPENSL_BUFFERQUEUE(p) (*((SLBufferQueueItf)(p)))
-#endif
+ return config;
+}
-static ma_result ma_result_from_OpenSL(SLuint32 result)
+static ma_int32 ma_channel_converter_float_to_fixed(float x)
{
- switch (result)
- {
- case SL_RESULT_SUCCESS: return MA_SUCCESS;
- case SL_RESULT_PRECONDITIONS_VIOLATED: return MA_ERROR;
- case SL_RESULT_PARAMETER_INVALID: return MA_INVALID_ARGS;
- case SL_RESULT_MEMORY_FAILURE: return MA_OUT_OF_MEMORY;
- case SL_RESULT_RESOURCE_ERROR: return MA_INVALID_DATA;
- case SL_RESULT_RESOURCE_LOST: return MA_ERROR;
- case SL_RESULT_IO_ERROR: return MA_IO_ERROR;
- case SL_RESULT_BUFFER_INSUFFICIENT: return MA_NO_SPACE;
- case SL_RESULT_CONTENT_CORRUPTED: return MA_INVALID_DATA;
- case SL_RESULT_CONTENT_UNSUPPORTED: return MA_FORMAT_NOT_SUPPORTED;
- case SL_RESULT_CONTENT_NOT_FOUND: return MA_ERROR;
- case SL_RESULT_PERMISSION_DENIED: return MA_ACCESS_DENIED;
- case SL_RESULT_FEATURE_UNSUPPORTED: return MA_NOT_IMPLEMENTED;
- case SL_RESULT_INTERNAL_ERROR: return MA_ERROR;
- case SL_RESULT_UNKNOWN_ERROR: return MA_ERROR;
- case SL_RESULT_OPERATION_ABORTED: return MA_ERROR;
- case SL_RESULT_CONTROL_LOST: return MA_ERROR;
- default: return MA_ERROR;
- }
+ return (ma_int32)(x * (1<= MA_CHANNEL_AUX_0 && channelPosition <= MA_CHANNEL_AUX_31) {
+ return MA_FALSE;
}
-}
-/* Converts a channel mapping to an OpenSL-style channel mask. */
-static SLuint32 ma_channel_map_to_channel_mask__opensl(const ma_channel* pChannelMap, ma_uint32 channels)
-{
- SLuint32 channelMask = 0;
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < channels; ++iChannel) {
- channelMask |= ma_channel_id_to_opensl(pChannelMap[iChannel]);
+ for (i = 0; i < 6; ++i) { /* Each side of a cube. */
+ if (g_maChannelPlaneRatios[channelPosition][i] != 0) {
+ return MA_TRUE;
+ }
}
- return channelMask;
+ return MA_FALSE;
}
-/* Converts an OpenSL-style channel mask to a miniaudio channel map. */
-static void ma_channel_mask_to_channel_map__opensl(SLuint32 channelMask, ma_uint32 channels, ma_channel* pChannelMap)
+
+static ma_bool32 ma_channel_map_is_passthrough(const ma_channel* pChannelMapIn, ma_uint32 channelsIn, const ma_channel* pChannelMapOut, ma_uint32 channelsOut)
{
- if (channels == 1 && channelMask == 0) {
- pChannelMap[0] = MA_CHANNEL_MONO;
- } else if (channels == 2 && channelMask == 0) {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
+ if (channelsOut == channelsIn) {
+ return ma_channel_map_is_equal(pChannelMapOut, pChannelMapIn, channelsOut);
} else {
- if (channels == 1 && (channelMask & SL_SPEAKER_FRONT_CENTER) != 0) {
- pChannelMap[0] = MA_CHANNEL_MONO;
- } else {
- /* Just iterate over each bit. */
- ma_uint32 iChannel = 0;
- ma_uint32 iBit;
- for (iBit = 0; iBit < 32 && iChannel < channels; ++iBit) {
- SLuint32 bitValue = (channelMask & (1UL << iBit));
- if (bitValue != 0) {
- /* The bit is set. */
- pChannelMap[iChannel] = ma_channel_id_to_ma__opensl(bitValue);
- iChannel += 1;
- }
- }
- }
+ return MA_FALSE; /* Channel counts differ, so cannot be a passthrough. */
}
}
-static SLuint32 ma_round_to_standard_sample_rate__opensl(SLuint32 samplesPerSec)
+static ma_channel_conversion_path ma_channel_map_get_conversion_path(const ma_channel* pChannelMapIn, ma_uint32 channelsIn, const ma_channel* pChannelMapOut, ma_uint32 channelsOut, ma_channel_mix_mode mode)
{
- if (samplesPerSec <= SL_SAMPLINGRATE_8) {
- return SL_SAMPLINGRATE_8;
- }
- if (samplesPerSec <= SL_SAMPLINGRATE_11_025) {
- return SL_SAMPLINGRATE_11_025;
- }
- if (samplesPerSec <= SL_SAMPLINGRATE_12) {
- return SL_SAMPLINGRATE_12;
- }
- if (samplesPerSec <= SL_SAMPLINGRATE_16) {
- return SL_SAMPLINGRATE_16;
- }
- if (samplesPerSec <= SL_SAMPLINGRATE_22_05) {
- return SL_SAMPLINGRATE_22_05;
- }
- if (samplesPerSec <= SL_SAMPLINGRATE_24) {
- return SL_SAMPLINGRATE_24;
- }
- if (samplesPerSec <= SL_SAMPLINGRATE_32) {
- return SL_SAMPLINGRATE_32;
- }
- if (samplesPerSec <= SL_SAMPLINGRATE_44_1) {
- return SL_SAMPLINGRATE_44_1;
- }
- if (samplesPerSec <= SL_SAMPLINGRATE_48) {
- return SL_SAMPLINGRATE_48;
+ if (ma_channel_map_is_passthrough(pChannelMapIn, channelsIn, pChannelMapOut, channelsOut)) {
+ return ma_channel_conversion_path_passthrough;
}
- /* Android doesn't support more than 48000. */
-#ifndef MA_ANDROID
- if (samplesPerSec <= SL_SAMPLINGRATE_64) {
- return SL_SAMPLINGRATE_64;
- }
- if (samplesPerSec <= SL_SAMPLINGRATE_88_2) {
- return SL_SAMPLINGRATE_88_2;
- }
- if (samplesPerSec <= SL_SAMPLINGRATE_96) {
- return SL_SAMPLINGRATE_96;
- }
- if (samplesPerSec <= SL_SAMPLINGRATE_192) {
- return SL_SAMPLINGRATE_192;
+ if (channelsOut == 1 && (pChannelMapOut == NULL || pChannelMapOut[0] == MA_CHANNEL_MONO)) {
+ return ma_channel_conversion_path_mono_out;
}
-#endif
-
- return SL_SAMPLINGRATE_16;
-}
+ if (channelsIn == 1 && (pChannelMapIn == NULL || pChannelMapIn[0] == MA_CHANNEL_MONO)) {
+ return ma_channel_conversion_path_mono_in;
+ }
-static SLint32 ma_to_stream_type__opensl(ma_opensl_stream_type streamType)
-{
- switch (streamType) {
- case ma_opensl_stream_type_voice: return SL_ANDROID_STREAM_VOICE;
- case ma_opensl_stream_type_system: return SL_ANDROID_STREAM_SYSTEM;
- case ma_opensl_stream_type_ring: return SL_ANDROID_STREAM_RING;
- case ma_opensl_stream_type_media: return SL_ANDROID_STREAM_MEDIA;
- case ma_opensl_stream_type_alarm: return SL_ANDROID_STREAM_ALARM;
- case ma_opensl_stream_type_notification: return SL_ANDROID_STREAM_NOTIFICATION;
- default: break;
+ if (mode == ma_channel_mix_mode_custom_weights) {
+ return ma_channel_conversion_path_weights;
}
- return SL_ANDROID_STREAM_VOICE;
-}
+ /*
+ We can use a simple shuffle if both channel maps have the same channel count and all channel
+ positions are present in both.
+ */
+ if (channelsIn == channelsOut) {
+ ma_uint32 iChannelIn;
+ ma_bool32 areAllChannelPositionsPresent = MA_TRUE;
+ for (iChannelIn = 0; iChannelIn < channelsIn; ++iChannelIn) {
+ ma_bool32 isInputChannelPositionInOutput = MA_FALSE;
+ if (ma_channel_map_contains_channel_position(channelsOut, pChannelMapOut, ma_channel_map_get_channel(pChannelMapIn, channelsIn, iChannelIn))) {
+ isInputChannelPositionInOutput = MA_TRUE;
+ break;
+ }
-static SLint32 ma_to_recording_preset__opensl(ma_opensl_recording_preset recordingPreset)
-{
- switch (recordingPreset) {
- case ma_opensl_recording_preset_generic: return SL_ANDROID_RECORDING_PRESET_GENERIC;
- case ma_opensl_recording_preset_camcorder: return SL_ANDROID_RECORDING_PRESET_CAMCORDER;
- case ma_opensl_recording_preset_voice_recognition: return SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION;
- case ma_opensl_recording_preset_voice_communication: return SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION;
- case ma_opensl_recording_preset_voice_unprocessed: return SL_ANDROID_RECORDING_PRESET_UNPROCESSED;
- default: break;
+ if (!isInputChannelPositionInOutput) {
+ areAllChannelPositionsPresent = MA_FALSE;
+ break;
+ }
+ }
+
+ if (areAllChannelPositionsPresent) {
+ return ma_channel_conversion_path_shuffle;
+ }
}
- return SL_ANDROID_RECORDING_PRESET_NONE;
+ /* Getting here means we'll need to use weights. */
+ return ma_channel_conversion_path_weights;
}
-static ma_result ma_context_enumerate_devices__opensl(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+static ma_result ma_channel_map_build_shuffle_table(const ma_channel* pChannelMapIn, ma_uint32 channelCountIn, const ma_channel* pChannelMapOut, ma_uint32 channelCountOut, ma_uint8* pShuffleTable)
{
- ma_bool32 cbResult;
-
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(callback != NULL);
+ ma_uint32 iChannelIn;
+ ma_uint32 iChannelOut;
- MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to enumerate devices. */
- if (g_maOpenSLInitCounter == 0) {
- return MA_INVALID_OPERATION;
+ if (pShuffleTable == NULL || channelCountIn == 0 || channelCountOut == 0) {
+ return MA_INVALID_ARGS;
}
/*
- TODO: Test Me.
-
- This is currently untested, so for now we are just returning default devices.
+ When building the shuffle table we just do a 1:1 mapping based on the first occurance of a channel. If the
+ input channel has more than one occurance of a channel position, the second one will be ignored.
*/
-#if 0 && !defined(MA_ANDROID)
- ma_bool32 isTerminated = MA_FALSE;
+ for (iChannelOut = 0; iChannelOut < channelCountOut; iChannelOut += 1) {
+ ma_channel channelOut;
- SLuint32 pDeviceIDs[128];
- SLint32 deviceCount = sizeof(pDeviceIDs) / sizeof(pDeviceIDs[0]);
+ /* Default to MA_CHANNEL_INDEX_NULL so that if a mapping is not found it'll be set appropriately. */
+ pShuffleTable[iChannelOut] = MA_CHANNEL_INDEX_NULL;
- SLAudioIODeviceCapabilitiesItf deviceCaps;
- SLresult resultSL = (*g_maEngineObjectSL)->GetInterface(g_maEngineObjectSL, (SLInterfaceID)pContext->opensl.SL_IID_AUDIOIODEVICECAPABILITIES, &deviceCaps);
- if (resultSL != SL_RESULT_SUCCESS) {
- /* The interface may not be supported so just report a default device. */
- goto return_default_device;
- }
+ channelOut = ma_channel_map_get_channel(pChannelMapOut, channelCountOut, iChannelOut);
+ for (iChannelIn = 0; iChannelIn < channelCountIn; iChannelIn += 1) {
+ ma_channel channelIn;
- /* Playback */
- if (!isTerminated) {
- resultSL = (*deviceCaps)->GetAvailableAudioOutputs(deviceCaps, &deviceCount, pDeviceIDs);
- if (resultSL != SL_RESULT_SUCCESS) {
- return ma_result_from_OpenSL(resultSL);
- }
+ channelIn = ma_channel_map_get_channel(pChannelMapIn, channelCountIn, iChannelIn);
+ if (channelOut == channelIn) {
+ pShuffleTable[iChannelOut] = (ma_uint8)iChannelIn;
+ break;
+ }
- for (SLint32 iDevice = 0; iDevice < deviceCount; ++iDevice) {
- ma_device_info deviceInfo;
- MA_ZERO_OBJECT(&deviceInfo);
- deviceInfo.id.opensl = pDeviceIDs[iDevice];
+ /*
+ Getting here means the channels don't exactly match, but we are going to support some
+ relaxed matching for practicality. If, for example, there are two stereo channel maps,
+ but one uses front left/right and the other uses side left/right, it makes logical
+ sense to just map these. The way we'll do it is we'll check if there is a logical
+ corresponding mapping, and if so, apply it, but we will *not* break from the loop,
+ thereby giving the loop a chance to find an exact match later which will take priority.
+ */
+ switch (channelOut)
+ {
+ /* Left channels. */
+ case MA_CHANNEL_FRONT_LEFT:
+ case MA_CHANNEL_SIDE_LEFT:
+ {
+ switch (channelIn) {
+ case MA_CHANNEL_FRONT_LEFT:
+ case MA_CHANNEL_SIDE_LEFT:
+ {
+ pShuffleTable[iChannelOut] = (ma_uint8)iChannelIn;
+ } break;
+ }
+ } break;
- SLAudioOutputDescriptor desc;
- resultSL = (*deviceCaps)->QueryAudioOutputCapabilities(deviceCaps, deviceInfo.id.opensl, &desc);
- if (resultSL == SL_RESULT_SUCCESS) {
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), (const char*)desc.pDeviceName, (size_t)-1);
+ /* Right channels. */
+ case MA_CHANNEL_FRONT_RIGHT:
+ case MA_CHANNEL_SIDE_RIGHT:
+ {
+ switch (channelIn) {
+ case MA_CHANNEL_FRONT_RIGHT:
+ case MA_CHANNEL_SIDE_RIGHT:
+ {
+ pShuffleTable[iChannelOut] = (ma_uint8)iChannelIn;
+ } break;
+ }
+ } break;
- ma_bool32 cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
- if (cbResult == MA_FALSE) {
- isTerminated = MA_TRUE;
- break;
- }
+ default: break;
}
}
}
- /* Capture */
- if (!isTerminated) {
- resultSL = (*deviceCaps)->GetAvailableAudioInputs(deviceCaps, &deviceCount, pDeviceIDs);
- if (resultSL != SL_RESULT_SUCCESS) {
- return ma_result_from_OpenSL(resultSL);
- }
+ return MA_SUCCESS;
+}
- for (SLint32 iDevice = 0; iDevice < deviceCount; ++iDevice) {
- ma_device_info deviceInfo;
- MA_ZERO_OBJECT(&deviceInfo);
- deviceInfo.id.opensl = pDeviceIDs[iDevice];
- SLAudioInputDescriptor desc;
- resultSL = (*deviceCaps)->QueryAudioInputCapabilities(deviceCaps, deviceInfo.id.opensl, &desc);
- if (resultSL == SL_RESULT_SUCCESS) {
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), (const char*)desc.deviceName, (size_t)-1);
+static void ma_channel_map_apply_shuffle_table_u8(ma_uint8* pFramesOut, ma_uint32 channelsOut, const ma_uint8* pFramesIn, ma_uint32 channelsIn, ma_uint64 frameCount, const ma_uint8* pShuffleTable)
+{
+ ma_uint64 iFrame;
+ ma_uint32 iChannelOut;
- ma_bool32 cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
- if (cbResult == MA_FALSE) {
- isTerminated = MA_TRUE;
- break;
- }
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
+ ma_uint8 iChannelIn = pShuffleTable[iChannelOut];
+ if (iChannelIn < channelsIn) { /* For safety, and to deal with MA_CHANNEL_INDEX_NULL. */
+ pFramesOut[iChannelOut] = pFramesIn[iChannelIn];
+ } else {
+ pFramesOut[iChannelOut] = 0;
}
}
+
+ pFramesOut += channelsOut;
+ pFramesIn += channelsIn;
}
+}
- return MA_SUCCESS;
-#else
- goto return_default_device;
-#endif
+static void ma_channel_map_apply_shuffle_table_s16(ma_int16* pFramesOut, ma_uint32 channelsOut, const ma_int16* pFramesIn, ma_uint32 channelsIn, ma_uint64 frameCount, const ma_uint8* pShuffleTable)
+{
+ ma_uint64 iFrame;
+ ma_uint32 iChannelOut;
-return_default_device:;
- cbResult = MA_TRUE;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
+ ma_uint8 iChannelIn = pShuffleTable[iChannelOut];
+ if (iChannelIn < channelsIn) { /* For safety, and to deal with MA_CHANNEL_INDEX_NULL. */
+ pFramesOut[iChannelOut] = pFramesIn[iChannelIn];
+ } else {
+ pFramesOut[iChannelOut] = 0;
+ }
+ }
- /* Playback. */
- if (cbResult) {
- ma_device_info deviceInfo;
- MA_ZERO_OBJECT(&deviceInfo);
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
- cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
+ pFramesOut += channelsOut;
+ pFramesIn += channelsIn;
}
+}
- /* Capture. */
- if (cbResult) {
- ma_device_info deviceInfo;
- MA_ZERO_OBJECT(&deviceInfo);
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
- cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
- }
+static void ma_channel_map_apply_shuffle_table_s24(ma_uint8* pFramesOut, ma_uint32 channelsOut, const ma_uint8* pFramesIn, ma_uint32 channelsIn, ma_uint64 frameCount, const ma_uint8* pShuffleTable)
+{
+ ma_uint64 iFrame;
+ ma_uint32 iChannelOut;
- return MA_SUCCESS;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
+ ma_uint8 iChannelIn = pShuffleTable[iChannelOut];
+ if (iChannelIn < channelsIn) { /* For safety, and to deal with MA_CHANNEL_INDEX_NULL. */
+ pFramesOut[iChannelOut*3 + 0] = pFramesIn[iChannelIn*3 + 0];
+ pFramesOut[iChannelOut*3 + 1] = pFramesIn[iChannelIn*3 + 1];
+ pFramesOut[iChannelOut*3 + 2] = pFramesIn[iChannelIn*3 + 2];
+ } else {
+ pFramesOut[iChannelOut*3 + 0] = 0;
+ } pFramesOut[iChannelOut*3 + 1] = 0;
+ } pFramesOut[iChannelOut*3 + 2] = 0;
+
+ pFramesOut += channelsOut*3;
+ pFramesIn += channelsIn*3;
+ }
}
-static void ma_context_add_data_format_ex__opensl(ma_context* pContext, ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_device_info* pDeviceInfo)
+static void ma_channel_map_apply_shuffle_table_s32(ma_int32* pFramesOut, ma_uint32 channelsOut, const ma_int32* pFramesIn, ma_uint32 channelsIn, ma_uint64 frameCount, const ma_uint8* pShuffleTable)
{
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pDeviceInfo != NULL);
+ ma_uint64 iFrame;
+ ma_uint32 iChannelOut;
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].format = format;
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].channels = channels;
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = sampleRate;
- pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags = 0;
- pDeviceInfo->nativeDataFormatCount += 1;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
+ ma_uint8 iChannelIn = pShuffleTable[iChannelOut];
+ if (iChannelIn < channelsIn) { /* For safety, and to deal with MA_CHANNEL_INDEX_NULL. */
+ pFramesOut[iChannelOut] = pFramesIn[iChannelIn];
+ } else {
+ pFramesOut[iChannelOut] = 0;
+ }
+ }
+
+ pFramesOut += channelsOut;
+ pFramesIn += channelsIn;
+ }
}
-static void ma_context_add_data_format__opensl(ma_context* pContext, ma_format format, ma_device_info* pDeviceInfo)
+static void ma_channel_map_apply_shuffle_table_f32(float* pFramesOut, ma_uint32 channelsOut, const float* pFramesIn, ma_uint32 channelsIn, ma_uint64 frameCount, const ma_uint8* pShuffleTable)
{
- ma_uint32 minChannels = 1;
- ma_uint32 maxChannels = 2;
- ma_uint32 minSampleRate = (ma_uint32)ma_standard_sample_rate_8000;
- ma_uint32 maxSampleRate = (ma_uint32)ma_standard_sample_rate_48000;
- ma_uint32 iChannel;
- ma_uint32 iSampleRate;
-
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pDeviceInfo != NULL);
+ ma_uint64 iFrame;
+ ma_uint32 iChannelOut;
- /*
- Each sample format can support mono and stereo, and we'll support a small subset of standard
- rates (up to 48000). A better solution would be to somehow find a native sample rate.
- */
- for (iChannel = minChannels; iChannel < maxChannels; iChannel += 1) {
- for (iSampleRate = 0; iSampleRate < ma_countof(g_maStandardSampleRatePriorities); iSampleRate += 1) {
- ma_uint32 standardSampleRate = g_maStandardSampleRatePriorities[iSampleRate];
- if (standardSampleRate >= minSampleRate && standardSampleRate <= maxSampleRate) {
- ma_context_add_data_format_ex__opensl(pContext, format, iChannel, standardSampleRate, pDeviceInfo);
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
+ ma_uint8 iChannelIn = pShuffleTable[iChannelOut];
+ if (iChannelIn < channelsIn) { /* For safety, and to deal with MA_CHANNEL_INDEX_NULL. */
+ pFramesOut[iChannelOut] = pFramesIn[iChannelIn];
+ } else {
+ pFramesOut[iChannelOut] = 0;
}
}
+
+ pFramesOut += channelsOut;
+ pFramesIn += channelsIn;
}
}
-static ma_result ma_context_get_device_info__opensl(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+static ma_result ma_channel_map_apply_shuffle_table(void* pFramesOut, ma_uint32 channelsOut, const void* pFramesIn, ma_uint32 channelsIn, ma_uint64 frameCount, const ma_uint8* pShuffleTable, ma_format format)
{
- MA_ASSERT(pContext != NULL);
-
- MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to get device info. */
- if (g_maOpenSLInitCounter == 0) {
- return MA_INVALID_OPERATION;
+ if (pFramesOut == NULL || pFramesIn == NULL || channelsOut == 0 || pShuffleTable == NULL) {
+ return MA_INVALID_ARGS;
}
- /*
- TODO: Test Me.
+ switch (format)
+ {
+ case ma_format_u8:
+ {
+ ma_channel_map_apply_shuffle_table_u8((ma_uint8*)pFramesOut, channelsOut, (const ma_uint8*)pFramesIn, channelsIn, frameCount, pShuffleTable);
+ } break;
- This is currently untested, so for now we are just returning default devices.
- */
-#if 0 && !defined(MA_ANDROID)
- SLAudioIODeviceCapabilitiesItf deviceCaps;
- SLresult resultSL = (*g_maEngineObjectSL)->GetInterface(g_maEngineObjectSL, (SLInterfaceID)pContext->opensl.SL_IID_AUDIOIODEVICECAPABILITIES, &deviceCaps);
- if (resultSL != SL_RESULT_SUCCESS) {
- /* The interface may not be supported so just report a default device. */
- goto return_default_device;
+ case ma_format_s16:
+ {
+ ma_channel_map_apply_shuffle_table_s16((ma_int16*)pFramesOut, channelsOut, (const ma_int16*)pFramesIn, channelsIn, frameCount, pShuffleTable);
+ } break;
+
+ case ma_format_s24:
+ {
+ ma_channel_map_apply_shuffle_table_s24((ma_uint8*)pFramesOut, channelsOut, (const ma_uint8*)pFramesIn, channelsIn, frameCount, pShuffleTable);
+ } break;
+
+ case ma_format_s32:
+ {
+ ma_channel_map_apply_shuffle_table_s32((ma_int32*)pFramesOut, channelsOut, (const ma_int32*)pFramesIn, channelsIn, frameCount, pShuffleTable);
+ } break;
+
+ case ma_format_f32:
+ {
+ ma_channel_map_apply_shuffle_table_f32((float*)pFramesOut, channelsOut, (const float*)pFramesIn, channelsIn, frameCount, pShuffleTable);
+ } break;
+
+ default: return MA_INVALID_ARGS; /* Unknown format. */
}
- if (deviceType == ma_device_type_playback) {
- SLAudioOutputDescriptor desc;
- resultSL = (*deviceCaps)->QueryAudioOutputCapabilities(deviceCaps, pDeviceID->opensl, &desc);
- if (resultSL != SL_RESULT_SUCCESS) {
- return ma_result_from_OpenSL(resultSL);
- }
+ return MA_SUCCESS;
+}
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), (const char*)desc.pDeviceName, (size_t)-1);
- } else {
- SLAudioInputDescriptor desc;
- resultSL = (*deviceCaps)->QueryAudioInputCapabilities(deviceCaps, pDeviceID->opensl, &desc);
- if (resultSL != SL_RESULT_SUCCESS) {
- return ma_result_from_OpenSL(resultSL);
- }
+static ma_result ma_channel_map_apply_mono_out_f32(float* pFramesOut, const float* pFramesIn, const ma_channel* pChannelMapIn, ma_uint32 channelsIn, ma_uint64 frameCount)
+{
+ ma_uint64 iFrame;
+ ma_uint32 iChannelIn;
+ ma_uint32 accumulationCount;
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), (const char*)desc.deviceName, (size_t)-1);
+ if (pFramesOut == NULL || pFramesIn == NULL || channelsIn == 0) {
+ return MA_INVALID_ARGS;
}
- goto return_detailed_info;
-#else
- goto return_default_device;
-#endif
+ /* In this case the output stream needs to be the average of all channels, ignoring NONE. */
-return_default_device:
- if (pDeviceID != NULL) {
- if ((deviceType == ma_device_type_playback && pDeviceID->opensl != SL_DEFAULTDEVICEID_AUDIOOUTPUT) ||
- (deviceType == ma_device_type_capture && pDeviceID->opensl != SL_DEFAULTDEVICEID_AUDIOINPUT)) {
- return MA_NO_DEVICE; /* Don't know the device. */
+ /* A quick pre-processing step to get the accumulation counter since we're ignoring NONE channels. */
+ accumulationCount = 0;
+ for (iChannelIn = 0; iChannelIn < channelsIn; iChannelIn += 1) {
+ if (ma_channel_map_get_channel(pChannelMapIn, channelsIn, iChannelIn) != MA_CHANNEL_NONE) {
+ accumulationCount += 1;
}
}
- /* Name / Description */
- if (deviceType == ma_device_type_playback) {
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
+ if (accumulationCount > 0) { /* <-- Prevent a division by zero. */
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ float accumulation = 0;
+
+ for (iChannelIn = 0; iChannelIn < channelsIn; iChannelIn += 1) {
+ ma_channel channelIn = ma_channel_map_get_channel(pChannelMapIn, channelsIn, iChannelIn);
+ if (channelIn != MA_CHANNEL_NONE) {
+ accumulation += pFramesIn[iChannelIn];
+ }
+ }
+
+ pFramesOut[0] = accumulation / accumulationCount;
+ pFramesOut += 1;
+ pFramesIn += channelsIn;
+ }
} else {
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
+ ma_silence_pcm_frames(pFramesOut, frameCount, ma_format_f32, 1);
}
- pDeviceInfo->isDefault = MA_TRUE;
+ return MA_SUCCESS;
+}
- goto return_detailed_info;
+static ma_result ma_channel_map_apply_mono_in_f32(float* pFramesOut, const ma_channel* pChannelMapOut, ma_uint32 channelsOut, const float* pFramesIn, ma_uint64 frameCount, ma_mono_expansion_mode monoExpansionMode)
+{
+ ma_uint64 iFrame;
+ ma_uint32 iChannelOut;
+ if (pFramesOut == NULL || channelsOut == 0 || pFramesIn == NULL) {
+ return MA_INVALID_ARGS;
+ }
-return_detailed_info:
+ /* Note that the MA_CHANNEL_NONE channel must be ignored in all cases. */
+ switch (monoExpansionMode)
+ {
+ case ma_mono_expansion_mode_average:
+ {
+ float weight;
+ ma_uint32 validChannelCount = 0;
- /*
- For now we're just outputting a set of values that are supported by the API but not necessarily supported
- by the device natively. Later on we should work on this so that it more closely reflects the device's
- actual native format.
- */
- pDeviceInfo->nativeDataFormatCount = 0;
-#if defined(MA_ANDROID) && __ANDROID_API__ >= 21
- ma_context_add_data_format__opensl(pContext, ma_format_f32, pDeviceInfo);
-#endif
- ma_context_add_data_format__opensl(pContext, ma_format_s16, pDeviceInfo);
- ma_context_add_data_format__opensl(pContext, ma_format_u8, pDeviceInfo);
+ for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
+ ma_channel channelOut = ma_channel_map_get_channel(pChannelMapOut, channelsOut, iChannelOut);
+ if (channelOut != MA_CHANNEL_NONE) {
+ validChannelCount += 1;
+ }
+ }
- return MA_SUCCESS;
-}
+ weight = 1.0f / validChannelCount;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
+ ma_channel channelOut = ma_channel_map_get_channel(pChannelMapOut, channelsOut, iChannelOut);
+ if (channelOut != MA_CHANNEL_NONE) {
+ pFramesOut[iChannelOut] = pFramesIn[0] * weight;
+ }
+ }
-#ifdef MA_ANDROID
-/*void ma_buffer_queue_callback_capture__opensl_android(SLAndroidSimpleBufferQueueItf pBufferQueue, SLuint32 eventFlags, const void* pBuffer, SLuint32 bufferSize, SLuint32 dataUsed, void* pContext)*/
-static void ma_buffer_queue_callback_capture__opensl_android(SLAndroidSimpleBufferQueueItf pBufferQueue, void* pUserData)
-{
- ma_device* pDevice = (ma_device*)pUserData;
- size_t periodSizeInBytes;
- ma_uint8* pBuffer;
- SLresult resultSL;
+ pFramesOut += channelsOut;
+ pFramesIn += 1;
+ }
+ } break;
- MA_ASSERT(pDevice != NULL);
+ case ma_mono_expansion_mode_stereo_only:
+ {
+ if (channelsOut >= 2) {
+ ma_uint32 iChannelLeft = (ma_uint32)-1;
+ ma_uint32 iChannelRight = (ma_uint32)-1;
- (void)pBufferQueue;
+ /*
+ We first need to find our stereo channels. We prefer front-left and front-right, but
+ if they're not available, we'll also try side-left and side-right. If neither are
+ available we'll fall through to the default case below.
+ */
+ for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
+ ma_channel channelOut = ma_channel_map_get_channel(pChannelMapOut, channelsOut, iChannelOut);
+ if (channelOut == MA_CHANNEL_SIDE_LEFT) {
+ iChannelLeft = iChannelOut;
+ }
+ if (channelOut == MA_CHANNEL_SIDE_RIGHT) {
+ iChannelRight = iChannelOut;
+ }
+ }
- /*
- For now, don't do anything unless the buffer was fully processed. From what I can tell, it looks like
- OpenSL|ES 1.1 improves on buffer queues to the point that we could much more intelligently handle this,
- but unfortunately it looks like Android is only supporting OpenSL|ES 1.0.1 for now :(
- */
+ for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
+ ma_channel channelOut = ma_channel_map_get_channel(pChannelMapOut, channelsOut, iChannelOut);
+ if (channelOut == MA_CHANNEL_FRONT_LEFT) {
+ iChannelLeft = iChannelOut;
+ }
+ if (channelOut == MA_CHANNEL_FRONT_RIGHT) {
+ iChannelRight = iChannelOut;
+ }
+ }
- /* Don't do anything if the device is not started. */
- if (ma_device_get_state(pDevice) != MA_STATE_STARTED) {
- return;
- }
- /* Don't do anything if the device is being drained. */
- if (pDevice->opensl.isDrainingCapture) {
- return;
- }
+ if (iChannelLeft != (ma_uint32)-1 && iChannelRight != (ma_uint32)-1) {
+ /* We found our stereo channels so we can duplicate the signal across those channels. */
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
+ ma_channel channelOut = ma_channel_map_get_channel(pChannelMapOut, channelsOut, iChannelOut);
+ if (channelOut != MA_CHANNEL_NONE) {
+ if (iChannelOut == iChannelLeft || iChannelOut == iChannelRight) {
+ pFramesOut[iChannelOut] = pFramesIn[0];
+ } else {
+ pFramesOut[iChannelOut] = 0.0f;
+ }
+ }
+ }
- periodSizeInBytes = pDevice->capture.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
- pBuffer = pDevice->opensl.pBufferCapture + (pDevice->opensl.currentBufferIndexCapture * periodSizeInBytes);
+ pFramesOut += channelsOut;
+ pFramesIn += 1;
+ }
- ma_device_handle_backend_data_callback(pDevice, NULL, pBuffer, pDevice->capture.internalPeriodSizeInFrames);
+ break; /* Get out of the switch. */
+ } else {
+ /* Fallthrough. Does not have left and right channels. */
+ goto default_handler;
+ }
+ } else {
+ /* Fallthrough. Does not have stereo channels. */
+ goto default_handler;
+ }
+ }; /* Fallthrough. See comments above. */
- resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture, pBuffer, periodSizeInBytes);
- if (resultSL != SL_RESULT_SUCCESS) {
- return;
+ case ma_mono_expansion_mode_duplicate:
+ default:
+ {
+ default_handler:
+ {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
+ ma_channel channelOut = ma_channel_map_get_channel(pChannelMapOut, channelsOut, iChannelOut);
+ if (channelOut != MA_CHANNEL_NONE) {
+ pFramesOut[iChannelOut] = pFramesIn[0];
+ }
+ }
+
+ pFramesOut += channelsOut;
+ pFramesIn += 1;
+ }
+ }
+ } break;
}
- pDevice->opensl.currentBufferIndexCapture = (pDevice->opensl.currentBufferIndexCapture + 1) % pDevice->capture.internalPeriods;
+ return MA_SUCCESS;
}
-static void ma_buffer_queue_callback_playback__opensl_android(SLAndroidSimpleBufferQueueItf pBufferQueue, void* pUserData)
+static void ma_channel_map_apply_f32(float* pFramesOut, const ma_channel* pChannelMapOut, ma_uint32 channelsOut, const float* pFramesIn, const ma_channel* pChannelMapIn, ma_uint32 channelsIn, ma_uint64 frameCount, ma_channel_mix_mode mode, ma_mono_expansion_mode monoExpansionMode)
{
- ma_device* pDevice = (ma_device*)pUserData;
- size_t periodSizeInBytes;
- ma_uint8* pBuffer;
- SLresult resultSL;
-
- MA_ASSERT(pDevice != NULL);
+ ma_channel_conversion_path conversionPath = ma_channel_map_get_conversion_path(pChannelMapIn, channelsIn, pChannelMapOut, channelsOut, mode);
- (void)pBufferQueue;
+ /* Optimized Path: Passthrough */
+ if (conversionPath == ma_channel_conversion_path_passthrough) {
+ ma_copy_pcm_frames(pFramesOut, pFramesIn, frameCount, ma_format_f32, channelsOut);
+ return;
+ }
- /* Don't do anything if the device is not started. */
- if (ma_device_get_state(pDevice) != MA_STATE_STARTED) {
+ /* Special Path: Mono Output. */
+ if (conversionPath == ma_channel_conversion_path_mono_out) {
+ ma_channel_map_apply_mono_out_f32(pFramesOut, pFramesIn, pChannelMapIn, channelsIn, frameCount);
return;
}
- /* Don't do anything if the device is being drained. */
- if (pDevice->opensl.isDrainingPlayback) {
+ /* Special Path: Mono Input. */
+ if (conversionPath == ma_channel_conversion_path_mono_in) {
+ ma_channel_map_apply_mono_in_f32(pFramesOut, pChannelMapOut, channelsOut, pFramesIn, frameCount, monoExpansionMode);
return;
}
- periodSizeInBytes = pDevice->playback.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
- pBuffer = pDevice->opensl.pBufferPlayback + (pDevice->opensl.currentBufferIndexPlayback * periodSizeInBytes);
+ /* Getting here means we aren't running on an optimized conversion path. */
+ if (channelsOut <= MA_MAX_CHANNELS) {
+ ma_result result;
- ma_device_handle_backend_data_callback(pDevice, pBuffer, NULL, pDevice->playback.internalPeriodSizeInFrames);
+ if (mode == ma_channel_mix_mode_simple) {
+ ma_channel shuffleTable[MA_MAX_CHANNELS];
- resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback, pBuffer, periodSizeInBytes);
- if (resultSL != SL_RESULT_SUCCESS) {
- return;
- }
+ result = ma_channel_map_build_shuffle_table(pChannelMapIn, channelsIn, pChannelMapOut, channelsOut, shuffleTable);
+ if (result != MA_SUCCESS) {
+ return;
+ }
- pDevice->opensl.currentBufferIndexPlayback = (pDevice->opensl.currentBufferIndexPlayback + 1) % pDevice->playback.internalPeriods;
-}
-#endif
+ result = ma_channel_map_apply_shuffle_table(pFramesOut, channelsOut, pFramesIn, channelsIn, frameCount, shuffleTable, ma_format_f32);
+ if (result != MA_SUCCESS) {
+ return;
+ }
+ } else {
+ ma_uint32 iFrame;
+ ma_uint32 iChannelOut;
+ ma_uint32 iChannelIn;
+ float weights[32][32]; /* Do not use MA_MAX_CHANNELS here! */
-static ma_result ma_device_uninit__opensl(ma_device* pDevice)
-{
- MA_ASSERT(pDevice != NULL);
+ /*
+ If we have a small enough number of channels, pre-compute the weights. Otherwise we'll just need to
+ fall back to a slower path because otherwise we'll run out of stack space.
+ */
+ if (channelsIn <= ma_countof(weights) && channelsOut <= ma_countof(weights)) {
+ /* Pre-compute weights. */
+ for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
+ ma_channel channelOut = ma_channel_map_get_channel(pChannelMapOut, channelsOut, iChannelOut);
+ for (iChannelIn = 0; iChannelIn < channelsIn; iChannelIn += 1) {
+ ma_channel channelIn = ma_channel_map_get_channel(pChannelMapIn, channelsIn, iChannelIn);
+ weights[iChannelOut][iChannelIn] = ma_calculate_channel_position_rectangular_weight(channelOut, channelIn);
+ }
+ }
- MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it before uninitializing the device. */
- if (g_maOpenSLInitCounter == 0) {
- return MA_INVALID_OPERATION;
- }
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
+ float accumulation = 0;
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- if (pDevice->opensl.pAudioRecorderObj) {
- MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->Destroy((SLObjectItf)pDevice->opensl.pAudioRecorderObj);
- }
+ for (iChannelIn = 0; iChannelIn < channelsIn; iChannelIn += 1) {
+ accumulation += pFramesIn[iChannelIn] * weights[iChannelOut][iChannelIn];
+ }
- ma__free_from_callbacks(pDevice->opensl.pBufferCapture, &pDevice->pContext->allocationCallbacks);
- }
+ pFramesOut[iChannelOut] = accumulation;
+ }
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- if (pDevice->opensl.pAudioPlayerObj) {
- MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->Destroy((SLObjectItf)pDevice->opensl.pAudioPlayerObj);
- }
- if (pDevice->opensl.pOutputMixObj) {
- MA_OPENSL_OBJ(pDevice->opensl.pOutputMixObj)->Destroy((SLObjectItf)pDevice->opensl.pOutputMixObj);
- }
+ pFramesOut += channelsOut;
+ pFramesIn += channelsIn;
+ }
+ } else {
+ /* Cannot pre-compute weights because not enough room in stack-allocated buffer. */
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
+ float accumulation = 0;
+ ma_channel channelOut = ma_channel_map_get_channel(pChannelMapOut, channelsOut, iChannelOut);
+
+ for (iChannelIn = 0; iChannelIn < channelsIn; iChannelIn += 1) {
+ ma_channel channelIn = ma_channel_map_get_channel(pChannelMapIn, channelsIn, iChannelIn);
+ accumulation += pFramesIn[iChannelIn] * ma_calculate_channel_position_rectangular_weight(channelOut, channelIn);
+ }
- ma__free_from_callbacks(pDevice->opensl.pBufferPlayback, &pDevice->pContext->allocationCallbacks);
- }
+ pFramesOut[iChannelOut] = accumulation;
+ }
- return MA_SUCCESS;
+ pFramesOut += channelsOut;
+ pFramesIn += channelsIn;
+ }
+ }
+ }
+ } else {
+ /* Fall back to silence. If you hit this, what are you doing with so many channels?! */
+ ma_silence_pcm_frames(pFramesOut, frameCount, ma_format_f32, channelsOut);
+ }
}
-#if defined(MA_ANDROID) && __ANDROID_API__ >= 21
-typedef SLAndroidDataFormat_PCM_EX ma_SLDataFormat_PCM;
-#else
-typedef SLDataFormat_PCM ma_SLDataFormat_PCM;
-#endif
-static ma_result ma_SLDataFormat_PCM_init__opensl(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, const ma_channel* channelMap, ma_SLDataFormat_PCM* pDataFormat)
+typedef struct
{
- /* We need to convert our format/channels/rate so that they aren't set to default. */
- if (format == ma_format_unknown) {
- format = MA_DEFAULT_FORMAT;
+ size_t sizeInBytes;
+ size_t channelMapInOffset;
+ size_t channelMapOutOffset;
+ size_t shuffleTableOffset;
+ size_t weightsOffset;
+} ma_channel_converter_heap_layout;
+
+static ma_channel_conversion_path ma_channel_converter_config_get_conversion_path(const ma_channel_converter_config* pConfig)
+{
+ return ma_channel_map_get_conversion_path(pConfig->pChannelMapIn, pConfig->channelsIn, pConfig->pChannelMapOut, pConfig->channelsOut, pConfig->mixingMode);
+}
+
+static ma_result ma_channel_converter_get_heap_layout(const ma_channel_converter_config* pConfig, ma_channel_converter_heap_layout* pHeapLayout)
+{
+ ma_channel_conversion_path conversionPath;
+
+ MA_ASSERT(pHeapLayout != NULL);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- if (channels == 0) {
- channels = MA_DEFAULT_CHANNELS;
+
+ if (pConfig->channelsIn == 0 || pConfig->channelsOut == 0) {
+ return MA_INVALID_ARGS;
}
- if (sampleRate == 0) {
- sampleRate = MA_DEFAULT_SAMPLE_RATE;
+
+ if (!ma_channel_map_is_valid(pConfig->pChannelMapIn, pConfig->channelsIn)) {
+ return MA_INVALID_ARGS;
}
-#if defined(MA_ANDROID) && __ANDROID_API__ >= 21
- if (format == ma_format_f32) {
- pDataFormat->formatType = SL_ANDROID_DATAFORMAT_PCM_EX;
- pDataFormat->representation = SL_ANDROID_PCM_REPRESENTATION_FLOAT;
- } else {
- pDataFormat->formatType = SL_DATAFORMAT_PCM;
+ if (!ma_channel_map_is_valid(pConfig->pChannelMapOut, pConfig->channelsOut)) {
+ return MA_INVALID_ARGS;
}
-#else
- pDataFormat->formatType = SL_DATAFORMAT_PCM;
-#endif
- pDataFormat->numChannels = channels;
- ((SLDataFormat_PCM*)pDataFormat)->samplesPerSec = ma_round_to_standard_sample_rate__opensl(sampleRate) * 1000; /* In millihertz. Annoyingly, the sample rate variable is named differently between SLAndroidDataFormat_PCM_EX and SLDataFormat_PCM */
- pDataFormat->bitsPerSample = ma_get_bytes_per_sample(format)*8;
- pDataFormat->channelMask = ma_channel_map_to_channel_mask__opensl(channelMap, channels);
- pDataFormat->endianness = (ma_is_little_endian()) ? SL_BYTEORDER_LITTLEENDIAN : SL_BYTEORDER_BIGENDIAN;
+ pHeapLayout->sizeInBytes = 0;
- /*
- Android has a few restrictions on the format as documented here: https://developer.android.com/ndk/guides/audio/opensl-for-android.html
- - Only mono and stereo is supported.
- - Only u8 and s16 formats are supported.
- - Maximum sample rate of 48000.
- */
-#ifdef MA_ANDROID
- if (pDataFormat->numChannels > 2) {
- pDataFormat->numChannels = 2;
+ /* Input channel map. Only need to allocate this if we have an input channel map (otherwise default channel map is assumed). */
+ pHeapLayout->channelMapInOffset = pHeapLayout->sizeInBytes;
+ if (pConfig->pChannelMapIn != NULL) {
+ pHeapLayout->sizeInBytes += sizeof(ma_channel) * pConfig->channelsIn;
}
-#if __ANDROID_API__ >= 21
- if (pDataFormat->formatType == SL_ANDROID_DATAFORMAT_PCM_EX) {
- /* It's floating point. */
- MA_ASSERT(pDataFormat->representation == SL_ANDROID_PCM_REPRESENTATION_FLOAT);
- if (pDataFormat->bitsPerSample > 32) {
- pDataFormat->bitsPerSample = 32;
- }
- } else {
- if (pDataFormat->bitsPerSample > 16) {
- pDataFormat->bitsPerSample = 16;
- }
+
+ /* Output channel map. Only need to allocate this if we have an output channel map (otherwise default channel map is assumed). */
+ pHeapLayout->channelMapOutOffset = pHeapLayout->sizeInBytes;
+ if (pConfig->pChannelMapOut != NULL) {
+ pHeapLayout->sizeInBytes += sizeof(ma_channel) * pConfig->channelsOut;
}
-#else
- if (pDataFormat->bitsPerSample > 16) {
- pDataFormat->bitsPerSample = 16;
+
+ /* Alignment for the next section. */
+ pHeapLayout->sizeInBytes = ma_align_64(pHeapLayout->sizeInBytes);
+
+ /* Whether or not we use weights of a shuffle table depends on the channel map themselves and the algorithm we've chosen. */
+ conversionPath = ma_channel_converter_config_get_conversion_path(pConfig);
+
+ /* Shuffle table */
+ pHeapLayout->shuffleTableOffset = pHeapLayout->sizeInBytes;
+ if (conversionPath == ma_channel_conversion_path_shuffle) {
+ pHeapLayout->sizeInBytes += sizeof(ma_uint8) * pConfig->channelsOut;
}
-#endif
- if (((SLDataFormat_PCM*)pDataFormat)->samplesPerSec > SL_SAMPLINGRATE_48) {
- ((SLDataFormat_PCM*)pDataFormat)->samplesPerSec = SL_SAMPLINGRATE_48;
+
+ /* Weights */
+ pHeapLayout->weightsOffset = pHeapLayout->sizeInBytes;
+ if (conversionPath == ma_channel_conversion_path_weights) {
+ pHeapLayout->sizeInBytes += sizeof(float*) * pConfig->channelsIn;
+ pHeapLayout->sizeInBytes += sizeof(float ) * pConfig->channelsIn * pConfig->channelsOut;
}
-#endif
- pDataFormat->containerSize = pDataFormat->bitsPerSample; /* Always tightly packed for now. */
+ /* Make sure allocation size is aligned. */
+ pHeapLayout->sizeInBytes = ma_align_64(pHeapLayout->sizeInBytes);
return MA_SUCCESS;
}
-static ma_result ma_deconstruct_SLDataFormat_PCM__opensl(ma_SLDataFormat_PCM* pDataFormat, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
+MA_API ma_result ma_channel_converter_get_heap_size(const ma_channel_converter_config* pConfig, size_t* pHeapSizeInBytes)
{
- ma_bool32 isFloatingPoint = MA_FALSE;
-#if defined(MA_ANDROID) && __ANDROID_API__ >= 21
- if (pDataFormat->formatType == SL_ANDROID_DATAFORMAT_PCM_EX) {
- MA_ASSERT(pDataFormat->representation == SL_ANDROID_PCM_REPRESENTATION_FLOAT);
- isFloatingPoint = MA_TRUE;
+ ma_result result;
+ ma_channel_converter_heap_layout heapLayout;
+
+ if (pHeapSizeInBytes == NULL) {
+ return MA_INVALID_ARGS;
}
-#endif
- if (isFloatingPoint) {
- if (pDataFormat->bitsPerSample == 32) {
- *pFormat = ma_format_f32;
- }
- } else {
- if (pDataFormat->bitsPerSample == 8) {
- *pFormat = ma_format_u8;
- } else if (pDataFormat->bitsPerSample == 16) {
- *pFormat = ma_format_s16;
- } else if (pDataFormat->bitsPerSample == 24) {
- *pFormat = ma_format_s24;
- } else if (pDataFormat->bitsPerSample == 32) {
- *pFormat = ma_format_s32;
- }
+
+ *pHeapSizeInBytes = 0;
+
+ result = ma_channel_converter_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
}
- *pChannels = pDataFormat->numChannels;
- *pSampleRate = ((SLDataFormat_PCM*)pDataFormat)->samplesPerSec / 1000;
- ma_channel_mask_to_channel_map__opensl(pDataFormat->channelMask, ma_min(pDataFormat->numChannels, channelMapCap), pChannelMap);
+ *pHeapSizeInBytes = heapLayout.sizeInBytes;
return MA_SUCCESS;
}
-static ma_result ma_device_init__opensl(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+MA_API ma_result ma_channel_converter_init_preallocated(const ma_channel_converter_config* pConfig, void* pHeap, ma_channel_converter* pConverter)
{
-#ifdef MA_ANDROID
- SLDataLocator_AndroidSimpleBufferQueue queue;
- SLresult resultSL;
- size_t bufferSizeInBytes;
- SLInterfaceID itfIDs[2];
- const SLboolean itfIDsRequired[] = {
- SL_BOOLEAN_TRUE, /* SL_IID_ANDROIDSIMPLEBUFFERQUEUE */
- SL_BOOLEAN_FALSE /* SL_IID_ANDROIDCONFIGURATION */
- };
-#endif
+ ma_result result;
+ ma_channel_converter_heap_layout heapLayout;
- MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to initialize a new device. */
- if (g_maOpenSLInitCounter == 0) {
- return MA_INVALID_OPERATION;
+ if (pConverter == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pConfig->deviceType == ma_device_type_loopback) {
- return MA_DEVICE_TYPE_NOT_SUPPORTED;
+ MA_ZERO_OBJECT(pConverter);
+
+ result = ma_channel_converter_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
}
- /*
- For now, only supporting Android implementations of OpenSL|ES since that's the only one I've
- been able to test with and I currently depend on Android-specific extensions (simple buffer
- queues).
- */
-#ifdef MA_ANDROID
- itfIDs[0] = (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDSIMPLEBUFFERQUEUE;
- itfIDs[1] = (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDCONFIGURATION;
+ pConverter->_pHeap = pHeap;
+ MA_ZERO_MEMORY(pConverter->_pHeap, heapLayout.sizeInBytes);
- /* No exclusive mode with OpenSL|ES. */
- if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) ||
- ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) {
- return MA_SHARE_MODE_NOT_SUPPORTED;
+ pConverter->format = pConfig->format;
+ pConverter->channelsIn = pConfig->channelsIn;
+ pConverter->channelsOut = pConfig->channelsOut;
+ pConverter->mixingMode = pConfig->mixingMode;
+
+ if (pConfig->pChannelMapIn != NULL) {
+ pConverter->pChannelMapIn = (ma_channel*)ma_offset_ptr(pHeap, heapLayout.channelMapInOffset);
+ ma_channel_map_copy_or_default(pConverter->pChannelMapIn, pConfig->channelsIn, pConfig->pChannelMapIn, pConfig->channelsIn);
+ } else {
+ pConverter->pChannelMapIn = NULL; /* Use default channel map. */
}
- /* Now we can start initializing the device properly. */
- MA_ASSERT(pDevice != NULL);
- MA_ZERO_OBJECT(&pDevice->opensl);
+ if (pConfig->pChannelMapOut != NULL) {
+ pConverter->pChannelMapOut = (ma_channel*)ma_offset_ptr(pHeap, heapLayout.channelMapOutOffset);
+ ma_channel_map_copy_or_default(pConverter->pChannelMapOut, pConfig->channelsOut, pConfig->pChannelMapOut, pConfig->channelsOut);
+ } else {
+ pConverter->pChannelMapOut = NULL; /* Use default channel map. */
+ }
- queue.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
+ pConverter->conversionPath = ma_channel_converter_config_get_conversion_path(pConfig);
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
- ma_SLDataFormat_PCM pcm;
- SLDataLocator_IODevice locatorDevice;
- SLDataSource source;
- SLDataSink sink;
- SLAndroidConfigurationItf pRecorderConfig;
+ if (pConverter->conversionPath == ma_channel_conversion_path_shuffle) {
+ pConverter->pShuffleTable = (ma_uint8*)ma_offset_ptr(pHeap, heapLayout.shuffleTableOffset);
+ ma_channel_map_build_shuffle_table(pConverter->pChannelMapIn, pConverter->channelsIn, pConverter->pChannelMapOut, pConverter->channelsOut, pConverter->pShuffleTable);
+ }
- ma_SLDataFormat_PCM_init__opensl(pDescriptorCapture->format, pDescriptorCapture->channels, pDescriptorCapture->sampleRate, pDescriptorCapture->channelMap, &pcm);
+ if (pConverter->conversionPath == ma_channel_conversion_path_weights) {
+ ma_uint32 iChannelIn;
+ ma_uint32 iChannelOut;
- locatorDevice.locatorType = SL_DATALOCATOR_IODEVICE;
- locatorDevice.deviceType = SL_IODEVICE_AUDIOINPUT;
- locatorDevice.deviceID = (pDescriptorCapture->pDeviceID == NULL) ? SL_DEFAULTDEVICEID_AUDIOINPUT : pDescriptorCapture->pDeviceID->opensl;
- locatorDevice.device = NULL;
+ if (pConverter->format == ma_format_f32) {
+ pConverter->weights.f32 = (float** )ma_offset_ptr(pHeap, heapLayout.weightsOffset);
+ for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; iChannelIn += 1) {
+ pConverter->weights.f32[iChannelIn] = (float*)ma_offset_ptr(pHeap, heapLayout.weightsOffset + ((sizeof(float*) * pConverter->channelsIn) + (sizeof(float) * pConverter->channelsOut * iChannelIn)));
+ }
+ } else {
+ pConverter->weights.s16 = (ma_int32**)ma_offset_ptr(pHeap, heapLayout.weightsOffset);
+ for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; iChannelIn += 1) {
+ pConverter->weights.s16[iChannelIn] = (ma_int32*)ma_offset_ptr(pHeap, heapLayout.weightsOffset + ((sizeof(ma_int32*) * pConverter->channelsIn) + (sizeof(ma_int32) * pConverter->channelsOut * iChannelIn)));
+ }
+ }
- source.pLocator = &locatorDevice;
- source.pFormat = NULL;
+ /* Silence our weights by default. */
+ for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; iChannelIn += 1) {
+ for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; iChannelOut += 1) {
+ if (pConverter->format == ma_format_f32) {
+ pConverter->weights.f32[iChannelIn][iChannelOut] = 0.0f;
+ } else {
+ pConverter->weights.s16[iChannelIn][iChannelOut] = 0;
+ }
+ }
+ }
- queue.numBuffers = pDescriptorCapture->periodCount;
+ /*
+ We now need to fill out our weights table. This is determined by the mixing mode.
+ */
+ switch (pConverter->mixingMode)
+ {
+ case ma_channel_mix_mode_custom_weights:
+ {
+ if (pConfig->ppWeights == NULL) {
+ return MA_INVALID_ARGS; /* Config specified a custom weights mixing mode, but no custom weights have been specified. */
+ }
- sink.pLocator = &queue;
- sink.pFormat = (SLDataFormat_PCM*)&pcm;
+ for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; iChannelIn += 1) {
+ for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; iChannelOut += 1) {
+ float weight = pConfig->ppWeights[iChannelIn][iChannelOut];
- resultSL = (*g_maEngineSL)->CreateAudioRecorder(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioRecorderObj, &source, &sink, ma_countof(itfIDs), itfIDs, itfIDsRequired);
- if (resultSL == SL_RESULT_CONTENT_UNSUPPORTED) {
- /* Unsupported format. Fall back to something safer and try again. If this fails, just abort. */
- pcm.formatType = SL_DATAFORMAT_PCM;
- pcm.numChannels = 1;
- ((SLDataFormat_PCM*)&pcm)->samplesPerSec = SL_SAMPLINGRATE_16; /* The name of the sample rate variable is different between SLAndroidDataFormat_PCM_EX and SLDataFormat_PCM. */
- pcm.bitsPerSample = 16;
- pcm.containerSize = pcm.bitsPerSample; /* Always tightly packed for now. */
- pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
- resultSL = (*g_maEngineSL)->CreateAudioRecorder(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioRecorderObj, &source, &sink, ma_countof(itfIDs), itfIDs, itfIDsRequired);
- }
+ if (pConverter->format == ma_format_f32) {
+ pConverter->weights.f32[iChannelIn][iChannelOut] = weight;
+ } else {
+ pConverter->weights.s16[iChannelIn][iChannelOut] = ma_channel_converter_float_to_fixed(weight);
+ }
+ }
+ }
+ } break;
- if (resultSL != SL_RESULT_SUCCESS) {
- ma_device_uninit__opensl(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to create audio recorder.", ma_result_from_OpenSL(resultSL));
- }
+ case ma_channel_mix_mode_simple:
+ {
+ /* In simple mode, excess channels need to be silenced or dropped. */
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < ma_min(pConverter->channelsIn, pConverter->channelsOut); iChannel += 1) {
+ if (pConverter->format == ma_format_f32) {
+ if (pConverter->weights.f32[iChannel][iChannel] == 0) {
+ pConverter->weights.f32[iChannel][iChannel] = 1;
+ }
+ } else {
+ if (pConverter->weights.s16[iChannel][iChannel] == 0) {
+ pConverter->weights.s16[iChannel][iChannel] = ma_channel_converter_float_to_fixed(1);
+ }
+ }
+ }
+ } break;
+ case ma_channel_mix_mode_rectangular:
+ default:
+ {
+ /* Unmapped input channels. */
+ for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
+ ma_channel channelPosIn = pConverter->pChannelMapIn[iChannelIn];
- /* Set the recording preset before realizing the player. */
- if (pConfig->opensl.recordingPreset != ma_opensl_recording_preset_default) {
- resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioRecorderObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDCONFIGURATION, &pRecorderConfig);
- if (resultSL == SL_RESULT_SUCCESS) {
- SLint32 recordingPreset = ma_to_recording_preset__opensl(pConfig->opensl.recordingPreset);
- resultSL = (*pRecorderConfig)->SetConfiguration(pRecorderConfig, SL_ANDROID_KEY_RECORDING_PRESET, &recordingPreset, sizeof(SLint32));
- if (resultSL != SL_RESULT_SUCCESS) {
- /* Failed to set the configuration. Just keep going. */
+ if (ma_is_spatial_channel_position(channelPosIn)) {
+ if (!ma_channel_map_contains_channel_position(pConverter->channelsOut, pConverter->pChannelMapOut, channelPosIn)) {
+ for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
+ ma_channel channelPosOut = pConverter->pChannelMapOut[iChannelOut];
+
+ if (ma_is_spatial_channel_position(channelPosOut)) {
+ float weight = 0;
+ if (pConverter->mixingMode == ma_channel_mix_mode_rectangular) {
+ weight = ma_calculate_channel_position_rectangular_weight(channelPosIn, channelPosOut);
+ }
+
+ /* Only apply the weight if we haven't already got some contribution from the respective channels. */
+ if (pConverter->format == ma_format_f32) {
+ if (pConverter->weights.f32[iChannelIn][iChannelOut] == 0) {
+ pConverter->weights.f32[iChannelIn][iChannelOut] = weight;
+ }
+ } else {
+ if (pConverter->weights.s16[iChannelIn][iChannelOut] == 0) {
+ pConverter->weights.s16[iChannelIn][iChannelOut] = ma_channel_converter_float_to_fixed(weight);
+ }
+ }
+ }
+ }
+ }
+ }
}
- }
- }
- resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->Realize((SLObjectItf)pDevice->opensl.pAudioRecorderObj, SL_BOOLEAN_FALSE);
- if (resultSL != SL_RESULT_SUCCESS) {
- ma_device_uninit__opensl(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to realize audio recorder.", ma_result_from_OpenSL(resultSL));
- }
+ /* Unmapped output channels. */
+ for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
+ ma_channel channelPosOut = pConverter->pChannelMapOut[iChannelOut];
- resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioRecorderObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_RECORD, &pDevice->opensl.pAudioRecorder);
- if (resultSL != SL_RESULT_SUCCESS) {
- ma_device_uninit__opensl(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_RECORD interface.", ma_result_from_OpenSL(resultSL));
- }
+ if (ma_is_spatial_channel_position(channelPosOut)) {
+ if (!ma_channel_map_contains_channel_position(pConverter->channelsIn, pConverter->pChannelMapIn, channelPosOut)) {
+ for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
+ ma_channel channelPosIn = pConverter->pChannelMapIn[iChannelIn];
- resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioRecorderObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &pDevice->opensl.pBufferQueueCapture);
- if (resultSL != SL_RESULT_SUCCESS) {
- ma_device_uninit__opensl(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_ANDROIDSIMPLEBUFFERQUEUE interface.", ma_result_from_OpenSL(resultSL));
- }
+ if (ma_is_spatial_channel_position(channelPosIn)) {
+ float weight = 0;
+ if (pConverter->mixingMode == ma_channel_mix_mode_rectangular) {
+ weight = ma_calculate_channel_position_rectangular_weight(channelPosIn, channelPosOut);
+ }
- resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->RegisterCallback((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture, ma_buffer_queue_callback_capture__opensl_android, pDevice);
- if (resultSL != SL_RESULT_SUCCESS) {
- ma_device_uninit__opensl(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to register buffer queue callback.", ma_result_from_OpenSL(resultSL));
+ /* Only apply the weight if we haven't already got some contribution from the respective channels. */
+ if (pConverter->format == ma_format_f32) {
+ if (pConverter->weights.f32[iChannelIn][iChannelOut] == 0) {
+ pConverter->weights.f32[iChannelIn][iChannelOut] = weight;
+ }
+ } else {
+ if (pConverter->weights.s16[iChannelIn][iChannelOut] == 0) {
+ pConverter->weights.s16[iChannelIn][iChannelOut] = ma_channel_converter_float_to_fixed(weight);
+ }
+ }
+ }
+ }
+ }
+ }
+ }
+ } break;
}
+ }
- /* The internal format is determined by the "pcm" object. */
- ma_deconstruct_SLDataFormat_PCM__opensl(&pcm, &pDescriptorCapture->format, &pDescriptorCapture->channels, &pDescriptorCapture->sampleRate, pDescriptorCapture->channelMap, ma_countof(pDescriptorCapture->channelMap));
+ return MA_SUCCESS;
+}
- /* Buffer. */
- pDescriptorCapture->periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptorCapture, pDescriptorCapture->sampleRate, pConfig->performanceProfile);
- pDevice->opensl.currentBufferIndexCapture = 0;
+MA_API ma_result ma_channel_converter_init(const ma_channel_converter_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_channel_converter* pConverter)
+{
+ ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
- bufferSizeInBytes = pDescriptorCapture->periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels) * pDescriptorCapture->periodCount;
- pDevice->opensl.pBufferCapture = (ma_uint8*)ma__calloc_from_callbacks(bufferSizeInBytes, &pDevice->pContext->allocationCallbacks);
- if (pDevice->opensl.pBufferCapture == NULL) {
- ma_device_uninit__opensl(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to allocate memory for data buffer.", MA_OUT_OF_MEMORY);
+ result = ma_channel_converter_get_heap_size(pConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
}
- MA_ZERO_MEMORY(pDevice->opensl.pBufferCapture, bufferSizeInBytes);
+ } else {
+ pHeap = NULL;
}
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- ma_SLDataFormat_PCM pcm;
- SLDataSource source;
- SLDataLocator_OutputMix outmixLocator;
- SLDataSink sink;
- SLAndroidConfigurationItf pPlayerConfig;
+ result = ma_channel_converter_init_preallocated(pConfig, pHeap, pConverter);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
+ }
- ma_SLDataFormat_PCM_init__opensl(pDescriptorPlayback->format, pDescriptorPlayback->channels, pDescriptorPlayback->sampleRate, pDescriptorPlayback->channelMap, &pcm);
+ pConverter->_ownsHeap = MA_TRUE;
+ return MA_SUCCESS;
+}
- resultSL = (*g_maEngineSL)->CreateOutputMix(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pOutputMixObj, 0, NULL, NULL);
- if (resultSL != SL_RESULT_SUCCESS) {
- ma_device_uninit__opensl(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to create output mix.", ma_result_from_OpenSL(resultSL));
- }
+MA_API void ma_channel_converter_uninit(ma_channel_converter* pConverter, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pConverter == NULL) {
+ return;
+ }
- resultSL = MA_OPENSL_OBJ(pDevice->opensl.pOutputMixObj)->Realize((SLObjectItf)pDevice->opensl.pOutputMixObj, SL_BOOLEAN_FALSE);
- if (resultSL != SL_RESULT_SUCCESS) {
- ma_device_uninit__opensl(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to realize output mix object.", ma_result_from_OpenSL(resultSL));
- }
+ if (pConverter->_ownsHeap) {
+ ma_free(pConverter->_pHeap, pAllocationCallbacks);
+ }
+}
- resultSL = MA_OPENSL_OBJ(pDevice->opensl.pOutputMixObj)->GetInterface((SLObjectItf)pDevice->opensl.pOutputMixObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_OUTPUTMIX, &pDevice->opensl.pOutputMix);
- if (resultSL != SL_RESULT_SUCCESS) {
- ma_device_uninit__opensl(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_OUTPUTMIX interface.", ma_result_from_OpenSL(resultSL));
- }
+static ma_result ma_channel_converter_process_pcm_frames__passthrough(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+{
+ MA_ASSERT(pConverter != NULL);
+ MA_ASSERT(pFramesOut != NULL);
+ MA_ASSERT(pFramesIn != NULL);
- /* Set the output device. */
- if (pDescriptorPlayback->pDeviceID != NULL) {
- SLuint32 deviceID_OpenSL = pDescriptorPlayback->pDeviceID->opensl;
- MA_OPENSL_OUTPUTMIX(pDevice->opensl.pOutputMix)->ReRoute((SLOutputMixItf)pDevice->opensl.pOutputMix, 1, &deviceID_OpenSL);
- }
+ ma_copy_memory_64(pFramesOut, pFramesIn, frameCount * ma_get_bytes_per_frame(pConverter->format, pConverter->channelsOut));
+ return MA_SUCCESS;
+}
- queue.numBuffers = pDescriptorPlayback->periodCount;
+static ma_result ma_channel_converter_process_pcm_frames__shuffle(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+{
+ MA_ASSERT(pConverter != NULL);
+ MA_ASSERT(pFramesOut != NULL);
+ MA_ASSERT(pFramesIn != NULL);
+ MA_ASSERT(pConverter->channelsIn == pConverter->channelsOut);
- source.pLocator = &queue;
- source.pFormat = (SLDataFormat_PCM*)&pcm;
+ return ma_channel_map_apply_shuffle_table(pFramesOut, pConverter->channelsOut, pFramesIn, pConverter->channelsIn, frameCount, pConverter->pShuffleTable, pConverter->format);
+}
- outmixLocator.locatorType = SL_DATALOCATOR_OUTPUTMIX;
- outmixLocator.outputMix = (SLObjectItf)pDevice->opensl.pOutputMixObj;
+static ma_result ma_channel_converter_process_pcm_frames__mono_in(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+{
+ ma_uint64 iFrame;
- sink.pLocator = &outmixLocator;
- sink.pFormat = NULL;
+ MA_ASSERT(pConverter != NULL);
+ MA_ASSERT(pFramesOut != NULL);
+ MA_ASSERT(pFramesIn != NULL);
+ MA_ASSERT(pConverter->channelsIn == 1);
- resultSL = (*g_maEngineSL)->CreateAudioPlayer(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioPlayerObj, &source, &sink, ma_countof(itfIDs), itfIDs, itfIDsRequired);
- if (resultSL == SL_RESULT_CONTENT_UNSUPPORTED) {
- /* Unsupported format. Fall back to something safer and try again. If this fails, just abort. */
- pcm.formatType = SL_DATAFORMAT_PCM;
- pcm.numChannels = 2;
- ((SLDataFormat_PCM*)&pcm)->samplesPerSec = SL_SAMPLINGRATE_16;
- pcm.bitsPerSample = 16;
- pcm.containerSize = pcm.bitsPerSample; /* Always tightly packed for now. */
- pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
- resultSL = (*g_maEngineSL)->CreateAudioPlayer(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioPlayerObj, &source, &sink, ma_countof(itfIDs), itfIDs, itfIDsRequired);
- }
+ switch (pConverter->format)
+ {
+ case ma_format_u8:
+ {
+ /* */ ma_uint8* pFramesOutU8 = ( ma_uint8*)pFramesOut;
+ const ma_uint8* pFramesInU8 = (const ma_uint8*)pFramesIn;
- if (resultSL != SL_RESULT_SUCCESS) {
- ma_device_uninit__opensl(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to create audio player.", ma_result_from_OpenSL(resultSL));
- }
+ for (iFrame = 0; iFrame < frameCount; ++iFrame) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < pConverter->channelsOut; iChannel += 1) {
+ pFramesOutU8[iFrame*pConverter->channelsOut + iChannel] = pFramesInU8[iFrame];
+ }
+ }
+ } break;
+ case ma_format_s16:
+ {
+ /* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut;
+ const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn;
- /* Set the stream type before realizing the player. */
- if (pConfig->opensl.streamType != ma_opensl_stream_type_default) {
- resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioPlayerObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDCONFIGURATION, &pPlayerConfig);
- if (resultSL == SL_RESULT_SUCCESS) {
- SLint32 streamType = ma_to_stream_type__opensl(pConfig->opensl.streamType);
- resultSL = (*pPlayerConfig)->SetConfiguration(pPlayerConfig, SL_ANDROID_KEY_STREAM_TYPE, &streamType, sizeof(SLint32));
- if (resultSL != SL_RESULT_SUCCESS) {
- /* Failed to set the configuration. Just keep going. */
+ if (pConverter->channelsOut == 2) {
+ for (iFrame = 0; iFrame < frameCount; ++iFrame) {
+ pFramesOutS16[iFrame*2 + 0] = pFramesInS16[iFrame];
+ pFramesOutS16[iFrame*2 + 1] = pFramesInS16[iFrame];
+ }
+ } else {
+ for (iFrame = 0; iFrame < frameCount; ++iFrame) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < pConverter->channelsOut; iChannel += 1) {
+ pFramesOutS16[iFrame*pConverter->channelsOut + iChannel] = pFramesInS16[iFrame];
+ }
}
}
- }
+ } break;
- resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->Realize((SLObjectItf)pDevice->opensl.pAudioPlayerObj, SL_BOOLEAN_FALSE);
- if (resultSL != SL_RESULT_SUCCESS) {
- ma_device_uninit__opensl(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to realize audio player.", ma_result_from_OpenSL(resultSL));
- }
+ case ma_format_s24:
+ {
+ /* */ ma_uint8* pFramesOutS24 = ( ma_uint8*)pFramesOut;
+ const ma_uint8* pFramesInS24 = (const ma_uint8*)pFramesIn;
- resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioPlayerObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_PLAY, &pDevice->opensl.pAudioPlayer);
- if (resultSL != SL_RESULT_SUCCESS) {
- ma_device_uninit__opensl(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_PLAY interface.", ma_result_from_OpenSL(resultSL));
- }
+ for (iFrame = 0; iFrame < frameCount; ++iFrame) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < pConverter->channelsOut; iChannel += 1) {
+ ma_uint64 iSampleOut = iFrame*pConverter->channelsOut + iChannel;
+ ma_uint64 iSampleIn = iFrame;
+ pFramesOutS24[iSampleOut*3 + 0] = pFramesInS24[iSampleIn*3 + 0];
+ pFramesOutS24[iSampleOut*3 + 1] = pFramesInS24[iSampleIn*3 + 1];
+ pFramesOutS24[iSampleOut*3 + 2] = pFramesInS24[iSampleIn*3 + 2];
+ }
+ }
+ } break;
- resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioPlayerObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &pDevice->opensl.pBufferQueuePlayback);
- if (resultSL != SL_RESULT_SUCCESS) {
- ma_device_uninit__opensl(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_ANDROIDSIMPLEBUFFERQUEUE interface.", ma_result_from_OpenSL(resultSL));
- }
+ case ma_format_s32:
+ {
+ /* */ ma_int32* pFramesOutS32 = ( ma_int32*)pFramesOut;
+ const ma_int32* pFramesInS32 = (const ma_int32*)pFramesIn;
- resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->RegisterCallback((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback, ma_buffer_queue_callback_playback__opensl_android, pDevice);
- if (resultSL != SL_RESULT_SUCCESS) {
- ma_device_uninit__opensl(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to register buffer queue callback.", ma_result_from_OpenSL(resultSL));
- }
+ for (iFrame = 0; iFrame < frameCount; ++iFrame) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < pConverter->channelsOut; iChannel += 1) {
+ pFramesOutS32[iFrame*pConverter->channelsOut + iChannel] = pFramesInS32[iFrame];
+ }
+ }
+ } break;
- /* The internal format is determined by the "pcm" object. */
- ma_deconstruct_SLDataFormat_PCM__opensl(&pcm, &pDescriptorPlayback->format, &pDescriptorPlayback->channels, &pDescriptorPlayback->sampleRate, pDescriptorPlayback->channelMap, ma_countof(pDescriptorPlayback->channelMap));
+ case ma_format_f32:
+ {
+ /* */ float* pFramesOutF32 = ( float*)pFramesOut;
+ const float* pFramesInF32 = (const float*)pFramesIn;
- /* Buffer. */
- pDescriptorPlayback->periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptorPlayback, pDescriptorPlayback->sampleRate, pConfig->performanceProfile);
- pDevice->opensl.currentBufferIndexPlayback = 0;
+ if (pConverter->channelsOut == 2) {
+ for (iFrame = 0; iFrame < frameCount; ++iFrame) {
+ pFramesOutF32[iFrame*2 + 0] = pFramesInF32[iFrame];
+ pFramesOutF32[iFrame*2 + 1] = pFramesInF32[iFrame];
+ }
+ } else {
+ for (iFrame = 0; iFrame < frameCount; ++iFrame) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < pConverter->channelsOut; iChannel += 1) {
+ pFramesOutF32[iFrame*pConverter->channelsOut + iChannel] = pFramesInF32[iFrame];
+ }
+ }
+ }
+ } break;
- bufferSizeInBytes = pDescriptorPlayback->periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorPlayback->format, pDescriptorPlayback->channels) * pDescriptorPlayback->periodCount;
- pDevice->opensl.pBufferPlayback = (ma_uint8*)ma__calloc_from_callbacks(bufferSizeInBytes, &pDevice->pContext->allocationCallbacks);
- if (pDevice->opensl.pBufferPlayback == NULL) {
- ma_device_uninit__opensl(pDevice);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to allocate memory for data buffer.", MA_OUT_OF_MEMORY);
- }
- MA_ZERO_MEMORY(pDevice->opensl.pBufferPlayback, bufferSizeInBytes);
+ default: return MA_INVALID_OPERATION; /* Unknown format. */
}
return MA_SUCCESS;
-#else
- return MA_NO_BACKEND; /* Non-Android implementations are not supported. */
-#endif
}
-static ma_result ma_device_start__opensl(ma_device* pDevice)
+static ma_result ma_channel_converter_process_pcm_frames__mono_out(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
- SLresult resultSL;
- size_t periodSizeInBytes;
- ma_uint32 iPeriod;
+ ma_uint64 iFrame;
+ ma_uint32 iChannel;
- MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(pConverter != NULL);
+ MA_ASSERT(pFramesOut != NULL);
+ MA_ASSERT(pFramesIn != NULL);
+ MA_ASSERT(pConverter->channelsOut == 1);
- MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to start the device. */
- if (g_maOpenSLInitCounter == 0) {
- return MA_INVALID_OPERATION;
- }
+ switch (pConverter->format)
+ {
+ case ma_format_u8:
+ {
+ /* */ ma_uint8* pFramesOutU8 = ( ma_uint8*)pFramesOut;
+ const ma_uint8* pFramesInU8 = (const ma_uint8*)pFramesIn;
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- resultSL = MA_OPENSL_RECORD(pDevice->opensl.pAudioRecorder)->SetRecordState((SLRecordItf)pDevice->opensl.pAudioRecorder, SL_RECORDSTATE_RECORDING);
- if (resultSL != SL_RESULT_SUCCESS) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to start internal capture device.", ma_result_from_OpenSL(resultSL));
- }
+ for (iFrame = 0; iFrame < frameCount; ++iFrame) {
+ ma_int32 t = 0;
+ for (iChannel = 0; iChannel < pConverter->channelsIn; iChannel += 1) {
+ t += ma_pcm_sample_u8_to_s16_no_scale(pFramesInU8[iFrame*pConverter->channelsIn + iChannel]);
+ }
- periodSizeInBytes = pDevice->capture.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
- for (iPeriod = 0; iPeriod < pDevice->capture.internalPeriods; ++iPeriod) {
- resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture, pDevice->opensl.pBufferCapture + (periodSizeInBytes * iPeriod), periodSizeInBytes);
- if (resultSL != SL_RESULT_SUCCESS) {
- MA_OPENSL_RECORD(pDevice->opensl.pAudioRecorder)->SetRecordState((SLRecordItf)pDevice->opensl.pAudioRecorder, SL_RECORDSTATE_STOPPED);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to enqueue buffer for capture device.", ma_result_from_OpenSL(resultSL));
+ pFramesOutU8[iFrame] = ma_clip_u8(t / pConverter->channelsOut);
}
- }
- }
+ } break;
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- resultSL = MA_OPENSL_PLAY(pDevice->opensl.pAudioPlayer)->SetPlayState((SLPlayItf)pDevice->opensl.pAudioPlayer, SL_PLAYSTATE_PLAYING);
- if (resultSL != SL_RESULT_SUCCESS) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to start internal playback device.", ma_result_from_OpenSL(resultSL));
- }
+ case ma_format_s16:
+ {
+ /* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut;
+ const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn;
- /* In playback mode (no duplex) we need to load some initial buffers. In duplex mode we need to enqueu silent buffers. */
- if (pDevice->type == ma_device_type_duplex) {
- MA_ZERO_MEMORY(pDevice->opensl.pBufferPlayback, pDevice->playback.internalPeriodSizeInFrames * pDevice->playback.internalPeriods * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels));
- } else {
- ma_device__read_frames_from_client(pDevice, pDevice->playback.internalPeriodSizeInFrames * pDevice->playback.internalPeriods, pDevice->opensl.pBufferPlayback);
- }
+ for (iFrame = 0; iFrame < frameCount; ++iFrame) {
+ ma_int32 t = 0;
+ for (iChannel = 0; iChannel < pConverter->channelsIn; iChannel += 1) {
+ t += pFramesInS16[iFrame*pConverter->channelsIn + iChannel];
+ }
- periodSizeInBytes = pDevice->playback.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
- for (iPeriod = 0; iPeriod < pDevice->playback.internalPeriods; ++iPeriod) {
- resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback, pDevice->opensl.pBufferPlayback + (periodSizeInBytes * iPeriod), periodSizeInBytes);
- if (resultSL != SL_RESULT_SUCCESS) {
- MA_OPENSL_PLAY(pDevice->opensl.pAudioPlayer)->SetPlayState((SLPlayItf)pDevice->opensl.pAudioPlayer, SL_PLAYSTATE_STOPPED);
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to enqueue buffer for playback device.", ma_result_from_OpenSL(resultSL));
+ pFramesOutS16[iFrame] = (ma_int16)(t / pConverter->channelsIn);
}
- }
- }
+ } break;
- return MA_SUCCESS;
-}
+ case ma_format_s24:
+ {
+ /* */ ma_uint8* pFramesOutS24 = ( ma_uint8*)pFramesOut;
+ const ma_uint8* pFramesInS24 = (const ma_uint8*)pFramesIn;
-static ma_result ma_device_drain__opensl(ma_device* pDevice, ma_device_type deviceType)
-{
- SLAndroidSimpleBufferQueueItf pBufferQueue;
+ for (iFrame = 0; iFrame < frameCount; ++iFrame) {
+ ma_int64 t = 0;
+ for (iChannel = 0; iChannel < pConverter->channelsIn; iChannel += 1) {
+ t += ma_pcm_sample_s24_to_s32_no_scale(&pFramesInS24[(iFrame*pConverter->channelsIn + iChannel)*3]);
+ }
- MA_ASSERT(deviceType == ma_device_type_capture || deviceType == ma_device_type_playback);
+ ma_pcm_sample_s32_to_s24_no_scale(t / pConverter->channelsIn, &pFramesOutS24[iFrame*3]);
+ }
+ } break;
- if (pDevice->type == ma_device_type_capture) {
- pBufferQueue = (SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture;
- pDevice->opensl.isDrainingCapture = MA_TRUE;
- } else {
- pBufferQueue = (SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback;
- pDevice->opensl.isDrainingPlayback = MA_TRUE;
- }
+ case ma_format_s32:
+ {
+ /* */ ma_int32* pFramesOutS32 = ( ma_int32*)pFramesOut;
+ const ma_int32* pFramesInS32 = (const ma_int32*)pFramesIn;
- for (;;) {
- SLAndroidSimpleBufferQueueState state;
+ for (iFrame = 0; iFrame < frameCount; ++iFrame) {
+ ma_int64 t = 0;
+ for (iChannel = 0; iChannel < pConverter->channelsIn; iChannel += 1) {
+ t += pFramesInS32[iFrame*pConverter->channelsIn + iChannel];
+ }
- MA_OPENSL_BUFFERQUEUE(pBufferQueue)->GetState(pBufferQueue, &state);
- if (state.count == 0) {
- break;
- }
+ pFramesOutS32[iFrame] = (ma_int32)(t / pConverter->channelsIn);
+ }
+ } break;
- ma_sleep(10);
- }
+ case ma_format_f32:
+ {
+ /* */ float* pFramesOutF32 = ( float*)pFramesOut;
+ const float* pFramesInF32 = (const float*)pFramesIn;
- if (pDevice->type == ma_device_type_capture) {
- pDevice->opensl.isDrainingCapture = MA_FALSE;
- } else {
- pDevice->opensl.isDrainingPlayback = MA_FALSE;
+ for (iFrame = 0; iFrame < frameCount; ++iFrame) {
+ float t = 0;
+ for (iChannel = 0; iChannel < pConverter->channelsIn; iChannel += 1) {
+ t += pFramesInF32[iFrame*pConverter->channelsIn + iChannel];
+ }
+
+ pFramesOutF32[iFrame] = t / pConverter->channelsIn;
+ }
+ } break;
+
+ default: return MA_INVALID_OPERATION; /* Unknown format. */
}
return MA_SUCCESS;
}
-static ma_result ma_device_stop__opensl(ma_device* pDevice)
+static ma_result ma_channel_converter_process_pcm_frames__weights(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
- SLresult resultSL;
- ma_stop_proc onStop;
+ ma_uint32 iFrame;
+ ma_uint32 iChannelIn;
+ ma_uint32 iChannelOut;
- MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(pConverter != NULL);
+ MA_ASSERT(pFramesOut != NULL);
+ MA_ASSERT(pFramesIn != NULL);
- MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it before stopping/uninitializing the device. */
- if (g_maOpenSLInitCounter == 0) {
- return MA_INVALID_OPERATION;
- }
+ /* This is the more complicated case. Each of the output channels is accumulated with 0 or more input channels. */
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- ma_device_drain__opensl(pDevice, ma_device_type_capture);
+ /* Clear. */
+ ma_zero_memory_64(pFramesOut, frameCount * ma_get_bytes_per_frame(pConverter->format, pConverter->channelsOut));
- resultSL = MA_OPENSL_RECORD(pDevice->opensl.pAudioRecorder)->SetRecordState((SLRecordItf)pDevice->opensl.pAudioRecorder, SL_RECORDSTATE_STOPPED);
- if (resultSL != SL_RESULT_SUCCESS) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to stop internal capture device.", ma_result_from_OpenSL(resultSL));
- }
+ /* Accumulate. */
+ switch (pConverter->format)
+ {
+ case ma_format_u8:
+ {
+ /* */ ma_uint8* pFramesOutU8 = ( ma_uint8*)pFramesOut;
+ const ma_uint8* pFramesInU8 = (const ma_uint8*)pFramesIn;
- MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->Clear((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture);
- }
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
+ for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
+ ma_int16 u8_O = ma_pcm_sample_u8_to_s16_no_scale(pFramesOutU8[iFrame*pConverter->channelsOut + iChannelOut]);
+ ma_int16 u8_I = ma_pcm_sample_u8_to_s16_no_scale(pFramesInU8 [iFrame*pConverter->channelsIn + iChannelIn ]);
+ ma_int32 s = (ma_int32)ma_clamp(u8_O + ((u8_I * pConverter->weights.s16[iChannelIn][iChannelOut]) >> MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT), -128, 127);
+ pFramesOutU8[iFrame*pConverter->channelsOut + iChannelOut] = ma_clip_u8((ma_int16)s);
+ }
+ }
+ }
+ } break;
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- ma_device_drain__opensl(pDevice, ma_device_type_playback);
+ case ma_format_s16:
+ {
+ /* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut;
+ const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn;
- resultSL = MA_OPENSL_PLAY(pDevice->opensl.pAudioPlayer)->SetPlayState((SLPlayItf)pDevice->opensl.pAudioPlayer, SL_PLAYSTATE_STOPPED);
- if (resultSL != SL_RESULT_SUCCESS) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to stop internal playback device.", ma_result_from_OpenSL(resultSL));
- }
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
+ for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
+ ma_int32 s = pFramesOutS16[iFrame*pConverter->channelsOut + iChannelOut];
+ s += (pFramesInS16[iFrame*pConverter->channelsIn + iChannelIn] * pConverter->weights.s16[iChannelIn][iChannelOut]) >> MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT;
- MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->Clear((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback);
- }
+ pFramesOutS16[iFrame*pConverter->channelsOut + iChannelOut] = (ma_int16)ma_clamp(s, -32768, 32767);
+ }
+ }
+ }
+ } break;
- /* Make sure the client is aware that the device has stopped. There may be an OpenSL|ES callback for this, but I haven't found it. */
- onStop = pDevice->onStop;
- if (onStop) {
- onStop(pDevice);
- }
+ case ma_format_s24:
+ {
+ /* */ ma_uint8* pFramesOutS24 = ( ma_uint8*)pFramesOut;
+ const ma_uint8* pFramesInS24 = (const ma_uint8*)pFramesIn;
- return MA_SUCCESS;
-}
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
+ for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
+ ma_int64 s24_O = ma_pcm_sample_s24_to_s32_no_scale(&pFramesOutS24[(iFrame*pConverter->channelsOut + iChannelOut)*3]);
+ ma_int64 s24_I = ma_pcm_sample_s24_to_s32_no_scale(&pFramesInS24 [(iFrame*pConverter->channelsIn + iChannelIn )*3]);
+ ma_int64 s24 = (ma_int32)ma_clamp(s24_O + ((s24_I * pConverter->weights.s16[iChannelIn][iChannelOut]) >> MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT), -8388608, 8388607);
+ ma_pcm_sample_s32_to_s24_no_scale(s24, &pFramesOutS24[(iFrame*pConverter->channelsOut + iChannelOut)*3]);
+ }
+ }
+ }
+ } break;
+ case ma_format_s32:
+ {
+ /* */ ma_int32* pFramesOutS32 = ( ma_int32*)pFramesOut;
+ const ma_int32* pFramesInS32 = (const ma_int32*)pFramesIn;
-static ma_result ma_context_uninit__opensl(ma_context* pContext)
-{
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pContext->backend == ma_backend_opensl);
- (void)pContext;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
+ for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
+ ma_int64 s = pFramesOutS32[iFrame*pConverter->channelsOut + iChannelOut];
+ s += ((ma_int64)pFramesInS32[iFrame*pConverter->channelsIn + iChannelIn] * pConverter->weights.s16[iChannelIn][iChannelOut]) >> MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT;
- /* Uninit global data. */
- ma_spinlock_lock(&g_maOpenSLSpinlock);
- {
- MA_ASSERT(g_maOpenSLInitCounter > 0); /* If you've triggered this, it means you have ma_context_init/uninit mismatch. Each successful call to ma_context_init() must be matched up with a call to ma_context_uninit(). */
+ pFramesOutS32[iFrame*pConverter->channelsOut + iChannelOut] = ma_clip_s32(s);
+ }
+ }
+ }
+ } break;
- g_maOpenSLInitCounter -= 1;
- if (g_maOpenSLInitCounter == 0) {
- (*g_maEngineObjectSL)->Destroy(g_maEngineObjectSL);
- }
- }
- ma_spinlock_unlock(&g_maOpenSLSpinlock);
+ case ma_format_f32:
+ {
+ /* */ float* pFramesOutF32 = ( float*)pFramesOut;
+ const float* pFramesInF32 = (const float*)pFramesIn;
- return MA_SUCCESS;
-}
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
+ for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
+ pFramesOutF32[iFrame*pConverter->channelsOut + iChannelOut] += pFramesInF32[iFrame*pConverter->channelsIn + iChannelIn] * pConverter->weights.f32[iChannelIn][iChannelOut];
+ }
+ }
+ }
+ } break;
-static ma_result ma_dlsym_SLInterfaceID__opensl(ma_context* pContext, const char* pName, ma_handle* pHandle)
-{
- /* We need to return an error if the symbol cannot be found. This is important because there have been reports that some symbols do not exist. */
- ma_handle* p = (ma_handle*)ma_dlsym(pContext, pContext->opensl.libOpenSLES, pName);
- if (p == NULL) {
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_INFO, "[OpenSL|ES] Cannot find symbol %s", pName);
- return MA_NO_BACKEND;
+ default: return MA_INVALID_OPERATION; /* Unknown format. */
}
- *pHandle = *p;
return MA_SUCCESS;
}
-static ma_result ma_context_init_engine_nolock__opensl(ma_context* pContext)
+MA_API ma_result ma_channel_converter_process_pcm_frames(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
- g_maOpenSLInitCounter += 1;
- if (g_maOpenSLInitCounter == 1) {
- SLresult resultSL;
+ if (pConverter == NULL) {
+ return MA_INVALID_ARGS;
+ }
- resultSL = ((ma_slCreateEngine_proc)pContext->opensl.slCreateEngine)(&g_maEngineObjectSL, 0, NULL, 0, NULL, NULL);
- if (resultSL != SL_RESULT_SUCCESS) {
- g_maOpenSLInitCounter -= 1;
- return ma_result_from_OpenSL(resultSL);
- }
+ if (pFramesOut == NULL) {
+ return MA_INVALID_ARGS;
+ }
- (*g_maEngineObjectSL)->Realize(g_maEngineObjectSL, SL_BOOLEAN_FALSE);
+ if (pFramesIn == NULL) {
+ ma_zero_memory_64(pFramesOut, frameCount * ma_get_bytes_per_frame(pConverter->format, pConverter->channelsOut));
+ return MA_SUCCESS;
+ }
- resultSL = (*g_maEngineObjectSL)->GetInterface(g_maEngineObjectSL, (SLInterfaceID)pContext->opensl.SL_IID_ENGINE, &g_maEngineSL);
- if (resultSL != SL_RESULT_SUCCESS) {
- (*g_maEngineObjectSL)->Destroy(g_maEngineObjectSL);
- g_maOpenSLInitCounter -= 1;
- return ma_result_from_OpenSL(resultSL);
+ switch (pConverter->conversionPath)
+ {
+ case ma_channel_conversion_path_passthrough: return ma_channel_converter_process_pcm_frames__passthrough(pConverter, pFramesOut, pFramesIn, frameCount);
+ case ma_channel_conversion_path_mono_out: return ma_channel_converter_process_pcm_frames__mono_out(pConverter, pFramesOut, pFramesIn, frameCount);
+ case ma_channel_conversion_path_mono_in: return ma_channel_converter_process_pcm_frames__mono_in(pConverter, pFramesOut, pFramesIn, frameCount);
+ case ma_channel_conversion_path_shuffle: return ma_channel_converter_process_pcm_frames__shuffle(pConverter, pFramesOut, pFramesIn, frameCount);
+ case ma_channel_conversion_path_weights:
+ default:
+ {
+ return ma_channel_converter_process_pcm_frames__weights(pConverter, pFramesOut, pFramesIn, frameCount);
}
}
-
- return MA_SUCCESS;
}
-static ma_result ma_context_init__opensl(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+MA_API ma_result ma_channel_converter_get_input_channel_map(const ma_channel_converter* pConverter, ma_channel* pChannelMap, size_t channelMapCap)
{
- ma_result result;
-
-#if !defined(MA_NO_RUNTIME_LINKING)
- size_t i;
- const char* libOpenSLESNames[] = {
- "libOpenSLES.so"
- };
-#endif
-
- MA_ASSERT(pContext != NULL);
-
- (void)pConfig;
-
-#if !defined(MA_NO_RUNTIME_LINKING)
- /*
- Dynamically link against libOpenSLES.so. I have now had multiple reports that SL_IID_ANDROIDSIMPLEBUFFERQUEUE cannot be found. One
- report was happening at compile time and another at runtime. To try working around this, I'm going to link to libOpenSLES at runtime
- and extract the symbols rather than reference them directly. This should, hopefully, fix these issues as the compiler won't see any
- references to the symbols and will hopefully skip the checks.
- */
- for (i = 0; i < ma_countof(libOpenSLESNames); i += 1) {
- pContext->opensl.libOpenSLES = ma_dlopen(pContext, libOpenSLESNames[i]);
- if (pContext->opensl.libOpenSLES != NULL) {
- break;
- }
+ if (pConverter == NULL || pChannelMap == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pContext->opensl.libOpenSLES == NULL) {
- ma_post_log_message(pContext, NULL, MA_LOG_LEVEL_INFO, "[OpenSL|ES] Could not find libOpenSLES.so");
- return MA_NO_BACKEND;
- }
+ ma_channel_map_copy_or_default(pChannelMap, channelMapCap, pConverter->pChannelMapIn, pConverter->channelsIn);
- result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_ENGINE", &pContext->opensl.SL_IID_ENGINE);
- if (result != MA_SUCCESS) {
- ma_dlclose(pContext, pContext->opensl.libOpenSLES);
- return result;
- }
+ return MA_SUCCESS;
+}
- result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_AUDIOIODEVICECAPABILITIES", &pContext->opensl.SL_IID_AUDIOIODEVICECAPABILITIES);
- if (result != MA_SUCCESS) {
- ma_dlclose(pContext, pContext->opensl.libOpenSLES);
- return result;
+MA_API ma_result ma_channel_converter_get_output_channel_map(const ma_channel_converter* pConverter, ma_channel* pChannelMap, size_t channelMapCap)
+{
+ if (pConverter == NULL || pChannelMap == NULL) {
+ return MA_INVALID_ARGS;
}
- result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_ANDROIDSIMPLEBUFFERQUEUE", &pContext->opensl.SL_IID_ANDROIDSIMPLEBUFFERQUEUE);
- if (result != MA_SUCCESS) {
- ma_dlclose(pContext, pContext->opensl.libOpenSLES);
- return result;
- }
+ ma_channel_map_copy_or_default(pChannelMap, channelMapCap, pConverter->pChannelMapOut, pConverter->channelsOut);
- result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_RECORD", &pContext->opensl.SL_IID_RECORD);
- if (result != MA_SUCCESS) {
- ma_dlclose(pContext, pContext->opensl.libOpenSLES);
- return result;
- }
+ return MA_SUCCESS;
+}
- result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_PLAY", &pContext->opensl.SL_IID_PLAY);
- if (result != MA_SUCCESS) {
- ma_dlclose(pContext, pContext->opensl.libOpenSLES);
- return result;
- }
- result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_OUTPUTMIX", &pContext->opensl.SL_IID_OUTPUTMIX);
- if (result != MA_SUCCESS) {
- ma_dlclose(pContext, pContext->opensl.libOpenSLES);
- return result;
- }
+/**************************************************************************************************************************************************************
- result = ma_dlsym_SLInterfaceID__opensl(pContext, "SL_IID_ANDROIDCONFIGURATION", &pContext->opensl.SL_IID_ANDROIDCONFIGURATION);
- if (result != MA_SUCCESS) {
- ma_dlclose(pContext, pContext->opensl.libOpenSLES);
- return result;
- }
+Data Conversion
- pContext->opensl.slCreateEngine = (ma_proc)ma_dlsym(pContext, pContext->opensl.libOpenSLES, "slCreateEngine");
- if (pContext->opensl.slCreateEngine == NULL) {
- ma_dlclose(pContext, pContext->opensl.libOpenSLES);
- ma_post_log_message(pContext, NULL, MA_LOG_LEVEL_INFO, "[OpenSL|ES] Cannot find symbol slCreateEngine.");
- return MA_NO_BACKEND;
- }
-#else
- pContext->opensl.SL_IID_ENGINE = (ma_handle)SL_IID_ENGINE;
- pContext->opensl.SL_IID_AUDIOIODEVICECAPABILITIES = (ma_handle)SL_IID_AUDIOIODEVICECAPABILITIES;
- pContext->opensl.SL_IID_ANDROIDSIMPLEBUFFERQUEUE = (ma_handle)SL_IID_ANDROIDSIMPLEBUFFERQUEUE;
- pContext->opensl.SL_IID_RECORD = (ma_handle)SL_IID_RECORD;
- pContext->opensl.SL_IID_PLAY = (ma_handle)SL_IID_PLAY;
- pContext->opensl.SL_IID_OUTPUTMIX = (ma_handle)SL_IID_OUTPUTMIX;
- pContext->opensl.SL_IID_ANDROIDCONFIGURATION = (ma_handle)SL_IID_ANDROIDCONFIGURATION;
- pContext->opensl.slCreateEngine = (ma_proc)slCreateEngine;
-#endif
+**************************************************************************************************************************************************************/
+MA_API ma_data_converter_config ma_data_converter_config_init_default()
+{
+ ma_data_converter_config config;
+ MA_ZERO_OBJECT(&config);
+ config.ditherMode = ma_dither_mode_none;
+ config.resampling.algorithm = ma_resample_algorithm_linear;
+ config.allowDynamicSampleRate = MA_FALSE; /* Disable dynamic sample rates by default because dynamic rate adjustments should be quite rare and it allows an optimization for cases when the in and out sample rates are the same. */
- /* Initialize global data first if applicable. */
- ma_spinlock_lock(&g_maOpenSLSpinlock);
- {
- result = ma_context_init_engine_nolock__opensl(pContext);
- }
- ma_spinlock_unlock(&g_maOpenSLSpinlock);
+ /* Linear resampling defaults. */
+ config.resampling.linear.lpfOrder = 1;
- if (result != MA_SUCCESS) {
- ma_dlclose(pContext, pContext->opensl.libOpenSLES);
- ma_post_log_message(pContext, NULL, MA_LOG_LEVEL_INFO, "[OpenSL|ES] Failed to initialize OpenSL engine.");
- return result;
- }
+ return config;
+}
- pCallbacks->onContextInit = ma_context_init__opensl;
- pCallbacks->onContextUninit = ma_context_uninit__opensl;
- pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__opensl;
- pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__opensl;
- pCallbacks->onDeviceInit = ma_device_init__opensl;
- pCallbacks->onDeviceUninit = ma_device_uninit__opensl;
- pCallbacks->onDeviceStart = ma_device_start__opensl;
- pCallbacks->onDeviceStop = ma_device_stop__opensl;
- pCallbacks->onDeviceRead = NULL; /* Not needed because OpenSL|ES is asynchronous. */
- pCallbacks->onDeviceWrite = NULL; /* Not needed because OpenSL|ES is asynchronous. */
- pCallbacks->onDeviceDataLoop = NULL; /* Not needed because OpenSL|ES is asynchronous. */
+MA_API ma_data_converter_config ma_data_converter_config_init(ma_format formatIn, ma_format formatOut, ma_uint32 channelsIn, ma_uint32 channelsOut, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut)
+{
+ ma_data_converter_config config = ma_data_converter_config_init_default();
+ config.formatIn = formatIn;
+ config.formatOut = formatOut;
+ config.channelsIn = channelsIn;
+ config.channelsOut = channelsOut;
+ config.sampleRateIn = sampleRateIn;
+ config.sampleRateOut = sampleRateOut;
- return MA_SUCCESS;
+ return config;
}
-#endif /* OpenSL|ES */
-
-
-/******************************************************************************
-Web Audio Backend
-******************************************************************************/
-#ifdef MA_HAS_WEBAUDIO
-#include
+typedef struct
+{
+ size_t sizeInBytes;
+ size_t channelConverterOffset;
+ size_t resamplerOffset;
+} ma_data_converter_heap_layout;
-static ma_bool32 ma_is_capture_supported__webaudio()
+static ma_bool32 ma_data_converter_config_is_resampler_required(const ma_data_converter_config* pConfig)
{
- return EM_ASM_INT({
- return (navigator.mediaDevices !== undefined && navigator.mediaDevices.getUserMedia !== undefined);
- }, 0) != 0; /* Must pass in a dummy argument for C99 compatibility. */
+ MA_ASSERT(pConfig != NULL);
+
+ return pConfig->allowDynamicSampleRate || pConfig->sampleRateIn != pConfig->sampleRateOut;
}
-#ifdef __cplusplus
-extern "C" {
-#endif
-void EMSCRIPTEN_KEEPALIVE ma_device_process_pcm_frames_capture__webaudio(ma_device* pDevice, int frameCount, float* pFrames)
+static ma_format ma_data_converter_config_get_mid_format(const ma_data_converter_config* pConfig)
{
- ma_device_handle_backend_data_callback(pDevice, NULL, pFrames, (ma_uint32)frameCount);
+ MA_ASSERT(pConfig != NULL);
+
+ /*
+ We want to avoid as much data conversion as possible. The channel converter and linear
+ resampler both support s16 and f32 natively. We need to decide on the format to use for this
+ stage. We call this the mid format because it's used in the middle stage of the conversion
+ pipeline. If the output format is either s16 or f32 we use that one. If that is not the case it
+ will do the same thing for the input format. If it's neither we just use f32. If we are using a
+ custom resampling backend, we can only guarantee that f32 will be supported so we'll be forced
+ to use that if resampling is required.
+ */
+ if (ma_data_converter_config_is_resampler_required(pConfig) && pConfig->resampling.algorithm != ma_resample_algorithm_linear) {
+ return ma_format_f32; /* <-- Force f32 since that is the only one we can guarantee will be supported by the resampler. */
+ } else {
+ /* */ if (pConfig->formatOut == ma_format_s16 || pConfig->formatOut == ma_format_f32) {
+ return pConfig->formatOut;
+ } else if (pConfig->formatIn == ma_format_s16 || pConfig->formatIn == ma_format_f32) {
+ return pConfig->formatIn;
+ } else {
+ return ma_format_f32;
+ }
+ }
}
-void EMSCRIPTEN_KEEPALIVE ma_device_process_pcm_frames_playback__webaudio(ma_device* pDevice, int frameCount, float* pFrames)
+static ma_channel_converter_config ma_channel_converter_config_init_from_data_converter_config(const ma_data_converter_config* pConfig)
{
- ma_device_handle_backend_data_callback(pDevice, pFrames, NULL, (ma_uint32)frameCount);
-}
-#ifdef __cplusplus
+ ma_channel_converter_config channelConverterConfig;
+
+ MA_ASSERT(pConfig != NULL);
+
+ channelConverterConfig = ma_channel_converter_config_init(ma_data_converter_config_get_mid_format(pConfig), pConfig->channelsIn, pConfig->pChannelMapIn, pConfig->channelsOut, pConfig->pChannelMapOut, pConfig->channelMixMode);
+ channelConverterConfig.ppWeights = pConfig->ppChannelWeights;
+
+ return channelConverterConfig;
}
-#endif
-static ma_result ma_context_enumerate_devices__webaudio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+static ma_resampler_config ma_resampler_config_init_from_data_converter_config(const ma_data_converter_config* pConfig)
{
- ma_bool32 cbResult = MA_TRUE;
-
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(callback != NULL);
+ ma_resampler_config resamplerConfig;
+ ma_uint32 resamplerChannels;
- /* Only supporting default devices for now. */
+ MA_ASSERT(pConfig != NULL);
- /* Playback. */
- if (cbResult) {
- ma_device_info deviceInfo;
- MA_ZERO_OBJECT(&deviceInfo);
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
- deviceInfo.isDefault = MA_TRUE; /* Only supporting default devices. */
- cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
+ /* The resampler is the most expensive part of the conversion process, so we need to do it at the stage where the channel count is at it's lowest. */
+ if (pConfig->channelsIn < pConfig->channelsOut) {
+ resamplerChannels = pConfig->channelsIn;
+ } else {
+ resamplerChannels = pConfig->channelsOut;
}
- /* Capture. */
- if (cbResult) {
- if (ma_is_capture_supported__webaudio()) {
- ma_device_info deviceInfo;
- MA_ZERO_OBJECT(&deviceInfo);
- ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
- deviceInfo.isDefault = MA_TRUE; /* Only supporting default devices. */
- cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
- }
- }
+ resamplerConfig = ma_resampler_config_init(ma_data_converter_config_get_mid_format(pConfig), resamplerChannels, pConfig->sampleRateIn, pConfig->sampleRateOut, pConfig->resampling.algorithm);
+ resamplerConfig.linear = pConfig->resampling.linear;
+ resamplerConfig.pBackendVTable = pConfig->resampling.pBackendVTable;
+ resamplerConfig.pBackendUserData = pConfig->resampling.pBackendUserData;
- return MA_SUCCESS;
+ return resamplerConfig;
}
-static ma_result ma_context_get_device_info__webaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+static ma_result ma_data_converter_get_heap_layout(const ma_data_converter_config* pConfig, ma_data_converter_heap_layout* pHeapLayout)
{
- MA_ASSERT(pContext != NULL);
+ ma_result result;
- if (deviceType == ma_device_type_capture && !ma_is_capture_supported__webaudio()) {
- return MA_NO_DEVICE;
+ MA_ASSERT(pHeapLayout != NULL);
+
+ MA_ZERO_OBJECT(pHeapLayout);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- MA_ZERO_MEMORY(pDeviceInfo->id.webaudio, sizeof(pDeviceInfo->id.webaudio));
+ if (pConfig->channelsIn == 0 || pConfig->channelsOut == 0) {
+ return MA_INVALID_ARGS;
+ }
- /* Only supporting default devices for now. */
- (void)pDeviceID;
- if (deviceType == ma_device_type_playback) {
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
- } else {
- ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
+ pHeapLayout->sizeInBytes = 0;
+
+ /* Channel converter. */
+ pHeapLayout->channelConverterOffset = pHeapLayout->sizeInBytes;
+ {
+ size_t heapSizeInBytes;
+ ma_channel_converter_config channelConverterConfig = ma_channel_converter_config_init_from_data_converter_config(pConfig);
+
+ result = ma_channel_converter_get_heap_size(&channelConverterConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ pHeapLayout->sizeInBytes += heapSizeInBytes;
}
- /* Only supporting default devices. */
- pDeviceInfo->isDefault = MA_TRUE;
+ /* Resampler. */
+ pHeapLayout->resamplerOffset = pHeapLayout->sizeInBytes;
+ if (ma_data_converter_config_is_resampler_required(pConfig)) {
+ size_t heapSizeInBytes;
+ ma_resampler_config resamplerConfig = ma_resampler_config_init_from_data_converter_config(pConfig);
- /* Web Audio can support any number of channels and sample rates. It only supports f32 formats, however. */
- pDeviceInfo->nativeDataFormats[0].flags = 0;
- pDeviceInfo->nativeDataFormats[0].format = ma_format_unknown;
- pDeviceInfo->nativeDataFormats[0].channels = 0; /* All channels are supported. */
- pDeviceInfo->nativeDataFormats[0].sampleRate = EM_ASM_INT({
- try {
- var temp = new (window.AudioContext || window.webkitAudioContext)();
- var sampleRate = temp.sampleRate;
- temp.close();
- return sampleRate;
- } catch(e) {
- return 0;
+ result = ma_resampler_get_heap_size(&resamplerConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
}
- }, 0); /* Must pass in a dummy argument for C99 compatibility. */
- if (pDeviceInfo->nativeDataFormats[0].sampleRate == 0) {
- return MA_NO_DEVICE;
+ pHeapLayout->sizeInBytes += heapSizeInBytes;
}
- pDeviceInfo->nativeDataFormatCount = 1;
+ /* Make sure allocation size is aligned. */
+ pHeapLayout->sizeInBytes = ma_align_64(pHeapLayout->sizeInBytes);
return MA_SUCCESS;
}
-
-static void ma_device_uninit_by_index__webaudio(ma_device* pDevice, ma_device_type deviceType, int deviceIndex)
+MA_API ma_result ma_data_converter_get_heap_size(const ma_data_converter_config* pConfig, size_t* pHeapSizeInBytes)
{
- MA_ASSERT(pDevice != NULL);
+ ma_result result;
+ ma_data_converter_heap_layout heapLayout;
- EM_ASM({
- var device = miniaudio.get_device_by_index($0);
+ if (pHeapSizeInBytes == NULL) {
+ return MA_INVALID_ARGS;
+ }
- /* Make sure all nodes are disconnected and marked for collection. */
- if (device.scriptNode !== undefined) {
- device.scriptNode.onaudioprocess = function(e) {}; /* We want to reset the callback to ensure it doesn't get called after AudioContext.close() has returned. Shouldn't happen since we're disconnecting, but just to be safe... */
- device.scriptNode.disconnect();
- device.scriptNode = undefined;
- }
- if (device.streamNode !== undefined) {
- device.streamNode.disconnect();
- device.streamNode = undefined;
- }
+ *pHeapSizeInBytes = 0;
- /*
- Stop the device. I think there is a chance the callback could get fired after calling this, hence why we want
- to clear the callback before closing.
- */
- device.webaudio.close();
- device.webaudio = undefined;
+ result = ma_data_converter_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- /* Can't forget to free the intermediary buffer. This is the buffer that's shared between JavaScript and C. */
- if (device.intermediaryBuffer !== undefined) {
- Module._free(device.intermediaryBuffer);
- device.intermediaryBuffer = undefined;
- device.intermediaryBufferView = undefined;
- device.intermediaryBufferSizeInBytes = undefined;
- }
+ *pHeapSizeInBytes = heapLayout.sizeInBytes;
- /* Make sure the device is untracked so the slot can be reused later. */
- miniaudio.untrack_device_by_index($0);
- }, deviceIndex, deviceType);
+ return MA_SUCCESS;
}
-static ma_result ma_device_uninit__webaudio(ma_device* pDevice)
+MA_API ma_result ma_data_converter_init_preallocated(const ma_data_converter_config* pConfig, void* pHeap, ma_data_converter* pConverter)
{
- MA_ASSERT(pDevice != NULL);
+ ma_result result;
+ ma_data_converter_heap_layout heapLayout;
+ ma_format midFormat;
+ ma_bool32 isResamplingRequired;
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- ma_device_uninit_by_index__webaudio(pDevice, ma_device_type_capture, pDevice->webaudio.indexCapture);
+ if (pConverter == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- ma_device_uninit_by_index__webaudio(pDevice, ma_device_type_playback, pDevice->webaudio.indexPlayback);
+ MA_ZERO_OBJECT(pConverter);
+
+ result = ma_data_converter_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
}
- return MA_SUCCESS;
-}
+ pConverter->_pHeap = pHeap;
+ MA_ZERO_MEMORY(pHeap, heapLayout.sizeInBytes);
+
+ pConverter->formatIn = pConfig->formatIn;
+ pConverter->formatOut = pConfig->formatOut;
+ pConverter->channelsIn = pConfig->channelsIn;
+ pConverter->channelsOut = pConfig->channelsOut;
+ pConverter->sampleRateIn = pConfig->sampleRateIn;
+ pConverter->sampleRateOut = pConfig->sampleRateOut;
+ pConverter->ditherMode = pConfig->ditherMode;
-static ma_uint32 ma_calculate_period_size_in_frames_from_descriptor__webaudio(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile)
-{
/*
- There have been reports of the default buffer size being too small on some browsers. There have been reports of the default buffer
- size being too small on some browsers. If we're using default buffer size, we'll make sure the period size is a big biffer than our
- standard defaults.
+ Determine if resampling is required. We need to do this so we can determine an appropriate
+ mid format to use. If resampling is required, the mid format must be ma_format_f32 since
+ that is the only one that is guaranteed to supported by custom resampling backends.
*/
- ma_uint32 periodSizeInFrames;
+ isResamplingRequired = ma_data_converter_config_is_resampler_required(pConfig);
+ midFormat = ma_data_converter_config_get_mid_format(pConfig);
- if (pDescriptor->periodSizeInFrames == 0) {
- if (pDescriptor->periodSizeInMilliseconds == 0) {
- if (performanceProfile == ma_performance_profile_low_latency) {
- periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(33, nativeSampleRate); /* 1 frame @ 30 FPS */
- } else {
- periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(333, nativeSampleRate);
- }
+
+ /* Channel converter. We always initialize this, but we check if it configures itself as a passthrough to determine whether or not it's needed. */
+ {
+ ma_channel_converter_config channelConverterConfig = ma_channel_converter_config_init_from_data_converter_config(pConfig);
+
+ result = ma_channel_converter_init_preallocated(&channelConverterConfig, ma_offset_ptr(pHeap, heapLayout.channelConverterOffset), &pConverter->channelConverter);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ /* If the channel converter is not a passthrough we need to enable it. Otherwise we can skip it. */
+ if (pConverter->channelConverter.conversionPath != ma_channel_conversion_path_passthrough) {
+ pConverter->hasChannelConverter = MA_TRUE;
+ }
+ }
+
+
+ /* Resampler. */
+ if (isResamplingRequired) {
+ ma_resampler_config resamplerConfig = ma_resampler_config_init_from_data_converter_config(pConfig);
+
+ result = ma_resampler_init_preallocated(&resamplerConfig, ma_offset_ptr(pHeap, heapLayout.resamplerOffset), &pConverter->resampler);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ pConverter->hasResampler = MA_TRUE;
+ }
+
+
+ /* We can simplify pre- and post-format conversion if we have neither channel conversion nor resampling. */
+ if (pConverter->hasChannelConverter == MA_FALSE && pConverter->hasResampler == MA_FALSE) {
+ /* We have neither channel conversion nor resampling so we'll only need one of pre- or post-format conversion, or none if the input and output formats are the same. */
+ if (pConverter->formatIn == pConverter->formatOut) {
+ /* The formats are the same so we can just pass through. */
+ pConverter->hasPreFormatConversion = MA_FALSE;
+ pConverter->hasPostFormatConversion = MA_FALSE;
} else {
- periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pDescriptor->periodSizeInMilliseconds, nativeSampleRate);
+ /* The formats are different so we need to do either pre- or post-format conversion. It doesn't matter which. */
+ pConverter->hasPreFormatConversion = MA_FALSE;
+ pConverter->hasPostFormatConversion = MA_TRUE;
}
} else {
- periodSizeInFrames = pDescriptor->periodSizeInFrames;
+ /* We have a channel converter and/or resampler so we'll need channel conversion based on the mid format. */
+ if (pConverter->formatIn != midFormat) {
+ pConverter->hasPreFormatConversion = MA_TRUE;
+ }
+ if (pConverter->formatOut != midFormat) {
+ pConverter->hasPostFormatConversion = MA_TRUE;
+ }
}
- /* The size of the buffer must be a power of 2 and between 256 and 16384. */
- if (periodSizeInFrames < 256) {
- periodSizeInFrames = 256;
- } else if (periodSizeInFrames > 16384) {
- periodSizeInFrames = 16384;
+ /* We can enable passthrough optimizations if applicable. Note that we'll only be able to do this if the sample rate is static. */
+ if (pConverter->hasPreFormatConversion == MA_FALSE &&
+ pConverter->hasPostFormatConversion == MA_FALSE &&
+ pConverter->hasChannelConverter == MA_FALSE &&
+ pConverter->hasResampler == MA_FALSE) {
+ pConverter->isPassthrough = MA_TRUE;
+ }
+
+
+ /* We now need to determine our execution path. */
+ if (pConverter->isPassthrough) {
+ pConverter->executionPath = ma_data_converter_execution_path_passthrough;
} else {
- periodSizeInFrames = ma_next_power_of_2(periodSizeInFrames);
+ if (pConverter->channelsIn < pConverter->channelsOut) {
+ /* Do resampling first, if necessary. */
+ MA_ASSERT(pConverter->hasChannelConverter == MA_TRUE);
+
+ if (pConverter->hasResampler) {
+ pConverter->executionPath = ma_data_converter_execution_path_resample_first;
+ } else {
+ pConverter->executionPath = ma_data_converter_execution_path_channels_only;
+ }
+ } else {
+ /* Do channel conversion first, if necessary. */
+ if (pConverter->hasChannelConverter) {
+ if (pConverter->hasResampler) {
+ pConverter->executionPath = ma_data_converter_execution_path_channels_first;
+ } else {
+ pConverter->executionPath = ma_data_converter_execution_path_channels_only;
+ }
+ } else {
+ /* Channel routing not required. */
+ if (pConverter->hasResampler) {
+ pConverter->executionPath = ma_data_converter_execution_path_resample_only;
+ } else {
+ pConverter->executionPath = ma_data_converter_execution_path_format_only;
+ }
+ }
+ }
}
- return periodSizeInFrames;
+ return MA_SUCCESS;
}
-static ma_result ma_device_init_by_type__webaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptor, ma_device_type deviceType)
+MA_API ma_result ma_data_converter_init(const ma_data_converter_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_converter* pConverter)
{
- int deviceIndex;
- ma_uint32 channels;
- ma_uint32 sampleRate;
- ma_uint32 periodSizeInFrames;
+ ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
- MA_ASSERT(pDevice != NULL);
- MA_ASSERT(pConfig != NULL);
- MA_ASSERT(deviceType != ma_device_type_duplex);
+ result = ma_data_converter_get_heap_size(pConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- if (deviceType == ma_device_type_capture && !ma_is_capture_supported__webaudio()) {
- return MA_NO_DEVICE;
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+ } else {
+ pHeap = NULL;
}
- /* We're going to calculate some stuff in C just to simplify the JS code. */
- channels = (pDescriptor->channels > 0) ? pDescriptor->channels : MA_DEFAULT_CHANNELS;
- sampleRate = (pDescriptor->sampleRate > 0) ? pDescriptor->sampleRate : MA_DEFAULT_SAMPLE_RATE;
- periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__webaudio(pDescriptor, sampleRate, pConfig->performanceProfile);
+ result = ma_data_converter_init_preallocated(pConfig, pHeap, pConverter);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
+ }
+ pConverter->_ownsHeap = MA_TRUE;
+ return MA_SUCCESS;
+}
- /* We create the device on the JavaScript side and reference it using an index. We use this to make it possible to reference the device between JavaScript and C. */
- deviceIndex = EM_ASM_INT({
- var channels = $0;
- var sampleRate = $1;
- var bufferSize = $2; /* In PCM frames. */
- var isCapture = $3;
- var pDevice = $4;
+MA_API void ma_data_converter_uninit(ma_data_converter* pConverter, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pConverter == NULL) {
+ return;
+ }
- if (typeof(miniaudio) === 'undefined') {
- return -1; /* Context not initialized. */
- }
+ if (pConverter->hasResampler) {
+ ma_resampler_uninit(&pConverter->resampler, pAllocationCallbacks);
+ }
- var device = {};
+ ma_channel_converter_uninit(&pConverter->channelConverter, pAllocationCallbacks);
- /* The AudioContext must be created in a suspended state. */
- device.webaudio = new (window.AudioContext || window.webkitAudioContext)({sampleRate:sampleRate});
- device.webaudio.suspend();
- device.state = 1; /* MA_STATE_STOPPED */
+ if (pConverter->_ownsHeap) {
+ ma_free(pConverter->_pHeap, pAllocationCallbacks);
+ }
+}
- /*
- We need an intermediary buffer which we use for JavaScript and C interop. This buffer stores interleaved f32 PCM data. Because it's passed between
- JavaScript and C it needs to be allocated and freed using Module._malloc() and Module._free().
- */
- device.intermediaryBufferSizeInBytes = channels * bufferSize * 4;
- device.intermediaryBuffer = Module._malloc(device.intermediaryBufferSizeInBytes);
- device.intermediaryBufferView = new Float32Array(Module.HEAPF32.buffer, device.intermediaryBuffer, device.intermediaryBufferSizeInBytes);
+static ma_result ma_data_converter_process_pcm_frames__passthrough(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+{
+ ma_uint64 frameCountIn;
+ ma_uint64 frameCountOut;
+ ma_uint64 frameCount;
- /*
- Both playback and capture devices use a ScriptProcessorNode for performing per-sample operations.
+ MA_ASSERT(pConverter != NULL);
- ScriptProcessorNode is actually deprecated so this is likely to be temporary. The way this works for playback is very simple. You just set a callback
- that's periodically fired, just like a normal audio callback function. But apparently this design is "flawed" and is now deprecated in favour of
- something called AudioWorklets which _forces_ you to load a _separate_ .js file at run time... nice... Hopefully ScriptProcessorNode will continue to
- work for years to come, but this may need to change to use AudioSourceBufferNode instead, which I think is what Emscripten uses for it's built-in SDL
- implementation. I'll be avoiding that insane AudioWorklet API like the plague...
+ frameCountIn = 0;
+ if (pFrameCountIn != NULL) {
+ frameCountIn = *pFrameCountIn;
+ }
- For capture it is a bit unintuitive. We use the ScriptProccessorNode _only_ to get the raw PCM data. It is connected to an AudioContext just like the
- playback case, however we just output silence to the AudioContext instead of passing any real data. It would make more sense to me to use the
- MediaRecorder API, but unfortunately you need to specify a MIME time (Opus, Vorbis, etc.) for the binary blob that's returned to the client, but I've
- been unable to figure out how to get this as raw PCM. The closest I can think is to use the MIME type for WAV files and just parse it, but I don't know
- how well this would work. Although ScriptProccessorNode is deprecated, in practice it seems to have pretty good browser support so I'm leaving it like
- this for now. If anyone knows how I could get raw PCM data using the MediaRecorder API please let me know!
- */
- device.scriptNode = device.webaudio.createScriptProcessor(bufferSize, channels, channels);
+ frameCountOut = 0;
+ if (pFrameCountOut != NULL) {
+ frameCountOut = *pFrameCountOut;
+ }
- if (isCapture) {
- device.scriptNode.onaudioprocess = function(e) {
- if (device.intermediaryBuffer === undefined) {
- return; /* This means the device has been uninitialized. */
- }
+ frameCount = ma_min(frameCountIn, frameCountOut);
- if(device.intermediaryBufferView.length == 0) {
- /* Recreate intermediaryBufferView when losing reference to the underlying buffer, probably due to emscripten resizing heap. */
- device.intermediaryBufferView = new Float32Array(Module.HEAPF32.buffer, device.intermediaryBuffer, device.intermediaryBufferSizeInBytes);
- }
+ if (pFramesOut != NULL) {
+ if (pFramesIn != NULL) {
+ ma_copy_memory_64(pFramesOut, pFramesIn, frameCount * ma_get_bytes_per_frame(pConverter->formatOut, pConverter->channelsOut));
+ } else {
+ ma_zero_memory_64(pFramesOut, frameCount * ma_get_bytes_per_frame(pConverter->formatOut, pConverter->channelsOut));
+ }
+ }
- /* Make sure silence it output to the AudioContext destination. Not doing this will cause sound to come out of the speakers! */
- for (var iChannel = 0; iChannel < e.outputBuffer.numberOfChannels; ++iChannel) {
- e.outputBuffer.getChannelData(iChannel).fill(0.0);
- }
+ if (pFrameCountIn != NULL) {
+ *pFrameCountIn = frameCount;
+ }
+ if (pFrameCountOut != NULL) {
+ *pFrameCountOut = frameCount;
+ }
- /* There are some situations where we may want to send silence to the client. */
- var sendSilence = false;
- if (device.streamNode === undefined) {
- sendSilence = true;
- }
+ return MA_SUCCESS;
+}
- /* Sanity check. This will never happen, right? */
- if (e.inputBuffer.numberOfChannels != channels) {
- console.log("Capture: Channel count mismatch. " + e.inputBufer.numberOfChannels + " != " + channels + ". Sending silence.");
- sendSilence = true;
- }
+static ma_result ma_data_converter_process_pcm_frames__format_only(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+{
+ ma_uint64 frameCountIn;
+ ma_uint64 frameCountOut;
+ ma_uint64 frameCount;
- /* This looped design guards against the situation where e.inputBuffer is a different size to the original buffer size. Should never happen in practice. */
- var totalFramesProcessed = 0;
- while (totalFramesProcessed < e.inputBuffer.length) {
- var framesRemaining = e.inputBuffer.length - totalFramesProcessed;
- var framesToProcess = framesRemaining;
- if (framesToProcess > (device.intermediaryBufferSizeInBytes/channels/4)) {
- framesToProcess = (device.intermediaryBufferSizeInBytes/channels/4);
- }
+ MA_ASSERT(pConverter != NULL);
- /* We need to do the reverse of the playback case. We need to interleave the input data and copy it into the intermediary buffer. Then we send it to the client. */
- if (sendSilence) {
- device.intermediaryBufferView.fill(0.0);
- } else {
- for (var iFrame = 0; iFrame < framesToProcess; ++iFrame) {
- for (var iChannel = 0; iChannel < e.inputBuffer.numberOfChannels; ++iChannel) {
- device.intermediaryBufferView[iFrame*channels + iChannel] = e.inputBuffer.getChannelData(iChannel)[totalFramesProcessed + iFrame];
- }
- }
- }
+ frameCountIn = 0;
+ if (pFrameCountIn != NULL) {
+ frameCountIn = *pFrameCountIn;
+ }
- /* Send data to the client from our intermediary buffer. */
- ccall("ma_device_process_pcm_frames_capture__webaudio", "undefined", ["number", "number", "number"], [pDevice, framesToProcess, device.intermediaryBuffer]);
+ frameCountOut = 0;
+ if (pFrameCountOut != NULL) {
+ frameCountOut = *pFrameCountOut;
+ }
- totalFramesProcessed += framesToProcess;
- }
- };
+ frameCount = ma_min(frameCountIn, frameCountOut);
- navigator.mediaDevices.getUserMedia({audio:true, video:false})
- .then(function(stream) {
- device.streamNode = device.webaudio.createMediaStreamSource(stream);
- device.streamNode.connect(device.scriptNode);
- device.scriptNode.connect(device.webaudio.destination);
- })
- .catch(function(error) {
- /* I think this should output silence... */
- device.scriptNode.connect(device.webaudio.destination);
- });
+ if (pFramesOut != NULL) {
+ if (pFramesIn != NULL) {
+ ma_convert_pcm_frames_format(pFramesOut, pConverter->formatOut, pFramesIn, pConverter->formatIn, frameCount, pConverter->channelsIn, pConverter->ditherMode);
} else {
- device.scriptNode.onaudioprocess = function(e) {
- if (device.intermediaryBuffer === undefined) {
- return; /* This means the device has been uninitialized. */
- }
+ ma_zero_memory_64(pFramesOut, frameCount * ma_get_bytes_per_frame(pConverter->formatOut, pConverter->channelsOut));
+ }
+ }
- if(device.intermediaryBufferView.length == 0) {
- /* Recreate intermediaryBufferView when losing reference to the underlying buffer, probably due to emscripten resizing heap. */
- device.intermediaryBufferView = new Float32Array(Module.HEAPF32.buffer, device.intermediaryBuffer, device.intermediaryBufferSizeInBytes);
- }
+ if (pFrameCountIn != NULL) {
+ *pFrameCountIn = frameCount;
+ }
+ if (pFrameCountOut != NULL) {
+ *pFrameCountOut = frameCount;
+ }
- var outputSilence = false;
+ return MA_SUCCESS;
+}
- /* Sanity check. This will never happen, right? */
- if (e.outputBuffer.numberOfChannels != channels) {
- console.log("Playback: Channel count mismatch. " + e.outputBufer.numberOfChannels + " != " + channels + ". Outputting silence.");
- outputSilence = true;
- return;
- }
- /* This looped design guards against the situation where e.outputBuffer is a different size to the original buffer size. Should never happen in practice. */
- var totalFramesProcessed = 0;
- while (totalFramesProcessed < e.outputBuffer.length) {
- var framesRemaining = e.outputBuffer.length - totalFramesProcessed;
- var framesToProcess = framesRemaining;
- if (framesToProcess > (device.intermediaryBufferSizeInBytes/channels/4)) {
- framesToProcess = (device.intermediaryBufferSizeInBytes/channels/4);
- }
+static ma_result ma_data_converter_process_pcm_frames__resample_with_format_conversion(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+{
+ ma_result result = MA_SUCCESS;
+ ma_uint64 frameCountIn;
+ ma_uint64 frameCountOut;
+ ma_uint64 framesProcessedIn;
+ ma_uint64 framesProcessedOut;
- /* Read data from the client into our intermediary buffer. */
- ccall("ma_device_process_pcm_frames_playback__webaudio", "undefined", ["number", "number", "number"], [pDevice, framesToProcess, device.intermediaryBuffer]);
+ MA_ASSERT(pConverter != NULL);
- /* At this point we'll have data in our intermediary buffer which we now need to deinterleave and copy over to the output buffers. */
- if (outputSilence) {
- for (var iChannel = 0; iChannel < e.outputBuffer.numberOfChannels; ++iChannel) {
- e.outputBuffer.getChannelData(iChannel).fill(0.0);
- }
- } else {
- for (var iChannel = 0; iChannel < e.outputBuffer.numberOfChannels; ++iChannel) {
- for (var iFrame = 0; iFrame < framesToProcess; ++iFrame) {
- e.outputBuffer.getChannelData(iChannel)[totalFramesProcessed + iFrame] = device.intermediaryBufferView[iFrame*channels + iChannel];
- }
- }
- }
+ frameCountIn = 0;
+ if (pFrameCountIn != NULL) {
+ frameCountIn = *pFrameCountIn;
+ }
- totalFramesProcessed += framesToProcess;
- }
- };
+ frameCountOut = 0;
+ if (pFrameCountOut != NULL) {
+ frameCountOut = *pFrameCountOut;
+ }
- device.scriptNode.connect(device.webaudio.destination);
+ framesProcessedIn = 0;
+ framesProcessedOut = 0;
+
+ while (framesProcessedOut < frameCountOut) {
+ ma_uint8 pTempBufferOut[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
+ const ma_uint32 tempBufferOutCap = sizeof(pTempBufferOut) / ma_get_bytes_per_frame(pConverter->resampler.format, pConverter->resampler.channels);
+ const void* pFramesInThisIteration;
+ /* */ void* pFramesOutThisIteration;
+ ma_uint64 frameCountInThisIteration;
+ ma_uint64 frameCountOutThisIteration;
+
+ if (pFramesIn != NULL) {
+ pFramesInThisIteration = ma_offset_ptr(pFramesIn, framesProcessedIn * ma_get_bytes_per_frame(pConverter->formatIn, pConverter->channelsIn));
+ } else {
+ pFramesInThisIteration = NULL;
}
- return miniaudio.track_device(device);
- }, channels, sampleRate, periodSizeInFrames, deviceType == ma_device_type_capture, pDevice);
+ if (pFramesOut != NULL) {
+ pFramesOutThisIteration = ma_offset_ptr(pFramesOut, framesProcessedOut * ma_get_bytes_per_frame(pConverter->formatOut, pConverter->channelsOut));
+ } else {
+ pFramesOutThisIteration = NULL;
+ }
+
+ /* Do a pre format conversion if necessary. */
+ if (pConverter->hasPreFormatConversion) {
+ ma_uint8 pTempBufferIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
+ const ma_uint32 tempBufferInCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->resampler.format, pConverter->resampler.channels);
+
+ frameCountInThisIteration = (frameCountIn - framesProcessedIn);
+ if (frameCountInThisIteration > tempBufferInCap) {
+ frameCountInThisIteration = tempBufferInCap;
+ }
- if (deviceIndex < 0) {
- return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
- }
+ if (pConverter->hasPostFormatConversion) {
+ if (frameCountInThisIteration > tempBufferOutCap) {
+ frameCountInThisIteration = tempBufferOutCap;
+ }
+ }
- if (deviceType == ma_device_type_capture) {
- pDevice->webaudio.indexCapture = deviceIndex;
- } else {
- pDevice->webaudio.indexPlayback = deviceIndex;
- }
+ if (pFramesInThisIteration != NULL) {
+ ma_convert_pcm_frames_format(pTempBufferIn, pConverter->resampler.format, pFramesInThisIteration, pConverter->formatIn, frameCountInThisIteration, pConverter->channelsIn, pConverter->ditherMode);
+ } else {
+ MA_ZERO_MEMORY(pTempBufferIn, sizeof(pTempBufferIn));
+ }
- pDescriptor->format = ma_format_f32;
- pDescriptor->channels = channels;
- ma_get_standard_channel_map(ma_standard_channel_map_webaudio, pDescriptor->channels, pDescriptor->channelMap);
- pDescriptor->sampleRate = EM_ASM_INT({ return miniaudio.get_device_by_index($0).webaudio.sampleRate; }, deviceIndex);
- pDescriptor->periodSizeInFrames = periodSizeInFrames;
- pDescriptor->periodCount = 1;
+ frameCountOutThisIteration = (frameCountOut - framesProcessedOut);
- return MA_SUCCESS;
-}
+ if (pConverter->hasPostFormatConversion) {
+ /* Both input and output conversion required. Output to the temp buffer. */
+ if (frameCountOutThisIteration > tempBufferOutCap) {
+ frameCountOutThisIteration = tempBufferOutCap;
+ }
-static ma_result ma_device_init__webaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
-{
- ma_result result;
+ result = ma_resampler_process_pcm_frames(&pConverter->resampler, pTempBufferIn, &frameCountInThisIteration, pTempBufferOut, &frameCountOutThisIteration);
+ } else {
+ /* Only pre-format required. Output straight to the output buffer. */
+ result = ma_resampler_process_pcm_frames(&pConverter->resampler, pTempBufferIn, &frameCountInThisIteration, pFramesOutThisIteration, &frameCountOutThisIteration);
+ }
- if (pConfig->deviceType == ma_device_type_loopback) {
- return MA_DEVICE_TYPE_NOT_SUPPORTED;
- }
+ if (result != MA_SUCCESS) {
+ break;
+ }
+ } else {
+ /* No pre-format required. Just read straight from the input buffer. */
+ MA_ASSERT(pConverter->hasPostFormatConversion == MA_TRUE);
- /* No exclusive mode with Web Audio. */
- if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) ||
- ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) {
- return MA_SHARE_MODE_NOT_SUPPORTED;
- }
+ frameCountInThisIteration = (frameCountIn - framesProcessedIn);
+ frameCountOutThisIteration = (frameCountOut - framesProcessedOut);
+ if (frameCountOutThisIteration > tempBufferOutCap) {
+ frameCountOutThisIteration = tempBufferOutCap;
+ }
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
- result = ma_device_init_by_type__webaudio(pDevice, pConfig, pDescriptorCapture, ma_device_type_capture);
- if (result != MA_SUCCESS) {
- return result;
+ result = ma_resampler_process_pcm_frames(&pConverter->resampler, pFramesInThisIteration, &frameCountInThisIteration, pTempBufferOut, &frameCountOutThisIteration);
+ if (result != MA_SUCCESS) {
+ break;
+ }
}
- }
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- result = ma_device_init_by_type__webaudio(pDevice, pConfig, pDescriptorPlayback, ma_device_type_playback);
- if (result != MA_SUCCESS) {
- if (pConfig->deviceType == ma_device_type_duplex) {
- ma_device_uninit_by_index__webaudio(pDevice, ma_device_type_capture, pDevice->webaudio.indexCapture);
+ /* If we are doing a post format conversion we need to do that now. */
+ if (pConverter->hasPostFormatConversion) {
+ if (pFramesOutThisIteration != NULL) {
+ ma_convert_pcm_frames_format(pFramesOutThisIteration, pConverter->formatOut, pTempBufferOut, pConverter->resampler.format, frameCountOutThisIteration, pConverter->resampler.channels, pConverter->ditherMode);
}
- return result;
}
- }
- return MA_SUCCESS;
-}
+ framesProcessedIn += frameCountInThisIteration;
+ framesProcessedOut += frameCountOutThisIteration;
-static ma_result ma_device_start__webaudio(ma_device* pDevice)
-{
- MA_ASSERT(pDevice != NULL);
+ MA_ASSERT(framesProcessedIn <= frameCountIn);
+ MA_ASSERT(framesProcessedOut <= frameCountOut);
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- EM_ASM({
- var device = miniaudio.get_device_by_index($0);
- device.webaudio.resume();
- device.state = 2; /* MA_STATE_STARTED */
- }, pDevice->webaudio.indexCapture);
+ if (frameCountOutThisIteration == 0) {
+ break; /* Consumed all of our input data. */
+ }
}
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- EM_ASM({
- var device = miniaudio.get_device_by_index($0);
- device.webaudio.resume();
- device.state = 2; /* MA_STATE_STARTED */
- }, pDevice->webaudio.indexPlayback);
+ if (pFrameCountIn != NULL) {
+ *pFrameCountIn = framesProcessedIn;
+ }
+ if (pFrameCountOut != NULL) {
+ *pFrameCountOut = framesProcessedOut;
}
- return MA_SUCCESS;
+ return result;
}
-static ma_result ma_device_stop__webaudio(ma_device* pDevice)
+static ma_result ma_data_converter_process_pcm_frames__resample_only(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
- MA_ASSERT(pDevice != NULL);
-
- /*
- From the WebAudio API documentation for AudioContext.suspend():
+ MA_ASSERT(pConverter != NULL);
- Suspends the progression of AudioContext's currentTime, allows any current context processing blocks that are already processed to be played to the
- destination, and then allows the system to release its claim on audio hardware.
+ if (pConverter->hasPreFormatConversion == MA_FALSE && pConverter->hasPostFormatConversion == MA_FALSE) {
+ /* Neither pre- nor post-format required. This is simple case where only resampling is required. */
+ return ma_resampler_process_pcm_frames(&pConverter->resampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
+ } else {
+ /* Format conversion required. */
+ return ma_data_converter_process_pcm_frames__resample_with_format_conversion(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
+ }
+}
- I read this to mean that "any current context processing blocks" are processed by suspend() - i.e. They they are drained. We therefore shouldn't need to
- do any kind of explicit draining.
- */
+static ma_result ma_data_converter_process_pcm_frames__channels_only(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+{
+ ma_result result;
+ ma_uint64 frameCountIn;
+ ma_uint64 frameCountOut;
+ ma_uint64 frameCount;
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- EM_ASM({
- var device = miniaudio.get_device_by_index($0);
- device.webaudio.suspend();
- device.state = 1; /* MA_STATE_STOPPED */
- }, pDevice->webaudio.indexCapture);
- }
+ MA_ASSERT(pConverter != NULL);
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- EM_ASM({
- var device = miniaudio.get_device_by_index($0);
- device.webaudio.suspend();
- device.state = 1; /* MA_STATE_STOPPED */
- }, pDevice->webaudio.indexPlayback);
+ frameCountIn = 0;
+ if (pFrameCountIn != NULL) {
+ frameCountIn = *pFrameCountIn;
}
- ma_stop_proc onStop = pDevice->onStop;
- if (onStop) {
- onStop(pDevice);
+ frameCountOut = 0;
+ if (pFrameCountOut != NULL) {
+ frameCountOut = *pFrameCountOut;
}
- return MA_SUCCESS;
-}
-
-static ma_result ma_context_uninit__webaudio(ma_context* pContext)
-{
- MA_ASSERT(pContext != NULL);
- MA_ASSERT(pContext->backend == ma_backend_webaudio);
-
- /* Nothing needs to be done here. */
- (void)pContext;
+ frameCount = ma_min(frameCountIn, frameCountOut);
- return MA_SUCCESS;
-}
+ if (pConverter->hasPreFormatConversion == MA_FALSE && pConverter->hasPostFormatConversion == MA_FALSE) {
+ /* No format conversion required. */
+ result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pFramesOut, pFramesIn, frameCount);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ } else {
+ /* Format conversion required. */
+ ma_uint64 framesProcessed = 0;
-static ma_result ma_context_init__webaudio(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
-{
- int resultFromJS;
+ while (framesProcessed < frameCount) {
+ ma_uint8 pTempBufferOut[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
+ const ma_uint32 tempBufferOutCap = sizeof(pTempBufferOut) / ma_get_bytes_per_frame(pConverter->channelConverter.format, pConverter->channelConverter.channelsOut);
+ const void* pFramesInThisIteration;
+ /* */ void* pFramesOutThisIteration;
+ ma_uint64 frameCountThisIteration;
- MA_ASSERT(pContext != NULL);
+ if (pFramesIn != NULL) {
+ pFramesInThisIteration = ma_offset_ptr(pFramesIn, framesProcessed * ma_get_bytes_per_frame(pConverter->formatIn, pConverter->channelsIn));
+ } else {
+ pFramesInThisIteration = NULL;
+ }
- (void)pConfig; /* Unused. */
+ if (pFramesOut != NULL) {
+ pFramesOutThisIteration = ma_offset_ptr(pFramesOut, framesProcessed * ma_get_bytes_per_frame(pConverter->formatOut, pConverter->channelsOut));
+ } else {
+ pFramesOutThisIteration = NULL;
+ }
- /* Here is where our global JavaScript object is initialized. */
- resultFromJS = EM_ASM_INT({
- if ((window.AudioContext || window.webkitAudioContext) === undefined) {
- return 0; /* Web Audio not supported. */
- }
+ /* Do a pre format conversion if necessary. */
+ if (pConverter->hasPreFormatConversion) {
+ ma_uint8 pTempBufferIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
+ const ma_uint32 tempBufferInCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->channelConverter.format, pConverter->channelConverter.channelsIn);
- if (typeof(miniaudio) === 'undefined') {
- miniaudio = {};
- miniaudio.devices = []; /* Device cache for mapping devices to indexes for JavaScript/C interop. */
+ frameCountThisIteration = (frameCount - framesProcessed);
+ if (frameCountThisIteration > tempBufferInCap) {
+ frameCountThisIteration = tempBufferInCap;
+ }
- miniaudio.track_device = function(device) {
- /* Try inserting into a free slot first. */
- for (var iDevice = 0; iDevice < miniaudio.devices.length; ++iDevice) {
- if (miniaudio.devices[iDevice] == null) {
- miniaudio.devices[iDevice] = device;
- return iDevice;
+ if (pConverter->hasPostFormatConversion) {
+ if (frameCountThisIteration > tempBufferOutCap) {
+ frameCountThisIteration = tempBufferOutCap;
}
}
- /* Getting here means there is no empty slots in the array so we just push to the end. */
- miniaudio.devices.push(device);
- return miniaudio.devices.length - 1;
- };
-
- miniaudio.untrack_device_by_index = function(deviceIndex) {
- /* We just set the device's slot to null. The slot will get reused in the next call to ma_track_device. */
- miniaudio.devices[deviceIndex] = null;
+ if (pFramesInThisIteration != NULL) {
+ ma_convert_pcm_frames_format(pTempBufferIn, pConverter->channelConverter.format, pFramesInThisIteration, pConverter->formatIn, frameCountThisIteration, pConverter->channelsIn, pConverter->ditherMode);
+ } else {
+ MA_ZERO_MEMORY(pTempBufferIn, sizeof(pTempBufferIn));
+ }
- /* Trim the array if possible. */
- while (miniaudio.devices.length > 0) {
- if (miniaudio.devices[miniaudio.devices.length-1] == null) {
- miniaudio.devices.pop();
- } else {
- break;
- }
+ if (pConverter->hasPostFormatConversion) {
+ /* Both input and output conversion required. Output to the temp buffer. */
+ result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pTempBufferOut, pTempBufferIn, frameCountThisIteration);
+ } else {
+ /* Only pre-format required. Output straight to the output buffer. */
+ result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pFramesOutThisIteration, pTempBufferIn, frameCountThisIteration);
}
- };
- miniaudio.untrack_device = function(device) {
- for (var iDevice = 0; iDevice < miniaudio.devices.length; ++iDevice) {
- if (miniaudio.devices[iDevice] == device) {
- return miniaudio.untrack_device_by_index(iDevice);
- }
+ if (result != MA_SUCCESS) {
+ break;
}
- };
+ } else {
+ /* No pre-format required. Just read straight from the input buffer. */
+ MA_ASSERT(pConverter->hasPostFormatConversion == MA_TRUE);
- miniaudio.get_device_by_index = function(deviceIndex) {
- return miniaudio.devices[deviceIndex];
- };
+ frameCountThisIteration = (frameCount - framesProcessed);
+ if (frameCountThisIteration > tempBufferOutCap) {
+ frameCountThisIteration = tempBufferOutCap;
+ }
- miniaudio.unlock_event_types = (function(){
- return ['touchstart', 'touchend', 'click'];
- })();
+ result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pTempBufferOut, pFramesInThisIteration, frameCountThisIteration);
+ if (result != MA_SUCCESS) {
+ break;
+ }
+ }
- miniaudio.unlock = function() {
- for(var i = 0; i < miniaudio.devices.length; ++i) {
- var device = miniaudio.devices[i];
- if (device != null && device.webaudio != null && device.state === 2 /* MA_STATE_STARTED */) {
- device.webaudio.resume();
- }
+ /* If we are doing a post format conversion we need to do that now. */
+ if (pConverter->hasPostFormatConversion) {
+ if (pFramesOutThisIteration != NULL) {
+ ma_convert_pcm_frames_format(pFramesOutThisIteration, pConverter->formatOut, pTempBufferOut, pConverter->channelConverter.format, frameCountThisIteration, pConverter->channelConverter.channelsOut, pConverter->ditherMode);
}
- miniaudio.unlock_event_types.map(function(event_type) {
- document.removeEventListener(event_type, miniaudio.unlock, true);
- });
- };
+ }
- miniaudio.unlock_event_types.map(function(event_type) {
- document.addEventListener(event_type, miniaudio.unlock, true);
- });
+ framesProcessed += frameCountThisIteration;
}
-
- return 1;
- }, 0); /* Must pass in a dummy argument for C99 compatibility. */
-
- if (resultFromJS != 1) {
- return MA_FAILED_TO_INIT_BACKEND;
}
- pCallbacks->onContextInit = ma_context_init__webaudio;
- pCallbacks->onContextUninit = ma_context_uninit__webaudio;
- pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__webaudio;
- pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__webaudio;
- pCallbacks->onDeviceInit = ma_device_init__webaudio;
- pCallbacks->onDeviceUninit = ma_device_uninit__webaudio;
- pCallbacks->onDeviceStart = ma_device_start__webaudio;
- pCallbacks->onDeviceStop = ma_device_stop__webaudio;
- pCallbacks->onDeviceRead = NULL; /* Not needed because WebAudio is asynchronous. */
- pCallbacks->onDeviceWrite = NULL; /* Not needed because WebAudio is asynchronous. */
- pCallbacks->onDeviceDataLoop = NULL; /* Not needed because WebAudio is asynchronous. */
+ if (pFrameCountIn != NULL) {
+ *pFrameCountIn = frameCount;
+ }
+ if (pFrameCountOut != NULL) {
+ *pFrameCountOut = frameCount;
+ }
return MA_SUCCESS;
}
-#endif /* Web Audio */
-
-
-static ma_bool32 ma__is_channel_map_valid(const ma_channel* channelMap, ma_uint32 channels)
+static ma_result ma_data_converter_process_pcm_frames__resample_first(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
- /* A blank channel map should be allowed, in which case it should use an appropriate default which will depend on context. */
- if (channelMap[0] != MA_CHANNEL_NONE) {
- ma_uint32 iChannel;
+ ma_result result;
+ ma_uint64 frameCountIn;
+ ma_uint64 frameCountOut;
+ ma_uint64 framesProcessedIn;
+ ma_uint64 framesProcessedOut;
+ ma_uint8 pTempBufferIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In resampler format. */
+ ma_uint64 tempBufferInCap;
+ ma_uint8 pTempBufferMid[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In resampler format, channel converter input format. */
+ ma_uint64 tempBufferMidCap;
+ ma_uint8 pTempBufferOut[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In channel converter output format. */
+ ma_uint64 tempBufferOutCap;
- if (channels == 0 || channels > MA_MAX_CHANNELS) {
- return MA_FALSE; /* Channel count out of range. */
- }
+ MA_ASSERT(pConverter != NULL);
+ MA_ASSERT(pConverter->resampler.format == pConverter->channelConverter.format);
+ MA_ASSERT(pConverter->resampler.channels == pConverter->channelConverter.channelsIn);
+ MA_ASSERT(pConverter->resampler.channels < pConverter->channelConverter.channelsOut);
- /* A channel cannot be present in the channel map more than once. */
- for (iChannel = 0; iChannel < channels; ++iChannel) {
- ma_uint32 jChannel;
- for (jChannel = iChannel + 1; jChannel < channels; ++jChannel) {
- if (channelMap[iChannel] == channelMap[jChannel]) {
- return MA_FALSE;
- }
- }
- }
+ frameCountIn = 0;
+ if (pFrameCountIn != NULL) {
+ frameCountIn = *pFrameCountIn;
}
- return MA_TRUE;
-}
+ frameCountOut = 0;
+ if (pFrameCountOut != NULL) {
+ frameCountOut = *pFrameCountOut;
+ }
+ framesProcessedIn = 0;
+ framesProcessedOut = 0;
-static ma_result ma_device__post_init_setup(ma_device* pDevice, ma_device_type deviceType)
-{
- ma_result result;
+ tempBufferInCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->resampler.format, pConverter->resampler.channels);
+ tempBufferMidCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->resampler.format, pConverter->resampler.channels);
+ tempBufferOutCap = sizeof(pTempBufferOut) / ma_get_bytes_per_frame(pConverter->channelConverter.format, pConverter->channelConverter.channelsOut);
- MA_ASSERT(pDevice != NULL);
+ while (framesProcessedOut < frameCountOut) {
+ ma_uint64 frameCountInThisIteration;
+ ma_uint64 frameCountOutThisIteration;
+ const void* pRunningFramesIn = NULL;
+ void* pRunningFramesOut = NULL;
+ const void* pResampleBufferIn;
+ void* pChannelsBufferOut;
- if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex || deviceType == ma_device_type_loopback) {
- if (pDevice->capture.format == ma_format_unknown) {
- pDevice->capture.format = pDevice->capture.internalFormat;
+ if (pFramesIn != NULL) {
+ pRunningFramesIn = ma_offset_ptr(pFramesIn, framesProcessedIn * ma_get_bytes_per_frame(pConverter->formatIn, pConverter->channelsIn));
}
- if (pDevice->capture.channels == 0) {
- pDevice->capture.channels = pDevice->capture.internalChannels;
+ if (pFramesOut != NULL) {
+ pRunningFramesOut = ma_offset_ptr(pFramesOut, framesProcessedOut * ma_get_bytes_per_frame(pConverter->formatOut, pConverter->channelsOut));
}
- if (pDevice->capture.channelMap[0] == MA_CHANNEL_NONE) {
- MA_ASSERT(pDevice->capture.channels <= MA_MAX_CHANNELS);
- if (pDevice->capture.internalChannels == pDevice->capture.channels) {
- ma_channel_map_copy(pDevice->capture.channelMap, pDevice->capture.internalChannelMap, pDevice->capture.channels);
- } else {
- if (pDevice->capture.channelMixMode == ma_channel_mix_mode_simple) {
- ma_channel_map_init_blank(pDevice->capture.channels, pDevice->capture.channelMap);
- } else {
- ma_get_standard_channel_map(ma_standard_channel_map_default, pDevice->capture.channels, pDevice->capture.channelMap);
- }
+
+ /* Run input data through the resampler and output it to the temporary buffer. */
+ frameCountInThisIteration = (frameCountIn - framesProcessedIn);
+
+ if (pConverter->hasPreFormatConversion) {
+ if (frameCountInThisIteration > tempBufferInCap) {
+ frameCountInThisIteration = tempBufferInCap;
}
}
- }
- if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) {
- if (pDevice->playback.format == ma_format_unknown) {
- pDevice->playback.format = pDevice->playback.internalFormat;
+ frameCountOutThisIteration = (frameCountOut - framesProcessedOut);
+ if (frameCountOutThisIteration > tempBufferMidCap) {
+ frameCountOutThisIteration = tempBufferMidCap;
}
- if (pDevice->playback.channels == 0) {
- pDevice->playback.channels = pDevice->playback.internalChannels;
+
+ /* We can't read more frames than can fit in the output buffer. */
+ if (pConverter->hasPostFormatConversion) {
+ if (frameCountOutThisIteration > tempBufferOutCap) {
+ frameCountOutThisIteration = tempBufferOutCap;
+ }
}
- if (pDevice->playback.channelMap[0] == MA_CHANNEL_NONE) {
- MA_ASSERT(pDevice->playback.channels <= MA_MAX_CHANNELS);
- if (pDevice->playback.internalChannels == pDevice->playback.channels) {
- ma_channel_map_copy(pDevice->playback.channelMap, pDevice->playback.internalChannelMap, pDevice->playback.channels);
- } else {
- if (pDevice->playback.channelMixMode == ma_channel_mix_mode_simple) {
- ma_channel_map_init_blank(pDevice->playback.channels, pDevice->playback.channelMap);
- } else {
- ma_get_standard_channel_map(ma_standard_channel_map_default, pDevice->playback.channels, pDevice->playback.channelMap);
- }
+
+ /* We need to ensure we don't try to process too many input frames that we run out of room in the output buffer. If this happens we'll end up glitching. */
+
+ /*
+ We need to try to predict how many input frames will be required for the resampler. If the
+ resampler can tell us, we'll use that. Otherwise we'll need to make a best guess. The further
+ off we are from this, the more wasted format conversions we'll end up doing.
+ */
+ #if 1
+ {
+ ma_uint64 requiredInputFrameCount;
+
+ result = ma_resampler_get_required_input_frame_count(&pConverter->resampler, frameCountOutThisIteration, &requiredInputFrameCount);
+ if (result != MA_SUCCESS) {
+ /* Fall back to a best guess. */
+ requiredInputFrameCount = (frameCountOutThisIteration * pConverter->resampler.sampleRateIn) / pConverter->resampler.sampleRateOut;
+ }
+
+ if (frameCountInThisIteration > requiredInputFrameCount) {
+ frameCountInThisIteration = requiredInputFrameCount;
}
}
- }
+ #endif
- if (pDevice->sampleRate == 0) {
- if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex || deviceType == ma_device_type_loopback) {
- pDevice->sampleRate = pDevice->capture.internalSampleRate;
+ if (pConverter->hasPreFormatConversion) {
+ if (pFramesIn != NULL) {
+ ma_convert_pcm_frames_format(pTempBufferIn, pConverter->resampler.format, pRunningFramesIn, pConverter->formatIn, frameCountInThisIteration, pConverter->channelsIn, pConverter->ditherMode);
+ pResampleBufferIn = pTempBufferIn;
+ } else {
+ pResampleBufferIn = NULL;
+ }
} else {
- pDevice->sampleRate = pDevice->playback.internalSampleRate;
+ pResampleBufferIn = pRunningFramesIn;
}
- }
- /* Data converters. */
- if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex || deviceType == ma_device_type_loopback) {
- /* Converting from internal device format to client format. */
- ma_data_converter_config converterConfig = ma_data_converter_config_init_default();
- converterConfig.formatIn = pDevice->capture.internalFormat;
- converterConfig.channelsIn = pDevice->capture.internalChannels;
- converterConfig.sampleRateIn = pDevice->capture.internalSampleRate;
- ma_channel_map_copy(converterConfig.channelMapIn, pDevice->capture.internalChannelMap, ma_min(pDevice->capture.internalChannels, MA_MAX_CHANNELS));
- converterConfig.formatOut = pDevice->capture.format;
- converterConfig.channelsOut = pDevice->capture.channels;
- converterConfig.sampleRateOut = pDevice->sampleRate;
- ma_channel_map_copy(converterConfig.channelMapOut, pDevice->capture.channelMap, ma_min(pDevice->capture.channels, MA_MAX_CHANNELS));
- converterConfig.channelMixMode = pDevice->capture.channelMixMode;
- converterConfig.resampling.allowDynamicSampleRate = MA_FALSE;
- converterConfig.resampling.algorithm = pDevice->resampling.algorithm;
- converterConfig.resampling.linear.lpfOrder = pDevice->resampling.linear.lpfOrder;
- converterConfig.resampling.speex.quality = pDevice->resampling.speex.quality;
-
- result = ma_data_converter_init(&converterConfig, &pDevice->capture.converter);
+ result = ma_resampler_process_pcm_frames(&pConverter->resampler, pResampleBufferIn, &frameCountInThisIteration, pTempBufferMid, &frameCountOutThisIteration);
if (result != MA_SUCCESS) {
return result;
}
- }
- if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) {
- /* Converting from client format to device format. */
- ma_data_converter_config converterConfig = ma_data_converter_config_init_default();
- converterConfig.formatIn = pDevice->playback.format;
- converterConfig.channelsIn = pDevice->playback.channels;
- converterConfig.sampleRateIn = pDevice->sampleRate;
- ma_channel_map_copy(converterConfig.channelMapIn, pDevice->playback.channelMap, ma_min(pDevice->playback.channels, MA_MAX_CHANNELS));
- converterConfig.formatOut = pDevice->playback.internalFormat;
- converterConfig.channelsOut = pDevice->playback.internalChannels;
- converterConfig.sampleRateOut = pDevice->playback.internalSampleRate;
- ma_channel_map_copy(converterConfig.channelMapOut, pDevice->playback.internalChannelMap, ma_min(pDevice->playback.internalChannels, MA_MAX_CHANNELS));
- converterConfig.channelMixMode = pDevice->playback.channelMixMode;
- converterConfig.resampling.allowDynamicSampleRate = MA_FALSE;
- converterConfig.resampling.algorithm = pDevice->resampling.algorithm;
- converterConfig.resampling.linear.lpfOrder = pDevice->resampling.linear.lpfOrder;
- converterConfig.resampling.speex.quality = pDevice->resampling.speex.quality;
-
- result = ma_data_converter_init(&converterConfig, &pDevice->playback.converter);
- if (result != MA_SUCCESS) {
- return result;
+
+ /*
+ The input data has been resampled so now we need to run it through the channel converter. The input data is always contained in pTempBufferMid. We only need to do
+ this part if we have an output buffer.
+ */
+ if (pFramesOut != NULL) {
+ if (pConverter->hasPostFormatConversion) {
+ pChannelsBufferOut = pTempBufferOut;
+ } else {
+ pChannelsBufferOut = pRunningFramesOut;
+ }
+
+ result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pChannelsBufferOut, pTempBufferMid, frameCountOutThisIteration);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ /* Finally we do post format conversion. */
+ if (pConverter->hasPostFormatConversion) {
+ ma_convert_pcm_frames_format(pRunningFramesOut, pConverter->formatOut, pChannelsBufferOut, pConverter->channelConverter.format, frameCountOutThisIteration, pConverter->channelConverter.channelsOut, pConverter->ditherMode);
+ }
+ }
+
+
+ framesProcessedIn += frameCountInThisIteration;
+ framesProcessedOut += frameCountOutThisIteration;
+
+ MA_ASSERT(framesProcessedIn <= frameCountIn);
+ MA_ASSERT(framesProcessedOut <= frameCountOut);
+
+ if (frameCountOutThisIteration == 0) {
+ break; /* Consumed all of our input data. */
}
}
+ if (pFrameCountIn != NULL) {
+ *pFrameCountIn = framesProcessedIn;
+ }
+ if (pFrameCountOut != NULL) {
+ *pFrameCountOut = framesProcessedOut;
+ }
+
return MA_SUCCESS;
}
+static ma_result ma_data_converter_process_pcm_frames__channels_first(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+{
+ ma_result result;
+ ma_uint64 frameCountIn;
+ ma_uint64 frameCountOut;
+ ma_uint64 framesProcessedIn;
+ ma_uint64 framesProcessedOut;
+ ma_uint8 pTempBufferIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In resampler format. */
+ ma_uint64 tempBufferInCap;
+ ma_uint8 pTempBufferMid[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In resampler format, channel converter input format. */
+ ma_uint64 tempBufferMidCap;
+ ma_uint8 pTempBufferOut[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In channel converter output format. */
+ ma_uint64 tempBufferOutCap;
+
+ MA_ASSERT(pConverter != NULL);
+ MA_ASSERT(pConverter->resampler.format == pConverter->channelConverter.format);
+ MA_ASSERT(pConverter->resampler.channels == pConverter->channelConverter.channelsOut);
+ MA_ASSERT(pConverter->resampler.channels <= pConverter->channelConverter.channelsIn);
+
+ frameCountIn = 0;
+ if (pFrameCountIn != NULL) {
+ frameCountIn = *pFrameCountIn;
+ }
+
+ frameCountOut = 0;
+ if (pFrameCountOut != NULL) {
+ frameCountOut = *pFrameCountOut;
+ }
+
+ framesProcessedIn = 0;
+ framesProcessedOut = 0;
+
+ tempBufferInCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->channelConverter.format, pConverter->channelConverter.channelsIn);
+ tempBufferMidCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->channelConverter.format, pConverter->channelConverter.channelsOut);
+ tempBufferOutCap = sizeof(pTempBufferOut) / ma_get_bytes_per_frame(pConverter->resampler.format, pConverter->resampler.channels);
+
+ while (framesProcessedOut < frameCountOut) {
+ ma_uint64 frameCountInThisIteration;
+ ma_uint64 frameCountOutThisIteration;
+ const void* pRunningFramesIn = NULL;
+ void* pRunningFramesOut = NULL;
+ const void* pChannelsBufferIn;
+ void* pResampleBufferOut;
+
+ if (pFramesIn != NULL) {
+ pRunningFramesIn = ma_offset_ptr(pFramesIn, framesProcessedIn * ma_get_bytes_per_frame(pConverter->formatIn, pConverter->channelsIn));
+ }
+ if (pFramesOut != NULL) {
+ pRunningFramesOut = ma_offset_ptr(pFramesOut, framesProcessedOut * ma_get_bytes_per_frame(pConverter->formatOut, pConverter->channelsOut));
+ }
-static ma_thread_result MA_THREADCALL ma_worker_thread(void* pData)
-{
- ma_device* pDevice = (ma_device*)pData;
- MA_ASSERT(pDevice != NULL);
+ /*
+ Before doing any processing we need to determine how many frames we should try processing
+ this iteration, for both input and output. The resampler requires us to perform format and
+ channel conversion before passing any data into it. If we get our input count wrong, we'll
+ end up peforming redundant pre-processing. This isn't the end of the world, but it does
+ result in some inefficiencies proportionate to how far our estimates are off.
+
+ If the resampler has a means to calculate exactly how much we'll need, we'll use that.
+ Otherwise we'll make a best guess. In order to do this, we'll need to calculate the output
+ frame count first.
+ */
+ frameCountOutThisIteration = (frameCountOut - framesProcessedOut);
+ if (frameCountOutThisIteration > tempBufferMidCap) {
+ frameCountOutThisIteration = tempBufferMidCap;
+ }
-#ifdef MA_WIN32
- ma_CoInitializeEx(pDevice->pContext, NULL, MA_COINIT_VALUE);
-#endif
+ if (pConverter->hasPostFormatConversion) {
+ if (frameCountOutThisIteration > tempBufferOutCap) {
+ frameCountOutThisIteration = tempBufferOutCap;
+ }
+ }
- /*
- When the device is being initialized it's initial state is set to MA_STATE_UNINITIALIZED. Before returning from
- ma_device_init(), the state needs to be set to something valid. In miniaudio the device's default state immediately
- after initialization is stopped, so therefore we need to mark the device as such. miniaudio will wait on the worker
- thread to signal an event to know when the worker thread is ready for action.
- */
- ma_device__set_state(pDevice, MA_STATE_STOPPED);
- ma_event_signal(&pDevice->stopEvent);
+ /* Now that we have the output frame count we can determine the input frame count. */
+ frameCountInThisIteration = (frameCountIn - framesProcessedIn);
+ if (pConverter->hasPreFormatConversion) {
+ if (frameCountInThisIteration > tempBufferInCap) {
+ frameCountInThisIteration = tempBufferInCap;
+ }
+ }
- for (;;) { /* <-- This loop just keeps the thread alive. The main audio loop is inside. */
- ma_result startResult;
- ma_result stopResult; /* <-- This will store the result from onDeviceStop(). If it returns an error, we don't fire the onStop callback. */
+ if (frameCountInThisIteration > tempBufferMidCap) {
+ frameCountInThisIteration = tempBufferMidCap;
+ }
- /* We wait on an event to know when something has requested that the device be started and the main loop entered. */
- ma_event_wait(&pDevice->wakeupEvent);
+ #if 1
+ {
+ ma_uint64 requiredInputFrameCount;
- /* Default result code. */
- pDevice->workResult = MA_SUCCESS;
+ result = ma_resampler_get_required_input_frame_count(&pConverter->resampler, frameCountOutThisIteration, &requiredInputFrameCount);
+ if (result != MA_SUCCESS) {
+ /* Fall back to a best guess. */
+ requiredInputFrameCount = (frameCountOutThisIteration * pConverter->resampler.sampleRateIn) / pConverter->resampler.sampleRateOut;
+ }
- /* If the reason for the wake up is that we are terminating, just break from the loop. */
- if (ma_device_get_state(pDevice) == MA_STATE_UNINITIALIZED) {
- break;
+ if (frameCountInThisIteration > requiredInputFrameCount) {
+ frameCountInThisIteration = requiredInputFrameCount;
+ }
}
+ #endif
- /*
- Getting to this point means the device is wanting to get started. The function that has requested that the device
- be started will be waiting on an event (pDevice->startEvent) which means we need to make sure we signal the event
- in both the success and error case. It's important that the state of the device is set _before_ signaling the event.
- */
- MA_ASSERT(ma_device_get_state(pDevice) == MA_STATE_STARTING);
- /* If the device has a start callback, start it now. */
- if (pDevice->pContext->callbacks.onDeviceStart != NULL) {
- startResult = pDevice->pContext->callbacks.onDeviceStart(pDevice);
+ /* Pre format conversion. */
+ if (pConverter->hasPreFormatConversion) {
+ if (pRunningFramesIn != NULL) {
+ ma_convert_pcm_frames_format(pTempBufferIn, pConverter->channelConverter.format, pRunningFramesIn, pConverter->formatIn, frameCountInThisIteration, pConverter->channelsIn, pConverter->ditherMode);
+ pChannelsBufferIn = pTempBufferIn;
+ } else {
+ pChannelsBufferIn = NULL;
+ }
} else {
- startResult = MA_SUCCESS;
+ pChannelsBufferIn = pRunningFramesIn;
}
- if (startResult != MA_SUCCESS) {
- pDevice->workResult = startResult;
- continue; /* Failed to start. Loop back to the start and wait for something to happen (pDevice->wakeupEvent). */
+
+ /* Channel conversion. */
+ result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pTempBufferMid, pChannelsBufferIn, frameCountInThisIteration);
+ if (result != MA_SUCCESS) {
+ return result;
}
- /* Make sure the state is set appropriately. */
- ma_device__set_state(pDevice, MA_STATE_STARTED);
- ma_event_signal(&pDevice->startEvent);
- if (pDevice->pContext->callbacks.onDeviceDataLoop != NULL) {
- pDevice->pContext->callbacks.onDeviceDataLoop(pDevice);
+ /* Resampling. */
+ if (pConverter->hasPostFormatConversion) {
+ pResampleBufferOut = pTempBufferOut;
} else {
- /* The backend is not using a custom main loop implementation, so now fall back to the blocking read-write implementation. */
- ma_device_audio_thread__default_read_write(pDevice);
+ pResampleBufferOut = pRunningFramesOut;
}
- /* Getting here means we have broken from the main loop which happens the application has requested that device be stopped. */
- if (pDevice->pContext->callbacks.onDeviceStop != NULL) {
- stopResult = pDevice->pContext->callbacks.onDeviceStop(pDevice);
- } else {
- stopResult = MA_SUCCESS; /* No stop callback with the backend. Just assume successful. */
+ result = ma_resampler_process_pcm_frames(&pConverter->resampler, pTempBufferMid, &frameCountInThisIteration, pResampleBufferOut, &frameCountOutThisIteration);
+ if (result != MA_SUCCESS) {
+ return result;
}
- /*
- After the device has stopped, make sure an event is posted. Don't post an onStop event if
- stopping failed. This can happen on some backends when the underlying stream has been
- stopped due to the device being physically unplugged or disabled via an OS setting.
- */
- if (pDevice->onStop && stopResult != MA_SUCCESS) {
- pDevice->onStop(pDevice);
+
+ /* Post format conversion. */
+ if (pConverter->hasPostFormatConversion) {
+ if (pRunningFramesOut != NULL) {
+ ma_convert_pcm_frames_format(pRunningFramesOut, pConverter->formatOut, pResampleBufferOut, pConverter->resampler.format, frameCountOutThisIteration, pConverter->channelsOut, pConverter->ditherMode);
+ }
}
- /* A function somewhere is waiting for the device to have stopped for real so we need to signal an event to allow it to continue. */
- ma_device__set_state(pDevice, MA_STATE_STOPPED);
- ma_event_signal(&pDevice->stopEvent);
- }
-#ifdef MA_WIN32
- ma_CoUninitialize(pDevice->pContext);
-#endif
+ framesProcessedIn += frameCountInThisIteration;
+ framesProcessedOut += frameCountOutThisIteration;
- return (ma_thread_result)0;
-}
+ MA_ASSERT(framesProcessedIn <= frameCountIn);
+ MA_ASSERT(framesProcessedOut <= frameCountOut);
+ if (frameCountOutThisIteration == 0) {
+ break; /* Consumed all of our input data. */
+ }
+ }
-/* Helper for determining whether or not the given device is initialized. */
-static ma_bool32 ma_device__is_initialized(ma_device* pDevice)
-{
- if (pDevice == NULL) {
- return MA_FALSE;
+ if (pFrameCountIn != NULL) {
+ *pFrameCountIn = framesProcessedIn;
+ }
+ if (pFrameCountOut != NULL) {
+ *pFrameCountOut = framesProcessedOut;
}
- return ma_device_get_state(pDevice) != MA_STATE_UNINITIALIZED;
+ return MA_SUCCESS;
}
-
-#ifdef MA_WIN32
-static ma_result ma_context_uninit_backend_apis__win32(ma_context* pContext)
+MA_API ma_result ma_data_converter_process_pcm_frames(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
- ma_CoUninitialize(pContext);
- ma_dlclose(pContext, pContext->win32.hUser32DLL);
- ma_dlclose(pContext, pContext->win32.hOle32DLL);
- ma_dlclose(pContext, pContext->win32.hAdvapi32DLL);
+ if (pConverter == NULL) {
+ return MA_INVALID_ARGS;
+ }
- return MA_SUCCESS;
+ switch (pConverter->executionPath)
+ {
+ case ma_data_converter_execution_path_passthrough: return ma_data_converter_process_pcm_frames__passthrough(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
+ case ma_data_converter_execution_path_format_only: return ma_data_converter_process_pcm_frames__format_only(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
+ case ma_data_converter_execution_path_channels_only: return ma_data_converter_process_pcm_frames__channels_only(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
+ case ma_data_converter_execution_path_resample_only: return ma_data_converter_process_pcm_frames__resample_only(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
+ case ma_data_converter_execution_path_resample_first: return ma_data_converter_process_pcm_frames__resample_first(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
+ case ma_data_converter_execution_path_channels_first: return ma_data_converter_process_pcm_frames__channels_first(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
+ default: return MA_INVALID_OPERATION; /* Should never hit this. */
+ }
}
-static ma_result ma_context_init_backend_apis__win32(ma_context* pContext)
+MA_API ma_result ma_data_converter_set_rate(ma_data_converter* pConverter, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut)
{
-#ifdef MA_WIN32_DESKTOP
- /* Ole32.dll */
- pContext->win32.hOle32DLL = ma_dlopen(pContext, "ole32.dll");
- if (pContext->win32.hOle32DLL == NULL) {
- return MA_FAILED_TO_INIT_BACKEND;
+ if (pConverter == NULL) {
+ return MA_INVALID_ARGS;
}
- pContext->win32.CoInitializeEx = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "CoInitializeEx");
- pContext->win32.CoUninitialize = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "CoUninitialize");
- pContext->win32.CoCreateInstance = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "CoCreateInstance");
- pContext->win32.CoTaskMemFree = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "CoTaskMemFree");
- pContext->win32.PropVariantClear = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "PropVariantClear");
- pContext->win32.StringFromGUID2 = (ma_proc)ma_dlsym(pContext, pContext->win32.hOle32DLL, "StringFromGUID2");
+ if (pConverter->hasResampler == MA_FALSE) {
+ return MA_INVALID_OPERATION; /* Dynamic resampling not enabled. */
+ }
+ return ma_resampler_set_rate(&pConverter->resampler, sampleRateIn, sampleRateOut);
+}
- /* User32.dll */
- pContext->win32.hUser32DLL = ma_dlopen(pContext, "user32.dll");
- if (pContext->win32.hUser32DLL == NULL) {
- return MA_FAILED_TO_INIT_BACKEND;
+MA_API ma_result ma_data_converter_set_rate_ratio(ma_data_converter* pConverter, float ratioInOut)
+{
+ if (pConverter == NULL) {
+ return MA_INVALID_ARGS;
}
- pContext->win32.GetForegroundWindow = (ma_proc)ma_dlsym(pContext, pContext->win32.hUser32DLL, "GetForegroundWindow");
- pContext->win32.GetDesktopWindow = (ma_proc)ma_dlsym(pContext, pContext->win32.hUser32DLL, "GetDesktopWindow");
+ if (pConverter->hasResampler == MA_FALSE) {
+ return MA_INVALID_OPERATION; /* Dynamic resampling not enabled. */
+ }
+ return ma_resampler_set_rate_ratio(&pConverter->resampler, ratioInOut);
+}
- /* Advapi32.dll */
- pContext->win32.hAdvapi32DLL = ma_dlopen(pContext, "advapi32.dll");
- if (pContext->win32.hAdvapi32DLL == NULL) {
- return MA_FAILED_TO_INIT_BACKEND;
+MA_API ma_uint64 ma_data_converter_get_input_latency(const ma_data_converter* pConverter)
+{
+ if (pConverter == NULL) {
+ return 0;
}
- pContext->win32.RegOpenKeyExA = (ma_proc)ma_dlsym(pContext, pContext->win32.hAdvapi32DLL, "RegOpenKeyExA");
- pContext->win32.RegCloseKey = (ma_proc)ma_dlsym(pContext, pContext->win32.hAdvapi32DLL, "RegCloseKey");
- pContext->win32.RegQueryValueExA = (ma_proc)ma_dlsym(pContext, pContext->win32.hAdvapi32DLL, "RegQueryValueExA");
-#endif
+ if (pConverter->hasResampler) {
+ return ma_resampler_get_input_latency(&pConverter->resampler);
+ }
- ma_CoInitializeEx(pContext, NULL, MA_COINIT_VALUE);
- return MA_SUCCESS;
+ return 0; /* No latency without a resampler. */
}
-#else
-static ma_result ma_context_uninit_backend_apis__nix(ma_context* pContext)
+
+MA_API ma_uint64 ma_data_converter_get_output_latency(const ma_data_converter* pConverter)
{
-#if defined(MA_USE_RUNTIME_LINKING_FOR_PTHREAD) && !defined(MA_NO_RUNTIME_LINKING)
- ma_dlclose(pContext, pContext->posix.pthreadSO);
-#else
- (void)pContext;
-#endif
+ if (pConverter == NULL) {
+ return 0;
+ }
- return MA_SUCCESS;
+ if (pConverter->hasResampler) {
+ return ma_resampler_get_output_latency(&pConverter->resampler);
+ }
+
+ return 0; /* No latency without a resampler. */
}
-static ma_result ma_context_init_backend_apis__nix(ma_context* pContext)
+MA_API ma_result ma_data_converter_get_required_input_frame_count(const ma_data_converter* pConverter, ma_uint64 outputFrameCount, ma_uint64* pInputFrameCount)
{
- /* pthread */
-#if defined(MA_USE_RUNTIME_LINKING_FOR_PTHREAD) && !defined(MA_NO_RUNTIME_LINKING)
- const char* libpthreadFileNames[] = {
- "libpthread.so",
- "libpthread.so.0",
- "libpthread.dylib"
- };
- size_t i;
-
- for (i = 0; i < sizeof(libpthreadFileNames) / sizeof(libpthreadFileNames[0]); ++i) {
- pContext->posix.pthreadSO = ma_dlopen(pContext, libpthreadFileNames[i]);
- if (pContext->posix.pthreadSO != NULL) {
- break;
- }
+ if (pInputFrameCount == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pContext->posix.pthreadSO == NULL) {
- return MA_FAILED_TO_INIT_BACKEND;
- }
+ *pInputFrameCount = 0;
- pContext->posix.pthread_create = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_create");
- pContext->posix.pthread_join = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_join");
- pContext->posix.pthread_mutex_init = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_mutex_init");
- pContext->posix.pthread_mutex_destroy = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_mutex_destroy");
- pContext->posix.pthread_mutex_lock = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_mutex_lock");
- pContext->posix.pthread_mutex_unlock = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_mutex_unlock");
- pContext->posix.pthread_cond_init = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_cond_init");
- pContext->posix.pthread_cond_destroy = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_cond_destroy");
- pContext->posix.pthread_cond_wait = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_cond_wait");
- pContext->posix.pthread_cond_signal = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_cond_signal");
- pContext->posix.pthread_attr_init = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_attr_init");
- pContext->posix.pthread_attr_destroy = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_attr_destroy");
- pContext->posix.pthread_attr_setschedpolicy = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_attr_setschedpolicy");
- pContext->posix.pthread_attr_getschedparam = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_attr_getschedparam");
- pContext->posix.pthread_attr_setschedparam = (ma_proc)ma_dlsym(pContext, pContext->posix.pthreadSO, "pthread_attr_setschedparam");
-#else
- pContext->posix.pthread_create = (ma_proc)pthread_create;
- pContext->posix.pthread_join = (ma_proc)pthread_join;
- pContext->posix.pthread_mutex_init = (ma_proc)pthread_mutex_init;
- pContext->posix.pthread_mutex_destroy = (ma_proc)pthread_mutex_destroy;
- pContext->posix.pthread_mutex_lock = (ma_proc)pthread_mutex_lock;
- pContext->posix.pthread_mutex_unlock = (ma_proc)pthread_mutex_unlock;
- pContext->posix.pthread_cond_init = (ma_proc)pthread_cond_init;
- pContext->posix.pthread_cond_destroy = (ma_proc)pthread_cond_destroy;
- pContext->posix.pthread_cond_wait = (ma_proc)pthread_cond_wait;
- pContext->posix.pthread_cond_signal = (ma_proc)pthread_cond_signal;
- pContext->posix.pthread_attr_init = (ma_proc)pthread_attr_init;
- pContext->posix.pthread_attr_destroy = (ma_proc)pthread_attr_destroy;
-#if !defined(__EMSCRIPTEN__)
- pContext->posix.pthread_attr_setschedpolicy = (ma_proc)pthread_attr_setschedpolicy;
- pContext->posix.pthread_attr_getschedparam = (ma_proc)pthread_attr_getschedparam;
- pContext->posix.pthread_attr_setschedparam = (ma_proc)pthread_attr_setschedparam;
-#endif
-#endif
+ if (pConverter == NULL) {
+ return MA_INVALID_ARGS;
+ }
- return MA_SUCCESS;
+ if (pConverter->hasResampler) {
+ return ma_resampler_get_required_input_frame_count(&pConverter->resampler, outputFrameCount, pInputFrameCount);
+ } else {
+ *pInputFrameCount = outputFrameCount; /* 1:1 */
+ return MA_SUCCESS;
+ }
}
-#endif
-static ma_result ma_context_init_backend_apis(ma_context* pContext)
+MA_API ma_result ma_data_converter_get_expected_output_frame_count(const ma_data_converter* pConverter, ma_uint64 inputFrameCount, ma_uint64* pOutputFrameCount)
{
- ma_result result;
-#ifdef MA_WIN32
- result = ma_context_init_backend_apis__win32(pContext);
-#else
- result = ma_context_init_backend_apis__nix(pContext);
-#endif
+ if (pOutputFrameCount == NULL) {
+ return MA_INVALID_ARGS;
+ }
- return result;
-}
+ *pOutputFrameCount = 0;
-static ma_result ma_context_uninit_backend_apis(ma_context* pContext)
-{
- ma_result result;
-#ifdef MA_WIN32
- result = ma_context_uninit_backend_apis__win32(pContext);
-#else
- result = ma_context_uninit_backend_apis__nix(pContext);
-#endif
+ if (pConverter == NULL) {
+ return MA_INVALID_ARGS;
+ }
- return result;
+ if (pConverter->hasResampler) {
+ return ma_resampler_get_expected_output_frame_count(&pConverter->resampler, inputFrameCount, pOutputFrameCount);
+ } else {
+ *pOutputFrameCount = inputFrameCount; /* 1:1 */
+ return MA_SUCCESS;
+ }
}
-
-static ma_bool32 ma_context_is_backend_asynchronous(ma_context* pContext)
+MA_API ma_result ma_data_converter_get_input_channel_map(const ma_data_converter* pConverter, ma_channel* pChannelMap, size_t channelMapCap)
{
- MA_ASSERT(pContext != NULL);
+ if (pConverter == NULL || pChannelMap == NULL) {
+ return MA_INVALID_ARGS;
+ }
- if (pContext->callbacks.onDeviceRead == NULL && pContext->callbacks.onDeviceWrite == NULL) {
- if (pContext->callbacks.onDeviceDataLoop == NULL) {
- return MA_TRUE;
- } else {
- return MA_FALSE;
- }
+ if (pConverter->hasChannelConverter) {
+ ma_channel_converter_get_output_channel_map(&pConverter->channelConverter, pChannelMap, channelMapCap);
} else {
- return MA_FALSE;
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, pConverter->channelsOut);
}
-}
+ return MA_SUCCESS;
+}
-MA_API ma_context_config ma_context_config_init()
+MA_API ma_result ma_data_converter_get_output_channel_map(const ma_data_converter* pConverter, ma_channel* pChannelMap, size_t channelMapCap)
{
- ma_context_config config;
- MA_ZERO_OBJECT(&config);
+ if (pConverter == NULL || pChannelMap == NULL) {
+ return MA_INVALID_ARGS;
+ }
- return config;
+ if (pConverter->hasChannelConverter) {
+ ma_channel_converter_get_input_channel_map(&pConverter->channelConverter, pChannelMap, channelMapCap);
+ } else {
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, pConverter->channelsIn);
+ }
+
+ return MA_SUCCESS;
}
-MA_API ma_result ma_context_init(const ma_backend backends[], ma_uint32 backendCount, const ma_context_config* pConfig, ma_context* pContext)
-{
- ma_result result;
- ma_context_config defaultConfig;
- ma_backend defaultBackends[ma_backend_null+1];
- ma_uint32 iBackend;
- ma_backend* pBackendsToIterate;
- ma_uint32 backendsToIterateCount;
- if (pContext == NULL) {
- return MA_INVALID_ARGS;
- }
- MA_ZERO_OBJECT(pContext);
+/**************************************************************************************************************************************************************
- /* Always make sure the config is set first to ensure properties are available as soon as possible. */
- if (pConfig == NULL) {
- defaultConfig = ma_context_config_init();
- pConfig = &defaultConfig;
- }
+Channel Maps
- /* Allocation callbacks need to come first because they'll be passed around to other areas. */
- result = ma_allocation_callbacks_init_copy(&pContext->allocationCallbacks, &pConfig->allocationCallbacks);
- if (result != MA_SUCCESS) {
- return result;
- }
+**************************************************************************************************************************************************************/
+static ma_channel ma_channel_map_init_standard_channel(ma_standard_channel_map standardChannelMap, ma_uint32 channelCount, ma_uint32 channelIndex);
- /* Get a lot set up first so we can start logging ASAP. */
- if (pConfig->pLog != NULL) {
- pContext->pLog = pConfig->pLog;
+MA_API ma_channel ma_channel_map_get_channel(const ma_channel* pChannelMap, ma_uint32 channelCount, ma_uint32 channelIndex)
+{
+ if (pChannelMap == NULL) {
+ return ma_channel_map_init_standard_channel(ma_standard_channel_map_default, channelCount, channelIndex);
} else {
- result = ma_log_init(&pContext->allocationCallbacks, &pContext->log);
- if (result == MA_SUCCESS) {
- pContext->pLog = &pContext->log;
- } else {
- pContext->pLog = NULL; /* Logging is not available. */
+ if (channelIndex >= channelCount) {
+ return MA_CHANNEL_NONE;
}
- }
- pContext->logCallback = pConfig->logCallback;
- pContext->threadPriority = pConfig->threadPriority;
- pContext->threadStackSize = pConfig->threadStackSize;
- pContext->pUserData = pConfig->pUserData;
-
- /* Backend APIs need to be initialized first. This is where external libraries will be loaded and linked. */
- result = ma_context_init_backend_apis(pContext);
- if (result != MA_SUCCESS) {
- return result;
+ return pChannelMap[channelIndex];
}
+}
- for (iBackend = 0; iBackend <= ma_backend_null; ++iBackend) {
- defaultBackends[iBackend] = (ma_backend)iBackend;
+MA_API void ma_channel_map_init_blank(ma_channel* pChannelMap, ma_uint32 channels)
+{
+ if (pChannelMap == NULL) {
+ return;
}
- pBackendsToIterate = (ma_backend*)backends;
- backendsToIterateCount = backendCount;
- if (pBackendsToIterate == NULL) {
- pBackendsToIterate = (ma_backend*)defaultBackends;
- backendsToIterateCount = ma_countof(defaultBackends);
+ MA_ZERO_MEMORY(pChannelMap, sizeof(*pChannelMap) * channels);
+}
+
+
+static ma_channel ma_channel_map_init_standard_channel_microsoft(ma_uint32 channelCount, ma_uint32 channelIndex)
+{
+ if (channelCount == 0 || channelIndex >= channelCount) {
+ return MA_CHANNEL_NONE;
}
- MA_ASSERT(pBackendsToIterate != NULL);
+ /* This is the Microsoft channel map. Based off the speaker configurations mentioned here: https://docs.microsoft.com/en-us/windows-hardware/drivers/ddi/content/ksmedia/ns-ksmedia-ksaudio_channel_config */
+ switch (channelCount)
+ {
+ case 0: return MA_CHANNEL_NONE;
- for (iBackend = 0; iBackend < backendsToIterateCount; iBackend += 1) {
- ma_backend backend = pBackendsToIterate[iBackend];
+ case 1:
+ {
+ return MA_CHANNEL_MONO;
+ } break;
- /* Make sure all callbacks are reset so we don't accidentally drag in any from previously failed initialization attempts. */
- MA_ZERO_OBJECT(&pContext->callbacks);
+ case 2:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ }
+ } break;
- /* These backends are using the new callback system. */
- switch (backend) {
- #ifdef MA_HAS_WASAPI
- case ma_backend_wasapi:
- {
- pContext->callbacks.onContextInit = ma_context_init__wasapi;
- } break;
- #endif
- #ifdef MA_HAS_DSOUND
- case ma_backend_dsound:
- {
- pContext->callbacks.onContextInit = ma_context_init__dsound;
- } break;
- #endif
- #ifdef MA_HAS_WINMM
- case ma_backend_winmm:
- {
- pContext->callbacks.onContextInit = ma_context_init__winmm;
- } break;
- #endif
- #ifdef MA_HAS_COREAUDIO
- case ma_backend_coreaudio:
- {
- pContext->callbacks.onContextInit = ma_context_init__coreaudio;
- } break;
- #endif
- #ifdef MA_HAS_SNDIO
- case ma_backend_sndio:
- {
- pContext->callbacks.onContextInit = ma_context_init__sndio;
- } break;
- #endif
- #ifdef MA_HAS_AUDIO4
- case ma_backend_audio4:
- {
- pContext->callbacks.onContextInit = ma_context_init__audio4;
- } break;
- #endif
- #ifdef MA_HAS_OSS
- case ma_backend_oss:
- {
- pContext->callbacks.onContextInit = ma_context_init__oss;
- } break;
- #endif
- #ifdef MA_HAS_PULSEAUDIO
- case ma_backend_pulseaudio:
- {
- pContext->callbacks.onContextInit = ma_context_init__pulse;
- } break;
- #endif
- #ifdef MA_HAS_ALSA
- case ma_backend_alsa:
- {
- pContext->callbacks.onContextInit = ma_context_init__alsa;
- } break;
- #endif
- #ifdef MA_HAS_JACK
- case ma_backend_jack:
- {
- pContext->callbacks.onContextInit = ma_context_init__jack;
- } break;
- #endif
- #ifdef MA_HAS_AAUDIO
- case ma_backend_aaudio:
- {
- pContext->callbacks.onContextInit = ma_context_init__aaudio;
- } break;
- #endif
- #ifdef MA_HAS_OPENSL
- case ma_backend_opensl:
- {
- pContext->callbacks.onContextInit = ma_context_init__opensl;
- } break;
- #endif
- #ifdef MA_HAS_WEBAUDIO
- case ma_backend_webaudio:
- {
- pContext->callbacks.onContextInit = ma_context_init__webaudio;
- } break;
- #endif
- #ifdef MA_HAS_CUSTOM
- case ma_backend_custom:
- {
- /* Slightly different logic for custom backends. Custom backends can optionally set all of their callbacks in the config. */
- pContext->callbacks = pConfig->custom;
- } break;
- #endif
- #ifdef MA_HAS_NULL
- case ma_backend_null:
- {
- pContext->callbacks.onContextInit = ma_context_init__null;
- } break;
- #endif
+ case 3: /* No defined, but best guess. */
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_FRONT_CENTER;
+ }
+ } break;
- default: break;
- }
+ case 4:
+ {
+ switch (channelIndex) {
+ #ifndef MA_USE_QUAD_MICROSOFT_CHANNEL_MAP
+ /* Surround. Using the Surround profile has the advantage of the 3rd channel (MA_CHANNEL_FRONT_CENTER) mapping nicely with higher channel counts. */
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_FRONT_CENTER;
+ case 3: return MA_CHANNEL_BACK_CENTER;
+ #else
+ /* Quad. */
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_BACK_LEFT;
+ case 3: return MA_CHANNEL_BACK_RIGHT;
+ #endif
+ }
+ } break;
- if (pContext->callbacks.onContextInit != NULL) {
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "Attempting to initialize %s backend...\n", ma_get_backend_name(backend));
- result = pContext->callbacks.onContextInit(pContext, pConfig, &pContext->callbacks);
- } else {
- result = MA_NO_BACKEND;
- }
+ case 5: /* Not defined, but best guess. */
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_FRONT_CENTER;
+ case 3: return MA_CHANNEL_BACK_LEFT;
+ case 4: return MA_CHANNEL_BACK_RIGHT;
+ }
+ } break;
- /* If this iteration was successful, return. */
- if (result == MA_SUCCESS) {
- result = ma_mutex_init(&pContext->deviceEnumLock);
- if (result != MA_SUCCESS) {
- ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_WARNING, "Failed to initialize mutex for device enumeration. ma_context_get_devices() is not thread safe.\n", result);
+ case 6:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_FRONT_CENTER;
+ case 3: return MA_CHANNEL_LFE;
+ case 4: return MA_CHANNEL_SIDE_LEFT;
+ case 5: return MA_CHANNEL_SIDE_RIGHT;
}
+ } break;
- result = ma_mutex_init(&pContext->deviceInfoLock);
- if (result != MA_SUCCESS) {
- ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_WARNING, "Failed to initialize mutex for device info retrieval. ma_context_get_device_info() is not thread safe.\n", result);
+ case 7: /* Not defined, but best guess. */
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_FRONT_CENTER;
+ case 3: return MA_CHANNEL_LFE;
+ case 4: return MA_CHANNEL_BACK_CENTER;
+ case 5: return MA_CHANNEL_SIDE_LEFT;
+ case 6: return MA_CHANNEL_SIDE_RIGHT;
}
+ } break;
- #ifdef MA_DEBUG_OUTPUT
- {
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[miniaudio] Endian: %s\n", ma_is_little_endian() ? "LE" : "BE");
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[miniaudio] SSE2: %s\n", ma_has_sse2() ? "YES" : "NO");
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[miniaudio] AVX2: %s\n", ma_has_avx2() ? "YES" : "NO");
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[miniaudio] AVX512F: %s\n", ma_has_avx512f() ? "YES" : "NO");
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "[miniaudio] NEON: %s\n", ma_has_neon() ? "YES" : "NO");
+ case 8:
+ default:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_FRONT_CENTER;
+ case 3: return MA_CHANNEL_LFE;
+ case 4: return MA_CHANNEL_BACK_LEFT;
+ case 5: return MA_CHANNEL_BACK_RIGHT;
+ case 6: return MA_CHANNEL_SIDE_LEFT;
+ case 7: return MA_CHANNEL_SIDE_RIGHT;
}
- #endif
+ } break;
+ }
- pContext->backend = backend;
- return result;
- } else {
- ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_DEBUG, "Failed to initialize %s backend.\n", ma_get_backend_name(backend));
+ if (channelCount > 8) {
+ if (channelIndex < 32) { /* We have 32 AUX channels. */
+ return (ma_channel)(MA_CHANNEL_AUX_0 + (channelIndex - 8));
}
}
- /* If we get here it means an error occurred. */
- MA_ZERO_OBJECT(pContext); /* Safety. */
- return MA_NO_BACKEND;
+ /* Getting here means we don't know how to map the channel position so just return MA_CHANNEL_NONE. */
+ return MA_CHANNEL_NONE;
}
-MA_API ma_result ma_context_uninit(ma_context* pContext)
+static ma_channel ma_channel_map_init_standard_channel_alsa(ma_uint32 channelCount, ma_uint32 channelIndex)
{
- if (pContext == NULL) {
- return MA_INVALID_ARGS;
- }
-
- if (pContext->callbacks.onContextUninit != NULL) {
- pContext->callbacks.onContextUninit(pContext);
- }
-
- ma_mutex_uninit(&pContext->deviceEnumLock);
- ma_mutex_uninit(&pContext->deviceInfoLock);
- ma__free_from_callbacks(pContext->pDeviceInfos, &pContext->allocationCallbacks);
- ma_context_uninit_backend_apis(pContext);
-
- if (pContext->pLog == &pContext->log) {
- ma_log_uninit(&pContext->log);
- }
+ switch (channelCount)
+ {
+ case 0: return MA_CHANNEL_NONE;
- return MA_SUCCESS;
-}
+ case 1:
+ {
+ return MA_CHANNEL_MONO;
+ } break;
-MA_API size_t ma_context_sizeof()
-{
- return sizeof(ma_context);
-}
+ case 2:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ }
+ } break;
+ case 3:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_FRONT_CENTER;
+ }
+ } break;
-MA_API ma_log* ma_context_get_log(ma_context* pContext)
-{
- if (pContext == NULL) {
- return NULL;
- }
+ case 4:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_BACK_LEFT;
+ case 3: return MA_CHANNEL_BACK_RIGHT;
+ }
+ } break;
- return pContext->pLog;
-}
+ case 5:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_BACK_LEFT;
+ case 3: return MA_CHANNEL_BACK_RIGHT;
+ case 4: return MA_CHANNEL_FRONT_CENTER;
+ }
+ } break;
+ case 6:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_BACK_LEFT;
+ case 3: return MA_CHANNEL_BACK_RIGHT;
+ case 4: return MA_CHANNEL_FRONT_CENTER;
+ case 5: return MA_CHANNEL_LFE;
+ }
+ } break;
-MA_API ma_result ma_context_enumerate_devices(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
-{
- ma_result result;
+ case 7:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_BACK_LEFT;
+ case 3: return MA_CHANNEL_BACK_RIGHT;
+ case 4: return MA_CHANNEL_FRONT_CENTER;
+ case 5: return MA_CHANNEL_LFE;
+ case 6: return MA_CHANNEL_BACK_CENTER;
+ }
+ } break;
- if (pContext == NULL || callback == NULL) {
- return MA_INVALID_ARGS;
+ case 8:
+ default:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_BACK_LEFT;
+ case 3: return MA_CHANNEL_BACK_RIGHT;
+ case 4: return MA_CHANNEL_FRONT_CENTER;
+ case 5: return MA_CHANNEL_LFE;
+ case 6: return MA_CHANNEL_SIDE_LEFT;
+ case 7: return MA_CHANNEL_SIDE_RIGHT;
+ }
+ } break;
}
- if (pContext->callbacks.onContextEnumerateDevices == NULL) {
- return MA_INVALID_OPERATION;
+ if (channelCount > 8) {
+ if (channelIndex < 32) { /* We have 32 AUX channels. */
+ return (ma_channel)(MA_CHANNEL_AUX_0 + (channelIndex - 8));
+ }
}
- ma_mutex_lock(&pContext->deviceEnumLock);
+ /* Getting here means we don't know how to map the channel position so just return MA_CHANNEL_NONE. */
+ return MA_CHANNEL_NONE;
+}
+
+static ma_channel ma_channel_map_init_standard_channel_rfc3551(ma_uint32 channelCount, ma_uint32 channelIndex)
+{
+ switch (channelCount)
{
- result = pContext->callbacks.onContextEnumerateDevices(pContext, callback, pUserData);
- }
- ma_mutex_unlock(&pContext->deviceEnumLock);
+ case 0: return MA_CHANNEL_NONE;
- return result;
-}
+ case 1:
+ {
+ return MA_CHANNEL_MONO;
+ } break;
+ case 2:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ }
+ } break;
-static ma_bool32 ma_context_get_devices__enum_callback(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pInfo, void* pUserData)
-{
- /*
- We need to insert the device info into our main internal buffer. Where it goes depends on the device type. If it's a capture device
- it's just appended to the end. If it's a playback device it's inserted just before the first capture device.
- */
+ case 3:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_FRONT_CENTER;
+ }
+ } break;
- /*
- First make sure we have room. Since the number of devices we add to the list is usually relatively small I've decided to use a
- simple fixed size increment for buffer expansion.
- */
- const ma_uint32 bufferExpansionCount = 2;
- const ma_uint32 totalDeviceInfoCount = pContext->playbackDeviceInfoCount + pContext->captureDeviceInfoCount;
+ case 4:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 2: return MA_CHANNEL_FRONT_CENTER;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 3: return MA_CHANNEL_BACK_CENTER;
+ }
+ } break;
- if (totalDeviceInfoCount >= pContext->deviceInfoCapacity) {
- ma_uint32 oldCapacity = pContext->deviceInfoCapacity;
- ma_uint32 newCapacity = oldCapacity + bufferExpansionCount;
- ma_device_info* pNewInfos = (ma_device_info*)ma__realloc_from_callbacks(pContext->pDeviceInfos, sizeof(*pContext->pDeviceInfos)*newCapacity, sizeof(*pContext->pDeviceInfos)*oldCapacity, &pContext->allocationCallbacks);
- if (pNewInfos == NULL) {
- return MA_FALSE; /* Out of memory. */
- }
+ case 5:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_FRONT_CENTER;
+ case 3: return MA_CHANNEL_BACK_LEFT;
+ case 4: return MA_CHANNEL_BACK_RIGHT;
+ }
+ } break;
- pContext->pDeviceInfos = pNewInfos;
- pContext->deviceInfoCapacity = newCapacity;
+ case 6:
+ default:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_SIDE_LEFT;
+ case 2: return MA_CHANNEL_FRONT_CENTER;
+ case 3: return MA_CHANNEL_FRONT_RIGHT;
+ case 4: return MA_CHANNEL_SIDE_RIGHT;
+ case 5: return MA_CHANNEL_BACK_CENTER;
+ }
+ } break;
}
- if (deviceType == ma_device_type_playback) {
- /* Playback. Insert just before the first capture device. */
-
- /* The first thing to do is move all of the capture devices down a slot. */
- ma_uint32 iFirstCaptureDevice = pContext->playbackDeviceInfoCount;
- size_t iCaptureDevice;
- for (iCaptureDevice = totalDeviceInfoCount; iCaptureDevice > iFirstCaptureDevice; --iCaptureDevice) {
- pContext->pDeviceInfos[iCaptureDevice] = pContext->pDeviceInfos[iCaptureDevice-1];
+ if (channelCount > 6) {
+ if (channelIndex < 32) { /* We have 32 AUX channels. */
+ return (ma_channel)(MA_CHANNEL_AUX_0 + (channelIndex - 6));
}
-
- /* Now just insert where the first capture device was before moving it down a slot. */
- pContext->pDeviceInfos[iFirstCaptureDevice] = *pInfo;
- pContext->playbackDeviceInfoCount += 1;
- } else {
- /* Capture. Insert at the end. */
- pContext->pDeviceInfos[totalDeviceInfoCount] = *pInfo;
- pContext->captureDeviceInfoCount += 1;
}
- (void)pUserData;
- return MA_TRUE;
+ /* Getting here means we don't know how to map the channel position so just return MA_CHANNEL_NONE. */
+ return MA_CHANNEL_NONE;
}
-MA_API ma_result ma_context_get_devices(ma_context* pContext, ma_device_info** ppPlaybackDeviceInfos, ma_uint32* pPlaybackDeviceCount, ma_device_info** ppCaptureDeviceInfos, ma_uint32* pCaptureDeviceCount)
+static ma_channel ma_channel_map_init_standard_channel_flac(ma_uint32 channelCount, ma_uint32 channelIndex)
{
- ma_result result;
+ switch (channelCount)
+ {
+ case 0: return MA_CHANNEL_NONE;
- /* Safety. */
- if (ppPlaybackDeviceInfos != NULL) *ppPlaybackDeviceInfos = NULL;
- if (pPlaybackDeviceCount != NULL) *pPlaybackDeviceCount = 0;
- if (ppCaptureDeviceInfos != NULL) *ppCaptureDeviceInfos = NULL;
- if (pCaptureDeviceCount != NULL) *pCaptureDeviceCount = 0;
+ case 1:
+ {
+ return MA_CHANNEL_MONO;
+ } break;
- if (pContext == NULL) {
- return MA_INVALID_ARGS;
- }
+ case 2:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ }
+ } break;
- if (pContext->callbacks.onContextEnumerateDevices == NULL) {
- return MA_INVALID_OPERATION;
- }
+ case 3:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_FRONT_CENTER;
+ }
+ } break;
- /* Note that we don't use ma_context_enumerate_devices() here because we want to do locking at a higher level. */
- ma_mutex_lock(&pContext->deviceEnumLock);
- {
- /* Reset everything first. */
- pContext->playbackDeviceInfoCount = 0;
- pContext->captureDeviceInfoCount = 0;
+ case 4:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_BACK_LEFT;
+ case 3: return MA_CHANNEL_BACK_RIGHT;
+ }
+ } break;
- /* Now enumerate over available devices. */
- result = pContext->callbacks.onContextEnumerateDevices(pContext, ma_context_get_devices__enum_callback, NULL);
- if (result == MA_SUCCESS) {
- /* Playback devices. */
- if (ppPlaybackDeviceInfos != NULL) {
- *ppPlaybackDeviceInfos = pContext->pDeviceInfos;
+ case 5:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_FRONT_CENTER;
+ case 3: return MA_CHANNEL_BACK_LEFT;
+ case 4: return MA_CHANNEL_BACK_RIGHT;
}
- if (pPlaybackDeviceCount != NULL) {
- *pPlaybackDeviceCount = pContext->playbackDeviceInfoCount;
+ } break;
+
+ case 6:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_FRONT_CENTER;
+ case 3: return MA_CHANNEL_LFE;
+ case 4: return MA_CHANNEL_BACK_LEFT;
+ case 5: return MA_CHANNEL_BACK_RIGHT;
}
+ } break;
- /* Capture devices. */
- if (ppCaptureDeviceInfos != NULL) {
- *ppCaptureDeviceInfos = pContext->pDeviceInfos + pContext->playbackDeviceInfoCount; /* Capture devices come after playback devices. */
+ case 7:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_FRONT_CENTER;
+ case 3: return MA_CHANNEL_LFE;
+ case 4: return MA_CHANNEL_BACK_CENTER;
+ case 5: return MA_CHANNEL_SIDE_LEFT;
+ case 6: return MA_CHANNEL_SIDE_RIGHT;
}
- if (pCaptureDeviceCount != NULL) {
- *pCaptureDeviceCount = pContext->captureDeviceInfoCount;
+ } break;
+
+ case 8:
+ default:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_FRONT_CENTER;
+ case 3: return MA_CHANNEL_LFE;
+ case 4: return MA_CHANNEL_BACK_LEFT;
+ case 5: return MA_CHANNEL_BACK_RIGHT;
+ case 6: return MA_CHANNEL_SIDE_LEFT;
+ case 7: return MA_CHANNEL_SIDE_RIGHT;
}
+ } break;
+ }
+
+ if (channelCount > 8) {
+ if (channelIndex < 32) { /* We have 32 AUX channels. */
+ return (ma_channel)(MA_CHANNEL_AUX_0 + (channelIndex - 8));
}
}
- ma_mutex_unlock(&pContext->deviceEnumLock);
- return result;
+ /* Getting here means we don't know how to map the channel position so just return MA_CHANNEL_NONE. */
+ return MA_CHANNEL_NONE;
}
-MA_API ma_result ma_context_get_device_info(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_share_mode shareMode, ma_device_info* pDeviceInfo)
+static ma_channel ma_channel_map_init_standard_channel_vorbis(ma_uint32 channelCount, ma_uint32 channelIndex)
{
- ma_result result;
- ma_device_info deviceInfo;
-
- (void)shareMode; /* Unused. This parameter will be removed in version 0.11. */
-
- /* NOTE: Do not clear pDeviceInfo on entry. The reason is the pDeviceID may actually point to pDeviceInfo->id which will break things. */
- if (pContext == NULL || pDeviceInfo == NULL) {
- return MA_INVALID_ARGS;
- }
-
- MA_ZERO_OBJECT(&deviceInfo);
-
- /* Help the backend out by copying over the device ID if we have one. */
- if (pDeviceID != NULL) {
- MA_COPY_MEMORY(&deviceInfo.id, pDeviceID, sizeof(*pDeviceID));
- }
+ switch (channelCount)
+ {
+ case 0: return MA_CHANNEL_NONE;
- if (pContext->callbacks.onContextGetDeviceInfo == NULL) {
- return MA_INVALID_OPERATION;
- }
+ case 1:
+ {
+ return MA_CHANNEL_MONO;
+ } break;
- ma_mutex_lock(&pContext->deviceInfoLock);
- {
- result = pContext->callbacks.onContextGetDeviceInfo(pContext, deviceType, pDeviceID, &deviceInfo);
- }
- ma_mutex_unlock(&pContext->deviceInfoLock);
+ case 2:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ }
+ } break;
- /*
- If the backend is using the new device info system, do a pass to fill out the old settings for backwards compatibility. This will be removed in
- the future when all backends have implemented the new device info system.
- */
- if (deviceInfo.nativeDataFormatCount > 0) {
- ma_uint32 iNativeFormat;
- ma_uint32 iSampleFormat;
+ case 3:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_CENTER;
+ case 2: return MA_CHANNEL_FRONT_RIGHT;
+ }
+ } break;
- deviceInfo.minChannels = 0xFFFFFFFF;
- deviceInfo.maxChannels = 0;
- deviceInfo.minSampleRate = 0xFFFFFFFF;
- deviceInfo.maxSampleRate = 0;
+ case 4:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_BACK_LEFT;
+ case 3: return MA_CHANNEL_BACK_RIGHT;
+ }
+ } break;
- for (iNativeFormat = 0; iNativeFormat < deviceInfo.nativeDataFormatCount; iNativeFormat += 1) {
- /* Formats. */
- if (deviceInfo.nativeDataFormats[iNativeFormat].format == ma_format_unknown) {
- /* All formats are supported. */
- deviceInfo.formats[0] = ma_format_u8;
- deviceInfo.formats[1] = ma_format_s16;
- deviceInfo.formats[2] = ma_format_s24;
- deviceInfo.formats[3] = ma_format_s32;
- deviceInfo.formats[4] = ma_format_f32;
- deviceInfo.formatCount = 5;
- } else {
- /* Make sure the format isn't already in the list. If so, skip. */
- ma_bool32 alreadyExists = MA_FALSE;
- for (iSampleFormat = 0; iSampleFormat < deviceInfo.formatCount; iSampleFormat += 1) {
- if (deviceInfo.formats[iSampleFormat] == deviceInfo.nativeDataFormats[iNativeFormat].format) {
- alreadyExists = MA_TRUE;
- break;
- }
- }
+ case 5:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_CENTER;
+ case 2: return MA_CHANNEL_FRONT_RIGHT;
+ case 3: return MA_CHANNEL_BACK_LEFT;
+ case 4: return MA_CHANNEL_BACK_RIGHT;
+ }
+ } break;
- if (!alreadyExists) {
- deviceInfo.formats[deviceInfo.formatCount++] = deviceInfo.nativeDataFormats[iNativeFormat].format;
- }
+ case 6:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_CENTER;
+ case 2: return MA_CHANNEL_FRONT_RIGHT;
+ case 3: return MA_CHANNEL_BACK_LEFT;
+ case 4: return MA_CHANNEL_BACK_RIGHT;
+ case 5: return MA_CHANNEL_LFE;
}
+ } break;
- /* Channels. */
- if (deviceInfo.nativeDataFormats[iNativeFormat].channels == 0) {
- /* All channels supported. */
- deviceInfo.minChannels = MA_MIN_CHANNELS;
- deviceInfo.maxChannels = MA_MAX_CHANNELS;
- } else {
- if (deviceInfo.minChannels > deviceInfo.nativeDataFormats[iNativeFormat].channels) {
- deviceInfo.minChannels = deviceInfo.nativeDataFormats[iNativeFormat].channels;
- }
- if (deviceInfo.maxChannels < deviceInfo.nativeDataFormats[iNativeFormat].channels) {
- deviceInfo.maxChannels = deviceInfo.nativeDataFormats[iNativeFormat].channels;
- }
+ case 7:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_CENTER;
+ case 2: return MA_CHANNEL_FRONT_RIGHT;
+ case 3: return MA_CHANNEL_SIDE_LEFT;
+ case 4: return MA_CHANNEL_SIDE_RIGHT;
+ case 5: return MA_CHANNEL_BACK_CENTER;
+ case 6: return MA_CHANNEL_LFE;
}
+ } break;
- /* Sample rate. */
- if (deviceInfo.nativeDataFormats[iNativeFormat].sampleRate == 0) {
- /* All sample rates supported. */
- deviceInfo.minSampleRate = (ma_uint32)ma_standard_sample_rate_min;
- deviceInfo.maxSampleRate = (ma_uint32)ma_standard_sample_rate_max;
- } else {
- if (deviceInfo.minSampleRate > deviceInfo.nativeDataFormats[iNativeFormat].sampleRate) {
- deviceInfo.minSampleRate = deviceInfo.nativeDataFormats[iNativeFormat].sampleRate;
- }
- if (deviceInfo.maxSampleRate < deviceInfo.nativeDataFormats[iNativeFormat].sampleRate) {
- deviceInfo.maxSampleRate = deviceInfo.nativeDataFormats[iNativeFormat].sampleRate;
- }
+ case 8:
+ default:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_CENTER;
+ case 2: return MA_CHANNEL_FRONT_RIGHT;
+ case 3: return MA_CHANNEL_SIDE_LEFT;
+ case 4: return MA_CHANNEL_SIDE_RIGHT;
+ case 5: return MA_CHANNEL_BACK_LEFT;
+ case 6: return MA_CHANNEL_BACK_RIGHT;
+ case 7: return MA_CHANNEL_LFE;
}
+ } break;
+ }
+
+ if (channelCount > 8) {
+ if (channelIndex < 32) { /* We have 32 AUX channels. */
+ return (ma_channel)(MA_CHANNEL_AUX_0 + (channelIndex - 8));
}
}
-
- /* Clamp ranges. */
- deviceInfo.minChannels = ma_max(deviceInfo.minChannels, MA_MIN_CHANNELS);
- deviceInfo.maxChannels = ma_min(deviceInfo.maxChannels, MA_MAX_CHANNELS);
- deviceInfo.minSampleRate = ma_max(deviceInfo.minSampleRate, (ma_uint32)ma_standard_sample_rate_min);
- deviceInfo.maxSampleRate = ma_min(deviceInfo.maxSampleRate, (ma_uint32)ma_standard_sample_rate_max);
-
- *pDeviceInfo = deviceInfo;
- return result;
+ /* Getting here means we don't know how to map the channel position so just return MA_CHANNEL_NONE. */
+ return MA_CHANNEL_NONE;
}
-MA_API ma_bool32 ma_context_is_loopback_supported(ma_context* pContext)
+static ma_channel ma_channel_map_init_standard_channel_sound4(ma_uint32 channelCount, ma_uint32 channelIndex)
{
- if (pContext == NULL) {
- return MA_FALSE;
- }
+ switch (channelCount)
+ {
+ case 0: return MA_CHANNEL_NONE;
- return ma_is_loopback_supported(pContext->backend);
-}
+ case 1:
+ {
+ return MA_CHANNEL_MONO;
+ } break;
+ case 2:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ }
+ } break;
-MA_API ma_device_config ma_device_config_init(ma_device_type deviceType)
-{
- ma_device_config config;
- MA_ZERO_OBJECT(&config);
- config.deviceType = deviceType;
+ case 3:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_FRONT_CENTER;
+ }
+ } break;
- /* Resampling defaults. We must never use the Speex backend by default because it uses licensed third party code. */
- config.resampling.algorithm = ma_resample_algorithm_linear;
- config.resampling.linear.lpfOrder = ma_min(MA_DEFAULT_RESAMPLER_LPF_ORDER, MA_MAX_FILTER_ORDER);
- config.resampling.speex.quality = 3;
+ case 4:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_BACK_LEFT;
+ case 3: return MA_CHANNEL_BACK_RIGHT;
+ }
+ } break;
- return config;
-}
+ case 5:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_FRONT_CENTER;
+ case 3: return MA_CHANNEL_BACK_LEFT;
+ case 4: return MA_CHANNEL_BACK_RIGHT;
+ }
+ } break;
-MA_API ma_result ma_device_init(ma_context* pContext, const ma_device_config* pConfig, ma_device* pDevice)
-{
- ma_result result;
- ma_device_descriptor descriptorPlayback;
- ma_device_descriptor descriptorCapture;
+ case 6:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_CENTER;
+ case 2: return MA_CHANNEL_FRONT_RIGHT;
+ case 3: return MA_CHANNEL_BACK_LEFT;
+ case 4: return MA_CHANNEL_BACK_RIGHT;
+ case 5: return MA_CHANNEL_LFE;
+ }
+ } break;
- /* The context can be null, in which case we self-manage it. */
- if (pContext == NULL) {
- return ma_device_init_ex(NULL, 0, NULL, pConfig, pDevice);
+ case 7:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_CENTER;
+ case 2: return MA_CHANNEL_FRONT_RIGHT;
+ case 3: return MA_CHANNEL_SIDE_LEFT;
+ case 4: return MA_CHANNEL_SIDE_RIGHT;
+ case 5: return MA_CHANNEL_BACK_CENTER;
+ case 6: return MA_CHANNEL_LFE;
+ }
+ } break;
+
+ case 8:
+ default:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_CENTER;
+ case 2: return MA_CHANNEL_FRONT_RIGHT;
+ case 3: return MA_CHANNEL_SIDE_LEFT;
+ case 4: return MA_CHANNEL_SIDE_RIGHT;
+ case 5: return MA_CHANNEL_BACK_LEFT;
+ case 6: return MA_CHANNEL_BACK_RIGHT;
+ case 7: return MA_CHANNEL_LFE;
+ }
+ } break;
}
- if (pDevice == NULL) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with invalid arguments (pDevice == NULL).", MA_INVALID_ARGS);
+ if (channelCount > 8) {
+ if (channelIndex < 32) { /* We have 32 AUX channels. */
+ return (ma_channel)(MA_CHANNEL_AUX_0 + (channelIndex - 8));
+ }
}
- MA_ZERO_OBJECT(pDevice);
+ /* Getting here means we don't know how to map the channel position so just return MA_CHANNEL_NONE. */
+ return MA_CHANNEL_NONE;
+}
- if (pConfig == NULL) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with invalid arguments (pConfig == NULL).", MA_INVALID_ARGS);
- }
+static ma_channel ma_channel_map_init_standard_channel_sndio(ma_uint32 channelCount, ma_uint32 channelIndex)
+{
+ switch (channelCount)
+ {
+ case 0: return MA_CHANNEL_NONE;
+ case 1:
+ {
+ return MA_CHANNEL_MONO;
+ } break;
- /* Check that we have our callbacks defined. */
- if (pContext->callbacks.onDeviceInit == NULL) {
- return MA_INVALID_OPERATION;
- }
+ case 2:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ }
+ } break;
+
+ case 3: /* No defined, but best guess. */
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_FRONT_CENTER;
+ }
+ } break;
+ case 4:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_BACK_LEFT;
+ case 3: return MA_CHANNEL_BACK_RIGHT;
+ }
+ } break;
- /* Basic config validation. */
- if (pConfig->deviceType != ma_device_type_playback && pConfig->deviceType != ma_device_type_capture && pConfig->deviceType != ma_device_type_duplex && pConfig->deviceType != ma_device_type_loopback) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with an invalid config. Device type is invalid. Make sure the device type has been set in the config.", MA_INVALID_DEVICE_CONFIG);
- }
+ case 5: /* Not defined, but best guess. */
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_BACK_LEFT;
+ case 3: return MA_CHANNEL_BACK_RIGHT;
+ case 4: return MA_CHANNEL_FRONT_CENTER;
+ }
+ } break;
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
- if (pConfig->capture.channels > MA_MAX_CHANNELS) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with an invalid config. Capture channel count cannot exceed 32.", MA_INVALID_DEVICE_CONFIG);
- }
- if (!ma__is_channel_map_valid(pConfig->capture.channelMap, pConfig->capture.channels)) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with invalid config. Capture channel map is invalid.", MA_INVALID_DEVICE_CONFIG);
- }
+ case 6:
+ default:
+ {
+ switch (channelIndex) {
+ case 0: return MA_CHANNEL_FRONT_LEFT;
+ case 1: return MA_CHANNEL_FRONT_RIGHT;
+ case 2: return MA_CHANNEL_BACK_LEFT;
+ case 3: return MA_CHANNEL_BACK_RIGHT;
+ case 4: return MA_CHANNEL_FRONT_CENTER;
+ case 5: return MA_CHANNEL_LFE;
+ }
+ } break;
}
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex || pConfig->deviceType == ma_device_type_loopback) {
- if (pConfig->playback.channels > MA_MAX_CHANNELS) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with an invalid config. Playback channel count cannot exceed 32.", MA_INVALID_DEVICE_CONFIG);
- }
- if (!ma__is_channel_map_valid(pConfig->playback.channelMap, pConfig->playback.channels)) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "ma_device_init() called with invalid config. Playback channel map is invalid.", MA_INVALID_DEVICE_CONFIG);
+ if (channelCount > 6) {
+ if (channelIndex < 32) { /* We have 32 AUX channels. */
+ return (ma_channel)(MA_CHANNEL_AUX_0 + (channelIndex - 6));
}
}
- pDevice->pContext = pContext;
-
- /* Set the user data and log callback ASAP to ensure it is available for the entire initialization process. */
- pDevice->pUserData = pConfig->pUserData;
- pDevice->onData = pConfig->dataCallback;
- pDevice->onStop = pConfig->stopCallback;
+ /* Getting here means we don't know how to map the channel position so just return MA_CHANNEL_NONE. */
+ return MA_CHANNEL_NONE;
+}
- if (((ma_uintptr)pDevice % sizeof(pDevice)) != 0) {
- if (pContext->logCallback) {
- pContext->logCallback(pContext, pDevice, MA_LOG_LEVEL_WARNING, "WARNING: ma_device_init() called for a device that is not properly aligned. Thread safety is not supported.");
- }
- }
- if (pConfig->playback.pDeviceID != NULL) {
- MA_COPY_MEMORY(&pDevice->playback.id, pConfig->playback.pDeviceID, sizeof(pDevice->playback.id));
+static ma_channel ma_channel_map_init_standard_channel(ma_standard_channel_map standardChannelMap, ma_uint32 channelCount, ma_uint32 channelIndex)
+{
+ if (channelCount == 0 || channelIndex >= channelCount) {
+ return MA_CHANNEL_NONE;
}
- if (pConfig->capture.pDeviceID != NULL) {
- MA_COPY_MEMORY(&pDevice->capture.id, pConfig->capture.pDeviceID, sizeof(pDevice->capture.id));
- }
+ switch (standardChannelMap)
+ {
+ case ma_standard_channel_map_alsa:
+ {
+ return ma_channel_map_init_standard_channel_alsa(channelCount, channelIndex);
+ } break;
- pDevice->noPreZeroedOutputBuffer = pConfig->noPreZeroedOutputBuffer;
- pDevice->noClip = pConfig->noClip;
- pDevice->masterVolumeFactor = 1;
+ case ma_standard_channel_map_rfc3551:
+ {
+ return ma_channel_map_init_standard_channel_rfc3551(channelCount, channelIndex);
+ } break;
- pDevice->type = pConfig->deviceType;
- pDevice->sampleRate = pConfig->sampleRate;
- pDevice->resampling.algorithm = pConfig->resampling.algorithm;
- pDevice->resampling.linear.lpfOrder = pConfig->resampling.linear.lpfOrder;
- pDevice->resampling.speex.quality = pConfig->resampling.speex.quality;
+ case ma_standard_channel_map_flac:
+ {
+ return ma_channel_map_init_standard_channel_flac(channelCount, channelIndex);
+ } break;
- pDevice->capture.shareMode = pConfig->capture.shareMode;
- pDevice->capture.format = pConfig->capture.format;
- pDevice->capture.channels = pConfig->capture.channels;
- ma_channel_map_copy(pDevice->capture.channelMap, pConfig->capture.channelMap, pConfig->capture.channels);
- pDevice->capture.channelMixMode = pConfig->capture.channelMixMode;
+ case ma_standard_channel_map_vorbis:
+ {
+ return ma_channel_map_init_standard_channel_vorbis(channelCount, channelIndex);
+ } break;
- pDevice->playback.shareMode = pConfig->playback.shareMode;
- pDevice->playback.format = pConfig->playback.format;
- pDevice->playback.channels = pConfig->playback.channels;
- ma_channel_map_copy(pDevice->playback.channelMap, pConfig->playback.channelMap, pConfig->playback.channels);
- pDevice->playback.channelMixMode = pConfig->playback.channelMixMode;
+ case ma_standard_channel_map_sound4:
+ {
+ return ma_channel_map_init_standard_channel_sound4(channelCount, channelIndex);
+ } break;
+ case ma_standard_channel_map_sndio:
+ {
+ return ma_channel_map_init_standard_channel_sndio(channelCount, channelIndex);
+ } break;
- result = ma_mutex_init(&pDevice->startStopLock);
- if (result != MA_SUCCESS) {
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "Failed to create mutex.", result);
+ case ma_standard_channel_map_microsoft: /* Also default. */
+ /*case ma_standard_channel_map_default;*/
+ default:
+ {
+ return ma_channel_map_init_standard_channel_microsoft(channelCount, channelIndex);
+ } break;
}
+}
- /*
- When the device is started, the worker thread is the one that does the actual startup of the backend device. We
- use a semaphore to wait for the background thread to finish the work. The same applies for stopping the device.
+MA_API void ma_channel_map_init_standard(ma_standard_channel_map standardChannelMap, ma_channel* pChannelMap, size_t channelMapCap, ma_uint32 channels)
+{
+ ma_uint32 iChannel;
- Each of these semaphores is released internally by the worker thread when the work is completed. The start
- semaphore is also used to wake up the worker thread.
- */
- result = ma_event_init(&pDevice->wakeupEvent);
- if (result != MA_SUCCESS) {
- ma_mutex_uninit(&pDevice->startStopLock);
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "Failed to create worker thread wakeup event.", result);
+ if (pChannelMap == NULL || channelMapCap == 0 || channels == 0) {
+ return;
}
- result = ma_event_init(&pDevice->startEvent);
- if (result != MA_SUCCESS) {
- ma_event_uninit(&pDevice->wakeupEvent);
- ma_mutex_uninit(&pDevice->startStopLock);
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "Failed to create worker thread start event.", result);
- }
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ if (channelMapCap == 0) {
+ break; /* Ran out of room. */
+ }
- result = ma_event_init(&pDevice->stopEvent);
- if (result != MA_SUCCESS) {
- ma_event_uninit(&pDevice->startEvent);
- ma_event_uninit(&pDevice->wakeupEvent);
- ma_mutex_uninit(&pDevice->startStopLock);
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "Failed to create worker thread stop event.", result);
+ pChannelMap[0] = ma_channel_map_init_standard_channel(standardChannelMap, channels, iChannel);
+ pChannelMap += 1;
+ channelMapCap -= 1;
}
+}
-
- MA_ZERO_OBJECT(&descriptorPlayback);
- descriptorPlayback.pDeviceID = pConfig->playback.pDeviceID;
- descriptorPlayback.shareMode = pConfig->playback.shareMode;
- descriptorPlayback.format = pConfig->playback.format;
- descriptorPlayback.channels = pConfig->playback.channels;
- descriptorPlayback.sampleRate = pConfig->sampleRate;
- ma_channel_map_copy(descriptorPlayback.channelMap, pConfig->playback.channelMap, pConfig->playback.channels);
- descriptorPlayback.periodSizeInFrames = pConfig->periodSizeInFrames;
- descriptorPlayback.periodSizeInMilliseconds = pConfig->periodSizeInMilliseconds;
- descriptorPlayback.periodCount = pConfig->periods;
-
- if (descriptorPlayback.periodCount == 0) {
- descriptorPlayback.periodCount = MA_DEFAULT_PERIODS;
+MA_API void ma_channel_map_copy(ma_channel* pOut, const ma_channel* pIn, ma_uint32 channels)
+{
+ if (pOut != NULL && pIn != NULL && channels > 0) {
+ MA_COPY_MEMORY(pOut, pIn, sizeof(*pOut) * channels);
}
+}
-
- MA_ZERO_OBJECT(&descriptorCapture);
- descriptorCapture.pDeviceID = pConfig->capture.pDeviceID;
- descriptorCapture.shareMode = pConfig->capture.shareMode;
- descriptorCapture.format = pConfig->capture.format;
- descriptorCapture.channels = pConfig->capture.channels;
- descriptorCapture.sampleRate = pConfig->sampleRate;
- ma_channel_map_copy(descriptorCapture.channelMap, pConfig->capture.channelMap, pConfig->capture.channels);
- descriptorCapture.periodSizeInFrames = pConfig->periodSizeInFrames;
- descriptorCapture.periodSizeInMilliseconds = pConfig->periodSizeInMilliseconds;
- descriptorCapture.periodCount = pConfig->periods;
-
- if (descriptorCapture.periodCount == 0) {
- descriptorCapture.periodCount = MA_DEFAULT_PERIODS;
+MA_API void ma_channel_map_copy_or_default(ma_channel* pOut, size_t channelMapCapOut, const ma_channel* pIn, ma_uint32 channels)
+{
+ if (pOut == NULL || channels == 0) {
+ return;
}
-
- result = pContext->callbacks.onDeviceInit(pDevice, pConfig, &descriptorPlayback, &descriptorCapture);
- if (result != MA_SUCCESS) {
- ma_event_uninit(&pDevice->startEvent);
- ma_event_uninit(&pDevice->wakeupEvent);
- ma_mutex_uninit(&pDevice->startStopLock);
- return result;
+ if (pIn != NULL) {
+ ma_channel_map_copy(pOut, pIn, channels);
+ } else {
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pOut, channelMapCapOut, channels);
}
+}
-
- /*
- On output the descriptors will contain the *actual* data format of the device. We need this to know how to convert the data between
- the requested format and the internal format.
- */
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex || pConfig->deviceType == ma_device_type_loopback) {
- if (!ma_device_descriptor_is_valid(&descriptorCapture)) {
- ma_device_uninit(pDevice);
- return MA_INVALID_ARGS;
- }
-
- pDevice->capture.internalFormat = descriptorCapture.format;
- pDevice->capture.internalChannels = descriptorCapture.channels;
- pDevice->capture.internalSampleRate = descriptorCapture.sampleRate;
- ma_channel_map_copy(pDevice->capture.internalChannelMap, descriptorCapture.channelMap, descriptorCapture.channels);
- pDevice->capture.internalPeriodSizeInFrames = descriptorCapture.periodSizeInFrames;
- pDevice->capture.internalPeriods = descriptorCapture.periodCount;
-
- if (pDevice->capture.internalPeriodSizeInFrames == 0) {
- pDevice->capture.internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(descriptorCapture.periodSizeInMilliseconds, descriptorCapture.sampleRate);
- }
+MA_API ma_bool32 ma_channel_map_is_valid(const ma_channel* pChannelMap, ma_uint32 channels)
+{
+ /* A channel count of 0 is invalid. */
+ if (channels == 0) {
+ return MA_FALSE;
}
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- if (!ma_device_descriptor_is_valid(&descriptorPlayback)) {
- ma_device_uninit(pDevice);
- return MA_INVALID_ARGS;
- }
-
- pDevice->playback.internalFormat = descriptorPlayback.format;
- pDevice->playback.internalChannels = descriptorPlayback.channels;
- pDevice->playback.internalSampleRate = descriptorPlayback.sampleRate;
- ma_channel_map_copy(pDevice->playback.internalChannelMap, descriptorPlayback.channelMap, descriptorPlayback.channels);
- pDevice->playback.internalPeriodSizeInFrames = descriptorPlayback.periodSizeInFrames;
- pDevice->playback.internalPeriods = descriptorPlayback.periodCount;
-
- if (pDevice->playback.internalPeriodSizeInFrames == 0) {
- pDevice->playback.internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(descriptorPlayback.periodSizeInMilliseconds, descriptorPlayback.sampleRate);
+ /* It does not make sense to have a mono channel when there is more than 1 channel. */
+ if (channels > 1) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < channels; ++iChannel) {
+ if (ma_channel_map_get_channel(pChannelMap, channels, iChannel) == MA_CHANNEL_MONO) {
+ return MA_FALSE;
+ }
}
}
+ return MA_TRUE;
+}
- /*
- The name of the device can be retrieved from device info. This may be temporary and replaced with a `ma_device_get_info(pDevice, deviceType)` instead.
- For loopback devices, we need to retrieve the name of the playback device.
- */
- {
- ma_device_info deviceInfo;
+MA_API ma_bool32 ma_channel_map_is_equal(const ma_channel* pChannelMapA, const ma_channel* pChannelMapB, ma_uint32 channels)
+{
+ ma_uint32 iChannel;
- if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex || pConfig->deviceType == ma_device_type_loopback) {
- result = ma_context_get_device_info(pContext, (pConfig->deviceType == ma_device_type_loopback) ? ma_device_type_playback : ma_device_type_capture, descriptorCapture.pDeviceID, descriptorCapture.shareMode, &deviceInfo);
- if (result == MA_SUCCESS) {
- ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), deviceInfo.name, (size_t)-1);
- } else {
- /* We failed to retrieve the device info. Fall back to a default name. */
- if (descriptorCapture.pDeviceID == NULL) {
- ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
- } else {
- ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), "Capture Device", (size_t)-1);
- }
- }
- }
+ if (pChannelMapA == pChannelMapB) {
+ return MA_TRUE;
+ }
- if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
- result = ma_context_get_device_info(pContext, ma_device_type_playback, descriptorPlayback.pDeviceID, descriptorPlayback.shareMode, &deviceInfo);
- if (result == MA_SUCCESS) {
- ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), deviceInfo.name, (size_t)-1);
- } else {
- /* We failed to retrieve the device info. Fall back to a default name. */
- if (descriptorPlayback.pDeviceID == NULL) {
- ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
- } else {
- ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), "Playback Device", (size_t)-1);
- }
- }
+ for (iChannel = 0; iChannel < channels; ++iChannel) {
+ if (ma_channel_map_get_channel(pChannelMapA, channels, iChannel) != ma_channel_map_get_channel(pChannelMapB, channels, iChannel)) {
+ return MA_FALSE;
}
}
+ return MA_TRUE;
+}
- ma_device__post_init_setup(pDevice, pConfig->deviceType);
-
-
- /* Some backends don't require the worker thread. */
- if (!ma_context_is_backend_asynchronous(pContext)) {
- /* The worker thread. */
- result = ma_thread_create(&pDevice->thread, pContext->threadPriority, pContext->threadStackSize, ma_worker_thread, pDevice, &pContext->allocationCallbacks);
- if (result != MA_SUCCESS) {
- ma_device_uninit(pDevice);
- return ma_context_post_error(pContext, NULL, MA_LOG_LEVEL_ERROR, "Failed to create worker thread.", result);
- }
-
- /* Wait for the worker thread to put the device into it's stopped state for real. */
- ma_event_wait(&pDevice->stopEvent);
- MA_ASSERT(ma_device_get_state(pDevice) == MA_STATE_STOPPED);
- } else {
- /*
- If the backend is asynchronous and the device is duplex, we'll need an intermediary ring buffer. Note that this needs to be done
- after ma_device__post_init_setup().
- */
- if (ma_context_is_backend_asynchronous(pContext)) {
- if (pConfig->deviceType == ma_device_type_duplex) {
- result = ma_duplex_rb_init(pDevice->capture.format, pDevice->capture.channels, pDevice->sampleRate, pDevice->capture.internalSampleRate, pDevice->capture.internalPeriodSizeInFrames, &pDevice->pContext->allocationCallbacks, &pDevice->duplexRB);
- if (result != MA_SUCCESS) {
- ma_device_uninit(pDevice);
- return result;
- }
- }
- }
+MA_API ma_bool32 ma_channel_map_is_blank(const ma_channel* pChannelMap, ma_uint32 channels)
+{
+ ma_uint32 iChannel;
- ma_device__set_state(pDevice, MA_STATE_STOPPED);
+ /* A null channel map is equivalent to the default channel map. */
+ if (pChannelMap == NULL) {
+ return MA_FALSE;
}
-
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[%s]\n", ma_get_backend_name(pDevice->pContext->backend));
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " %s (%s)\n", pDevice->capture.name, "Capture");
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Format: %s -> %s\n", ma_get_format_name(pDevice->capture.internalFormat), ma_get_format_name(pDevice->capture.format));
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Channels: %d -> %d\n", pDevice->capture.internalChannels, pDevice->capture.channels);
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Sample Rate: %d -> %d\n", pDevice->capture.internalSampleRate, pDevice->sampleRate);
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Buffer Size: %d*%d (%d)\n", pDevice->capture.internalPeriodSizeInFrames, pDevice->capture.internalPeriods, (pDevice->capture.internalPeriodSizeInFrames * pDevice->capture.internalPeriods));
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Conversion:\n");
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Pre Format Conversion: %s\n", pDevice->capture.converter.hasPreFormatConversion ? "YES" : "NO");
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Post Format Conversion: %s\n", pDevice->capture.converter.hasPostFormatConversion ? "YES" : "NO");
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Channel Routing: %s\n", pDevice->capture.converter.hasChannelConverter ? "YES" : "NO");
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Resampling: %s\n", pDevice->capture.converter.hasResampler ? "YES" : "NO");
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Passthrough: %s\n", pDevice->capture.converter.isPassthrough ? "YES" : "NO");
- }
- if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " %s (%s)\n", pDevice->playback.name, "Playback");
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Format: %s -> %s\n", ma_get_format_name(pDevice->playback.format), ma_get_format_name(pDevice->playback.internalFormat));
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Channels: %d -> %d\n", pDevice->playback.channels, pDevice->playback.internalChannels);
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Sample Rate: %d -> %d\n", pDevice->sampleRate, pDevice->playback.internalSampleRate);
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Buffer Size: %d*%d (%d)\n", pDevice->playback.internalPeriodSizeInFrames, pDevice->playback.internalPeriods, (pDevice->playback.internalPeriodSizeInFrames * pDevice->playback.internalPeriods));
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Conversion:\n");
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Pre Format Conversion: %s\n", pDevice->playback.converter.hasPreFormatConversion ? "YES" : "NO");
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Post Format Conversion: %s\n", pDevice->playback.converter.hasPostFormatConversion ? "YES" : "NO");
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Channel Routing: %s\n", pDevice->playback.converter.hasChannelConverter ? "YES" : "NO");
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Resampling: %s\n", pDevice->playback.converter.hasResampler ? "YES" : "NO");
- ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Passthrough: %s\n", pDevice->playback.converter.isPassthrough ? "YES" : "NO");
+ for (iChannel = 0; iChannel < channels; ++iChannel) {
+ if (pChannelMap[iChannel] != MA_CHANNEL_NONE) {
+ return MA_FALSE;
+ }
}
- MA_ASSERT(ma_device_get_state(pDevice) == MA_STATE_STOPPED);
- return MA_SUCCESS;
+ return MA_TRUE;
}
-MA_API ma_result ma_device_init_ex(const ma_backend backends[], ma_uint32 backendCount, const ma_context_config* pContextConfig, const ma_device_config* pConfig, ma_device* pDevice)
+MA_API ma_bool32 ma_channel_map_contains_channel_position(ma_uint32 channels, const ma_channel* pChannelMap, ma_channel channelPosition)
{
- ma_result result;
- ma_context* pContext;
- ma_backend defaultBackends[ma_backend_null+1];
- ma_uint32 iBackend;
- ma_backend* pBackendsToIterate;
- ma_uint32 backendsToIterateCount;
- ma_allocation_callbacks allocationCallbacks;
-
- if (pConfig == NULL) {
- return MA_INVALID_ARGS;
- }
+ ma_uint32 iChannel;
- if (pContextConfig != NULL) {
- result = ma_allocation_callbacks_init_copy(&allocationCallbacks, &pContextConfig->allocationCallbacks);
- if (result != MA_SUCCESS) {
- return result;
+ for (iChannel = 0; iChannel < channels; ++iChannel) {
+ if (ma_channel_map_get_channel(pChannelMap, channels, iChannel) == channelPosition) {
+ return MA_TRUE;
}
- } else {
- allocationCallbacks = ma_allocation_callbacks_init_default();
}
+ return MA_FALSE;
+}
- pContext = (ma_context*)ma__malloc_from_callbacks(sizeof(*pContext), &allocationCallbacks);
- if (pContext == NULL) {
- return MA_OUT_OF_MEMORY;
- }
- for (iBackend = 0; iBackend <= ma_backend_null; ++iBackend) {
- defaultBackends[iBackend] = (ma_backend)iBackend;
- }
- pBackendsToIterate = (ma_backend*)backends;
- backendsToIterateCount = backendCount;
- if (pBackendsToIterate == NULL) {
- pBackendsToIterate = (ma_backend*)defaultBackends;
- backendsToIterateCount = ma_countof(defaultBackends);
- }
+/**************************************************************************************************************************************************************
- result = MA_NO_BACKEND;
+Conversion Helpers
- for (iBackend = 0; iBackend < backendsToIterateCount; ++iBackend) {
- result = ma_context_init(&pBackendsToIterate[iBackend], 1, pContextConfig, pContext);
- if (result == MA_SUCCESS) {
- result = ma_device_init(pContext, pConfig, pDevice);
- if (result == MA_SUCCESS) {
- break; /* Success. */
- } else {
- ma_context_uninit(pContext); /* Failure. */
- }
- }
- }
+**************************************************************************************************************************************************************/
+MA_API ma_uint64 ma_convert_frames(void* pOut, ma_uint64 frameCountOut, ma_format formatOut, ma_uint32 channelsOut, ma_uint32 sampleRateOut, const void* pIn, ma_uint64 frameCountIn, ma_format formatIn, ma_uint32 channelsIn, ma_uint32 sampleRateIn)
+{
+ ma_data_converter_config config;
- if (result != MA_SUCCESS) {
- ma__free_from_callbacks(pContext, &allocationCallbacks);
- return result;
- }
+ config = ma_data_converter_config_init(formatIn, formatOut, channelsIn, channelsOut, sampleRateIn, sampleRateOut);
+ config.resampling.linear.lpfOrder = ma_min(MA_DEFAULT_RESAMPLER_LPF_ORDER, MA_MAX_FILTER_ORDER);
- pDevice->isOwnerOfContext = MA_TRUE;
- return result;
+ return ma_convert_frames_ex(pOut, frameCountOut, pIn, frameCountIn, &config);
}
-MA_API void ma_device_uninit(ma_device* pDevice)
+MA_API ma_uint64 ma_convert_frames_ex(void* pOut, ma_uint64 frameCountOut, const void* pIn, ma_uint64 frameCountIn, const ma_data_converter_config* pConfig)
{
- if (!ma_device__is_initialized(pDevice)) {
- return;
- }
+ ma_result result;
+ ma_data_converter converter;
- /* Make sure the device is stopped first. The backends will probably handle this naturally, but I like to do it explicitly for my own sanity. */
- if (ma_device_is_started(pDevice)) {
- ma_device_stop(pDevice);
+ if (frameCountIn == 0 || pConfig == NULL) {
+ return 0;
}
- /* Putting the device into an uninitialized state will make the worker thread return. */
- ma_device__set_state(pDevice, MA_STATE_UNINITIALIZED);
-
- /* Wake up the worker thread and wait for it to properly terminate. */
- if (!ma_context_is_backend_asynchronous(pDevice->pContext)) {
- ma_event_signal(&pDevice->wakeupEvent);
- ma_thread_wait(&pDevice->thread);
+ result = ma_data_converter_init(pConfig, NULL, &converter);
+ if (result != MA_SUCCESS) {
+ return 0; /* Failed to initialize the data converter. */
}
- if (pDevice->pContext->callbacks.onDeviceUninit != NULL) {
- pDevice->pContext->callbacks.onDeviceUninit(pDevice);
- }
+ if (pOut == NULL) {
+ result = ma_data_converter_get_expected_output_frame_count(&converter, frameCountIn, &frameCountOut);
+ if (result != MA_SUCCESS) {
+ if (result == MA_NOT_IMPLEMENTED) {
+ /* No way to calculate the number of frames, so we'll need to brute force it and loop. */
+ frameCountOut = 0;
+ while (frameCountIn > 0) {
+ ma_uint64 framesProcessedIn = frameCountIn;
+ ma_uint64 framesProcessedOut = 0xFFFFFFFF;
- ma_event_uninit(&pDevice->stopEvent);
- ma_event_uninit(&pDevice->startEvent);
- ma_event_uninit(&pDevice->wakeupEvent);
- ma_mutex_uninit(&pDevice->startStopLock);
+ result = ma_data_converter_process_pcm_frames(&converter, pIn, &framesProcessedIn, NULL, &framesProcessedOut);
+ if (result != MA_SUCCESS) {
+ break;
+ }
- if (ma_context_is_backend_asynchronous(pDevice->pContext)) {
- if (pDevice->type == ma_device_type_duplex) {
- ma_duplex_rb_uninit(&pDevice->duplexRB);
+ frameCountIn -= framesProcessedIn;
+ }
+ }
+ }
+ } else {
+ result = ma_data_converter_process_pcm_frames(&converter, pIn, &frameCountIn, pOut, &frameCountOut);
+ if (result != MA_SUCCESS) {
+ frameCountOut = 0;
}
}
- if (pDevice->isOwnerOfContext) {
- ma_allocation_callbacks allocationCallbacks = pDevice->pContext->allocationCallbacks;
+ ma_data_converter_uninit(&converter, NULL);
+ return frameCountOut;
+}
- ma_context_uninit(pDevice->pContext);
- ma__free_from_callbacks(pDevice->pContext, &allocationCallbacks);
- }
- MA_ZERO_OBJECT(pDevice);
+/**************************************************************************************************************************************************************
+
+Ring Buffer
+
+**************************************************************************************************************************************************************/
+static MA_INLINE ma_uint32 ma_rb__extract_offset_in_bytes(ma_uint32 encodedOffset)
+{
+ return encodedOffset & 0x7FFFFFFF;
}
-MA_API ma_context* ma_device_get_context(ma_device* pDevice)
+static MA_INLINE ma_uint32 ma_rb__extract_offset_loop_flag(ma_uint32 encodedOffset)
{
- if (pDevice == NULL) {
- return NULL;
- }
+ return encodedOffset & 0x80000000;
+}
- return pDevice->pContext;
+static MA_INLINE void* ma_rb__get_read_ptr(ma_rb* pRB)
+{
+ MA_ASSERT(pRB != NULL);
+ return ma_offset_ptr(pRB->pBuffer, ma_rb__extract_offset_in_bytes(c89atomic_load_32(&pRB->encodedReadOffset)));
}
-MA_API ma_log* ma_device_get_log(ma_device* pDevice)
+static MA_INLINE void* ma_rb__get_write_ptr(ma_rb* pRB)
{
- return ma_context_get_log(ma_device_get_context(pDevice));
+ MA_ASSERT(pRB != NULL);
+ return ma_offset_ptr(pRB->pBuffer, ma_rb__extract_offset_in_bytes(c89atomic_load_32(&pRB->encodedWriteOffset)));
}
-MA_API ma_result ma_device_start(ma_device* pDevice)
+static MA_INLINE ma_uint32 ma_rb__construct_offset(ma_uint32 offsetInBytes, ma_uint32 offsetLoopFlag)
+{
+ return offsetLoopFlag | offsetInBytes;
+}
+
+static MA_INLINE void ma_rb__deconstruct_offset(ma_uint32 encodedOffset, ma_uint32* pOffsetInBytes, ma_uint32* pOffsetLoopFlag)
+{
+ MA_ASSERT(pOffsetInBytes != NULL);
+ MA_ASSERT(pOffsetLoopFlag != NULL);
+
+ *pOffsetInBytes = ma_rb__extract_offset_in_bytes(encodedOffset);
+ *pOffsetLoopFlag = ma_rb__extract_offset_loop_flag(encodedOffset);
+}
+
+
+MA_API ma_result ma_rb_init_ex(size_t subbufferSizeInBytes, size_t subbufferCount, size_t subbufferStrideInBytes, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_rb* pRB)
{
ma_result result;
+ const ma_uint32 maxSubBufferSize = 0x7FFFFFFF - (MA_SIMD_ALIGNMENT-1);
- if (pDevice == NULL) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "ma_device_start() called with invalid arguments (pDevice == NULL).", MA_INVALID_ARGS);
+ if (pRB == NULL) {
+ return MA_INVALID_ARGS;
}
- if (ma_device_get_state(pDevice) == MA_STATE_UNINITIALIZED) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "ma_device_start() called for an uninitialized device.", MA_DEVICE_NOT_INITIALIZED);
+ if (subbufferSizeInBytes == 0 || subbufferCount == 0) {
+ return MA_INVALID_ARGS;
}
- if (ma_device_get_state(pDevice) == MA_STATE_STARTED) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_WARNING, "ma_device_start() called when the device is already started.", MA_INVALID_OPERATION); /* Already started. Returning an error to let the application know because it probably means they're doing something wrong. */
+ if (subbufferSizeInBytes > maxSubBufferSize) {
+ return MA_INVALID_ARGS; /* Maximum buffer size is ~2GB. The most significant bit is a flag for use internally. */
}
- ma_mutex_lock(&pDevice->startStopLock);
- {
- /* Starting and stopping are wrapped in a mutex which means we can assert that the device is in a stopped or paused state. */
- MA_ASSERT(ma_device_get_state(pDevice) == MA_STATE_STOPPED);
- ma_device__set_state(pDevice, MA_STATE_STARTING);
+ MA_ZERO_OBJECT(pRB);
- /* Asynchronous backends need to be handled differently. */
- if (ma_context_is_backend_asynchronous(pDevice->pContext)) {
- if (pDevice->pContext->callbacks.onDeviceStart != NULL) {
- result = pDevice->pContext->callbacks.onDeviceStart(pDevice);
- } else {
- result = MA_INVALID_OPERATION;
- }
+ result = ma_allocation_callbacks_init_copy(&pRB->allocationCallbacks, pAllocationCallbacks);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- if (result == MA_SUCCESS) {
- ma_device__set_state(pDevice, MA_STATE_STARTED);
- }
- } else {
- /*
- Synchronous backends are started by signaling an event that's being waited on in the worker thread. We first wake up the
- thread and then wait for the start event.
- */
- ma_event_signal(&pDevice->wakeupEvent);
+ pRB->subbufferSizeInBytes = (ma_uint32)subbufferSizeInBytes;
+ pRB->subbufferCount = (ma_uint32)subbufferCount;
- /*
- Wait for the worker thread to finish starting the device. Note that the worker thread will be the one who puts the device
- into the started state. Don't call ma_device__set_state() here.
- */
- ma_event_wait(&pDevice->startEvent);
- result = pDevice->workResult;
- }
+ if (pOptionalPreallocatedBuffer != NULL) {
+ pRB->subbufferStrideInBytes = (ma_uint32)subbufferStrideInBytes;
+ pRB->pBuffer = pOptionalPreallocatedBuffer;
+ } else {
+ size_t bufferSizeInBytes;
- /* We changed the state from stopped to started, so if we failed, make sure we put the state back to stopped. */
- if (result != MA_SUCCESS) {
- ma_device__set_state(pDevice, MA_STATE_STOPPED);
+ /*
+ Here is where we allocate our own buffer. We always want to align this to MA_SIMD_ALIGNMENT for future SIMD optimization opportunity. To do this
+ we need to make sure the stride is a multiple of MA_SIMD_ALIGNMENT.
+ */
+ pRB->subbufferStrideInBytes = (pRB->subbufferSizeInBytes + (MA_SIMD_ALIGNMENT-1)) & ~MA_SIMD_ALIGNMENT;
+
+ bufferSizeInBytes = (size_t)pRB->subbufferCount*pRB->subbufferStrideInBytes;
+ pRB->pBuffer = ma_aligned_malloc(bufferSizeInBytes, MA_SIMD_ALIGNMENT, &pRB->allocationCallbacks);
+ if (pRB->pBuffer == NULL) {
+ return MA_OUT_OF_MEMORY;
}
+
+ MA_ZERO_MEMORY(pRB->pBuffer, bufferSizeInBytes);
+ pRB->ownsBuffer = MA_TRUE;
}
- ma_mutex_unlock(&pDevice->startStopLock);
- return result;
+ return MA_SUCCESS;
}
-MA_API ma_result ma_device_stop(ma_device* pDevice)
+MA_API ma_result ma_rb_init(size_t bufferSizeInBytes, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_rb* pRB)
{
- ma_result result;
+ return ma_rb_init_ex(bufferSizeInBytes, 1, 0, pOptionalPreallocatedBuffer, pAllocationCallbacks, pRB);
+}
- if (pDevice == NULL) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "ma_device_stop() called with invalid arguments (pDevice == NULL).", MA_INVALID_ARGS);
+MA_API void ma_rb_uninit(ma_rb* pRB)
+{
+ if (pRB == NULL) {
+ return;
}
- if (ma_device_get_state(pDevice) == MA_STATE_UNINITIALIZED) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_ERROR, "ma_device_stop() called for an uninitialized device.", MA_DEVICE_NOT_INITIALIZED);
+ if (pRB->ownsBuffer) {
+ ma_aligned_free(pRB->pBuffer, &pRB->allocationCallbacks);
}
+}
- if (ma_device_get_state(pDevice) == MA_STATE_STOPPED) {
- return ma_post_error(pDevice, MA_LOG_LEVEL_WARNING, "ma_device_stop() called when the device is already stopped.", MA_INVALID_OPERATION); /* Already stopped. Returning an error to let the application know because it probably means they're doing something wrong. */
+MA_API void ma_rb_reset(ma_rb* pRB)
+{
+ if (pRB == NULL) {
+ return;
}
- ma_mutex_lock(&pDevice->startStopLock);
- {
- /* Starting and stopping are wrapped in a mutex which means we can assert that the device is in a started or paused state. */
- MA_ASSERT(ma_device_get_state(pDevice) == MA_STATE_STARTED);
+ c89atomic_exchange_32(&pRB->encodedReadOffset, 0);
+ c89atomic_exchange_32(&pRB->encodedWriteOffset, 0);
+}
- ma_device__set_state(pDevice, MA_STATE_STOPPING);
+MA_API ma_result ma_rb_acquire_read(ma_rb* pRB, size_t* pSizeInBytes, void** ppBufferOut)
+{
+ ma_uint32 writeOffset;
+ ma_uint32 writeOffsetInBytes;
+ ma_uint32 writeOffsetLoopFlag;
+ ma_uint32 readOffset;
+ ma_uint32 readOffsetInBytes;
+ ma_uint32 readOffsetLoopFlag;
+ size_t bytesAvailable;
+ size_t bytesRequested;
- /* Asynchronous backends need to be handled differently. */
- if (ma_context_is_backend_asynchronous(pDevice->pContext)) {
- /* Asynchronous backends must have a stop operation. */
- if (pDevice->pContext->callbacks.onDeviceStop != NULL) {
- result = pDevice->pContext->callbacks.onDeviceStop(pDevice);
- } else {
- result = MA_INVALID_OPERATION;
- }
+ if (pRB == NULL || pSizeInBytes == NULL || ppBufferOut == NULL) {
+ return MA_INVALID_ARGS;
+ }
- ma_device__set_state(pDevice, MA_STATE_STOPPED);
- } else {
- /*
- Synchronous backends. The stop callback is always called from the worker thread. Do not call the stop callback here. If
- the backend is implementing it's own audio thread loop we'll need to wake it up if required. Note that we need to make
- sure the state of the device is *not* playing right now, which it shouldn't be since we set it above. This is super
- important though, so I'm asserting it here as well for extra safety in case we accidentally change something later.
- */
- MA_ASSERT(ma_device_get_state(pDevice) != MA_STATE_STARTED);
+ /* The returned buffer should never move ahead of the write pointer. */
+ writeOffset = c89atomic_load_32(&pRB->encodedWriteOffset);
+ ma_rb__deconstruct_offset(writeOffset, &writeOffsetInBytes, &writeOffsetLoopFlag);
- if (pDevice->pContext->callbacks.onDeviceDataLoopWakeup != NULL) {
- pDevice->pContext->callbacks.onDeviceDataLoopWakeup(pDevice);
- }
+ readOffset = c89atomic_load_32(&pRB->encodedReadOffset);
+ ma_rb__deconstruct_offset(readOffset, &readOffsetInBytes, &readOffsetLoopFlag);
- /*
- We need to wait for the worker thread to become available for work before returning. Note that the worker thread will be
- the one who puts the device into the stopped state. Don't call ma_device__set_state() here.
- */
- ma_event_wait(&pDevice->stopEvent);
- result = MA_SUCCESS;
- }
+ /*
+ The number of bytes available depends on whether or not the read and write pointers are on the same loop iteration. If so, we
+ can only read up to the write pointer. If not, we can only read up to the end of the buffer.
+ */
+ if (readOffsetLoopFlag == writeOffsetLoopFlag) {
+ bytesAvailable = writeOffsetInBytes - readOffsetInBytes;
+ } else {
+ bytesAvailable = pRB->subbufferSizeInBytes - readOffsetInBytes;
}
- ma_mutex_unlock(&pDevice->startStopLock);
- return result;
-}
+ bytesRequested = *pSizeInBytes;
+ if (bytesRequested > bytesAvailable) {
+ bytesRequested = bytesAvailable;
+ }
-MA_API ma_bool32 ma_device_is_started(const ma_device* pDevice)
-{
- return ma_device_get_state(pDevice) == MA_STATE_STARTED;
+ *pSizeInBytes = bytesRequested;
+ (*ppBufferOut) = ma_rb__get_read_ptr(pRB);
+
+ return MA_SUCCESS;
}
-MA_API ma_uint32 ma_device_get_state(const ma_device* pDevice)
+MA_API ma_result ma_rb_commit_read(ma_rb* pRB, size_t sizeInBytes)
{
- if (pDevice == NULL) {
- return MA_STATE_UNINITIALIZED;
+ ma_uint32 readOffset;
+ ma_uint32 readOffsetInBytes;
+ ma_uint32 readOffsetLoopFlag;
+ ma_uint32 newReadOffsetInBytes;
+ ma_uint32 newReadOffsetLoopFlag;
+
+ if (pRB == NULL) {
+ return MA_INVALID_ARGS;
}
- return c89atomic_load_32((ma_uint32*)&pDevice->state); /* Naughty cast to get rid of a const warning. */
-}
+ readOffset = c89atomic_load_32(&pRB->encodedReadOffset);
+ ma_rb__deconstruct_offset(readOffset, &readOffsetInBytes, &readOffsetLoopFlag);
-MA_API ma_result ma_device_set_master_volume(ma_device* pDevice, float volume)
-{
- if (pDevice == NULL) {
- return MA_INVALID_ARGS;
+ /* Check that sizeInBytes is correct. It should never go beyond the end of the buffer. */
+ newReadOffsetInBytes = (ma_uint32)(readOffsetInBytes + sizeInBytes);
+ if (newReadOffsetInBytes > pRB->subbufferSizeInBytes) {
+ return MA_INVALID_ARGS; /* <-- sizeInBytes will cause the read offset to overflow. */
}
- if (volume < 0.0f || volume > 1.0f) {
- return MA_INVALID_ARGS;
+ /* Move the read pointer back to the start if necessary. */
+ newReadOffsetLoopFlag = readOffsetLoopFlag;
+ if (newReadOffsetInBytes == pRB->subbufferSizeInBytes) {
+ newReadOffsetInBytes = 0;
+ newReadOffsetLoopFlag ^= 0x80000000;
}
- c89atomic_exchange_f32(&pDevice->masterVolumeFactor, volume);
+ c89atomic_exchange_32(&pRB->encodedReadOffset, ma_rb__construct_offset(newReadOffsetLoopFlag, newReadOffsetInBytes));
- return MA_SUCCESS;
+ if (ma_rb_pointer_distance(pRB) == 0) {
+ return MA_AT_END;
+ } else {
+ return MA_SUCCESS;
+ }
}
-MA_API ma_result ma_device_get_master_volume(ma_device* pDevice, float* pVolume)
+MA_API ma_result ma_rb_acquire_write(ma_rb* pRB, size_t* pSizeInBytes, void** ppBufferOut)
{
- if (pVolume == NULL) {
+ ma_uint32 readOffset;
+ ma_uint32 readOffsetInBytes;
+ ma_uint32 readOffsetLoopFlag;
+ ma_uint32 writeOffset;
+ ma_uint32 writeOffsetInBytes;
+ ma_uint32 writeOffsetLoopFlag;
+ size_t bytesAvailable;
+ size_t bytesRequested;
+
+ if (pRB == NULL || pSizeInBytes == NULL || ppBufferOut == NULL) {
return MA_INVALID_ARGS;
}
- if (pDevice == NULL) {
- *pVolume = 0;
- return MA_INVALID_ARGS;
+ /* The returned buffer should never overtake the read buffer. */
+ readOffset = c89atomic_load_32(&pRB->encodedReadOffset);
+ ma_rb__deconstruct_offset(readOffset, &readOffsetInBytes, &readOffsetLoopFlag);
+
+ writeOffset = c89atomic_load_32(&pRB->encodedWriteOffset);
+ ma_rb__deconstruct_offset(writeOffset, &writeOffsetInBytes, &writeOffsetLoopFlag);
+
+ /*
+ In the case of writing, if the write pointer and the read pointer are on the same loop iteration we can only
+ write up to the end of the buffer. Otherwise we can only write up to the read pointer. The write pointer should
+ never overtake the read pointer.
+ */
+ if (writeOffsetLoopFlag == readOffsetLoopFlag) {
+ bytesAvailable = pRB->subbufferSizeInBytes - writeOffsetInBytes;
+ } else {
+ bytesAvailable = readOffsetInBytes - writeOffsetInBytes;
}
- *pVolume = c89atomic_load_f32(&pDevice->masterVolumeFactor);
+ bytesRequested = *pSizeInBytes;
+ if (bytesRequested > bytesAvailable) {
+ bytesRequested = bytesAvailable;
+ }
+
+ *pSizeInBytes = bytesRequested;
+ *ppBufferOut = ma_rb__get_write_ptr(pRB);
+
+ /* Clear the buffer if desired. */
+ if (pRB->clearOnWriteAcquire) {
+ MA_ZERO_MEMORY(*ppBufferOut, *pSizeInBytes);
+ }
return MA_SUCCESS;
}
-MA_API ma_result ma_device_set_master_gain_db(ma_device* pDevice, float gainDB)
+MA_API ma_result ma_rb_commit_write(ma_rb* pRB, size_t sizeInBytes)
{
- if (gainDB > 0) {
+ ma_uint32 writeOffset;
+ ma_uint32 writeOffsetInBytes;
+ ma_uint32 writeOffsetLoopFlag;
+ ma_uint32 newWriteOffsetInBytes;
+ ma_uint32 newWriteOffsetLoopFlag;
+
+ if (pRB == NULL) {
return MA_INVALID_ARGS;
}
- return ma_device_set_master_volume(pDevice, ma_gain_db_to_factor(gainDB));
+ writeOffset = c89atomic_load_32(&pRB->encodedWriteOffset);
+ ma_rb__deconstruct_offset(writeOffset, &writeOffsetInBytes, &writeOffsetLoopFlag);
+
+ /* Check that sizeInBytes is correct. It should never go beyond the end of the buffer. */
+ newWriteOffsetInBytes = (ma_uint32)(writeOffsetInBytes + sizeInBytes);
+ if (newWriteOffsetInBytes > pRB->subbufferSizeInBytes) {
+ return MA_INVALID_ARGS; /* <-- sizeInBytes will cause the read offset to overflow. */
+ }
+
+ /* Move the read pointer back to the start if necessary. */
+ newWriteOffsetLoopFlag = writeOffsetLoopFlag;
+ if (newWriteOffsetInBytes == pRB->subbufferSizeInBytes) {
+ newWriteOffsetInBytes = 0;
+ newWriteOffsetLoopFlag ^= 0x80000000;
+ }
+
+ c89atomic_exchange_32(&pRB->encodedWriteOffset, ma_rb__construct_offset(newWriteOffsetLoopFlag, newWriteOffsetInBytes));
+
+ if (ma_rb_pointer_distance(pRB) == 0) {
+ return MA_AT_END;
+ } else {
+ return MA_SUCCESS;
+ }
}
-MA_API ma_result ma_device_get_master_gain_db(ma_device* pDevice, float* pGainDB)
+MA_API ma_result ma_rb_seek_read(ma_rb* pRB, size_t offsetInBytes)
{
- float factor;
- ma_result result;
+ ma_uint32 readOffset;
+ ma_uint32 readOffsetInBytes;
+ ma_uint32 readOffsetLoopFlag;
+ ma_uint32 writeOffset;
+ ma_uint32 writeOffsetInBytes;
+ ma_uint32 writeOffsetLoopFlag;
+ ma_uint32 newReadOffsetInBytes;
+ ma_uint32 newReadOffsetLoopFlag;
- if (pGainDB == NULL) {
+ if (pRB == NULL || offsetInBytes > pRB->subbufferSizeInBytes) {
return MA_INVALID_ARGS;
}
- result = ma_device_get_master_volume(pDevice, &factor);
- if (result != MA_SUCCESS) {
- *pGainDB = 0;
- return result;
- }
+ readOffset = c89atomic_load_32(&pRB->encodedReadOffset);
+ ma_rb__deconstruct_offset(readOffset, &readOffsetInBytes, &readOffsetLoopFlag);
+
+ writeOffset = c89atomic_load_32(&pRB->encodedWriteOffset);
+ ma_rb__deconstruct_offset(writeOffset, &writeOffsetInBytes, &writeOffsetLoopFlag);
+
+ newReadOffsetLoopFlag = readOffsetLoopFlag;
- *pGainDB = ma_factor_to_gain_db(factor);
+ /* We cannot go past the write buffer. */
+ if (readOffsetLoopFlag == writeOffsetLoopFlag) {
+ if ((readOffsetInBytes + offsetInBytes) > writeOffsetInBytes) {
+ newReadOffsetInBytes = writeOffsetInBytes;
+ } else {
+ newReadOffsetInBytes = (ma_uint32)(readOffsetInBytes + offsetInBytes);
+ }
+ } else {
+ /* May end up looping. */
+ if ((readOffsetInBytes + offsetInBytes) >= pRB->subbufferSizeInBytes) {
+ newReadOffsetInBytes = (ma_uint32)(readOffsetInBytes + offsetInBytes) - pRB->subbufferSizeInBytes;
+ newReadOffsetLoopFlag ^= 0x80000000; /* <-- Looped. */
+ } else {
+ newReadOffsetInBytes = (ma_uint32)(readOffsetInBytes + offsetInBytes);
+ }
+ }
+ c89atomic_exchange_32(&pRB->encodedReadOffset, ma_rb__construct_offset(newReadOffsetInBytes, newReadOffsetLoopFlag));
return MA_SUCCESS;
}
-
-MA_API ma_result ma_device_handle_backend_data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
+MA_API ma_result ma_rb_seek_write(ma_rb* pRB, size_t offsetInBytes)
{
- if (pDevice == NULL) {
- return MA_INVALID_ARGS;
- }
+ ma_uint32 readOffset;
+ ma_uint32 readOffsetInBytes;
+ ma_uint32 readOffsetLoopFlag;
+ ma_uint32 writeOffset;
+ ma_uint32 writeOffsetInBytes;
+ ma_uint32 writeOffsetLoopFlag;
+ ma_uint32 newWriteOffsetInBytes;
+ ma_uint32 newWriteOffsetLoopFlag;
- if (pOutput == NULL && pInput == NULL) {
+ if (pRB == NULL) {
return MA_INVALID_ARGS;
}
- if (pDevice->type == ma_device_type_duplex) {
- if (pInput != NULL) {
- ma_device__handle_duplex_callback_capture(pDevice, frameCount, pInput, &pDevice->duplexRB.rb);
- }
-
- if (pOutput != NULL) {
- ma_device__handle_duplex_callback_playback(pDevice, frameCount, pOutput, &pDevice->duplexRB.rb);
- }
- } else {
- if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_loopback) {
- if (pInput == NULL) {
- return MA_INVALID_ARGS;
- }
+ readOffset = c89atomic_load_32(&pRB->encodedReadOffset);
+ ma_rb__deconstruct_offset(readOffset, &readOffsetInBytes, &readOffsetLoopFlag);
- ma_device__send_frames_to_client(pDevice, frameCount, pInput);
- }
+ writeOffset = c89atomic_load_32(&pRB->encodedWriteOffset);
+ ma_rb__deconstruct_offset(writeOffset, &writeOffsetInBytes, &writeOffsetLoopFlag);
- if (pDevice->type == ma_device_type_playback) {
- if (pOutput == NULL) {
- return MA_INVALID_ARGS;
- }
+ newWriteOffsetLoopFlag = writeOffsetLoopFlag;
- ma_device__read_frames_from_client(pDevice, frameCount, pOutput);
+ /* We cannot go past the write buffer. */
+ if (readOffsetLoopFlag == writeOffsetLoopFlag) {
+ /* May end up looping. */
+ if ((writeOffsetInBytes + offsetInBytes) >= pRB->subbufferSizeInBytes) {
+ newWriteOffsetInBytes = (ma_uint32)(writeOffsetInBytes + offsetInBytes) - pRB->subbufferSizeInBytes;
+ newWriteOffsetLoopFlag ^= 0x80000000; /* <-- Looped. */
+ } else {
+ newWriteOffsetInBytes = (ma_uint32)(writeOffsetInBytes + offsetInBytes);
+ }
+ } else {
+ if ((writeOffsetInBytes + offsetInBytes) > readOffsetInBytes) {
+ newWriteOffsetInBytes = readOffsetInBytes;
+ } else {
+ newWriteOffsetInBytes = (ma_uint32)(writeOffsetInBytes + offsetInBytes);
}
}
+ c89atomic_exchange_32(&pRB->encodedWriteOffset, ma_rb__construct_offset(newWriteOffsetInBytes, newWriteOffsetLoopFlag));
return MA_SUCCESS;
}
-MA_API ma_uint32 ma_calculate_buffer_size_in_frames_from_descriptor(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile)
+MA_API ma_int32 ma_rb_pointer_distance(ma_rb* pRB)
{
- if (pDescriptor == NULL) {
+ ma_uint32 readOffset;
+ ma_uint32 readOffsetInBytes;
+ ma_uint32 readOffsetLoopFlag;
+ ma_uint32 writeOffset;
+ ma_uint32 writeOffsetInBytes;
+ ma_uint32 writeOffsetLoopFlag;
+
+ if (pRB == NULL) {
return 0;
}
- /*
- We must have a non-0 native sample rate, but some backends don't allow retrieval of this at the
- time when the size of the buffer needs to be determined. In this case we need to just take a best
- guess and move on. We'll try using the sample rate in pDescriptor first. If that's not set we'll
- just fall back to MA_DEFAULT_SAMPLE_RATE.
- */
- if (nativeSampleRate == 0) {
- nativeSampleRate = pDescriptor->sampleRate;
- }
- if (nativeSampleRate == 0) {
- nativeSampleRate = MA_DEFAULT_SAMPLE_RATE;
- }
+ readOffset = c89atomic_load_32(&pRB->encodedReadOffset);
+ ma_rb__deconstruct_offset(readOffset, &readOffsetInBytes, &readOffsetLoopFlag);
- MA_ASSERT(nativeSampleRate != 0);
+ writeOffset = c89atomic_load_32(&pRB->encodedWriteOffset);
+ ma_rb__deconstruct_offset(writeOffset, &writeOffsetInBytes, &writeOffsetLoopFlag);
- if (pDescriptor->periodSizeInFrames == 0) {
- if (pDescriptor->periodSizeInMilliseconds == 0) {
- if (performanceProfile == ma_performance_profile_low_latency) {
- return ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY, nativeSampleRate);
- } else {
- return ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE, nativeSampleRate);
- }
- } else {
- return ma_calculate_buffer_size_in_frames_from_milliseconds(pDescriptor->periodSizeInMilliseconds, nativeSampleRate);
- }
+ if (readOffsetLoopFlag == writeOffsetLoopFlag) {
+ return writeOffsetInBytes - readOffsetInBytes;
} else {
- return pDescriptor->periodSizeInFrames;
+ return writeOffsetInBytes + (pRB->subbufferSizeInBytes - readOffsetInBytes);
}
}
-#endif /* MA_NO_DEVICE_IO */
-
-MA_API ma_uint32 ma_scale_buffer_size(ma_uint32 baseBufferSize, float scale)
+MA_API ma_uint32 ma_rb_available_read(ma_rb* pRB)
{
- return ma_max(1, (ma_uint32)(baseBufferSize*scale));
+ ma_int32 dist;
+
+ if (pRB == NULL) {
+ return 0;
+ }
+
+ dist = ma_rb_pointer_distance(pRB);
+ if (dist < 0) {
+ return 0;
+ }
+
+ return dist;
}
-MA_API ma_uint32 ma_calculate_buffer_size_in_milliseconds_from_frames(ma_uint32 bufferSizeInFrames, ma_uint32 sampleRate)
+MA_API ma_uint32 ma_rb_available_write(ma_rb* pRB)
{
- /* Prevent a division by zero. */
- if (sampleRate == 0) {
+ if (pRB == NULL) {
return 0;
}
- return bufferSizeInFrames*1000 / sampleRate;
+ return (ma_uint32)(ma_rb_get_subbuffer_size(pRB) - ma_rb_pointer_distance(pRB));
}
-MA_API ma_uint32 ma_calculate_buffer_size_in_frames_from_milliseconds(ma_uint32 bufferSizeInMilliseconds, ma_uint32 sampleRate)
+MA_API size_t ma_rb_get_subbuffer_size(ma_rb* pRB)
{
- /* Prevent a division by zero. */
- if (sampleRate == 0) {
+ if (pRB == NULL) {
return 0;
}
- return bufferSizeInMilliseconds*sampleRate / 1000;
+ return pRB->subbufferSizeInBytes;
}
-MA_API void ma_copy_pcm_frames(void* dst, const void* src, ma_uint64 frameCount, ma_format format, ma_uint32 channels)
+MA_API size_t ma_rb_get_subbuffer_stride(ma_rb* pRB)
{
- if (dst == src) {
- return; /* No-op. */
+ if (pRB == NULL) {
+ return 0;
}
- ma_copy_memory_64(dst, src, frameCount * ma_get_bytes_per_frame(format, channels));
+ if (pRB->subbufferStrideInBytes == 0) {
+ return (size_t)pRB->subbufferSizeInBytes;
+ }
+
+ return (size_t)pRB->subbufferStrideInBytes;
}
-MA_API void ma_silence_pcm_frames(void* p, ma_uint64 frameCount, ma_format format, ma_uint32 channels)
+MA_API size_t ma_rb_get_subbuffer_offset(ma_rb* pRB, size_t subbufferIndex)
{
- if (format == ma_format_u8) {
- ma_uint64 sampleCount = frameCount * channels;
- ma_uint64 iSample;
- for (iSample = 0; iSample < sampleCount; iSample += 1) {
- ((ma_uint8*)p)[iSample] = 128;
- }
- } else {
- ma_zero_memory_64(p, frameCount * ma_get_bytes_per_frame(format, channels));
+ if (pRB == NULL) {
+ return 0;
}
+
+ return subbufferIndex * ma_rb_get_subbuffer_stride(pRB);
}
-MA_API void* ma_offset_pcm_frames_ptr(void* p, ma_uint64 offsetInFrames, ma_format format, ma_uint32 channels)
+MA_API void* ma_rb_get_subbuffer_ptr(ma_rb* pRB, size_t subbufferIndex, void* pBuffer)
{
- return ma_offset_ptr(p, offsetInFrames * ma_get_bytes_per_frame(format, channels));
-}
+ if (pRB == NULL) {
+ return NULL;
+ }
-MA_API const void* ma_offset_pcm_frames_const_ptr(const void* p, ma_uint64 offsetInFrames, ma_format format, ma_uint32 channels)
-{
- return ma_offset_ptr(p, offsetInFrames * ma_get_bytes_per_frame(format, channels));
+ return ma_offset_ptr(pBuffer, ma_rb_get_subbuffer_offset(pRB, subbufferIndex));
}
-MA_API void ma_clip_samples_f32(float* p, ma_uint64 sampleCount)
+
+static MA_INLINE ma_uint32 ma_pcm_rb_get_bpf(ma_pcm_rb* pRB)
{
- ma_uint32 iSample;
+ MA_ASSERT(pRB != NULL);
- /* TODO: Research a branchless SSE implementation. */
- for (iSample = 0; iSample < sampleCount; iSample += 1) {
- p[iSample] = ma_clip_f32(p[iSample]);
- }
+ return ma_get_bytes_per_frame(pRB->format, pRB->channels);
}
-
-MA_API void ma_copy_and_apply_volume_factor_u8(ma_uint8* pSamplesOut, const ma_uint8* pSamplesIn, ma_uint64 sampleCount, float factor)
+MA_API ma_result ma_pcm_rb_init_ex(ma_format format, ma_uint32 channels, ma_uint32 subbufferSizeInFrames, ma_uint32 subbufferCount, ma_uint32 subbufferStrideInFrames, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_pcm_rb* pRB)
{
- ma_uint64 iSample;
+ ma_uint32 bpf;
+ ma_result result;
- if (pSamplesOut == NULL || pSamplesIn == NULL) {
- return;
+ if (pRB == NULL) {
+ return MA_INVALID_ARGS;
}
- for (iSample = 0; iSample < sampleCount; iSample += 1) {
- pSamplesOut[iSample] = (ma_uint8)(pSamplesIn[iSample] * factor);
+ MA_ZERO_OBJECT(pRB);
+
+ bpf = ma_get_bytes_per_frame(format, channels);
+ if (bpf == 0) {
+ return MA_INVALID_ARGS;
+ }
+
+ result = ma_rb_init_ex(subbufferSizeInFrames*bpf, subbufferCount, subbufferStrideInFrames*bpf, pOptionalPreallocatedBuffer, pAllocationCallbacks, &pRB->rb);
+ if (result != MA_SUCCESS) {
+ return result;
}
+
+ pRB->format = format;
+ pRB->channels = channels;
+
+ return MA_SUCCESS;
}
-MA_API void ma_copy_and_apply_volume_factor_s16(ma_int16* pSamplesOut, const ma_int16* pSamplesIn, ma_uint64 sampleCount, float factor)
+MA_API ma_result ma_pcm_rb_init(ma_format format, ma_uint32 channels, ma_uint32 bufferSizeInFrames, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_pcm_rb* pRB)
{
- ma_uint64 iSample;
+ return ma_pcm_rb_init_ex(format, channels, bufferSizeInFrames, 1, 0, pOptionalPreallocatedBuffer, pAllocationCallbacks, pRB);
+}
- if (pSamplesOut == NULL || pSamplesIn == NULL) {
+MA_API void ma_pcm_rb_uninit(ma_pcm_rb* pRB)
+{
+ if (pRB == NULL) {
return;
}
- for (iSample = 0; iSample < sampleCount; iSample += 1) {
- pSamplesOut[iSample] = (ma_int16)(pSamplesIn[iSample] * factor);
- }
+ ma_rb_uninit(&pRB->rb);
}
-MA_API void ma_copy_and_apply_volume_factor_s24(void* pSamplesOut, const void* pSamplesIn, ma_uint64 sampleCount, float factor)
+MA_API void ma_pcm_rb_reset(ma_pcm_rb* pRB)
{
- ma_uint64 iSample;
- ma_uint8* pSamplesOut8;
- ma_uint8* pSamplesIn8;
-
- if (pSamplesOut == NULL || pSamplesIn == NULL) {
+ if (pRB == NULL) {
return;
}
- pSamplesOut8 = (ma_uint8*)pSamplesOut;
- pSamplesIn8 = (ma_uint8*)pSamplesIn;
+ ma_rb_reset(&pRB->rb);
+}
- for (iSample = 0; iSample < sampleCount; iSample += 1) {
- ma_int32 sampleS32;
+MA_API ma_result ma_pcm_rb_acquire_read(ma_pcm_rb* pRB, ma_uint32* pSizeInFrames, void** ppBufferOut)
+{
+ size_t sizeInBytes;
+ ma_result result;
- sampleS32 = (ma_int32)(((ma_uint32)(pSamplesIn8[iSample*3+0]) << 8) | ((ma_uint32)(pSamplesIn8[iSample*3+1]) << 16) | ((ma_uint32)(pSamplesIn8[iSample*3+2])) << 24);
- sampleS32 = (ma_int32)(sampleS32 * factor);
+ if (pRB == NULL || pSizeInFrames == NULL) {
+ return MA_INVALID_ARGS;
+ }
- pSamplesOut8[iSample*3+0] = (ma_uint8)(((ma_uint32)sampleS32 & 0x0000FF00) >> 8);
- pSamplesOut8[iSample*3+1] = (ma_uint8)(((ma_uint32)sampleS32 & 0x00FF0000) >> 16);
- pSamplesOut8[iSample*3+2] = (ma_uint8)(((ma_uint32)sampleS32 & 0xFF000000) >> 24);
+ sizeInBytes = *pSizeInFrames * ma_pcm_rb_get_bpf(pRB);
+
+ result = ma_rb_acquire_read(&pRB->rb, &sizeInBytes, ppBufferOut);
+ if (result != MA_SUCCESS) {
+ return result;
}
+
+ *pSizeInFrames = (ma_uint32)(sizeInBytes / (size_t)ma_pcm_rb_get_bpf(pRB));
+ return MA_SUCCESS;
}
-MA_API void ma_copy_and_apply_volume_factor_s32(ma_int32* pSamplesOut, const ma_int32* pSamplesIn, ma_uint64 sampleCount, float factor)
+MA_API ma_result ma_pcm_rb_commit_read(ma_pcm_rb* pRB, ma_uint32 sizeInFrames)
{
- ma_uint64 iSample;
-
- if (pSamplesOut == NULL || pSamplesIn == NULL) {
- return;
+ if (pRB == NULL) {
+ return MA_INVALID_ARGS;
}
- for (iSample = 0; iSample < sampleCount; iSample += 1) {
- pSamplesOut[iSample] = (ma_int32)(pSamplesIn[iSample] * factor);
- }
+ return ma_rb_commit_read(&pRB->rb, sizeInFrames * ma_pcm_rb_get_bpf(pRB));
}
-MA_API void ma_copy_and_apply_volume_factor_f32(float* pSamplesOut, const float* pSamplesIn, ma_uint64 sampleCount, float factor)
+MA_API ma_result ma_pcm_rb_acquire_write(ma_pcm_rb* pRB, ma_uint32* pSizeInFrames, void** ppBufferOut)
{
- ma_uint64 iSample;
+ size_t sizeInBytes;
+ ma_result result;
- if (pSamplesOut == NULL || pSamplesIn == NULL) {
- return;
+ if (pRB == NULL) {
+ return MA_INVALID_ARGS;
}
- for (iSample = 0; iSample < sampleCount; iSample += 1) {
- pSamplesOut[iSample] = pSamplesIn[iSample] * factor;
+ sizeInBytes = *pSizeInFrames * ma_pcm_rb_get_bpf(pRB);
+
+ result = ma_rb_acquire_write(&pRB->rb, &sizeInBytes, ppBufferOut);
+ if (result != MA_SUCCESS) {
+ return result;
}
-}
-MA_API void ma_apply_volume_factor_u8(ma_uint8* pSamples, ma_uint64 sampleCount, float factor)
-{
- ma_copy_and_apply_volume_factor_u8(pSamples, pSamples, sampleCount, factor);
+ *pSizeInFrames = (ma_uint32)(sizeInBytes / ma_pcm_rb_get_bpf(pRB));
+ return MA_SUCCESS;
}
-MA_API void ma_apply_volume_factor_s16(ma_int16* pSamples, ma_uint64 sampleCount, float factor)
+MA_API ma_result ma_pcm_rb_commit_write(ma_pcm_rb* pRB, ma_uint32 sizeInFrames)
{
- ma_copy_and_apply_volume_factor_s16(pSamples, pSamples, sampleCount, factor);
-}
+ if (pRB == NULL) {
+ return MA_INVALID_ARGS;
+ }
-MA_API void ma_apply_volume_factor_s24(void* pSamples, ma_uint64 sampleCount, float factor)
-{
- ma_copy_and_apply_volume_factor_s24(pSamples, pSamples, sampleCount, factor);
+ return ma_rb_commit_write(&pRB->rb, sizeInFrames * ma_pcm_rb_get_bpf(pRB));
}
-MA_API void ma_apply_volume_factor_s32(ma_int32* pSamples, ma_uint64 sampleCount, float factor)
+MA_API ma_result ma_pcm_rb_seek_read(ma_pcm_rb* pRB, ma_uint32 offsetInFrames)
{
- ma_copy_and_apply_volume_factor_s32(pSamples, pSamples, sampleCount, factor);
-}
+ if (pRB == NULL) {
+ return MA_INVALID_ARGS;
+ }
-MA_API void ma_apply_volume_factor_f32(float* pSamples, ma_uint64 sampleCount, float factor)
-{
- ma_copy_and_apply_volume_factor_f32(pSamples, pSamples, sampleCount, factor);
+ return ma_rb_seek_read(&pRB->rb, offsetInFrames * ma_pcm_rb_get_bpf(pRB));
}
-MA_API void ma_copy_and_apply_volume_factor_pcm_frames_u8(ma_uint8* pPCMFramesOut, const ma_uint8* pPCMFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor)
+MA_API ma_result ma_pcm_rb_seek_write(ma_pcm_rb* pRB, ma_uint32 offsetInFrames)
{
- ma_copy_and_apply_volume_factor_u8(pPCMFramesOut, pPCMFramesIn, frameCount*channels, factor);
-}
+ if (pRB == NULL) {
+ return MA_INVALID_ARGS;
+ }
-MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s16(ma_int16* pPCMFramesOut, const ma_int16* pPCMFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor)
-{
- ma_copy_and_apply_volume_factor_s16(pPCMFramesOut, pPCMFramesIn, frameCount*channels, factor);
+ return ma_rb_seek_write(&pRB->rb, offsetInFrames * ma_pcm_rb_get_bpf(pRB));
}
-MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s24(void* pPCMFramesOut, const void* pPCMFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor)
+MA_API ma_int32 ma_pcm_rb_pointer_distance(ma_pcm_rb* pRB)
{
- ma_copy_and_apply_volume_factor_s24(pPCMFramesOut, pPCMFramesIn, frameCount*channels, factor);
-}
+ if (pRB == NULL) {
+ return 0;
+ }
-MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s32(ma_int32* pPCMFramesOut, const ma_int32* pPCMFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor)
-{
- ma_copy_and_apply_volume_factor_s32(pPCMFramesOut, pPCMFramesIn, frameCount*channels, factor);
+ return ma_rb_pointer_distance(&pRB->rb) / ma_pcm_rb_get_bpf(pRB);
}
-MA_API void ma_copy_and_apply_volume_factor_pcm_frames_f32(float* pPCMFramesOut, const float* pPCMFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor)
+MA_API ma_uint32 ma_pcm_rb_available_read(ma_pcm_rb* pRB)
{
- ma_copy_and_apply_volume_factor_f32(pPCMFramesOut, pPCMFramesIn, frameCount*channels, factor);
+ if (pRB == NULL) {
+ return 0;
+ }
+
+ return ma_rb_available_read(&pRB->rb) / ma_pcm_rb_get_bpf(pRB);
}
-MA_API void ma_copy_and_apply_volume_factor_pcm_frames(void* pPCMFramesOut, const void* pPCMFramesIn, ma_uint64 frameCount, ma_format format, ma_uint32 channels, float factor)
+MA_API ma_uint32 ma_pcm_rb_available_write(ma_pcm_rb* pRB)
{
- switch (format)
- {
- case ma_format_u8: ma_copy_and_apply_volume_factor_pcm_frames_u8 ((ma_uint8*)pPCMFramesOut, (const ma_uint8*)pPCMFramesIn, frameCount, channels, factor); return;
- case ma_format_s16: ma_copy_and_apply_volume_factor_pcm_frames_s16((ma_int16*)pPCMFramesOut, (const ma_int16*)pPCMFramesIn, frameCount, channels, factor); return;
- case ma_format_s24: ma_copy_and_apply_volume_factor_pcm_frames_s24( pPCMFramesOut, pPCMFramesIn, frameCount, channels, factor); return;
- case ma_format_s32: ma_copy_and_apply_volume_factor_pcm_frames_s32((ma_int32*)pPCMFramesOut, (const ma_int32*)pPCMFramesIn, frameCount, channels, factor); return;
- case ma_format_f32: ma_copy_and_apply_volume_factor_pcm_frames_f32( (float*)pPCMFramesOut, (const float*)pPCMFramesIn, frameCount, channels, factor); return;
- default: return; /* Do nothing. */
+ if (pRB == NULL) {
+ return 0;
}
-}
-MA_API void ma_apply_volume_factor_pcm_frames_u8(ma_uint8* pPCMFrames, ma_uint64 frameCount, ma_uint32 channels, float factor)
-{
- ma_copy_and_apply_volume_factor_pcm_frames_u8(pPCMFrames, pPCMFrames, frameCount, channels, factor);
+ return ma_rb_available_write(&pRB->rb) / ma_pcm_rb_get_bpf(pRB);
}
-MA_API void ma_apply_volume_factor_pcm_frames_s16(ma_int16* pPCMFrames, ma_uint64 frameCount, ma_uint32 channels, float factor)
+MA_API ma_uint32 ma_pcm_rb_get_subbuffer_size(ma_pcm_rb* pRB)
{
- ma_copy_and_apply_volume_factor_pcm_frames_s16(pPCMFrames, pPCMFrames, frameCount, channels, factor);
-}
+ if (pRB == NULL) {
+ return 0;
+ }
-MA_API void ma_apply_volume_factor_pcm_frames_s24(void* pPCMFrames, ma_uint64 frameCount, ma_uint32 channels, float factor)
-{
- ma_copy_and_apply_volume_factor_pcm_frames_s24(pPCMFrames, pPCMFrames, frameCount, channels, factor);
+ return (ma_uint32)(ma_rb_get_subbuffer_size(&pRB->rb) / ma_pcm_rb_get_bpf(pRB));
}
-MA_API void ma_apply_volume_factor_pcm_frames_s32(ma_int32* pPCMFrames, ma_uint64 frameCount, ma_uint32 channels, float factor)
+MA_API ma_uint32 ma_pcm_rb_get_subbuffer_stride(ma_pcm_rb* pRB)
{
- ma_copy_and_apply_volume_factor_pcm_frames_s32(pPCMFrames, pPCMFrames, frameCount, channels, factor);
-}
+ if (pRB == NULL) {
+ return 0;
+ }
-MA_API void ma_apply_volume_factor_pcm_frames_f32(float* pPCMFrames, ma_uint64 frameCount, ma_uint32 channels, float factor)
-{
- ma_copy_and_apply_volume_factor_pcm_frames_f32(pPCMFrames, pPCMFrames, frameCount, channels, factor);
+ return (ma_uint32)(ma_rb_get_subbuffer_stride(&pRB->rb) / ma_pcm_rb_get_bpf(pRB));
}
-MA_API void ma_apply_volume_factor_pcm_frames(void* pPCMFrames, ma_uint64 frameCount, ma_format format, ma_uint32 channels, float factor)
+MA_API ma_uint32 ma_pcm_rb_get_subbuffer_offset(ma_pcm_rb* pRB, ma_uint32 subbufferIndex)
{
- ma_copy_and_apply_volume_factor_pcm_frames(pPCMFrames, pPCMFrames, frameCount, format, channels, factor);
-}
-
+ if (pRB == NULL) {
+ return 0;
+ }
-MA_API float ma_factor_to_gain_db(float factor)
-{
- return (float)(20*ma_log10f(factor));
+ return (ma_uint32)(ma_rb_get_subbuffer_offset(&pRB->rb, subbufferIndex) / ma_pcm_rb_get_bpf(pRB));
}
-MA_API float ma_gain_db_to_factor(float gain)
+MA_API void* ma_pcm_rb_get_subbuffer_ptr(ma_pcm_rb* pRB, ma_uint32 subbufferIndex, void* pBuffer)
{
- return (float)ma_powf(10, gain/20.0f);
-}
-
-
-/**************************************************************************************************************************************************************
-
-Format Conversion
-
-**************************************************************************************************************************************************************/
+ if (pRB == NULL) {
+ return NULL;
+ }
-static MA_INLINE ma_int16 ma_pcm_sample_f32_to_s16(float x)
-{
- return (ma_int16)(x * 32767.0f);
+ return ma_rb_get_subbuffer_ptr(&pRB->rb, subbufferIndex, pBuffer);
}
-static MA_INLINE ma_int16 ma_pcm_sample_u8_to_s16_no_scale(ma_uint8 x)
-{
- return (ma_int16)((ma_int16)x - 128);
-}
-static MA_INLINE ma_int64 ma_pcm_sample_s24_to_s32_no_scale(const ma_uint8* x)
-{
- return (ma_int64)(((ma_uint64)x[0] << 40) | ((ma_uint64)x[1] << 48) | ((ma_uint64)x[2] << 56)) >> 40; /* Make sure the sign bits are maintained. */
-}
-static MA_INLINE void ma_pcm_sample_s32_to_s24_no_scale(ma_int64 x, ma_uint8* s24)
+MA_API ma_result ma_duplex_rb_init(ma_format captureFormat, ma_uint32 captureChannels, ma_uint32 sampleRate, ma_uint32 captureInternalSampleRate, ma_uint32 captureInternalPeriodSizeInFrames, const ma_allocation_callbacks* pAllocationCallbacks, ma_duplex_rb* pRB)
{
- s24[0] = (ma_uint8)((x & 0x000000FF) >> 0);
- s24[1] = (ma_uint8)((x & 0x0000FF00) >> 8);
- s24[2] = (ma_uint8)((x & 0x00FF0000) >> 16);
-}
+ ma_result result;
+ ma_uint32 sizeInFrames;
+ sizeInFrames = (ma_uint32)ma_calculate_frame_count_after_resampling(sampleRate, captureInternalSampleRate, captureInternalPeriodSizeInFrames * 5);
+ if (sizeInFrames == 0) {
+ return MA_INVALID_ARGS;
+ }
-static MA_INLINE ma_uint8 ma_clip_u8(ma_int16 x)
-{
- return (ma_uint8)(ma_clamp(x, -128, 127) + 128);
-}
+ result = ma_pcm_rb_init(captureFormat, captureChannels, sizeInFrames, NULL, pAllocationCallbacks, &pRB->rb);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
-static MA_INLINE ma_int16 ma_clip_s16(ma_int32 x)
-{
- return (ma_int16)ma_clamp(x, -32768, 32767);
-}
+ /* Seek forward a bit so we have a bit of a buffer in case of desyncs. */
+ ma_pcm_rb_seek_write((ma_pcm_rb*)pRB, captureInternalPeriodSizeInFrames * 2);
-static MA_INLINE ma_int64 ma_clip_s24(ma_int64 x)
-{
- return (ma_int64)ma_clamp(x, -8388608, 8388607);
+ return MA_SUCCESS;
}
-static MA_INLINE ma_int32 ma_clip_s32(ma_int64 x)
+MA_API ma_result ma_duplex_rb_uninit(ma_duplex_rb* pRB)
{
- /* This dance is to silence warnings with -std=c89. A good compiler should be able to optimize this away. */
- ma_int64 clipMin;
- ma_int64 clipMax;
- clipMin = -((ma_int64)2147483647 + 1);
- clipMax = (ma_int64)2147483647;
-
- return (ma_int32)ma_clamp(x, clipMin, clipMax);
+ ma_pcm_rb_uninit((ma_pcm_rb*)pRB);
+ return MA_SUCCESS;
}
-/* u8 */
-MA_API void ma_pcm_u8_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- (void)ditherMode;
- ma_copy_memory_64(dst, src, count * sizeof(ma_uint8));
-}
+/**************************************************************************************************************************************************************
-static MA_INLINE void ma_pcm_u8_to_s16__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+Miscellaneous Helpers
+
+**************************************************************************************************************************************************************/
+MA_API const char* ma_result_description(ma_result result)
{
- ma_int16* dst_s16 = (ma_int16*)dst;
- const ma_uint8* src_u8 = (const ma_uint8*)src;
+ switch (result)
+ {
+ case MA_SUCCESS: return "No error";
+ case MA_ERROR: return "Unknown error";
+ case MA_INVALID_ARGS: return "Invalid argument";
+ case MA_INVALID_OPERATION: return "Invalid operation";
+ case MA_OUT_OF_MEMORY: return "Out of memory";
+ case MA_OUT_OF_RANGE: return "Out of range";
+ case MA_ACCESS_DENIED: return "Permission denied";
+ case MA_DOES_NOT_EXIST: return "Resource does not exist";
+ case MA_ALREADY_EXISTS: return "Resource already exists";
+ case MA_TOO_MANY_OPEN_FILES: return "Too many open files";
+ case MA_INVALID_FILE: return "Invalid file";
+ case MA_TOO_BIG: return "Too large";
+ case MA_PATH_TOO_LONG: return "Path too long";
+ case MA_NAME_TOO_LONG: return "Name too long";
+ case MA_NOT_DIRECTORY: return "Not a directory";
+ case MA_IS_DIRECTORY: return "Is a directory";
+ case MA_DIRECTORY_NOT_EMPTY: return "Directory not empty";
+ case MA_AT_END: return "At end";
+ case MA_NO_SPACE: return "No space available";
+ case MA_BUSY: return "Device or resource busy";
+ case MA_IO_ERROR: return "Input/output error";
+ case MA_INTERRUPT: return "Interrupted";
+ case MA_UNAVAILABLE: return "Resource unavailable";
+ case MA_ALREADY_IN_USE: return "Resource already in use";
+ case MA_BAD_ADDRESS: return "Bad address";
+ case MA_BAD_SEEK: return "Illegal seek";
+ case MA_BAD_PIPE: return "Broken pipe";
+ case MA_DEADLOCK: return "Deadlock";
+ case MA_TOO_MANY_LINKS: return "Too many links";
+ case MA_NOT_IMPLEMENTED: return "Not implemented";
+ case MA_NO_MESSAGE: return "No message of desired type";
+ case MA_BAD_MESSAGE: return "Invalid message";
+ case MA_NO_DATA_AVAILABLE: return "No data available";
+ case MA_INVALID_DATA: return "Invalid data";
+ case MA_TIMEOUT: return "Timeout";
+ case MA_NO_NETWORK: return "Network unavailable";
+ case MA_NOT_UNIQUE: return "Not unique";
+ case MA_NOT_SOCKET: return "Socket operation on non-socket";
+ case MA_NO_ADDRESS: return "Destination address required";
+ case MA_BAD_PROTOCOL: return "Protocol wrong type for socket";
+ case MA_PROTOCOL_UNAVAILABLE: return "Protocol not available";
+ case MA_PROTOCOL_NOT_SUPPORTED: return "Protocol not supported";
+ case MA_PROTOCOL_FAMILY_NOT_SUPPORTED: return "Protocol family not supported";
+ case MA_ADDRESS_FAMILY_NOT_SUPPORTED: return "Address family not supported";
+ case MA_SOCKET_NOT_SUPPORTED: return "Socket type not supported";
+ case MA_CONNECTION_RESET: return "Connection reset";
+ case MA_ALREADY_CONNECTED: return "Already connected";
+ case MA_NOT_CONNECTED: return "Not connected";
+ case MA_CONNECTION_REFUSED: return "Connection refused";
+ case MA_NO_HOST: return "No host";
+ case MA_IN_PROGRESS: return "Operation in progress";
+ case MA_CANCELLED: return "Operation cancelled";
+ case MA_MEMORY_ALREADY_MAPPED: return "Memory already mapped";
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- ma_int16 x = src_u8[i];
- x = (ma_int16)(x - 128);
- x = (ma_int16)(x << 8);
- dst_s16[i] = x;
- }
+ case MA_FORMAT_NOT_SUPPORTED: return "Format not supported";
+ case MA_DEVICE_TYPE_NOT_SUPPORTED: return "Device type not supported";
+ case MA_SHARE_MODE_NOT_SUPPORTED: return "Share mode not supported";
+ case MA_NO_BACKEND: return "No backend";
+ case MA_NO_DEVICE: return "No device";
+ case MA_API_NOT_FOUND: return "API not found";
+ case MA_INVALID_DEVICE_CONFIG: return "Invalid device config";
- (void)ditherMode;
-}
+ case MA_DEVICE_NOT_INITIALIZED: return "Device not initialized";
+ case MA_DEVICE_NOT_STARTED: return "Device not started";
-static MA_INLINE void ma_pcm_u8_to_s16__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_u8_to_s16__reference(dst, src, count, ditherMode);
-}
+ case MA_FAILED_TO_INIT_BACKEND: return "Failed to initialize backend";
+ case MA_FAILED_TO_OPEN_BACKEND_DEVICE: return "Failed to open backend device";
+ case MA_FAILED_TO_START_BACKEND_DEVICE: return "Failed to start backend device";
+ case MA_FAILED_TO_STOP_BACKEND_DEVICE: return "Failed to stop backend device";
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE void ma_pcm_u8_to_s16__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_u8_to_s16__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE void ma_pcm_u8_to_s16__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_u8_to_s16__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE void ma_pcm_u8_to_s16__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_u8_to_s16__optimized(dst, src, count, ditherMode);
+ default: return "Unknown error";
+ }
}
-#endif
-MA_API void ma_pcm_u8_to_s16(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API void* ma_malloc(size_t sz, const ma_allocation_callbacks* pAllocationCallbacks)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_u8_to_s16__reference(dst, src, count, ditherMode);
-#else
- # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
- if (ma_has_avx2()) {
- ma_pcm_u8_to_s16__avx2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
- if (ma_has_sse2()) {
- ma_pcm_u8_to_s16__sse2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
- if (ma_has_neon()) {
- ma_pcm_u8_to_s16__neon(dst, src, count, ditherMode);
- } else
- #endif
- {
- ma_pcm_u8_to_s16__optimized(dst, src, count, ditherMode);
+ if (pAllocationCallbacks != NULL) {
+ if (pAllocationCallbacks->onMalloc != NULL) {
+ return pAllocationCallbacks->onMalloc(sz, pAllocationCallbacks->pUserData);
+ } else {
+ return NULL; /* Do not fall back to the default implementation. */
}
-#endif
+ } else {
+ return ma__malloc_default(sz, NULL);
+ }
}
-
-static MA_INLINE void ma_pcm_u8_to_s24__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API void* ma_calloc(size_t sz, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_uint8* dst_s24 = (ma_uint8*)dst;
- const ma_uint8* src_u8 = (const ma_uint8*)src;
-
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- ma_int16 x = src_u8[i];
- x = (ma_int16)(x - 128);
-
- dst_s24[i*3+0] = 0;
- dst_s24[i*3+1] = 0;
- dst_s24[i*3+2] = (ma_uint8)((ma_int8)x);
+ void* p = ma_malloc(sz, pAllocationCallbacks);
+ if (p != NULL) {
+ MA_ZERO_MEMORY(p, sz);
}
- (void)ditherMode;
+ return p;
}
-static MA_INLINE void ma_pcm_u8_to_s24__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API void* ma_realloc(void* p, size_t sz, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_pcm_u8_to_s24__reference(dst, src, count, ditherMode);
+ if (pAllocationCallbacks != NULL) {
+ if (pAllocationCallbacks->onRealloc != NULL) {
+ return pAllocationCallbacks->onRealloc(p, sz, pAllocationCallbacks->pUserData);
+ } else {
+ return NULL; /* Do not fall back to the default implementation. */
+ }
+ } else {
+ return ma__realloc_default(p, sz, NULL);
+ }
}
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE void ma_pcm_u8_to_s24__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_u8_to_s24__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE void ma_pcm_u8_to_s24__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_u8_to_s24__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE void ma_pcm_u8_to_s24__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API void ma_free(void* p, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_pcm_u8_to_s24__optimized(dst, src, count, ditherMode);
-}
-#endif
+ if (p == NULL) {
+ return;
+ }
-MA_API void ma_pcm_u8_to_s24(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_u8_to_s24__reference(dst, src, count, ditherMode);
-#else
- # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
- if (ma_has_avx2()) {
- ma_pcm_u8_to_s24__avx2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
- if (ma_has_sse2()) {
- ma_pcm_u8_to_s24__sse2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
- if (ma_has_neon()) {
- ma_pcm_u8_to_s24__neon(dst, src, count, ditherMode);
- } else
- #endif
- {
- ma_pcm_u8_to_s24__optimized(dst, src, count, ditherMode);
+ if (pAllocationCallbacks != NULL) {
+ if (pAllocationCallbacks->onFree != NULL) {
+ pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData);
+ } else {
+ return; /* Do no fall back to the default implementation. */
}
-#endif
+ } else {
+ ma__free_default(p, NULL);
+ }
}
-
-static MA_INLINE void ma_pcm_u8_to_s32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API void* ma_aligned_malloc(size_t sz, size_t alignment, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_int32* dst_s32 = (ma_int32*)dst;
- const ma_uint8* src_u8 = (const ma_uint8*)src;
+ size_t extraBytes;
+ void* pUnaligned;
+ void* pAligned;
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- ma_int32 x = src_u8[i];
- x = x - 128;
- x = x << 24;
- dst_s32[i] = x;
+ if (alignment == 0) {
+ return 0;
}
- (void)ditherMode;
+ extraBytes = alignment-1 + sizeof(void*);
+
+ pUnaligned = ma_malloc(sz + extraBytes, pAllocationCallbacks);
+ if (pUnaligned == NULL) {
+ return NULL;
+ }
+
+ pAligned = (void*)(((ma_uintptr)pUnaligned + extraBytes) & ~((ma_uintptr)(alignment-1)));
+ ((void**)pAligned)[-1] = pUnaligned;
+
+ return pAligned;
}
-static MA_INLINE void ma_pcm_u8_to_s32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API void ma_aligned_free(void* p, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_pcm_u8_to_s32__reference(dst, src, count, ditherMode);
+ ma_free(((void**)p)[-1], pAllocationCallbacks);
}
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE void ma_pcm_u8_to_s32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API const char* ma_get_format_name(ma_format format)
{
- ma_pcm_u8_to_s32__optimized(dst, src, count, ditherMode);
+ switch (format)
+ {
+ case ma_format_unknown: return "Unknown";
+ case ma_format_u8: return "8-bit Unsigned Integer";
+ case ma_format_s16: return "16-bit Signed Integer";
+ case ma_format_s24: return "24-bit Signed Integer (Tightly Packed)";
+ case ma_format_s32: return "32-bit Signed Integer";
+ case ma_format_f32: return "32-bit IEEE Floating Point";
+ default: return "Invalid";
+ }
}
-#endif
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE void ma_pcm_u8_to_s32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+
+MA_API void ma_blend_f32(float* pOut, float* pInA, float* pInB, float factor, ma_uint32 channels)
{
- ma_pcm_u8_to_s32__optimized(dst, src, count, ditherMode);
+ ma_uint32 i;
+ for (i = 0; i < channels; ++i) {
+ pOut[i] = ma_mix_f32(pInA[i], pInB[i], factor);
+ }
}
-#endif
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE void ma_pcm_u8_to_s32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+
+
+MA_API ma_uint32 ma_get_bytes_per_sample(ma_format format)
{
- ma_pcm_u8_to_s32__optimized(dst, src, count, ditherMode);
+ ma_uint32 sizes[] = {
+ 0, /* unknown */
+ 1, /* u8 */
+ 2, /* s16 */
+ 3, /* s24 */
+ 4, /* s32 */
+ 4, /* f32 */
+ };
+ return sizes[format];
}
-#endif
-MA_API void ma_pcm_u8_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+
+
+MA_API ma_data_source_config ma_data_source_config_init(void)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_u8_to_s32__reference(dst, src, count, ditherMode);
-#else
- # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
- if (ma_has_avx2()) {
- ma_pcm_u8_to_s32__avx2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
- if (ma_has_sse2()) {
- ma_pcm_u8_to_s32__sse2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
- if (ma_has_neon()) {
- ma_pcm_u8_to_s32__neon(dst, src, count, ditherMode);
- } else
- #endif
- {
- ma_pcm_u8_to_s32__optimized(dst, src, count, ditherMode);
- }
-#endif
+ ma_data_source_config config;
+
+ MA_ZERO_OBJECT(&config);
+
+ return config;
}
-static MA_INLINE void ma_pcm_u8_to_f32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_data_source_init(const ma_data_source_config* pConfig, ma_data_source* pDataSource)
{
- float* dst_f32 = (float*)dst;
- const ma_uint8* src_u8 = (const ma_uint8*)src;
+ ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- float x = (float)src_u8[i];
- x = x * 0.00784313725490196078f; /* 0..255 to 0..2 */
- x = x - 1; /* 0..2 to -1..1 */
+ if (pDataSource == NULL) {
+ return MA_INVALID_ARGS;
+ }
- dst_f32[i] = x;
+ MA_ZERO_OBJECT(pDataSourceBase);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- (void)ditherMode;
-}
+ pDataSourceBase->vtable = pConfig->vtable;
+ pDataSourceBase->rangeBegInFrames = 0;
+ pDataSourceBase->rangeEndInFrames = ~((ma_uint64)0);
+ pDataSourceBase->loopBegInFrames = 0;
+ pDataSourceBase->loopEndInFrames = ~((ma_uint64)0);
+ pDataSourceBase->pCurrent = pDataSource; /* Always read from ourself by default. */
+ pDataSourceBase->pNext = NULL;
+ pDataSourceBase->onGetNext = NULL;
-static MA_INLINE void ma_pcm_u8_to_f32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_u8_to_f32__reference(dst, src, count, ditherMode);
+ return MA_SUCCESS;
}
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE void ma_pcm_u8_to_f32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_u8_to_f32__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE void ma_pcm_u8_to_f32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_u8_to_f32__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE void ma_pcm_u8_to_f32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API void ma_data_source_uninit(ma_data_source* pDataSource)
{
- ma_pcm_u8_to_f32__optimized(dst, src, count, ditherMode);
-}
-#endif
+ if (pDataSource == NULL) {
+ return;
+ }
-MA_API void ma_pcm_u8_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_u8_to_f32__reference(dst, src, count, ditherMode);
-#else
- # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
- if (ma_has_avx2()) {
- ma_pcm_u8_to_f32__avx2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
- if (ma_has_sse2()) {
- ma_pcm_u8_to_f32__sse2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
- if (ma_has_neon()) {
- ma_pcm_u8_to_f32__neon(dst, src, count, ditherMode);
- } else
- #endif
- {
- ma_pcm_u8_to_f32__optimized(dst, src, count, ditherMode);
- }
-#endif
+ /*
+ This is placeholder in case we need this later. Data sources need to call this in their
+ uninitialization routine to ensure things work later on if something is added here.
+ */
}
-
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
-static MA_INLINE void ma_pcm_interleave_u8__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
+static ma_result ma_data_source_resolve_current(ma_data_source* pDataSource, ma_data_source** ppCurrentDataSource)
{
- ma_uint8* dst_u8 = (ma_uint8*)dst;
- const ma_uint8** src_u8 = (const ma_uint8**)src;
+ ma_data_source_base* pCurrentDataSource = (ma_data_source_base*)pDataSource;
- ma_uint64 iFrame;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- dst_u8[iFrame*channels + iChannel] = src_u8[iChannel][iFrame];
+ MA_ASSERT(pDataSource != NULL);
+ MA_ASSERT(ppCurrentDataSource != NULL);
+
+ if (pCurrentDataSource->pCurrent == NULL) {
+ /*
+ The current data source is NULL. If we're using this in the context of a chain we need to return NULL
+ here so that we don't end up looping. Otherwise we just return the data source itself.
+ */
+ if (pCurrentDataSource->pNext != NULL || pCurrentDataSource->onGetNext != NULL) {
+ pCurrentDataSource = NULL;
+ } else {
+ pCurrentDataSource = (ma_data_source_base*)pDataSource; /* Not being used in a chain. Make sure we just always read from the data source itself at all times. */
}
+ } else {
+ pCurrentDataSource = (ma_data_source_base*)pCurrentDataSource->pCurrent;
}
+
+ *ppCurrentDataSource = pCurrentDataSource;
+
+ return MA_SUCCESS;
}
-#else
-static MA_INLINE void ma_pcm_interleave_u8__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
+
+static ma_result ma_data_source_read_pcm_frames_within_range(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
- ma_uint8* dst_u8 = (ma_uint8*)dst;
- const ma_uint8** src_u8 = (const ma_uint8**)src;
+ ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
+ ma_result result;
+ ma_uint64 framesRead = 0;
+ ma_bool32 loop = ma_data_source_is_looping(pDataSource);
- if (channels == 1) {
- ma_copy_memory_64(dst, src[0], frameCount * sizeof(ma_uint8));
- } else if (channels == 2) {
- ma_uint64 iFrame;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- dst_u8[iFrame*2 + 0] = src_u8[0][iFrame];
- dst_u8[iFrame*2 + 1] = src_u8[1][iFrame];
- }
+ if (pDataSourceBase == NULL) {
+ return MA_AT_END;
+ }
+
+ if (frameCount == 0) {
+ return MA_INVALID_ARGS;
+ }
+
+ if ((pDataSourceBase->vtable->flags & MA_DATA_SOURCE_SELF_MANAGED_RANGE_AND_LOOP_POINT) != 0 || (pDataSourceBase->rangeEndInFrames == ~((ma_uint64)0) && (pDataSourceBase->loopEndInFrames == ~((ma_uint64)0) || loop == MA_FALSE))) {
+ /* Either the data source is self-managing the range, or no range is set - just read like normal. The data source itself will tell us when the end is reached. */
+ result = pDataSourceBase->vtable->onRead(pDataSourceBase, pFramesOut, frameCount, &framesRead);
} else {
- ma_uint64 iFrame;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- dst_u8[iFrame*channels + iChannel] = src_u8[iChannel][iFrame];
+ /* Need to clamp to within the range. */
+ ma_uint64 cursor;
+
+ result = ma_data_source_get_cursor_in_pcm_frames(pDataSourceBase, &cursor);
+ if (result != MA_SUCCESS) {
+ /* Failed to retrieve the cursor. Cannot read within a range or loop points. Just read like normal - this may happen for things like noise data sources where it doesn't really matter. */
+ result = pDataSourceBase->vtable->onRead(pDataSourceBase, pFramesOut, frameCount, &framesRead);
+ } else {
+ ma_uint64 rangeEnd;
+
+ /* We have the cursor. We need to make sure we don't read beyond our range. */
+ rangeEnd = pDataSourceBase->rangeEndInFrames;
+
+ /* If looping, make sure we're within range. */
+ if (loop) {
+ if (pDataSourceBase->loopEndInFrames != ~((ma_uint64)0)) {
+ rangeEnd = ma_min(rangeEnd, pDataSourceBase->rangeBegInFrames + pDataSourceBase->loopEndInFrames);
+ }
+ }
+
+ if (frameCount > (rangeEnd - cursor) && rangeEnd != ~((ma_uint64)0)) {
+ frameCount = (rangeEnd - cursor);
+ }
+
+ /*
+ If the cursor is sitting on the end of the range the frame count will be set to 0 which can
+ result in MA_INVALID_ARGS. In this case, we don't want to try reading, but instead return
+ MA_AT_END so the higher level function can know about it.
+ */
+ if (frameCount > 0) {
+ result = pDataSourceBase->vtable->onRead(pDataSourceBase, pFramesOut, frameCount, &framesRead);
+ } else {
+ result = MA_AT_END; /* The cursor is sitting on the end of the range which means we're at the end. */
}
}
}
+
+ if (pFramesRead != NULL) {
+ *pFramesRead = framesRead;
+ }
+
+ /* We need to make sure MA_AT_END is returned if we hit the end of the range. */
+ if (result == MA_SUCCESS && framesRead == 0) {
+ result = MA_AT_END;
+ }
+
+ return result;
}
-#endif
-MA_API void ma_pcm_interleave_u8(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
+MA_API ma_result ma_data_source_read_pcm_frames(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_interleave_u8__reference(dst, src, frameCount, channels);
-#else
- ma_pcm_interleave_u8__optimized(dst, src, frameCount, channels);
-#endif
-}
+ ma_result result = MA_SUCCESS;
+ ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
+ ma_data_source_base* pCurrentDataSource;
+ void* pRunningFramesOut = pFramesOut;
+ ma_uint64 totalFramesProcessed = 0;
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 emptyLoopCounter = 0; /* Keeps track of how many times 0 frames have been read. For infinite loop detection of sounds with no audio data. */
+ ma_bool32 loop;
+ if (pFramesRead != NULL) {
+ *pFramesRead = 0;
+ }
-static MA_INLINE void ma_pcm_deinterleave_u8__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
-{
- ma_uint8** dst_u8 = (ma_uint8**)dst;
- const ma_uint8* src_u8 = (const ma_uint8*)src;
+ if (frameCount == 0) {
+ return MA_INVALID_ARGS;
+ }
- ma_uint64 iFrame;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- dst_u8[iChannel][iFrame] = src_u8[iFrame*channels + iChannel];
+ if (pDataSourceBase == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ loop = ma_data_source_is_looping(pDataSource);
+
+ /*
+ We need to know the data format so we can advance the output buffer as we read frames. If this
+ fails, chaining will not work and we'll just read as much as we can from the current source.
+ */
+ if (ma_data_source_get_data_format(pDataSource, &format, &channels, NULL, NULL, 0) != MA_SUCCESS) {
+ result = ma_data_source_resolve_current(pDataSource, (ma_data_source**)&pCurrentDataSource);
+ if (result != MA_SUCCESS) {
+ return result;
}
+
+ return ma_data_source_read_pcm_frames_within_range(pCurrentDataSource, pFramesOut, frameCount, pFramesRead);
}
-}
-static MA_INLINE void ma_pcm_deinterleave_u8__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
-{
- ma_pcm_deinterleave_u8__reference(dst, src, frameCount, channels);
-}
+ /*
+ Looping is a bit of a special case. When the `loop` argument is true, chaining will not work and
+ only the current data source will be read from.
+ */
-MA_API void ma_pcm_deinterleave_u8(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
-{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_deinterleave_u8__reference(dst, src, frameCount, channels);
-#else
- ma_pcm_deinterleave_u8__optimized(dst, src, frameCount, channels);
-#endif
-}
+ /* Keep reading until we've read as many frames as possible. */
+ while (totalFramesProcessed < frameCount) {
+ ma_uint64 framesProcessed;
+ ma_uint64 framesRemaining = frameCount - totalFramesProcessed;
+
+ /* We need to resolve the data source that we'll actually be reading from. */
+ result = ma_data_source_resolve_current(pDataSource, (ma_data_source**)&pCurrentDataSource);
+ if (result != MA_SUCCESS) {
+ break;
+ }
+ if (pCurrentDataSource == NULL) {
+ break;
+ }
-/* s16 */
-static MA_INLINE void ma_pcm_s16_to_u8__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_uint8* dst_u8 = (ma_uint8*)dst;
- const ma_int16* src_s16 = (const ma_int16*)src;
+ result = ma_data_source_read_pcm_frames_within_range(pCurrentDataSource, pRunningFramesOut, framesRemaining, &framesProcessed);
+ totalFramesProcessed += framesProcessed;
- if (ditherMode == ma_dither_mode_none) {
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- ma_int16 x = src_s16[i];
- x = (ma_int16)(x >> 8);
- x = (ma_int16)(x + 128);
- dst_u8[i] = (ma_uint8)x;
+ /*
+ If we encounted an error from the read callback, make sure it's propagated to the caller. The caller may need to know whether or not MA_BUSY is returned which is
+ not necessarily considered an error.
+ */
+ if (result != MA_SUCCESS && result != MA_AT_END) {
+ break;
}
- } else {
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- ma_int16 x = src_s16[i];
- /* Dither. Don't overflow. */
- ma_int32 dither = ma_dither_s32(ditherMode, -0x80, 0x7F);
- if ((x + dither) <= 0x7FFF) {
- x = (ma_int16)(x + dither);
+ /*
+ We can determine if we've reached the end by checking if ma_data_source_read_pcm_frames_within_range() returned
+ MA_AT_END. To loop back to the start, all we need to do is seek back to the first frame.
+ */
+ if (result == MA_AT_END) {
+ /*
+ The result needs to be reset back to MA_SUCCESS (from MA_AT_END) so that we don't
+ accidentally return MA_AT_END when data has been read in prior loop iterations. at the
+ end of this function, the result will be checked for MA_SUCCESS, and if the total
+ number of frames processed is 0, will be explicitly set to MA_AT_END.
+ */
+ result = MA_SUCCESS;
+
+ /*
+ We reached the end. If we're looping, we just loop back to the start of the current
+ data source. If we're not looping we need to check if we have another in the chain, and
+ if so, switch to it.
+ */
+ if (loop) {
+ if (framesProcessed == 0) {
+ emptyLoopCounter += 1;
+ if (emptyLoopCounter > 1) {
+ break; /* Infinite loop detected. Get out. */
+ }
+ } else {
+ emptyLoopCounter = 0;
+ }
+
+ result = ma_data_source_seek_to_pcm_frame(pCurrentDataSource, pCurrentDataSource->loopBegInFrames);
+ if (result != MA_SUCCESS) {
+ break; /* Failed to loop. Abort. */
+ }
+
+ /* Don't return MA_AT_END for looping sounds. */
+ result = MA_SUCCESS;
} else {
- x = 0x7FFF;
+ if (pCurrentDataSource->pNext != NULL) {
+ pDataSourceBase->pCurrent = pCurrentDataSource->pNext;
+ } else if (pCurrentDataSource->onGetNext != NULL) {
+ pDataSourceBase->pCurrent = pCurrentDataSource->onGetNext(pCurrentDataSource);
+ if (pDataSourceBase->pCurrent == NULL) {
+ break; /* Our callback did not return a next data source. We're done. */
+ }
+ } else {
+ /* Reached the end of the chain. We're done. */
+ break;
+ }
+
+ /* The next data source needs to be rewound to ensure data is read in looping scenarios. */
+ result = ma_data_source_seek_to_pcm_frame(pDataSourceBase->pCurrent, 0);
+ if (result != MA_SUCCESS) {
+ break;
+ }
}
+ }
- x = (ma_int16)(x >> 8);
- x = (ma_int16)(x + 128);
- dst_u8[i] = (ma_uint8)x;
+ if (pRunningFramesOut != NULL) {
+ pRunningFramesOut = ma_offset_ptr(pRunningFramesOut, framesProcessed * ma_get_bytes_per_frame(format, channels));
}
}
-}
-static MA_INLINE void ma_pcm_s16_to_u8__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s16_to_u8__reference(dst, src, count, ditherMode);
+ if (pFramesRead != NULL) {
+ *pFramesRead = totalFramesProcessed;
+ }
+
+ MA_ASSERT(!(result == MA_AT_END && totalFramesProcessed > 0)); /* We should never be returning MA_AT_END if we read some data. */
+
+ if (result == MA_SUCCESS && totalFramesProcessed == 0) {
+ result = MA_AT_END;
+ }
+
+ return result;
}
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE void ma_pcm_s16_to_u8__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_data_source_seek_pcm_frames(ma_data_source* pDataSource, ma_uint64 frameCount, ma_uint64* pFramesSeeked)
{
- ma_pcm_s16_to_u8__optimized(dst, src, count, ditherMode);
+ return ma_data_source_read_pcm_frames(pDataSource, NULL, frameCount, pFramesSeeked);
}
-#endif
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE void ma_pcm_s16_to_u8__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+
+MA_API ma_result ma_data_source_seek_to_pcm_frame(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
- ma_pcm_s16_to_u8__optimized(dst, src, count, ditherMode);
+ ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
+
+ if (pDataSourceBase == NULL) {
+ return MA_SUCCESS;
+ }
+
+ if (pDataSourceBase->vtable->onSeek == NULL) {
+ return MA_NOT_IMPLEMENTED;
+ }
+
+ if (frameIndex > pDataSourceBase->rangeEndInFrames) {
+ return MA_INVALID_OPERATION; /* Trying to seek to far forward. */
+ }
+
+ return pDataSourceBase->vtable->onSeek(pDataSource, pDataSourceBase->rangeBegInFrames + frameIndex);
}
-#endif
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE void ma_pcm_s16_to_u8__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+
+MA_API ma_result ma_data_source_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
- ma_pcm_s16_to_u8__optimized(dst, src, count, ditherMode);
+ ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
+ ma_result result;
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+
+ /* Initialize to defaults for safety just in case the data source does not implement this callback. */
+ if (pFormat != NULL) {
+ *pFormat = ma_format_unknown;
+ }
+ if (pChannels != NULL) {
+ *pChannels = 0;
+ }
+ if (pSampleRate != NULL) {
+ *pSampleRate = 0;
+ }
+ if (pChannelMap != NULL) {
+ MA_ZERO_MEMORY(pChannelMap, sizeof(*pChannelMap) * channelMapCap);
+ }
+
+ if (pDataSourceBase == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (pDataSourceBase->vtable->onGetDataFormat == NULL) {
+ return MA_NOT_IMPLEMENTED;
+ }
+
+ result = pDataSourceBase->vtable->onGetDataFormat(pDataSource, &format, &channels, &sampleRate, pChannelMap, channelMapCap);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ if (pFormat != NULL) {
+ *pFormat = format;
+ }
+ if (pChannels != NULL) {
+ *pChannels = channels;
+ }
+ if (pSampleRate != NULL) {
+ *pSampleRate = sampleRate;
+ }
+
+ /* Channel map was passed in directly to the callback. This is safe due to the channelMapCap parameter. */
+
+ return MA_SUCCESS;
}
-#endif
-MA_API void ma_pcm_s16_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_data_source_get_cursor_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pCursor)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_s16_to_u8__reference(dst, src, count, ditherMode);
-#else
- # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
- if (ma_has_avx2()) {
- ma_pcm_s16_to_u8__avx2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
- if (ma_has_sse2()) {
- ma_pcm_s16_to_u8__sse2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
- if (ma_has_neon()) {
- ma_pcm_s16_to_u8__neon(dst, src, count, ditherMode);
- } else
- #endif
- {
- ma_pcm_s16_to_u8__optimized(dst, src, count, ditherMode);
- }
-#endif
-}
+ ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
+ ma_result result;
+ ma_uint64 cursor;
+ if (pCursor == NULL) {
+ return MA_INVALID_ARGS;
+ }
-MA_API void ma_pcm_s16_to_s16(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- (void)ditherMode;
- ma_copy_memory_64(dst, src, count * sizeof(ma_int16));
-}
+ *pCursor = 0;
+ if (pDataSourceBase == NULL) {
+ return MA_SUCCESS;
+ }
-static MA_INLINE void ma_pcm_s16_to_s24__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_uint8* dst_s24 = (ma_uint8*)dst;
- const ma_int16* src_s16 = (const ma_int16*)src;
+ if (pDataSourceBase->vtable->onGetCursor == NULL) {
+ return MA_NOT_IMPLEMENTED;
+ }
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- dst_s24[i*3+0] = 0;
- dst_s24[i*3+1] = (ma_uint8)(src_s16[i] & 0xFF);
- dst_s24[i*3+2] = (ma_uint8)(src_s16[i] >> 8);
+ result = pDataSourceBase->vtable->onGetCursor(pDataSourceBase, &cursor);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ /* The cursor needs to be made relative to the start of the range. */
+ if (cursor < pDataSourceBase->rangeBegInFrames) { /* Safety check so we don't return some huge number. */
+ *pCursor = 0;
+ } else {
+ *pCursor = cursor - pDataSourceBase->rangeBegInFrames;
}
- (void)ditherMode;
+ return MA_SUCCESS;
}
-static MA_INLINE void ma_pcm_s16_to_s24__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_data_source_get_length_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pLength)
{
- ma_pcm_s16_to_s24__reference(dst, src, count, ditherMode);
-}
+ ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE void ma_pcm_s16_to_s24__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s16_to_s24__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE void ma_pcm_s16_to_s24__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s16_to_s24__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE void ma_pcm_s16_to_s24__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s16_to_s24__optimized(dst, src, count, ditherMode);
-}
-#endif
+ if (pLength == NULL) {
+ return MA_INVALID_ARGS;
+ }
-MA_API void ma_pcm_s16_to_s24(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_s16_to_s24__reference(dst, src, count, ditherMode);
-#else
- # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
- if (ma_has_avx2()) {
- ma_pcm_s16_to_s24__avx2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
- if (ma_has_sse2()) {
- ma_pcm_s16_to_s24__sse2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
- if (ma_has_neon()) {
- ma_pcm_s16_to_s24__neon(dst, src, count, ditherMode);
- } else
- #endif
- {
- ma_pcm_s16_to_s24__optimized(dst, src, count, ditherMode);
- }
-#endif
-}
+ *pLength = 0;
+ if (pDataSourceBase == NULL) {
+ return MA_INVALID_ARGS;
+ }
-static MA_INLINE void ma_pcm_s16_to_s32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_int32* dst_s32 = (ma_int32*)dst;
- const ma_int16* src_s16 = (const ma_int16*)src;
+ /*
+ If we have a range defined we'll use that to determine the length. This is one of rare times
+ where we'll actually trust the caller. If they've set the range, I think it's mostly safe to
+ assume they've set it based on some higher level knowledge of the structure of the sound bank.
+ */
+ if (pDataSourceBase->rangeEndInFrames != ~((ma_uint64)0)) {
+ *pLength = pDataSourceBase->rangeEndInFrames - pDataSourceBase->rangeBegInFrames;
+ return MA_SUCCESS;
+ }
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- dst_s32[i] = src_s16[i] << 16;
+ /*
+ Getting here means a range is not defined so we'll need to get the data source itself to tell
+ us the length.
+ */
+ if (pDataSourceBase->vtable->onGetLength == NULL) {
+ return MA_NOT_IMPLEMENTED;
}
- (void)ditherMode;
+ return pDataSourceBase->vtable->onGetLength(pDataSource, pLength);
}
-static MA_INLINE void ma_pcm_s16_to_s32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_data_source_get_cursor_in_seconds(ma_data_source* pDataSource, float* pCursor)
{
- ma_pcm_s16_to_s32__reference(dst, src, count, ditherMode);
-}
+ ma_result result;
+ ma_uint64 cursorInPCMFrames;
+ ma_uint32 sampleRate;
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE void ma_pcm_s16_to_s32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s16_to_s32__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE void ma_pcm_s16_to_s32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s16_to_s32__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE void ma_pcm_s16_to_s32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s16_to_s32__optimized(dst, src, count, ditherMode);
-}
-#endif
+ if (pCursor == NULL) {
+ return MA_INVALID_ARGS;
+ }
-MA_API void ma_pcm_s16_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_s16_to_s32__reference(dst, src, count, ditherMode);
-#else
- # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
- if (ma_has_avx2()) {
- ma_pcm_s16_to_s32__avx2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
- if (ma_has_sse2()) {
- ma_pcm_s16_to_s32__sse2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
- if (ma_has_neon()) {
- ma_pcm_s16_to_s32__neon(dst, src, count, ditherMode);
- } else
- #endif
- {
- ma_pcm_s16_to_s32__optimized(dst, src, count, ditherMode);
- }
-#endif
-}
+ *pCursor = 0;
+
+ result = ma_data_source_get_cursor_in_pcm_frames(pDataSource, &cursorInPCMFrames);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ result = ma_data_source_get_data_format(pDataSource, NULL, NULL, &sampleRate, NULL, 0);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ *pCursor = cursorInPCMFrames / (float)sampleRate;
-static MA_INLINE void ma_pcm_s16_to_f32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+ return MA_SUCCESS;
+}
+
+MA_API ma_result ma_data_source_get_length_in_seconds(ma_data_source* pDataSource, float* pLength)
{
- float* dst_f32 = (float*)dst;
- const ma_int16* src_s16 = (const ma_int16*)src;
+ ma_result result;
+ ma_uint64 lengthInPCMFrames;
+ ma_uint32 sampleRate;
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- float x = (float)src_s16[i];
+ if (pLength == NULL) {
+ return MA_INVALID_ARGS;
+ }
-#if 0
- /* The accurate way. */
- x = x + 32768.0f; /* -32768..32767 to 0..65535 */
- x = x * 0.00003051804379339284f; /* 0..65535 to 0..2 */
- x = x - 1; /* 0..2 to -1..1 */
-#else
- /* The fast way. */
- x = x * 0.000030517578125f; /* -32768..32767 to -1..0.999969482421875 */
-#endif
+ *pLength = 0;
- dst_f32[i] = x;
+ result = ma_data_source_get_length_in_pcm_frames(pDataSource, &lengthInPCMFrames);
+ if (result != MA_SUCCESS) {
+ return result;
}
- (void)ditherMode;
-}
+ result = ma_data_source_get_data_format(pDataSource, NULL, NULL, &sampleRate, NULL, 0);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
-static MA_INLINE void ma_pcm_s16_to_f32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s16_to_f32__reference(dst, src, count, ditherMode);
-}
+ *pLength = lengthInPCMFrames / (float)sampleRate;
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE void ma_pcm_s16_to_f32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s16_to_f32__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE void ma_pcm_s16_to_f32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s16_to_f32__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE void ma_pcm_s16_to_f32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s16_to_f32__optimized(dst, src, count, ditherMode);
+ return MA_SUCCESS;
}
-#endif
-MA_API void ma_pcm_s16_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_data_source_set_looping(ma_data_source* pDataSource, ma_bool32 isLooping)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_s16_to_f32__reference(dst, src, count, ditherMode);
-#else
- # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
- if (ma_has_avx2()) {
- ma_pcm_s16_to_f32__avx2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
- if (ma_has_sse2()) {
- ma_pcm_s16_to_f32__sse2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
- if (ma_has_neon()) {
- ma_pcm_s16_to_f32__neon(dst, src, count, ditherMode);
- } else
- #endif
- {
- ma_pcm_s16_to_f32__optimized(dst, src, count, ditherMode);
- }
-#endif
-}
+ ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
+ if (pDataSource == NULL) {
+ return MA_INVALID_ARGS;
+ }
-static MA_INLINE void ma_pcm_interleave_s16__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
-{
- ma_int16* dst_s16 = (ma_int16*)dst;
- const ma_int16** src_s16 = (const ma_int16**)src;
+ c89atomic_exchange_32(&pDataSourceBase->isLooping, isLooping);
- ma_uint64 iFrame;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- dst_s16[iFrame*channels + iChannel] = src_s16[iChannel][iFrame];
- }
+ /* If there's no callback for this just treat it as a successful no-op. */
+ if (pDataSourceBase->vtable->onSetLooping == NULL) {
+ return MA_SUCCESS;
}
-}
-static MA_INLINE void ma_pcm_interleave_s16__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
-{
- ma_pcm_interleave_s16__reference(dst, src, frameCount, channels);
+ return pDataSourceBase->vtable->onSetLooping(pDataSource, isLooping);
}
-MA_API void ma_pcm_interleave_s16(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
+MA_API ma_bool32 ma_data_source_is_looping(ma_data_source* pDataSource)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_interleave_s16__reference(dst, src, frameCount, channels);
-#else
- ma_pcm_interleave_s16__optimized(dst, src, frameCount, channels);
-#endif
-}
+ ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
+
+ if (pDataSource == NULL) {
+ return MA_FALSE;
+ }
+ return c89atomic_load_32(&pDataSourceBase->isLooping);
+}
-static MA_INLINE void ma_pcm_deinterleave_s16__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
+MA_API ma_result ma_data_source_set_range_in_pcm_frames(ma_data_source* pDataSource, ma_uint64 rangeBegInFrames, ma_uint64 rangeEndInFrames)
{
- ma_int16** dst_s16 = (ma_int16**)dst;
- const ma_int16* src_s16 = (const ma_int16*)src;
+ ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
+ ma_result result;
+ ma_uint64 cursor;
+ ma_uint64 loopBegAbsolute;
+ ma_uint64 loopEndAbsolute;
- ma_uint64 iFrame;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- dst_s16[iChannel][iFrame] = src_s16[iFrame*channels + iChannel];
- }
+ if (pDataSource == NULL) {
+ return MA_INVALID_ARGS;
}
-}
-static MA_INLINE void ma_pcm_deinterleave_s16__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
-{
- ma_pcm_deinterleave_s16__reference(dst, src, frameCount, channels);
-}
+ if (rangeEndInFrames < rangeBegInFrames) {
+ return MA_INVALID_ARGS; /* The end of the range must come after the beginning. */
+ }
-MA_API void ma_pcm_deinterleave_s16(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
-{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_deinterleave_s16__reference(dst, src, frameCount, channels);
-#else
- ma_pcm_deinterleave_s16__optimized(dst, src, frameCount, channels);
-#endif
-}
+ /*
+ The loop points need to be updated. We'll be storing the loop points relative to the range. We'll update
+ these so that they maintain their absolute positioning. The loop points will then be clamped to the range.
+ */
+ loopBegAbsolute = pDataSourceBase->loopBegInFrames + pDataSourceBase->rangeBegInFrames;
+ loopEndAbsolute = pDataSourceBase->loopEndInFrames + ((pDataSourceBase->loopEndInFrames != ~((ma_uint64)0)) ? pDataSourceBase->rangeBegInFrames : 0);
+ pDataSourceBase->rangeBegInFrames = rangeBegInFrames;
+ pDataSourceBase->rangeEndInFrames = rangeEndInFrames;
-/* s24 */
-static MA_INLINE void ma_pcm_s24_to_u8__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_uint8* dst_u8 = (ma_uint8*)dst;
- const ma_uint8* src_s24 = (const ma_uint8*)src;
+ /* Make the loop points relative again, and make sure they're clamped to within the range. */
+ if (loopBegAbsolute > pDataSourceBase->rangeBegInFrames) {
+ pDataSourceBase->loopBegInFrames = loopBegAbsolute - pDataSourceBase->rangeBegInFrames;
+ } else {
+ pDataSourceBase->loopBegInFrames = 0;
+ }
- if (ditherMode == ma_dither_mode_none) {
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- dst_u8[i] = (ma_uint8)((ma_int8)src_s24[i*3 + 2] + 128);
+ if (pDataSourceBase->loopBegInFrames > pDataSourceBase->rangeEndInFrames) {
+ pDataSourceBase->loopBegInFrames = pDataSourceBase->rangeEndInFrames;
+ }
+
+ /* Only need to update the loop end point if it's not -1. */
+ if (loopEndAbsolute != ~((ma_uint64)0)) {
+ if (loopEndAbsolute > pDataSourceBase->rangeBegInFrames) {
+ pDataSourceBase->loopEndInFrames = loopEndAbsolute - pDataSourceBase->rangeBegInFrames;
+ } else {
+ pDataSourceBase->loopEndInFrames = 0;
}
- } else {
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- ma_int32 x = (ma_int32)(((ma_uint32)(src_s24[i*3+0]) << 8) | ((ma_uint32)(src_s24[i*3+1]) << 16) | ((ma_uint32)(src_s24[i*3+2])) << 24);
- /* Dither. Don't overflow. */
- ma_int32 dither = ma_dither_s32(ditherMode, -0x800000, 0x7FFFFF);
- if ((ma_int64)x + dither <= 0x7FFFFFFF) {
- x = x + dither;
- } else {
- x = 0x7FFFFFFF;
- }
+ if (pDataSourceBase->loopEndInFrames > pDataSourceBase->rangeEndInFrames && pDataSourceBase->loopEndInFrames) {
+ pDataSourceBase->loopEndInFrames = pDataSourceBase->rangeEndInFrames;
+ }
+ }
- x = x >> 24;
- x = x + 128;
- dst_u8[i] = (ma_uint8)x;
+
+ /* If the new range is past the current cursor position we need to seek to it. */
+ result = ma_data_source_get_cursor_in_pcm_frames(pDataSource, &cursor);
+ if (result == MA_SUCCESS) {
+ /* Seek to within range. Note that our seek positions here are relative to the new range. */
+ if (cursor < rangeBegInFrames) {
+ ma_data_source_seek_to_pcm_frame(pDataSource, 0);
+ } else if (cursor > rangeEndInFrames) {
+ ma_data_source_seek_to_pcm_frame(pDataSource, rangeEndInFrames - rangeBegInFrames);
}
+ } else {
+ /* We failed to get the cursor position. Probably means the data source has no notion of a cursor such a noise data source. Just pretend the seeking worked. */
}
-}
-static MA_INLINE void ma_pcm_s24_to_u8__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s24_to_u8__reference(dst, src, count, ditherMode);
+ return MA_SUCCESS;
}
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE void ma_pcm_s24_to_u8__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s24_to_u8__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE void ma_pcm_s24_to_u8__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s24_to_u8__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE void ma_pcm_s24_to_u8__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API void ma_data_source_get_range_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pRangeBegInFrames, ma_uint64* pRangeEndInFrames)
{
- ma_pcm_s24_to_u8__optimized(dst, src, count, ditherMode);
+ ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
+
+ if (pDataSource == NULL) {
+ return;
+ }
+
+ if (pRangeBegInFrames != NULL) {
+ *pRangeBegInFrames = pDataSourceBase->rangeBegInFrames;
+ }
+
+ if (pRangeEndInFrames != NULL) {
+ *pRangeEndInFrames = pDataSourceBase->rangeEndInFrames;
+ }
}
-#endif
-MA_API void ma_pcm_s24_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_data_source_set_loop_point_in_pcm_frames(ma_data_source* pDataSource, ma_uint64 loopBegInFrames, ma_uint64 loopEndInFrames)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_s24_to_u8__reference(dst, src, count, ditherMode);
-#else
- # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
- if (ma_has_avx2()) {
- ma_pcm_s24_to_u8__avx2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
- if (ma_has_sse2()) {
- ma_pcm_s24_to_u8__sse2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
- if (ma_has_neon()) {
- ma_pcm_s24_to_u8__neon(dst, src, count, ditherMode);
- } else
- #endif
- {
- ma_pcm_s24_to_u8__optimized(dst, src, count, ditherMode);
- }
-#endif
-}
+ ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
+ if (pDataSource == NULL) {
+ return MA_INVALID_ARGS;
+ }
-static MA_INLINE void ma_pcm_s24_to_s16__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_int16* dst_s16 = (ma_int16*)dst;
- const ma_uint8* src_s24 = (const ma_uint8*)src;
+ if (loopEndInFrames < loopBegInFrames) {
+ return MA_INVALID_ARGS; /* The end of the loop point must come after the beginning. */
+ }
- if (ditherMode == ma_dither_mode_none) {
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- ma_uint16 dst_lo = ((ma_uint16)src_s24[i*3 + 1]);
- ma_uint16 dst_hi = (ma_uint16)((ma_uint16)src_s24[i*3 + 2] << 8);
- dst_s16[i] = (ma_int16)(dst_lo | dst_hi);
- }
- } else {
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- ma_int32 x = (ma_int32)(((ma_uint32)(src_s24[i*3+0]) << 8) | ((ma_uint32)(src_s24[i*3+1]) << 16) | ((ma_uint32)(src_s24[i*3+2])) << 24);
+ if (loopEndInFrames > pDataSourceBase->rangeEndInFrames && loopEndInFrames != ~((ma_uint64)0)) {
+ return MA_INVALID_ARGS; /* The end of the loop point must not go beyond the range. */
+ }
- /* Dither. Don't overflow. */
- ma_int32 dither = ma_dither_s32(ditherMode, -0x8000, 0x7FFF);
- if ((ma_int64)x + dither <= 0x7FFFFFFF) {
- x = x + dither;
- } else {
- x = 0x7FFFFFFF;
- }
+ pDataSourceBase->loopBegInFrames = loopBegInFrames;
+ pDataSourceBase->loopEndInFrames = loopEndInFrames;
- x = x >> 16;
- dst_s16[i] = (ma_int16)x;
- }
+ /* The end cannot exceed the range. */
+ if (pDataSourceBase->loopEndInFrames > (pDataSourceBase->rangeEndInFrames - pDataSourceBase->rangeBegInFrames) && pDataSourceBase->loopEndInFrames != ~((ma_uint64)0)) {
+ pDataSourceBase->loopEndInFrames = (pDataSourceBase->rangeEndInFrames - pDataSourceBase->rangeBegInFrames);
}
-}
-static MA_INLINE void ma_pcm_s24_to_s16__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s24_to_s16__reference(dst, src, count, ditherMode);
+ return MA_SUCCESS;
}
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE void ma_pcm_s24_to_s16__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API void ma_data_source_get_loop_point_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pLoopBegInFrames, ma_uint64* pLoopEndInFrames)
{
- ma_pcm_s24_to_s16__optimized(dst, src, count, ditherMode);
+ ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
+
+ if (pDataSource == NULL) {
+ return;
+ }
+
+ if (pLoopBegInFrames != NULL) {
+ *pLoopBegInFrames = pDataSourceBase->loopBegInFrames;
+ }
+
+ if (pLoopEndInFrames != NULL) {
+ *pLoopEndInFrames = pDataSourceBase->loopEndInFrames;
+ }
}
-#endif
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE void ma_pcm_s24_to_s16__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+
+MA_API ma_result ma_data_source_set_current(ma_data_source* pDataSource, ma_data_source* pCurrentDataSource)
{
- ma_pcm_s24_to_s16__optimized(dst, src, count, ditherMode);
+ ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
+
+ if (pDataSource == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ pDataSourceBase->pCurrent = pCurrentDataSource;
+
+ return MA_SUCCESS;
}
-#endif
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE void ma_pcm_s24_to_s16__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+
+MA_API ma_data_source* ma_data_source_get_current(ma_data_source* pDataSource)
{
- ma_pcm_s24_to_s16__optimized(dst, src, count, ditherMode);
+ ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
+
+ if (pDataSource == NULL) {
+ return NULL;
+ }
+
+ return pDataSourceBase->pCurrent;
}
-#endif
-MA_API void ma_pcm_s24_to_s16(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_data_source_set_next(ma_data_source* pDataSource, ma_data_source* pNextDataSource)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_s24_to_s16__reference(dst, src, count, ditherMode);
-#else
- # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
- if (ma_has_avx2()) {
- ma_pcm_s24_to_s16__avx2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
- if (ma_has_sse2()) {
- ma_pcm_s24_to_s16__sse2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
- if (ma_has_neon()) {
- ma_pcm_s24_to_s16__neon(dst, src, count, ditherMode);
- } else
- #endif
- {
- ma_pcm_s24_to_s16__optimized(dst, src, count, ditherMode);
- }
-#endif
-}
+ ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
+ if (pDataSource == NULL) {
+ return MA_INVALID_ARGS;
+ }
-MA_API void ma_pcm_s24_to_s24(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- (void)ditherMode;
+ pDataSourceBase->pNext = pNextDataSource;
- ma_copy_memory_64(dst, src, count * 3);
+ return MA_SUCCESS;
}
-
-static MA_INLINE void ma_pcm_s24_to_s32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_data_source* ma_data_source_get_next(ma_data_source* pDataSource)
{
- ma_int32* dst_s32 = (ma_int32*)dst;
- const ma_uint8* src_s24 = (const ma_uint8*)src;
+ ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- dst_s32[i] = (ma_int32)(((ma_uint32)(src_s24[i*3+0]) << 8) | ((ma_uint32)(src_s24[i*3+1]) << 16) | ((ma_uint32)(src_s24[i*3+2])) << 24);
+ if (pDataSource == NULL) {
+ return NULL;
}
- (void)ditherMode;
+ return pDataSourceBase->pNext;
}
-static MA_INLINE void ma_pcm_s24_to_s32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_data_source_set_next_callback(ma_data_source* pDataSource, ma_data_source_get_next_proc onGetNext)
{
- ma_pcm_s24_to_s32__reference(dst, src, count, ditherMode);
-}
+ ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE void ma_pcm_s24_to_s32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s24_to_s32__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE void ma_pcm_s24_to_s32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s24_to_s32__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE void ma_pcm_s24_to_s32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s24_to_s32__optimized(dst, src, count, ditherMode);
+ if (pDataSource == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ pDataSourceBase->onGetNext = onGetNext;
+
+ return MA_SUCCESS;
}
-#endif
-MA_API void ma_pcm_s24_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_data_source_get_next_proc ma_data_source_get_next_callback(ma_data_source* pDataSource)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_s24_to_s32__reference(dst, src, count, ditherMode);
-#else
- # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
- if (ma_has_avx2()) {
- ma_pcm_s24_to_s32__avx2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
- if (ma_has_sse2()) {
- ma_pcm_s24_to_s32__sse2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
- if (ma_has_neon()) {
- ma_pcm_s24_to_s32__neon(dst, src, count, ditherMode);
- } else
- #endif
- {
- ma_pcm_s24_to_s32__optimized(dst, src, count, ditherMode);
- }
-#endif
+ ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
+
+ if (pDataSource == NULL) {
+ return NULL;
+ }
+
+ return pDataSourceBase->onGetNext;
}
-static MA_INLINE void ma_pcm_s24_to_f32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+static ma_result ma_audio_buffer_ref__data_source_on_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
- float* dst_f32 = (float*)dst;
- const ma_uint8* src_s24 = (const ma_uint8*)src;
-
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- float x = (float)(((ma_int32)(((ma_uint32)(src_s24[i*3+0]) << 8) | ((ma_uint32)(src_s24[i*3+1]) << 16) | ((ma_uint32)(src_s24[i*3+2])) << 24)) >> 8);
+ ma_audio_buffer_ref* pAudioBufferRef = (ma_audio_buffer_ref*)pDataSource;
+ ma_uint64 framesRead = ma_audio_buffer_ref_read_pcm_frames(pAudioBufferRef, pFramesOut, frameCount, MA_FALSE);
-#if 0
- /* The accurate way. */
- x = x + 8388608.0f; /* -8388608..8388607 to 0..16777215 */
- x = x * 0.00000011920929665621f; /* 0..16777215 to 0..2 */
- x = x - 1; /* 0..2 to -1..1 */
-#else
- /* The fast way. */
- x = x * 0.00000011920928955078125f; /* -8388608..8388607 to -1..0.999969482421875 */
-#endif
+ if (pFramesRead != NULL) {
+ *pFramesRead = framesRead;
+ }
- dst_f32[i] = x;
+ if (framesRead < frameCount || framesRead == 0) {
+ return MA_AT_END;
}
- (void)ditherMode;
+ return MA_SUCCESS;
}
-static MA_INLINE void ma_pcm_s24_to_f32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+static ma_result ma_audio_buffer_ref__data_source_on_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
- ma_pcm_s24_to_f32__reference(dst, src, count, ditherMode);
+ return ma_audio_buffer_ref_seek_to_pcm_frame((ma_audio_buffer_ref*)pDataSource, frameIndex);
}
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE void ma_pcm_s24_to_f32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+static ma_result ma_audio_buffer_ref__data_source_on_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
- ma_pcm_s24_to_f32__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE void ma_pcm_s24_to_f32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s24_to_f32__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE void ma_pcm_s24_to_f32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s24_to_f32__optimized(dst, src, count, ditherMode);
-}
-#endif
+ ma_audio_buffer_ref* pAudioBufferRef = (ma_audio_buffer_ref*)pDataSource;
-MA_API void ma_pcm_s24_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_s24_to_f32__reference(dst, src, count, ditherMode);
-#else
- # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
- if (ma_has_avx2()) {
- ma_pcm_s24_to_f32__avx2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
- if (ma_has_sse2()) {
- ma_pcm_s24_to_f32__sse2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
- if (ma_has_neon()) {
- ma_pcm_s24_to_f32__neon(dst, src, count, ditherMode);
- } else
- #endif
- {
- ma_pcm_s24_to_f32__optimized(dst, src, count, ditherMode);
- }
-#endif
-}
+ *pFormat = pAudioBufferRef->format;
+ *pChannels = pAudioBufferRef->channels;
+ *pSampleRate = 0; /* There is no notion of a sample rate with audio buffers. */
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, pAudioBufferRef->channels);
+ return MA_SUCCESS;
+}
-static MA_INLINE void ma_pcm_interleave_s24__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
+static ma_result ma_audio_buffer_ref__data_source_on_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
{
- ma_uint8* dst8 = (ma_uint8*)dst;
- const ma_uint8** src8 = (const ma_uint8**)src;
+ ma_audio_buffer_ref* pAudioBufferRef = (ma_audio_buffer_ref*)pDataSource;
- ma_uint64 iFrame;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- dst8[iFrame*3*channels + iChannel*3 + 0] = src8[iChannel][iFrame*3 + 0];
- dst8[iFrame*3*channels + iChannel*3 + 1] = src8[iChannel][iFrame*3 + 1];
- dst8[iFrame*3*channels + iChannel*3 + 2] = src8[iChannel][iFrame*3 + 2];
- }
- }
-}
+ *pCursor = pAudioBufferRef->cursor;
-static MA_INLINE void ma_pcm_interleave_s24__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
-{
- ma_pcm_interleave_s24__reference(dst, src, frameCount, channels);
+ return MA_SUCCESS;
}
-MA_API void ma_pcm_interleave_s24(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
+static ma_result ma_audio_buffer_ref__data_source_on_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_interleave_s24__reference(dst, src, frameCount, channels);
-#else
- ma_pcm_interleave_s24__optimized(dst, src, frameCount, channels);
-#endif
-}
-
+ ma_audio_buffer_ref* pAudioBufferRef = (ma_audio_buffer_ref*)pDataSource;
-static MA_INLINE void ma_pcm_deinterleave_s24__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
-{
- ma_uint8** dst8 = (ma_uint8**)dst;
- const ma_uint8* src8 = (const ma_uint8*)src;
+ *pLength = pAudioBufferRef->sizeInFrames;
- ma_uint32 iFrame;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- dst8[iChannel][iFrame*3 + 0] = src8[iFrame*3*channels + iChannel*3 + 0];
- dst8[iChannel][iFrame*3 + 1] = src8[iFrame*3*channels + iChannel*3 + 1];
- dst8[iChannel][iFrame*3 + 2] = src8[iFrame*3*channels + iChannel*3 + 2];
- }
- }
+ return MA_SUCCESS;
}
-static MA_INLINE void ma_pcm_deinterleave_s24__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
+static ma_data_source_vtable g_ma_audio_buffer_ref_data_source_vtable =
{
- ma_pcm_deinterleave_s24__reference(dst, src, frameCount, channels);
-}
+ ma_audio_buffer_ref__data_source_on_read,
+ ma_audio_buffer_ref__data_source_on_seek,
+ ma_audio_buffer_ref__data_source_on_get_data_format,
+ ma_audio_buffer_ref__data_source_on_get_cursor,
+ ma_audio_buffer_ref__data_source_on_get_length,
+ NULL, /* onSetLooping */
+ 0
+};
-MA_API void ma_pcm_deinterleave_s24(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
+MA_API ma_result ma_audio_buffer_ref_init(ma_format format, ma_uint32 channels, const void* pData, ma_uint64 sizeInFrames, ma_audio_buffer_ref* pAudioBufferRef)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_deinterleave_s24__reference(dst, src, frameCount, channels);
-#else
- ma_pcm_deinterleave_s24__optimized(dst, src, frameCount, channels);
-#endif
-}
+ ma_result result;
+ ma_data_source_config dataSourceConfig;
+ if (pAudioBufferRef == NULL) {
+ return MA_INVALID_ARGS;
+ }
+ MA_ZERO_OBJECT(pAudioBufferRef);
-/* s32 */
-static MA_INLINE void ma_pcm_s32_to_u8__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_uint8* dst_u8 = (ma_uint8*)dst;
- const ma_int32* src_s32 = (const ma_int32*)src;
+ dataSourceConfig = ma_data_source_config_init();
+ dataSourceConfig.vtable = &g_ma_audio_buffer_ref_data_source_vtable;
- if (ditherMode == ma_dither_mode_none) {
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- ma_int32 x = src_s32[i];
- x = x >> 24;
- x = x + 128;
- dst_u8[i] = (ma_uint8)x;
- }
- } else {
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- ma_int32 x = src_s32[i];
+ result = ma_data_source_init(&dataSourceConfig, &pAudioBufferRef->ds);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- /* Dither. Don't overflow. */
- ma_int32 dither = ma_dither_s32(ditherMode, -0x800000, 0x7FFFFF);
- if ((ma_int64)x + dither <= 0x7FFFFFFF) {
- x = x + dither;
- } else {
- x = 0x7FFFFFFF;
- }
+ pAudioBufferRef->format = format;
+ pAudioBufferRef->channels = channels;
+ pAudioBufferRef->cursor = 0;
+ pAudioBufferRef->sizeInFrames = sizeInFrames;
+ pAudioBufferRef->pData = pData;
- x = x >> 24;
- x = x + 128;
- dst_u8[i] = (ma_uint8)x;
- }
- }
+ return MA_SUCCESS;
}
-static MA_INLINE void ma_pcm_s32_to_u8__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API void ma_audio_buffer_ref_uninit(ma_audio_buffer_ref* pAudioBufferRef)
{
- ma_pcm_s32_to_u8__reference(dst, src, count, ditherMode);
-}
+ if (pAudioBufferRef == NULL) {
+ return;
+ }
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE void ma_pcm_s32_to_u8__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s32_to_u8__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE void ma_pcm_s32_to_u8__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s32_to_u8__optimized(dst, src, count, ditherMode);
+ ma_data_source_uninit(&pAudioBufferRef->ds);
}
-#endif
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE void ma_pcm_s32_to_u8__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+
+MA_API ma_result ma_audio_buffer_ref_set_data(ma_audio_buffer_ref* pAudioBufferRef, const void* pData, ma_uint64 sizeInFrames)
{
- ma_pcm_s32_to_u8__optimized(dst, src, count, ditherMode);
+ if (pAudioBufferRef == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ pAudioBufferRef->cursor = 0;
+ pAudioBufferRef->sizeInFrames = sizeInFrames;
+ pAudioBufferRef->pData = pData;
+
+ return MA_SUCCESS;
}
-#endif
-MA_API void ma_pcm_s32_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_uint64 ma_audio_buffer_ref_read_pcm_frames(ma_audio_buffer_ref* pAudioBufferRef, void* pFramesOut, ma_uint64 frameCount, ma_bool32 loop)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_s32_to_u8__reference(dst, src, count, ditherMode);
-#else
- # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
- if (ma_has_avx2()) {
- ma_pcm_s32_to_u8__avx2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
- if (ma_has_sse2()) {
- ma_pcm_s32_to_u8__sse2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
- if (ma_has_neon()) {
- ma_pcm_s32_to_u8__neon(dst, src, count, ditherMode);
- } else
- #endif
- {
- ma_pcm_s32_to_u8__optimized(dst, src, count, ditherMode);
- }
-#endif
-}
+ ma_uint64 totalFramesRead = 0;
+ if (pAudioBufferRef == NULL) {
+ return 0;
+ }
-static MA_INLINE void ma_pcm_s32_to_s16__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_int16* dst_s16 = (ma_int16*)dst;
- const ma_int32* src_s32 = (const ma_int32*)src;
+ if (frameCount == 0) {
+ return 0;
+ }
- if (ditherMode == ma_dither_mode_none) {
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- ma_int32 x = src_s32[i];
- x = x >> 16;
- dst_s16[i] = (ma_int16)x;
+ while (totalFramesRead < frameCount) {
+ ma_uint64 framesAvailable = pAudioBufferRef->sizeInFrames - pAudioBufferRef->cursor;
+ ma_uint64 framesRemaining = frameCount - totalFramesRead;
+ ma_uint64 framesToRead;
+
+ framesToRead = framesRemaining;
+ if (framesToRead > framesAvailable) {
+ framesToRead = framesAvailable;
}
- } else {
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- ma_int32 x = src_s32[i];
- /* Dither. Don't overflow. */
- ma_int32 dither = ma_dither_s32(ditherMode, -0x8000, 0x7FFF);
- if ((ma_int64)x + dither <= 0x7FFFFFFF) {
- x = x + dither;
+ if (pFramesOut != NULL) {
+ ma_copy_pcm_frames(pFramesOut, ma_offset_ptr(pAudioBufferRef->pData, pAudioBufferRef->cursor * ma_get_bytes_per_frame(pAudioBufferRef->format, pAudioBufferRef->channels)), framesToRead, pAudioBufferRef->format, pAudioBufferRef->channels);
+ }
+
+ totalFramesRead += framesToRead;
+
+ pAudioBufferRef->cursor += framesToRead;
+ if (pAudioBufferRef->cursor == pAudioBufferRef->sizeInFrames) {
+ if (loop) {
+ pAudioBufferRef->cursor = 0;
} else {
- x = 0x7FFFFFFF;
+ break; /* We've reached the end and we're not looping. Done. */
}
-
- x = x >> 16;
- dst_s16[i] = (ma_int16)x;
}
+
+ MA_ASSERT(pAudioBufferRef->cursor < pAudioBufferRef->sizeInFrames);
}
-}
-static MA_INLINE void ma_pcm_s32_to_s16__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s32_to_s16__reference(dst, src, count, ditherMode);
+ return totalFramesRead;
}
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE void ma_pcm_s32_to_s16__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s32_to_s16__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE void ma_pcm_s32_to_s16__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s32_to_s16__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE void ma_pcm_s32_to_s16__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_audio_buffer_ref_seek_to_pcm_frame(ma_audio_buffer_ref* pAudioBufferRef, ma_uint64 frameIndex)
{
- ma_pcm_s32_to_s16__optimized(dst, src, count, ditherMode);
+ if (pAudioBufferRef == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (frameIndex > pAudioBufferRef->sizeInFrames) {
+ return MA_INVALID_ARGS;
+ }
+
+ pAudioBufferRef->cursor = (size_t)frameIndex;
+
+ return MA_SUCCESS;
}
-#endif
-MA_API void ma_pcm_s32_to_s16(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_audio_buffer_ref_map(ma_audio_buffer_ref* pAudioBufferRef, void** ppFramesOut, ma_uint64* pFrameCount)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_s32_to_s16__reference(dst, src, count, ditherMode);
-#else
- # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
- if (ma_has_avx2()) {
- ma_pcm_s32_to_s16__avx2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
- if (ma_has_sse2()) {
- ma_pcm_s32_to_s16__sse2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
- if (ma_has_neon()) {
- ma_pcm_s32_to_s16__neon(dst, src, count, ditherMode);
- } else
- #endif
- {
- ma_pcm_s32_to_s16__optimized(dst, src, count, ditherMode);
- }
-#endif
-}
+ ma_uint64 framesAvailable;
+ ma_uint64 frameCount = 0;
+ if (ppFramesOut != NULL) {
+ *ppFramesOut = NULL; /* Safety. */
+ }
-static MA_INLINE void ma_pcm_s32_to_s24__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_uint8* dst_s24 = (ma_uint8*)dst;
- const ma_int32* src_s32 = (const ma_int32*)src;
+ if (pFrameCount != NULL) {
+ frameCount = *pFrameCount;
+ *pFrameCount = 0; /* Safety. */
+ }
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- ma_uint32 x = (ma_uint32)src_s32[i];
- dst_s24[i*3+0] = (ma_uint8)((x & 0x0000FF00) >> 8);
- dst_s24[i*3+1] = (ma_uint8)((x & 0x00FF0000) >> 16);
- dst_s24[i*3+2] = (ma_uint8)((x & 0xFF000000) >> 24);
+ if (pAudioBufferRef == NULL || ppFramesOut == NULL || pFrameCount == NULL) {
+ return MA_INVALID_ARGS;
}
- (void)ditherMode; /* No dithering for s32 -> s24. */
-}
+ framesAvailable = pAudioBufferRef->sizeInFrames - pAudioBufferRef->cursor;
+ if (frameCount > framesAvailable) {
+ frameCount = framesAvailable;
+ }
-static MA_INLINE void ma_pcm_s32_to_s24__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s32_to_s24__reference(dst, src, count, ditherMode);
-}
+ *ppFramesOut = ma_offset_ptr(pAudioBufferRef->pData, pAudioBufferRef->cursor * ma_get_bytes_per_frame(pAudioBufferRef->format, pAudioBufferRef->channels));
+ *pFrameCount = frameCount;
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE void ma_pcm_s32_to_s24__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s32_to_s24__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE void ma_pcm_s32_to_s24__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s32_to_s24__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE void ma_pcm_s32_to_s24__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s32_to_s24__optimized(dst, src, count, ditherMode);
+ return MA_SUCCESS;
}
-#endif
-MA_API void ma_pcm_s32_to_s24(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_audio_buffer_ref_unmap(ma_audio_buffer_ref* pAudioBufferRef, ma_uint64 frameCount)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_s32_to_s24__reference(dst, src, count, ditherMode);
-#else
- # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
- if (ma_has_avx2()) {
- ma_pcm_s32_to_s24__avx2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
- if (ma_has_sse2()) {
- ma_pcm_s32_to_s24__sse2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
- if (ma_has_neon()) {
- ma_pcm_s32_to_s24__neon(dst, src, count, ditherMode);
- } else
- #endif
- {
- ma_pcm_s32_to_s24__optimized(dst, src, count, ditherMode);
- }
-#endif
-}
+ ma_uint64 framesAvailable;
+
+ if (pAudioBufferRef == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ framesAvailable = pAudioBufferRef->sizeInFrames - pAudioBufferRef->cursor;
+ if (frameCount > framesAvailable) {
+ return MA_INVALID_ARGS; /* The frame count was too big. This should never happen in an unmapping. Need to make sure the caller is aware of this. */
+ }
+ pAudioBufferRef->cursor += frameCount;
-MA_API void ma_pcm_s32_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+ if (pAudioBufferRef->cursor == pAudioBufferRef->sizeInFrames) {
+ return MA_AT_END; /* Successful. Need to tell the caller that the end has been reached so that it can loop if desired. */
+ } else {
+ return MA_SUCCESS;
+ }
+}
+
+MA_API ma_bool32 ma_audio_buffer_ref_at_end(const ma_audio_buffer_ref* pAudioBufferRef)
{
- (void)ditherMode;
+ if (pAudioBufferRef == NULL) {
+ return MA_FALSE;
+ }
- ma_copy_memory_64(dst, src, count * sizeof(ma_int32));
+ return pAudioBufferRef->cursor == pAudioBufferRef->sizeInFrames;
}
-
-static MA_INLINE void ma_pcm_s32_to_f32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_audio_buffer_ref_get_cursor_in_pcm_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pCursor)
{
- float* dst_f32 = (float*)dst;
- const ma_int32* src_s32 = (const ma_int32*)src;
-
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- double x = src_s32[i];
+ if (pCursor == NULL) {
+ return MA_INVALID_ARGS;
+ }
-#if 0
- x = x + 2147483648.0;
- x = x * 0.0000000004656612873077392578125;
- x = x - 1;
-#else
- x = x / 2147483648.0;
-#endif
+ *pCursor = 0;
- dst_f32[i] = (float)x;
+ if (pAudioBufferRef == NULL) {
+ return MA_INVALID_ARGS;
}
- (void)ditherMode; /* No dithering for s32 -> f32. */
-}
+ *pCursor = pAudioBufferRef->cursor;
-static MA_INLINE void ma_pcm_s32_to_f32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s32_to_f32__reference(dst, src, count, ditherMode);
+ return MA_SUCCESS;
}
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE void ma_pcm_s32_to_f32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s32_to_f32__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE void ma_pcm_s32_to_f32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_s32_to_f32__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE void ma_pcm_s32_to_f32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_audio_buffer_ref_get_length_in_pcm_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pLength)
{
- ma_pcm_s32_to_f32__optimized(dst, src, count, ditherMode);
+ if (pLength == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ *pLength = 0;
+
+ if (pAudioBufferRef == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ *pLength = pAudioBufferRef->sizeInFrames;
+
+ return MA_SUCCESS;
}
-#endif
-MA_API void ma_pcm_s32_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_audio_buffer_ref_get_available_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pAvailableFrames)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_s32_to_f32__reference(dst, src, count, ditherMode);
-#else
- # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
- if (ma_has_avx2()) {
- ma_pcm_s32_to_f32__avx2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
- if (ma_has_sse2()) {
- ma_pcm_s32_to_f32__sse2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
- if (ma_has_neon()) {
- ma_pcm_s32_to_f32__neon(dst, src, count, ditherMode);
- } else
- #endif
- {
- ma_pcm_s32_to_f32__optimized(dst, src, count, ditherMode);
- }
-#endif
-}
+ if (pAvailableFrames == NULL) {
+ return MA_INVALID_ARGS;
+ }
+ *pAvailableFrames = 0;
-static MA_INLINE void ma_pcm_interleave_s32__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
-{
- ma_int32* dst_s32 = (ma_int32*)dst;
- const ma_int32** src_s32 = (const ma_int32**)src;
+ if (pAudioBufferRef == NULL) {
+ return MA_INVALID_ARGS;
+ }
- ma_uint64 iFrame;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- dst_s32[iFrame*channels + iChannel] = src_s32[iChannel][iFrame];
- }
+ if (pAudioBufferRef->sizeInFrames <= pAudioBufferRef->cursor) {
+ *pAvailableFrames = 0;
+ } else {
+ *pAvailableFrames = pAudioBufferRef->sizeInFrames - pAudioBufferRef->cursor;
}
-}
-static MA_INLINE void ma_pcm_interleave_s32__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
-{
- ma_pcm_interleave_s32__reference(dst, src, frameCount, channels);
+ return MA_SUCCESS;
}
-MA_API void ma_pcm_interleave_s32(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
-{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_interleave_s32__reference(dst, src, frameCount, channels);
-#else
- ma_pcm_interleave_s32__optimized(dst, src, frameCount, channels);
-#endif
-}
-static MA_INLINE void ma_pcm_deinterleave_s32__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
+
+MA_API ma_audio_buffer_config ma_audio_buffer_config_init(ma_format format, ma_uint32 channels, ma_uint64 sizeInFrames, const void* pData, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_int32** dst_s32 = (ma_int32**)dst;
- const ma_int32* src_s32 = (const ma_int32*)src;
+ ma_audio_buffer_config config;
- ma_uint64 iFrame;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- dst_s32[iChannel][iFrame] = src_s32[iFrame*channels + iChannel];
- }
- }
-}
+ MA_ZERO_OBJECT(&config);
+ config.format = format;
+ config.channels = channels;
+ config.sizeInFrames = sizeInFrames;
+ config.pData = pData;
+ ma_allocation_callbacks_init_copy(&config.allocationCallbacks, pAllocationCallbacks);
-static MA_INLINE void ma_pcm_deinterleave_s32__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
-{
- ma_pcm_deinterleave_s32__reference(dst, src, frameCount, channels);
+ return config;
}
-MA_API void ma_pcm_deinterleave_s32(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
+static ma_result ma_audio_buffer_init_ex(const ma_audio_buffer_config* pConfig, ma_bool32 doCopy, ma_audio_buffer* pAudioBuffer)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_deinterleave_s32__reference(dst, src, frameCount, channels);
-#else
- ma_pcm_deinterleave_s32__optimized(dst, src, frameCount, channels);
-#endif
-}
+ ma_result result;
+ if (pAudioBuffer == NULL) {
+ return MA_INVALID_ARGS;
+ }
-/* f32 */
-static MA_INLINE void ma_pcm_f32_to_u8__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_uint64 i;
+ MA_ZERO_MEMORY(pAudioBuffer, sizeof(*pAudioBuffer) - sizeof(pAudioBuffer->_pExtraData)); /* Safety. Don't overwrite the extra data. */
- ma_uint8* dst_u8 = (ma_uint8*)dst;
- const float* src_f32 = (const float*)src;
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
- float ditherMin = 0;
- float ditherMax = 0;
- if (ditherMode != ma_dither_mode_none) {
- ditherMin = 1.0f / -128;
- ditherMax = 1.0f / 127;
+ if (pConfig->sizeInFrames == 0) {
+ return MA_INVALID_ARGS; /* Not allowing buffer sizes of 0 frames. */
}
- for (i = 0; i < count; i += 1) {
- float x = src_f32[i];
- x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax);
- x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */
- x = x + 1; /* -1..1 to 0..2 */
- x = x * 127.5f; /* 0..2 to 0..255 */
+ result = ma_audio_buffer_ref_init(pConfig->format, pConfig->channels, NULL, 0, &pAudioBuffer->ref);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- dst_u8[i] = (ma_uint8)x;
+ ma_allocation_callbacks_init_copy(&pAudioBuffer->allocationCallbacks, &pConfig->allocationCallbacks);
+
+ if (doCopy) {
+ ma_uint64 allocationSizeInBytes;
+ void* pData;
+
+ allocationSizeInBytes = pConfig->sizeInFrames * ma_get_bytes_per_frame(pConfig->format, pConfig->channels);
+ if (allocationSizeInBytes > MA_SIZE_MAX) {
+ return MA_OUT_OF_MEMORY; /* Too big. */
+ }
+
+ pData = ma_malloc((size_t)allocationSizeInBytes, &pAudioBuffer->allocationCallbacks); /* Safe cast to size_t. */
+ if (pData == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+
+ if (pConfig->pData != NULL) {
+ ma_copy_pcm_frames(pData, pConfig->pData, pConfig->sizeInFrames, pConfig->format, pConfig->channels);
+ } else {
+ ma_silence_pcm_frames(pData, pConfig->sizeInFrames, pConfig->format, pConfig->channels);
+ }
+
+ ma_audio_buffer_ref_set_data(&pAudioBuffer->ref, pData, pConfig->sizeInFrames);
+ pAudioBuffer->ownsData = MA_TRUE;
+ } else {
+ ma_audio_buffer_ref_set_data(&pAudioBuffer->ref, pConfig->pData, pConfig->sizeInFrames);
+ pAudioBuffer->ownsData = MA_FALSE;
}
-}
-static MA_INLINE void ma_pcm_f32_to_u8__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_f32_to_u8__reference(dst, src, count, ditherMode);
+ return MA_SUCCESS;
}
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE void ma_pcm_f32_to_u8__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_f32_to_u8__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE void ma_pcm_f32_to_u8__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+static void ma_audio_buffer_uninit_ex(ma_audio_buffer* pAudioBuffer, ma_bool32 doFree)
{
- ma_pcm_f32_to_u8__optimized(dst, src, count, ditherMode);
+ if (pAudioBuffer == NULL) {
+ return;
+ }
+
+ if (pAudioBuffer->ownsData && pAudioBuffer->ref.pData != &pAudioBuffer->_pExtraData[0]) {
+ ma_free((void*)pAudioBuffer->ref.pData, &pAudioBuffer->allocationCallbacks); /* Naugty const cast, but OK in this case since we've guarded it with the ownsData check. */
+ }
+
+ if (doFree) {
+ ma_free(pAudioBuffer, &pAudioBuffer->allocationCallbacks);
+ }
+
+ ma_audio_buffer_ref_uninit(&pAudioBuffer->ref);
}
-#endif
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE void ma_pcm_f32_to_u8__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+
+MA_API ma_result ma_audio_buffer_init(const ma_audio_buffer_config* pConfig, ma_audio_buffer* pAudioBuffer)
{
- ma_pcm_f32_to_u8__optimized(dst, src, count, ditherMode);
+ return ma_audio_buffer_init_ex(pConfig, MA_FALSE, pAudioBuffer);
}
-#endif
-MA_API void ma_pcm_f32_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_audio_buffer_init_copy(const ma_audio_buffer_config* pConfig, ma_audio_buffer* pAudioBuffer)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_f32_to_u8__reference(dst, src, count, ditherMode);
-#else
- # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
- if (ma_has_avx2()) {
- ma_pcm_f32_to_u8__avx2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
- if (ma_has_sse2()) {
- ma_pcm_f32_to_u8__sse2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
- if (ma_has_neon()) {
- ma_pcm_f32_to_u8__neon(dst, src, count, ditherMode);
- } else
- #endif
- {
- ma_pcm_f32_to_u8__optimized(dst, src, count, ditherMode);
- }
-#endif
+ return ma_audio_buffer_init_ex(pConfig, MA_TRUE, pAudioBuffer);
}
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
-static MA_INLINE void ma_pcm_f32_to_s16__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_audio_buffer_alloc_and_init(const ma_audio_buffer_config* pConfig, ma_audio_buffer** ppAudioBuffer)
{
- ma_uint64 i;
+ ma_result result;
+ ma_audio_buffer* pAudioBuffer;
+ ma_audio_buffer_config innerConfig; /* We'll be making some changes to the config, so need to make a copy. */
+ ma_uint64 allocationSizeInBytes;
- ma_int16* dst_s16 = (ma_int16*)dst;
- const float* src_f32 = (const float*)src;
+ if (ppAudioBuffer == NULL) {
+ return MA_INVALID_ARGS;
+ }
- float ditherMin = 0;
- float ditherMax = 0;
- if (ditherMode != ma_dither_mode_none) {
- ditherMin = 1.0f / -32768;
- ditherMax = 1.0f / 32767;
+ *ppAudioBuffer = NULL; /* Safety. */
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- for (i = 0; i < count; i += 1) {
- float x = src_f32[i];
- x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax);
- x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */
+ innerConfig = *pConfig;
+ ma_allocation_callbacks_init_copy(&innerConfig.allocationCallbacks, &pConfig->allocationCallbacks);
-#if 0
- /* The accurate way. */
- x = x + 1; /* -1..1 to 0..2 */
- x = x * 32767.5f; /* 0..2 to 0..65535 */
- x = x - 32768.0f; /* 0...65535 to -32768..32767 */
-#else
- /* The fast way. */
- x = x * 32767.0f; /* -1..1 to -32767..32767 */
-#endif
+ allocationSizeInBytes = sizeof(*pAudioBuffer) - sizeof(pAudioBuffer->_pExtraData) + (pConfig->sizeInFrames * ma_get_bytes_per_frame(pConfig->format, pConfig->channels));
+ if (allocationSizeInBytes > MA_SIZE_MAX) {
+ return MA_OUT_OF_MEMORY; /* Too big. */
+ }
- dst_s16[i] = (ma_int16)x;
+ pAudioBuffer = (ma_audio_buffer*)ma_malloc((size_t)allocationSizeInBytes, &innerConfig.allocationCallbacks); /* Safe cast to size_t. */
+ if (pAudioBuffer == NULL) {
+ return MA_OUT_OF_MEMORY;
}
-}
-#else
-static MA_INLINE void ma_pcm_f32_to_s16__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_uint64 i;
- ma_uint64 i4;
- ma_uint64 count4;
- ma_int16* dst_s16 = (ma_int16*)dst;
- const float* src_f32 = (const float*)src;
+ if (pConfig->pData != NULL) {
+ ma_copy_pcm_frames(&pAudioBuffer->_pExtraData[0], pConfig->pData, pConfig->sizeInFrames, pConfig->format, pConfig->channels);
+ } else {
+ ma_silence_pcm_frames(&pAudioBuffer->_pExtraData[0], pConfig->sizeInFrames, pConfig->format, pConfig->channels);
+ }
- float ditherMin = 0;
- float ditherMax = 0;
- if (ditherMode != ma_dither_mode_none) {
- ditherMin = 1.0f / -32768;
- ditherMax = 1.0f / 32767;
+ innerConfig.pData = &pAudioBuffer->_pExtraData[0];
+
+ result = ma_audio_buffer_init_ex(&innerConfig, MA_FALSE, pAudioBuffer);
+ if (result != MA_SUCCESS) {
+ ma_free(pAudioBuffer, &innerConfig.allocationCallbacks);
+ return result;
}
- /* Unrolled. */
- i = 0;
- count4 = count >> 2;
- for (i4 = 0; i4 < count4; i4 += 1) {
- float d0 = ma_dither_f32(ditherMode, ditherMin, ditherMax);
- float d1 = ma_dither_f32(ditherMode, ditherMin, ditherMax);
- float d2 = ma_dither_f32(ditherMode, ditherMin, ditherMax);
- float d3 = ma_dither_f32(ditherMode, ditherMin, ditherMax);
+ *ppAudioBuffer = pAudioBuffer;
- float x0 = src_f32[i+0];
- float x1 = src_f32[i+1];
- float x2 = src_f32[i+2];
- float x3 = src_f32[i+3];
+ return MA_SUCCESS;
+}
- x0 = x0 + d0;
- x1 = x1 + d1;
- x2 = x2 + d2;
- x3 = x3 + d3;
+MA_API void ma_audio_buffer_uninit(ma_audio_buffer* pAudioBuffer)
+{
+ ma_audio_buffer_uninit_ex(pAudioBuffer, MA_FALSE);
+}
- x0 = ((x0 < -1) ? -1 : ((x0 > 1) ? 1 : x0));
- x1 = ((x1 < -1) ? -1 : ((x1 > 1) ? 1 : x1));
- x2 = ((x2 < -1) ? -1 : ((x2 > 1) ? 1 : x2));
- x3 = ((x3 < -1) ? -1 : ((x3 > 1) ? 1 : x3));
+MA_API void ma_audio_buffer_uninit_and_free(ma_audio_buffer* pAudioBuffer)
+{
+ ma_audio_buffer_uninit_ex(pAudioBuffer, MA_TRUE);
+}
- x0 = x0 * 32767.0f;
- x1 = x1 * 32767.0f;
- x2 = x2 * 32767.0f;
- x3 = x3 * 32767.0f;
+MA_API ma_uint64 ma_audio_buffer_read_pcm_frames(ma_audio_buffer* pAudioBuffer, void* pFramesOut, ma_uint64 frameCount, ma_bool32 loop)
+{
+ if (pAudioBuffer == NULL) {
+ return 0;
+ }
- dst_s16[i+0] = (ma_int16)x0;
- dst_s16[i+1] = (ma_int16)x1;
- dst_s16[i+2] = (ma_int16)x2;
- dst_s16[i+3] = (ma_int16)x3;
+ return ma_audio_buffer_ref_read_pcm_frames(&pAudioBuffer->ref, pFramesOut, frameCount, loop);
+}
- i += 4;
+MA_API ma_result ma_audio_buffer_seek_to_pcm_frame(ma_audio_buffer* pAudioBuffer, ma_uint64 frameIndex)
+{
+ if (pAudioBuffer == NULL) {
+ return MA_INVALID_ARGS;
}
- /* Leftover. */
- for (; i < count; i += 1) {
- float x = src_f32[i];
- x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax);
- x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */
- x = x * 32767.0f; /* -1..1 to -32767..32767 */
+ return ma_audio_buffer_ref_seek_to_pcm_frame(&pAudioBuffer->ref, frameIndex);
+}
- dst_s16[i] = (ma_int16)x;
+MA_API ma_result ma_audio_buffer_map(ma_audio_buffer* pAudioBuffer, void** ppFramesOut, ma_uint64* pFrameCount)
+{
+ if (ppFramesOut != NULL) {
+ *ppFramesOut = NULL; /* Safety. */
+ }
+
+ if (pAudioBuffer == NULL) {
+ if (pFrameCount != NULL) {
+ *pFrameCount = 0;
+ }
+
+ return MA_INVALID_ARGS;
}
+
+ return ma_audio_buffer_ref_map(&pAudioBuffer->ref, ppFramesOut, pFrameCount);
}
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE void ma_pcm_f32_to_s16__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_audio_buffer_unmap(ma_audio_buffer* pAudioBuffer, ma_uint64 frameCount)
{
- ma_uint64 i;
- ma_uint64 i8;
- ma_uint64 count8;
- ma_int16* dst_s16;
- const float* src_f32;
- float ditherMin;
- float ditherMax;
+ if (pAudioBuffer == NULL) {
+ return MA_INVALID_ARGS;
+ }
- /* Both the input and output buffers need to be aligned to 16 bytes. */
- if ((((ma_uintptr)dst & 15) != 0) || (((ma_uintptr)src & 15) != 0)) {
- ma_pcm_f32_to_s16__optimized(dst, src, count, ditherMode);
- return;
+ return ma_audio_buffer_ref_unmap(&pAudioBuffer->ref, frameCount);
+}
+
+MA_API ma_bool32 ma_audio_buffer_at_end(const ma_audio_buffer* pAudioBuffer)
+{
+ if (pAudioBuffer == NULL) {
+ return MA_FALSE;
}
- dst_s16 = (ma_int16*)dst;
- src_f32 = (const float*)src;
+ return ma_audio_buffer_ref_at_end(&pAudioBuffer->ref);
+}
- ditherMin = 0;
- ditherMax = 0;
- if (ditherMode != ma_dither_mode_none) {
- ditherMin = 1.0f / -32768;
- ditherMax = 1.0f / 32767;
+MA_API ma_result ma_audio_buffer_get_cursor_in_pcm_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pCursor)
+{
+ if (pAudioBuffer == NULL) {
+ return MA_INVALID_ARGS;
}
- i = 0;
+ return ma_audio_buffer_ref_get_cursor_in_pcm_frames(&pAudioBuffer->ref, pCursor);
+}
- /* SSE2. SSE allows us to output 8 s16's at a time which means our loop is unrolled 8 times. */
- count8 = count >> 3;
- for (i8 = 0; i8 < count8; i8 += 1) {
- __m128 d0;
- __m128 d1;
- __m128 x0;
- __m128 x1;
+MA_API ma_result ma_audio_buffer_get_length_in_pcm_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pLength)
+{
+ if (pAudioBuffer == NULL) {
+ return MA_INVALID_ARGS;
+ }
- if (ditherMode == ma_dither_mode_none) {
- d0 = _mm_set1_ps(0);
- d1 = _mm_set1_ps(0);
- } else if (ditherMode == ma_dither_mode_rectangle) {
- d0 = _mm_set_ps(
- ma_dither_f32_rectangle(ditherMin, ditherMax),
- ma_dither_f32_rectangle(ditherMin, ditherMax),
- ma_dither_f32_rectangle(ditherMin, ditherMax),
- ma_dither_f32_rectangle(ditherMin, ditherMax)
- );
- d1 = _mm_set_ps(
- ma_dither_f32_rectangle(ditherMin, ditherMax),
- ma_dither_f32_rectangle(ditherMin, ditherMax),
- ma_dither_f32_rectangle(ditherMin, ditherMax),
- ma_dither_f32_rectangle(ditherMin, ditherMax)
- );
- } else {
- d0 = _mm_set_ps(
- ma_dither_f32_triangle(ditherMin, ditherMax),
- ma_dither_f32_triangle(ditherMin, ditherMax),
- ma_dither_f32_triangle(ditherMin, ditherMax),
- ma_dither_f32_triangle(ditherMin, ditherMax)
- );
- d1 = _mm_set_ps(
- ma_dither_f32_triangle(ditherMin, ditherMax),
- ma_dither_f32_triangle(ditherMin, ditherMax),
- ma_dither_f32_triangle(ditherMin, ditherMax),
- ma_dither_f32_triangle(ditherMin, ditherMax)
- );
- }
+ return ma_audio_buffer_ref_get_length_in_pcm_frames(&pAudioBuffer->ref, pLength);
+}
+
+MA_API ma_result ma_audio_buffer_get_available_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pAvailableFrames)
+{
+ if (pAvailableFrames == NULL) {
+ return MA_INVALID_ARGS;
+ }
- x0 = *((__m128*)(src_f32 + i) + 0);
- x1 = *((__m128*)(src_f32 + i) + 1);
+ *pAvailableFrames = 0;
- x0 = _mm_add_ps(x0, d0);
- x1 = _mm_add_ps(x1, d1);
+ if (pAudioBuffer == NULL) {
+ return MA_INVALID_ARGS;
+ }
- x0 = _mm_mul_ps(x0, _mm_set1_ps(32767.0f));
- x1 = _mm_mul_ps(x1, _mm_set1_ps(32767.0f));
+ return ma_audio_buffer_ref_get_available_frames(&pAudioBuffer->ref, pAvailableFrames);
+}
- _mm_stream_si128(((__m128i*)(dst_s16 + i)), _mm_packs_epi32(_mm_cvttps_epi32(x0), _mm_cvttps_epi32(x1)));
- i += 8;
- }
- /* Leftover. */
- for (; i < count; i += 1) {
- float x = src_f32[i];
- x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax);
- x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */
- x = x * 32767.0f; /* -1..1 to -32767..32767 */
- dst_s16[i] = (ma_int16)x;
+MA_API ma_result ma_paged_audio_buffer_data_init(ma_format format, ma_uint32 channels, ma_paged_audio_buffer_data* pData)
+{
+ if (pData == NULL) {
+ return MA_INVALID_ARGS;
}
+
+ MA_ZERO_OBJECT(pData);
+
+ pData->format = format;
+ pData->channels = channels;
+ pData->pTail = &pData->head;
+
+ return MA_SUCCESS;
}
-#endif /* SSE2 */
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE void ma_pcm_f32_to_s16__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API void ma_paged_audio_buffer_data_uninit(ma_paged_audio_buffer_data* pData, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_uint64 i;
- ma_uint64 i16;
- ma_uint64 count16;
- ma_int16* dst_s16;
- const float* src_f32;
- float ditherMin;
- float ditherMax;
+ ma_paged_audio_buffer_page* pPage;
- /* Both the input and output buffers need to be aligned to 32 bytes. */
- if ((((ma_uintptr)dst & 31) != 0) || (((ma_uintptr)src & 31) != 0)) {
- ma_pcm_f32_to_s16__optimized(dst, src, count, ditherMode);
+ if (pData == NULL) {
return;
}
- dst_s16 = (ma_int16*)dst;
- src_f32 = (const float*)src;
+ /* All pages need to be freed. */
+ pPage = (ma_paged_audio_buffer_page*)c89atomic_load_ptr(&pData->head.pNext);
+ while (pPage != NULL) {
+ ma_paged_audio_buffer_page* pNext = (ma_paged_audio_buffer_page*)c89atomic_load_ptr(&pPage->pNext);
- ditherMin = 0;
- ditherMax = 0;
- if (ditherMode != ma_dither_mode_none) {
- ditherMin = 1.0f / -32768;
- ditherMax = 1.0f / 32767;
+ ma_free(pPage, pAllocationCallbacks);
+ pPage = pNext;
}
+}
- i = 0;
-
- /* AVX2. AVX2 allows us to output 16 s16's at a time which means our loop is unrolled 16 times. */
- count16 = count >> 4;
- for (i16 = 0; i16 < count16; i16 += 1) {
- __m256 d0;
- __m256 d1;
- __m256 x0;
- __m256 x1;
- __m256i i0;
- __m256i i1;
- __m256i p0;
- __m256i p1;
- __m256i r;
-
- if (ditherMode == ma_dither_mode_none) {
- d0 = _mm256_set1_ps(0);
- d1 = _mm256_set1_ps(0);
- } else if (ditherMode == ma_dither_mode_rectangle) {
- d0 = _mm256_set_ps(
- ma_dither_f32_rectangle(ditherMin, ditherMax),
- ma_dither_f32_rectangle(ditherMin, ditherMax),
- ma_dither_f32_rectangle(ditherMin, ditherMax),
- ma_dither_f32_rectangle(ditherMin, ditherMax),
- ma_dither_f32_rectangle(ditherMin, ditherMax),
- ma_dither_f32_rectangle(ditherMin, ditherMax),
- ma_dither_f32_rectangle(ditherMin, ditherMax),
- ma_dither_f32_rectangle(ditherMin, ditherMax)
- );
- d1 = _mm256_set_ps(
- ma_dither_f32_rectangle(ditherMin, ditherMax),
- ma_dither_f32_rectangle(ditherMin, ditherMax),
- ma_dither_f32_rectangle(ditherMin, ditherMax),
- ma_dither_f32_rectangle(ditherMin, ditherMax),
- ma_dither_f32_rectangle(ditherMin, ditherMax),
- ma_dither_f32_rectangle(ditherMin, ditherMax),
- ma_dither_f32_rectangle(ditherMin, ditherMax),
- ma_dither_f32_rectangle(ditherMin, ditherMax)
- );
- } else {
- d0 = _mm256_set_ps(
- ma_dither_f32_triangle(ditherMin, ditherMax),
- ma_dither_f32_triangle(ditherMin, ditherMax),
- ma_dither_f32_triangle(ditherMin, ditherMax),
- ma_dither_f32_triangle(ditherMin, ditherMax),
- ma_dither_f32_triangle(ditherMin, ditherMax),
- ma_dither_f32_triangle(ditherMin, ditherMax),
- ma_dither_f32_triangle(ditherMin, ditherMax),
- ma_dither_f32_triangle(ditherMin, ditherMax)
- );
- d1 = _mm256_set_ps(
- ma_dither_f32_triangle(ditherMin, ditherMax),
- ma_dither_f32_triangle(ditherMin, ditherMax),
- ma_dither_f32_triangle(ditherMin, ditherMax),
- ma_dither_f32_triangle(ditherMin, ditherMax),
- ma_dither_f32_triangle(ditherMin, ditherMax),
- ma_dither_f32_triangle(ditherMin, ditherMax),
- ma_dither_f32_triangle(ditherMin, ditherMax),
- ma_dither_f32_triangle(ditherMin, ditherMax)
- );
- }
-
- x0 = *((__m256*)(src_f32 + i) + 0);
- x1 = *((__m256*)(src_f32 + i) + 1);
+MA_API ma_paged_audio_buffer_page* ma_paged_audio_buffer_data_get_head(ma_paged_audio_buffer_data* pData)
+{
+ if (pData == NULL) {
+ return NULL;
+ }
- x0 = _mm256_add_ps(x0, d0);
- x1 = _mm256_add_ps(x1, d1);
+ return &pData->head;
+}
- x0 = _mm256_mul_ps(x0, _mm256_set1_ps(32767.0f));
- x1 = _mm256_mul_ps(x1, _mm256_set1_ps(32767.0f));
+MA_API ma_paged_audio_buffer_page* ma_paged_audio_buffer_data_get_tail(ma_paged_audio_buffer_data* pData)
+{
+ if (pData == NULL) {
+ return NULL;
+ }
- /* Computing the final result is a little more complicated for AVX2 than SSE2. */
- i0 = _mm256_cvttps_epi32(x0);
- i1 = _mm256_cvttps_epi32(x1);
- p0 = _mm256_permute2x128_si256(i0, i1, 0 | 32);
- p1 = _mm256_permute2x128_si256(i0, i1, 1 | 48);
- r = _mm256_packs_epi32(p0, p1);
+ return pData->pTail;
+}
- _mm256_stream_si256(((__m256i*)(dst_s16 + i)), r);
+MA_API ma_result ma_paged_audio_buffer_data_get_length_in_pcm_frames(ma_paged_audio_buffer_data* pData, ma_uint64* pLength)
+{
+ ma_paged_audio_buffer_page* pPage;
- i += 16;
+ if (pLength == NULL) {
+ return MA_INVALID_ARGS;
}
+ *pLength = 0;
- /* Leftover. */
- for (; i < count; i += 1) {
- float x = src_f32[i];
- x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax);
- x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */
- x = x * 32767.0f; /* -1..1 to -32767..32767 */
+ if (pData == NULL) {
+ return MA_INVALID_ARGS;
+ }
- dst_s16[i] = (ma_int16)x;
+ /* Calculate the length from the linked list. */
+ for (pPage = (ma_paged_audio_buffer_page*)c89atomic_load_ptr(&pData->head.pNext); pPage != NULL; pPage = (ma_paged_audio_buffer_page*)c89atomic_load_ptr(&pPage->pNext)) {
+ *pLength += pPage->sizeInFrames;
}
+
+ return MA_SUCCESS;
}
-#endif /* AVX2 */
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE void ma_pcm_f32_to_s16__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_paged_audio_buffer_data_allocate_page(ma_paged_audio_buffer_data* pData, ma_uint64 pageSizeInFrames, const void* pInitialData, const ma_allocation_callbacks* pAllocationCallbacks, ma_paged_audio_buffer_page** ppPage)
{
- ma_uint64 i;
- ma_uint64 i8;
- ma_uint64 count8;
- ma_int16* dst_s16;
- const float* src_f32;
- float ditherMin;
- float ditherMax;
-
- if (!ma_has_neon()) {
- return ma_pcm_f32_to_s16__optimized(dst, src, count, ditherMode);
- }
+ ma_paged_audio_buffer_page* pPage;
+ ma_uint64 allocationSize;
- /* Both the input and output buffers need to be aligned to 16 bytes. */
- if ((((ma_uintptr)dst & 15) != 0) || (((ma_uintptr)src & 15) != 0)) {
- ma_pcm_f32_to_s16__optimized(dst, src, count, ditherMode);
- return;
+ if (ppPage == NULL) {
+ return MA_INVALID_ARGS;
}
- dst_s16 = (ma_int16*)dst;
- src_f32 = (const float*)src;
+ *ppPage = NULL;
- ditherMin = 0;
- ditherMax = 0;
- if (ditherMode != ma_dither_mode_none) {
- ditherMin = 1.0f / -32768;
- ditherMax = 1.0f / 32767;
+ if (pData == NULL) {
+ return MA_INVALID_ARGS;
}
- i = 0;
+ allocationSize = sizeof(*pPage) + (pageSizeInFrames * ma_get_bytes_per_frame(pData->format, pData->channels));
+ if (allocationSize > MA_SIZE_MAX) {
+ return MA_OUT_OF_MEMORY; /* Too big. */
+ }
- /* NEON. NEON allows us to output 8 s16's at a time which means our loop is unrolled 8 times. */
- count8 = count >> 3;
- for (i8 = 0; i8 < count8; i8 += 1) {
- float32x4_t d0;
- float32x4_t d1;
- float32x4_t x0;
- float32x4_t x1;
- int32x4_t i0;
- int32x4_t i1;
+ pPage = (ma_paged_audio_buffer_page*)ma_malloc((size_t)allocationSize, pAllocationCallbacks); /* Safe cast to size_t. */
+ if (pPage == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
- if (ditherMode == ma_dither_mode_none) {
- d0 = vmovq_n_f32(0);
- d1 = vmovq_n_f32(0);
- } else if (ditherMode == ma_dither_mode_rectangle) {
- float d0v[4];
- d0v[0] = ma_dither_f32_rectangle(ditherMin, ditherMax);
- d0v[1] = ma_dither_f32_rectangle(ditherMin, ditherMax);
- d0v[2] = ma_dither_f32_rectangle(ditherMin, ditherMax);
- d0v[3] = ma_dither_f32_rectangle(ditherMin, ditherMax);
- d0 = vld1q_f32(d0v);
+ pPage->pNext = NULL;
+ pPage->sizeInFrames = pageSizeInFrames;
- float d1v[4];
- d1v[0] = ma_dither_f32_rectangle(ditherMin, ditherMax);
- d1v[1] = ma_dither_f32_rectangle(ditherMin, ditherMax);
- d1v[2] = ma_dither_f32_rectangle(ditherMin, ditherMax);
- d1v[3] = ma_dither_f32_rectangle(ditherMin, ditherMax);
- d1 = vld1q_f32(d1v);
- } else {
- float d0v[4];
- d0v[0] = ma_dither_f32_triangle(ditherMin, ditherMax);
- d0v[1] = ma_dither_f32_triangle(ditherMin, ditherMax);
- d0v[2] = ma_dither_f32_triangle(ditherMin, ditherMax);
- d0v[3] = ma_dither_f32_triangle(ditherMin, ditherMax);
- d0 = vld1q_f32(d0v);
+ if (pInitialData != NULL) {
+ ma_copy_pcm_frames(pPage->pAudioData, pInitialData, pageSizeInFrames, pData->format, pData->channels);
+ }
- float d1v[4];
- d1v[0] = ma_dither_f32_triangle(ditherMin, ditherMax);
- d1v[1] = ma_dither_f32_triangle(ditherMin, ditherMax);
- d1v[2] = ma_dither_f32_triangle(ditherMin, ditherMax);
- d1v[3] = ma_dither_f32_triangle(ditherMin, ditherMax);
- d1 = vld1q_f32(d1v);
- }
+ *ppPage = pPage;
- x0 = *((float32x4_t*)(src_f32 + i) + 0);
- x1 = *((float32x4_t*)(src_f32 + i) + 1);
+ return MA_SUCCESS;
+}
- x0 = vaddq_f32(x0, d0);
- x1 = vaddq_f32(x1, d1);
+MA_API ma_result ma_paged_audio_buffer_data_free_page(ma_paged_audio_buffer_data* pData, ma_paged_audio_buffer_page* pPage, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pData == NULL || pPage == NULL) {
+ return MA_INVALID_ARGS;
+ }
- x0 = vmulq_n_f32(x0, 32767.0f);
- x1 = vmulq_n_f32(x1, 32767.0f);
+ /* It's assumed the page is not attached to the list. */
+ ma_free(pPage, pAllocationCallbacks);
- i0 = vcvtq_s32_f32(x0);
- i1 = vcvtq_s32_f32(x1);
- *((int16x8_t*)(dst_s16 + i)) = vcombine_s16(vqmovn_s32(i0), vqmovn_s32(i1));
+ return MA_SUCCESS;
+}
- i += 8;
+MA_API ma_result ma_paged_audio_buffer_data_append_page(ma_paged_audio_buffer_data* pData, ma_paged_audio_buffer_page* pPage)
+{
+ if (pData == NULL || pPage == NULL) {
+ return MA_INVALID_ARGS;
}
+ /* This function assumes the page has been filled with audio data by this point. As soon as we append, the page will be available for reading. */
- /* Leftover. */
- for (; i < count; i += 1) {
- float x = src_f32[i];
- x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax);
- x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */
- x = x * 32767.0f; /* -1..1 to -32767..32767 */
+ /* First thing to do is update the tail. */
+ for (;;) {
+ ma_paged_audio_buffer_page* pOldTail = (ma_paged_audio_buffer_page*)c89atomic_load_ptr(&pData->pTail);
+ ma_paged_audio_buffer_page* pNewTail = pPage;
- dst_s16[i] = (ma_int16)x;
+ if (c89atomic_compare_exchange_weak_ptr((volatile void**)&pData->pTail, (void**)&pOldTail, pNewTail)) {
+ /* Here is where we append the page to the list. After this, the page is attached to the list and ready to be read from. */
+ c89atomic_exchange_ptr(&pOldTail->pNext, pPage);
+ break; /* Done. */
+ }
}
+
+ return MA_SUCCESS;
}
-#endif /* Neon */
-#endif /* MA_USE_REFERENCE_CONVERSION_APIS */
-MA_API void ma_pcm_f32_to_s16(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_paged_audio_buffer_data_allocate_and_append_page(ma_paged_audio_buffer_data* pData, ma_uint32 pageSizeInFrames, const void* pInitialData, const ma_allocation_callbacks* pAllocationCallbacks)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_f32_to_s16__reference(dst, src, count, ditherMode);
-#else
- # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
- if (ma_has_avx2()) {
- ma_pcm_f32_to_s16__avx2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
- if (ma_has_sse2()) {
- ma_pcm_f32_to_s16__sse2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
- if (ma_has_neon()) {
- ma_pcm_f32_to_s16__neon(dst, src, count, ditherMode);
- } else
- #endif
- {
- ma_pcm_f32_to_s16__optimized(dst, src, count, ditherMode);
- }
-#endif
-}
+ ma_result result;
+ ma_paged_audio_buffer_page* pPage;
+ result = ma_paged_audio_buffer_data_allocate_page(pData, pageSizeInFrames, pInitialData, pAllocationCallbacks, &pPage);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
-static MA_INLINE void ma_pcm_f32_to_s24__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_uint8* dst_s24 = (ma_uint8*)dst;
- const float* src_f32 = (const float*)src;
+ return ma_paged_audio_buffer_data_append_page(pData, pPage); /* <-- Should never fail. */
+}
- ma_uint64 i;
- for (i = 0; i < count; i += 1) {
- ma_int32 r;
- float x = src_f32[i];
- x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */
-#if 0
- /* The accurate way. */
- x = x + 1; /* -1..1 to 0..2 */
- x = x * 8388607.5f; /* 0..2 to 0..16777215 */
- x = x - 8388608.0f; /* 0..16777215 to -8388608..8388607 */
-#else
- /* The fast way. */
- x = x * 8388607.0f; /* -1..1 to -8388607..8388607 */
-#endif
+MA_API ma_paged_audio_buffer_config ma_paged_audio_buffer_config_init(ma_paged_audio_buffer_data* pData)
+{
+ ma_paged_audio_buffer_config config;
- r = (ma_int32)x;
- dst_s24[(i*3)+0] = (ma_uint8)((r & 0x0000FF) >> 0);
- dst_s24[(i*3)+1] = (ma_uint8)((r & 0x00FF00) >> 8);
- dst_s24[(i*3)+2] = (ma_uint8)((r & 0xFF0000) >> 16);
- }
+ MA_ZERO_OBJECT(&config);
+ config.pData = pData;
- (void)ditherMode; /* No dithering for f32 -> s24. */
+ return config;
}
-static MA_INLINE void ma_pcm_f32_to_s24__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+
+static ma_result ma_paged_audio_buffer__data_source_on_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
- ma_pcm_f32_to_s24__reference(dst, src, count, ditherMode);
+ return ma_paged_audio_buffer_read_pcm_frames((ma_paged_audio_buffer*)pDataSource, pFramesOut, frameCount, pFramesRead);
}
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE void ma_pcm_f32_to_s24__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+static ma_result ma_paged_audio_buffer__data_source_on_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
- ma_pcm_f32_to_s24__optimized(dst, src, count, ditherMode);
+ return ma_paged_audio_buffer_seek_to_pcm_frame((ma_paged_audio_buffer*)pDataSource, frameIndex);
}
-#endif
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE void ma_pcm_f32_to_s24__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+
+static ma_result ma_paged_audio_buffer__data_source_on_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
- ma_pcm_f32_to_s24__optimized(dst, src, count, ditherMode);
+ ma_paged_audio_buffer* pPagedAudioBuffer = (ma_paged_audio_buffer*)pDataSource;
+
+ *pFormat = pPagedAudioBuffer->pData->format;
+ *pChannels = pPagedAudioBuffer->pData->channels;
+ *pSampleRate = 0; /* There is no notion of a sample rate with audio buffers. */
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, pPagedAudioBuffer->pData->channels);
+
+ return MA_SUCCESS;
}
-#endif
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE void ma_pcm_f32_to_s24__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+
+static ma_result ma_paged_audio_buffer__data_source_on_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
{
- ma_pcm_f32_to_s24__optimized(dst, src, count, ditherMode);
+ return ma_paged_audio_buffer_get_cursor_in_pcm_frames((ma_paged_audio_buffer*)pDataSource, pCursor);
}
-#endif
-MA_API void ma_pcm_f32_to_s24(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+static ma_result ma_paged_audio_buffer__data_source_on_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_f32_to_s24__reference(dst, src, count, ditherMode);
-#else
- # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
- if (ma_has_avx2()) {
- ma_pcm_f32_to_s24__avx2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
- if (ma_has_sse2()) {
- ma_pcm_f32_to_s24__sse2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
- if (ma_has_neon()) {
- ma_pcm_f32_to_s24__neon(dst, src, count, ditherMode);
- } else
- #endif
- {
- ma_pcm_f32_to_s24__optimized(dst, src, count, ditherMode);
- }
-#endif
+ return ma_paged_audio_buffer_get_length_in_pcm_frames((ma_paged_audio_buffer*)pDataSource, pLength);
}
+static ma_data_source_vtable g_ma_paged_audio_buffer_data_source_vtable =
+{
+ ma_paged_audio_buffer__data_source_on_read,
+ ma_paged_audio_buffer__data_source_on_seek,
+ ma_paged_audio_buffer__data_source_on_get_data_format,
+ ma_paged_audio_buffer__data_source_on_get_cursor,
+ ma_paged_audio_buffer__data_source_on_get_length,
+ NULL, /* onSetLooping */
+ 0
+};
-static MA_INLINE void ma_pcm_f32_to_s32__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_paged_audio_buffer_init(const ma_paged_audio_buffer_config* pConfig, ma_paged_audio_buffer* pPagedAudioBuffer)
{
- ma_int32* dst_s32 = (ma_int32*)dst;
- const float* src_f32 = (const float*)src;
+ ma_result result;
+ ma_data_source_config dataSourceConfig;
- ma_uint32 i;
- for (i = 0; i < count; i += 1) {
- double x = src_f32[i];
- x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */
+ if (pPagedAudioBuffer == NULL) {
+ return MA_INVALID_ARGS;
+ }
-#if 0
- /* The accurate way. */
- x = x + 1; /* -1..1 to 0..2 */
- x = x * 2147483647.5; /* 0..2 to 0..4294967295 */
- x = x - 2147483648.0; /* 0...4294967295 to -2147483648..2147483647 */
-#else
- /* The fast way. */
- x = x * 2147483647.0; /* -1..1 to -2147483647..2147483647 */
-#endif
+ MA_ZERO_OBJECT(pPagedAudioBuffer);
- dst_s32[i] = (ma_int32)x;
+ /* A config is required for the format and channel count. */
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- (void)ditherMode; /* No dithering for f32 -> s32. */
-}
+ if (pConfig->pData == NULL) {
+ return MA_INVALID_ARGS; /* No underlying data specified. */
+ }
-static MA_INLINE void ma_pcm_f32_to_s32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_f32_to_s32__reference(dst, src, count, ditherMode);
-}
+ dataSourceConfig = ma_data_source_config_init();
+ dataSourceConfig.vtable = &g_ma_paged_audio_buffer_data_source_vtable;
-#if defined(MA_SUPPORT_SSE2)
-static MA_INLINE void ma_pcm_f32_to_s32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_f32_to_s32__optimized(dst, src, count, ditherMode);
-}
-#endif
-#if defined(MA_SUPPORT_AVX2)
-static MA_INLINE void ma_pcm_f32_to_s32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- ma_pcm_f32_to_s32__optimized(dst, src, count, ditherMode);
+ result = ma_data_source_init(&dataSourceConfig, &pPagedAudioBuffer->ds);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ pPagedAudioBuffer->pData = pConfig->pData;
+ pPagedAudioBuffer->pCurrent = ma_paged_audio_buffer_data_get_head(pConfig->pData);
+ pPagedAudioBuffer->relativeCursor = 0;
+ pPagedAudioBuffer->absoluteCursor = 0;
+
+ return MA_SUCCESS;
}
-#endif
-#if defined(MA_SUPPORT_NEON)
-static MA_INLINE void ma_pcm_f32_to_s32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+
+MA_API void ma_paged_audio_buffer_uninit(ma_paged_audio_buffer* pPagedAudioBuffer)
{
- ma_pcm_f32_to_s32__optimized(dst, src, count, ditherMode);
+ if (pPagedAudioBuffer == NULL) {
+ return;
+ }
+
+ /* Nothing to do. The data needs to be deleted separately. */
}
-#endif
-MA_API void ma_pcm_f32_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
+MA_API ma_result ma_paged_audio_buffer_read_pcm_frames(ma_paged_audio_buffer* pPagedAudioBuffer, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_f32_to_s32__reference(dst, src, count, ditherMode);
-#else
- # if MA_PREFERRED_SIMD == MA_SIMD_AVX2
- if (ma_has_avx2()) {
- ma_pcm_f32_to_s32__avx2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
- if (ma_has_sse2()) {
- ma_pcm_f32_to_s32__sse2(dst, src, count, ditherMode);
- } else
- #elif MA_PREFERRED_SIMD == MA_SIMD_NEON
- if (ma_has_neon()) {
- ma_pcm_f32_to_s32__neon(dst, src, count, ditherMode);
- } else
- #endif
- {
- ma_pcm_f32_to_s32__optimized(dst, src, count, ditherMode);
- }
-#endif
-}
+ ma_result result = MA_SUCCESS;
+ ma_uint64 totalFramesRead = 0;
+ ma_format format;
+ ma_uint32 channels;
+ if (pPagedAudioBuffer == NULL) {
+ return MA_INVALID_ARGS;
+ }
-MA_API void ma_pcm_f32_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
-{
- (void)ditherMode;
+ format = pPagedAudioBuffer->pData->format;
+ channels = pPagedAudioBuffer->pData->channels;
- ma_copy_memory_64(dst, src, count * sizeof(float));
-}
+ while (totalFramesRead < frameCount) {
+ /* Read from the current page. The buffer should never be in a state where this is NULL. */
+ ma_uint64 framesRemainingInCurrentPage;
+ ma_uint64 framesRemainingToRead = frameCount - totalFramesRead;
+ ma_uint64 framesToReadThisIteration;
+ MA_ASSERT(pPagedAudioBuffer->pCurrent != NULL);
-static void ma_pcm_interleave_f32__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
-{
- float* dst_f32 = (float*)dst;
- const float** src_f32 = (const float**)src;
+ framesRemainingInCurrentPage = pPagedAudioBuffer->pCurrent->sizeInFrames - pPagedAudioBuffer->relativeCursor;
- ma_uint64 iFrame;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- dst_f32[iFrame*channels + iChannel] = src_f32[iChannel][iFrame];
+ framesToReadThisIteration = ma_min(framesRemainingInCurrentPage, framesRemainingToRead);
+ ma_copy_pcm_frames(ma_offset_pcm_frames_ptr(pFramesOut, totalFramesRead, format, channels), ma_offset_pcm_frames_ptr(pPagedAudioBuffer->pCurrent->pAudioData, pPagedAudioBuffer->relativeCursor, format, channels), framesToReadThisIteration, format, channels);
+ totalFramesRead += framesToReadThisIteration;
+
+ pPagedAudioBuffer->absoluteCursor += framesToReadThisIteration;
+ pPagedAudioBuffer->relativeCursor += framesToReadThisIteration;
+
+ /* Move to the next page if necessary. If there's no more pages, we need to return MA_AT_END. */
+ MA_ASSERT(pPagedAudioBuffer->relativeCursor <= pPagedAudioBuffer->pCurrent->sizeInFrames);
+
+ if (pPagedAudioBuffer->relativeCursor == pPagedAudioBuffer->pCurrent->sizeInFrames) {
+ /* We reached the end of the page. Need to move to the next. If there's no more pages, we're done. */
+ ma_paged_audio_buffer_page* pNext = (ma_paged_audio_buffer_page*)c89atomic_load_ptr(&pPagedAudioBuffer->pCurrent->pNext);
+ if (pNext == NULL) {
+ result = MA_AT_END;
+ break; /* We've reached the end. */
+ } else {
+ pPagedAudioBuffer->pCurrent = pNext;
+ pPagedAudioBuffer->relativeCursor = 0;
+ }
}
}
-}
-static void ma_pcm_interleave_f32__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
-{
- ma_pcm_interleave_f32__reference(dst, src, frameCount, channels);
+ if (pFramesRead != NULL) {
+ *pFramesRead = totalFramesRead;
+ }
+
+ return result;
}
-MA_API void ma_pcm_interleave_f32(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
+MA_API ma_result ma_paged_audio_buffer_seek_to_pcm_frame(ma_paged_audio_buffer* pPagedAudioBuffer, ma_uint64 frameIndex)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_interleave_f32__reference(dst, src, frameCount, channels);
-#else
- ma_pcm_interleave_f32__optimized(dst, src, frameCount, channels);
-#endif
-}
+ if (pPagedAudioBuffer == NULL) {
+ return MA_INVALID_ARGS;
+ }
+ if (frameIndex == pPagedAudioBuffer->absoluteCursor) {
+ return MA_SUCCESS; /* Nothing to do. */
+ }
-static void ma_pcm_deinterleave_f32__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
-{
- float** dst_f32 = (float**)dst;
- const float* src_f32 = (const float*)src;
+ if (frameIndex < pPagedAudioBuffer->absoluteCursor) {
+ /* Moving backwards. Need to move the cursor back to the start, and then move forward. */
+ pPagedAudioBuffer->pCurrent = ma_paged_audio_buffer_data_get_head(pPagedAudioBuffer->pData);
+ pPagedAudioBuffer->absoluteCursor = 0;
+ pPagedAudioBuffer->relativeCursor = 0;
- ma_uint64 iFrame;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- dst_f32[iChannel][iFrame] = src_f32[iFrame*channels + iChannel];
+ /* Fall through to the forward seeking section below. */
+ }
+
+ if (frameIndex > pPagedAudioBuffer->absoluteCursor) {
+ /* Moving forward. */
+ ma_paged_audio_buffer_page* pPage;
+ ma_uint64 runningCursor = 0;
+
+ for (pPage = (ma_paged_audio_buffer_page*)c89atomic_load_ptr(&ma_paged_audio_buffer_data_get_head(pPagedAudioBuffer->pData)->pNext); pPage != NULL; pPage = (ma_paged_audio_buffer_page*)c89atomic_load_ptr(&pPage->pNext)) {
+ ma_uint64 pageRangeBeg = runningCursor;
+ ma_uint64 pageRangeEnd = pageRangeBeg + pPage->sizeInFrames;
+
+ if (frameIndex >= pageRangeBeg) {
+ if (frameIndex < pageRangeEnd || (frameIndex == pageRangeEnd && pPage == (ma_paged_audio_buffer_page*)c89atomic_load_ptr(ma_paged_audio_buffer_data_get_tail(pPagedAudioBuffer->pData)))) { /* A small edge case - allow seeking to the very end of the buffer. */
+ /* We found the page. */
+ pPagedAudioBuffer->pCurrent = pPage;
+ pPagedAudioBuffer->absoluteCursor = frameIndex;
+ pPagedAudioBuffer->relativeCursor = frameIndex - pageRangeBeg;
+ return MA_SUCCESS;
+ }
+ }
+
+ runningCursor = pageRangeEnd;
}
+
+ /* Getting here means we tried seeking too far forward. Don't change any state. */
+ return MA_BAD_SEEK;
+ }
+
+ return MA_SUCCESS;
+}
+
+MA_API ma_result ma_paged_audio_buffer_get_cursor_in_pcm_frames(ma_paged_audio_buffer* pPagedAudioBuffer, ma_uint64* pCursor)
+{
+ if (pCursor == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ *pCursor = 0; /* Safety. */
+
+ if (pPagedAudioBuffer == NULL) {
+ return MA_INVALID_ARGS;
}
-}
-static void ma_pcm_deinterleave_f32__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
-{
- ma_pcm_deinterleave_f32__reference(dst, src, frameCount, channels);
+ *pCursor = pPagedAudioBuffer->absoluteCursor;
+
+ return MA_SUCCESS;
}
-MA_API void ma_pcm_deinterleave_f32(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
+MA_API ma_result ma_paged_audio_buffer_get_length_in_pcm_frames(ma_paged_audio_buffer* pPagedAudioBuffer, ma_uint64* pLength)
{
-#ifdef MA_USE_REFERENCE_CONVERSION_APIS
- ma_pcm_deinterleave_f32__reference(dst, src, frameCount, channels);
-#else
- ma_pcm_deinterleave_f32__optimized(dst, src, frameCount, channels);
-#endif
+ return ma_paged_audio_buffer_data_get_length_in_pcm_frames(pPagedAudioBuffer->pData, pLength);
}
-MA_API void ma_pcm_convert(void* pOut, ma_format formatOut, const void* pIn, ma_format formatIn, ma_uint64 sampleCount, ma_dither_mode ditherMode)
+
+/**************************************************************************************************************************************************************
+
+VFS
+
+**************************************************************************************************************************************************************/
+MA_API ma_result ma_vfs_open(ma_vfs* pVFS, const char* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
{
- if (formatOut == formatIn) {
- ma_copy_memory_64(pOut, pIn, sampleCount * ma_get_bytes_per_sample(formatOut));
- return;
+ ma_vfs_callbacks* pCallbacks = (ma_vfs_callbacks*)pVFS;
+
+ if (pFile == NULL) {
+ return MA_INVALID_ARGS;
}
- switch (formatIn)
- {
- case ma_format_u8:
- {
- switch (formatOut)
- {
- case ma_format_s16: ma_pcm_u8_to_s16(pOut, pIn, sampleCount, ditherMode); return;
- case ma_format_s24: ma_pcm_u8_to_s24(pOut, pIn, sampleCount, ditherMode); return;
- case ma_format_s32: ma_pcm_u8_to_s32(pOut, pIn, sampleCount, ditherMode); return;
- case ma_format_f32: ma_pcm_u8_to_f32(pOut, pIn, sampleCount, ditherMode); return;
- default: break;
- }
- } break;
+ *pFile = NULL;
- case ma_format_s16:
- {
- switch (formatOut)
- {
- case ma_format_u8: ma_pcm_s16_to_u8( pOut, pIn, sampleCount, ditherMode); return;
- case ma_format_s24: ma_pcm_s16_to_s24(pOut, pIn, sampleCount, ditherMode); return;
- case ma_format_s32: ma_pcm_s16_to_s32(pOut, pIn, sampleCount, ditherMode); return;
- case ma_format_f32: ma_pcm_s16_to_f32(pOut, pIn, sampleCount, ditherMode); return;
- default: break;
- }
- } break;
+ if (pVFS == NULL || pFilePath == NULL || openMode == 0) {
+ return MA_INVALID_ARGS;
+ }
- case ma_format_s24:
- {
- switch (formatOut)
- {
- case ma_format_u8: ma_pcm_s24_to_u8( pOut, pIn, sampleCount, ditherMode); return;
- case ma_format_s16: ma_pcm_s24_to_s16(pOut, pIn, sampleCount, ditherMode); return;
- case ma_format_s32: ma_pcm_s24_to_s32(pOut, pIn, sampleCount, ditherMode); return;
- case ma_format_f32: ma_pcm_s24_to_f32(pOut, pIn, sampleCount, ditherMode); return;
- default: break;
- }
- } break;
+ if (pCallbacks->onOpen == NULL) {
+ return MA_NOT_IMPLEMENTED;
+ }
- case ma_format_s32:
- {
- switch (formatOut)
- {
- case ma_format_u8: ma_pcm_s32_to_u8( pOut, pIn, sampleCount, ditherMode); return;
- case ma_format_s16: ma_pcm_s32_to_s16(pOut, pIn, sampleCount, ditherMode); return;
- case ma_format_s24: ma_pcm_s32_to_s24(pOut, pIn, sampleCount, ditherMode); return;
- case ma_format_f32: ma_pcm_s32_to_f32(pOut, pIn, sampleCount, ditherMode); return;
- default: break;
- }
- } break;
+ return pCallbacks->onOpen(pVFS, pFilePath, openMode, pFile);
+}
- case ma_format_f32:
- {
- switch (formatOut)
- {
- case ma_format_u8: ma_pcm_f32_to_u8( pOut, pIn, sampleCount, ditherMode); return;
- case ma_format_s16: ma_pcm_f32_to_s16(pOut, pIn, sampleCount, ditherMode); return;
- case ma_format_s24: ma_pcm_f32_to_s24(pOut, pIn, sampleCount, ditherMode); return;
- case ma_format_s32: ma_pcm_f32_to_s32(pOut, pIn, sampleCount, ditherMode); return;
- default: break;
- }
- } break;
+MA_API ma_result ma_vfs_open_w(ma_vfs* pVFS, const wchar_t* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
+{
+ ma_vfs_callbacks* pCallbacks = (ma_vfs_callbacks*)pVFS;
- default: break;
+ if (pFile == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ *pFile = NULL;
+
+ if (pVFS == NULL || pFilePath == NULL || openMode == 0) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (pCallbacks->onOpenW == NULL) {
+ return MA_NOT_IMPLEMENTED;
}
+
+ return pCallbacks->onOpenW(pVFS, pFilePath, openMode, pFile);
}
-MA_API void ma_convert_pcm_frames_format(void* pOut, ma_format formatOut, const void* pIn, ma_format formatIn, ma_uint64 frameCount, ma_uint32 channels, ma_dither_mode ditherMode)
+MA_API ma_result ma_vfs_close(ma_vfs* pVFS, ma_vfs_file file)
{
- ma_pcm_convert(pOut, formatOut, pIn, formatIn, frameCount * channels, ditherMode);
+ ma_vfs_callbacks* pCallbacks = (ma_vfs_callbacks*)pVFS;
+
+ if (pVFS == NULL || file == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (pCallbacks->onClose == NULL) {
+ return MA_NOT_IMPLEMENTED;
+ }
+
+ return pCallbacks->onClose(pVFS, file);
}
-MA_API void ma_deinterleave_pcm_frames(ma_format format, ma_uint32 channels, ma_uint64 frameCount, const void* pInterleavedPCMFrames, void** ppDeinterleavedPCMFrames)
+MA_API ma_result ma_vfs_read(ma_vfs* pVFS, ma_vfs_file file, void* pDst, size_t sizeInBytes, size_t* pBytesRead)
{
- if (pInterleavedPCMFrames == NULL || ppDeinterleavedPCMFrames == NULL) {
- return; /* Invalid args. */
+ ma_vfs_callbacks* pCallbacks = (ma_vfs_callbacks*)pVFS;
+ ma_result result;
+ size_t bytesRead;
+
+ if (pBytesRead != NULL) {
+ *pBytesRead = 0;
}
- /* For efficiency we do this per format. */
- switch (format) {
- case ma_format_s16:
- {
- const ma_int16* pSrcS16 = (const ma_int16*)pInterleavedPCMFrames;
- ma_uint64 iPCMFrame;
- for (iPCMFrame = 0; iPCMFrame < frameCount; ++iPCMFrame) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < channels; ++iChannel) {
- ma_int16* pDstS16 = (ma_int16*)ppDeinterleavedPCMFrames[iChannel];
- pDstS16[iPCMFrame] = pSrcS16[iPCMFrame*channels+iChannel];
- }
- }
- } break;
+ if (pVFS == NULL || file == NULL || pDst == NULL) {
+ return MA_INVALID_ARGS;
+ }
- case ma_format_f32:
- {
- const float* pSrcF32 = (const float*)pInterleavedPCMFrames;
- ma_uint64 iPCMFrame;
- for (iPCMFrame = 0; iPCMFrame < frameCount; ++iPCMFrame) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < channels; ++iChannel) {
- float* pDstF32 = (float*)ppDeinterleavedPCMFrames[iChannel];
- pDstF32[iPCMFrame] = pSrcF32[iPCMFrame*channels+iChannel];
- }
- }
- } break;
+ if (pCallbacks->onRead == NULL) {
+ return MA_NOT_IMPLEMENTED;
+ }
- default:
- {
- ma_uint32 sampleSizeInBytes = ma_get_bytes_per_sample(format);
- ma_uint64 iPCMFrame;
- for (iPCMFrame = 0; iPCMFrame < frameCount; ++iPCMFrame) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < channels; ++iChannel) {
- void* pDst = ma_offset_ptr(ppDeinterleavedPCMFrames[iChannel], iPCMFrame*sampleSizeInBytes);
- const void* pSrc = ma_offset_ptr(pInterleavedPCMFrames, (iPCMFrame*channels+iChannel)*sampleSizeInBytes);
- memcpy(pDst, pSrc, sampleSizeInBytes);
- }
- }
- } break;
+ result = pCallbacks->onRead(pVFS, file, pDst, sizeInBytes, &bytesRead);
+
+ if (pBytesRead != NULL) {
+ *pBytesRead = bytesRead;
+ }
+
+ if (result == MA_SUCCESS && bytesRead == 0 && sizeInBytes > 0) {
+ result = MA_AT_END;
}
+
+ return result;
}
-MA_API void ma_interleave_pcm_frames(ma_format format, ma_uint32 channels, ma_uint64 frameCount, const void** ppDeinterleavedPCMFrames, void* pInterleavedPCMFrames)
+MA_API ma_result ma_vfs_write(ma_vfs* pVFS, ma_vfs_file file, const void* pSrc, size_t sizeInBytes, size_t* pBytesWritten)
{
- switch (format)
- {
- case ma_format_s16:
- {
- ma_int16* pDstS16 = (ma_int16*)pInterleavedPCMFrames;
- ma_uint64 iPCMFrame;
- for (iPCMFrame = 0; iPCMFrame < frameCount; ++iPCMFrame) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < channels; ++iChannel) {
- const ma_int16* pSrcS16 = (const ma_int16*)ppDeinterleavedPCMFrames[iChannel];
- pDstS16[iPCMFrame*channels+iChannel] = pSrcS16[iPCMFrame];
- }
- }
- } break;
+ ma_vfs_callbacks* pCallbacks = (ma_vfs_callbacks*)pVFS;
- case ma_format_f32:
- {
- float* pDstF32 = (float*)pInterleavedPCMFrames;
- ma_uint64 iPCMFrame;
- for (iPCMFrame = 0; iPCMFrame < frameCount; ++iPCMFrame) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < channels; ++iChannel) {
- const float* pSrcF32 = (const float*)ppDeinterleavedPCMFrames[iChannel];
- pDstF32[iPCMFrame*channels+iChannel] = pSrcF32[iPCMFrame];
- }
- }
- } break;
+ if (pBytesWritten != NULL) {
+ *pBytesWritten = 0;
+ }
- default:
- {
- ma_uint32 sampleSizeInBytes = ma_get_bytes_per_sample(format);
- ma_uint64 iPCMFrame;
- for (iPCMFrame = 0; iPCMFrame < frameCount; ++iPCMFrame) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < channels; ++iChannel) {
- void* pDst = ma_offset_ptr(pInterleavedPCMFrames, (iPCMFrame*channels+iChannel)*sampleSizeInBytes);
- const void* pSrc = ma_offset_ptr(ppDeinterleavedPCMFrames[iChannel], iPCMFrame*sampleSizeInBytes);
- memcpy(pDst, pSrc, sampleSizeInBytes);
- }
- }
- } break;
+ if (pVFS == NULL || file == NULL || pSrc == NULL) {
+ return MA_INVALID_ARGS;
}
-}
+ if (pCallbacks->onWrite == NULL) {
+ return MA_NOT_IMPLEMENTED;
+ }
-/**************************************************************************************************************************************************************
+ return pCallbacks->onWrite(pVFS, file, pSrc, sizeInBytes, pBytesWritten);
+}
-Biquad Filter
+MA_API ma_result ma_vfs_seek(ma_vfs* pVFS, ma_vfs_file file, ma_int64 offset, ma_seek_origin origin)
+{
+ ma_vfs_callbacks* pCallbacks = (ma_vfs_callbacks*)pVFS;
-**************************************************************************************************************************************************************/
-#ifndef MA_BIQUAD_FIXED_POINT_SHIFT
-#define MA_BIQUAD_FIXED_POINT_SHIFT 14
-#endif
+ if (pVFS == NULL || file == NULL) {
+ return MA_INVALID_ARGS;
+ }
-static ma_int32 ma_biquad_float_to_fp(double x)
-{
- return (ma_int32)(x * (1 << MA_BIQUAD_FIXED_POINT_SHIFT));
+ if (pCallbacks->onSeek == NULL) {
+ return MA_NOT_IMPLEMENTED;
+ }
+
+ return pCallbacks->onSeek(pVFS, file, offset, origin);
}
-MA_API ma_biquad_config ma_biquad_config_init(ma_format format, ma_uint32 channels, double b0, double b1, double b2, double a0, double a1, double a2)
+MA_API ma_result ma_vfs_tell(ma_vfs* pVFS, ma_vfs_file file, ma_int64* pCursor)
{
- ma_biquad_config config;
+ ma_vfs_callbacks* pCallbacks = (ma_vfs_callbacks*)pVFS;
- MA_ZERO_OBJECT(&config);
- config.format = format;
- config.channels = channels;
- config.b0 = b0;
- config.b1 = b1;
- config.b2 = b2;
- config.a0 = a0;
- config.a1 = a1;
- config.a2 = a2;
+ if (pCursor == NULL) {
+ return MA_INVALID_ARGS;
+ }
- return config;
+ *pCursor = 0;
+
+ if (pVFS == NULL || file == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (pCallbacks->onTell == NULL) {
+ return MA_NOT_IMPLEMENTED;
+ }
+
+ return pCallbacks->onTell(pVFS, file, pCursor);
}
-MA_API ma_result ma_biquad_init(const ma_biquad_config* pConfig, ma_biquad* pBQ)
+MA_API ma_result ma_vfs_info(ma_vfs* pVFS, ma_vfs_file file, ma_file_info* pInfo)
{
- if (pBQ == NULL) {
+ ma_vfs_callbacks* pCallbacks = (ma_vfs_callbacks*)pVFS;
+
+ if (pInfo == NULL) {
return MA_INVALID_ARGS;
}
- MA_ZERO_OBJECT(pBQ);
+ MA_ZERO_OBJECT(pInfo);
- if (pConfig == NULL) {
+ if (pVFS == NULL || file == NULL) {
return MA_INVALID_ARGS;
}
- if (pConfig->channels < MA_MIN_CHANNELS || pConfig->channels > MA_MAX_CHANNELS) {
- return MA_INVALID_ARGS;
+ if (pCallbacks->onInfo == NULL) {
+ return MA_NOT_IMPLEMENTED;
}
- return ma_biquad_reinit(pConfig, pBQ);
+ return pCallbacks->onInfo(pVFS, file, pInfo);
}
-MA_API ma_result ma_biquad_reinit(const ma_biquad_config* pConfig, ma_biquad* pBQ)
+
+static ma_result ma_vfs_open_and_read_file_ex(ma_vfs* pVFS, const char* pFilePath, const wchar_t* pFilePathW, void** ppData, size_t* pSize, const ma_allocation_callbacks* pAllocationCallbacks)
{
- if (pBQ == NULL || pConfig == NULL) {
+ ma_result result;
+ ma_vfs_file file;
+ ma_file_info info;
+ void* pData;
+ size_t bytesRead;
+
+ if (ppData != NULL) {
+ *ppData = NULL;
+ }
+ if (pSize != NULL) {
+ *pSize = 0;
+ }
+
+ if (ppData == NULL) {
return MA_INVALID_ARGS;
}
- if (pConfig->a0 == 0) {
- return MA_INVALID_ARGS; /* Division by zero. */
+ if (pFilePath != NULL) {
+ result = ma_vfs_open(pVFS, pFilePath, MA_OPEN_MODE_READ, &file);
+ } else {
+ result = ma_vfs_open_w(pVFS, pFilePathW, MA_OPEN_MODE_READ, &file);
+ }
+ if (result != MA_SUCCESS) {
+ return result;
}
- /* Only supporting f32 and s16. */
- if (pConfig->format != ma_format_f32 && pConfig->format != ma_format_s16) {
- return MA_INVALID_ARGS;
+ result = ma_vfs_info(pVFS, file, &info);
+ if (result != MA_SUCCESS) {
+ ma_vfs_close(pVFS, file);
+ return result;
}
- /* The format cannot be changed after initialization. */
- if (pBQ->format != ma_format_unknown && pBQ->format != pConfig->format) {
- return MA_INVALID_OPERATION;
+ if (info.sizeInBytes > MA_SIZE_MAX) {
+ ma_vfs_close(pVFS, file);
+ return MA_TOO_BIG;
}
- /* The channel count cannot be changed after initialization. */
- if (pBQ->channels != 0 && pBQ->channels != pConfig->channels) {
- return MA_INVALID_OPERATION;
+ pData = ma_malloc((size_t)info.sizeInBytes, pAllocationCallbacks); /* Safe cast. */
+ if (pData == NULL) {
+ ma_vfs_close(pVFS, file);
+ return result;
}
+ result = ma_vfs_read(pVFS, file, pData, (size_t)info.sizeInBytes, &bytesRead); /* Safe cast. */
+ ma_vfs_close(pVFS, file);
- pBQ->format = pConfig->format;
- pBQ->channels = pConfig->channels;
+ if (result != MA_SUCCESS) {
+ ma_free(pData, pAllocationCallbacks);
+ return result;
+ }
- /* Normalize. */
- if (pConfig->format == ma_format_f32) {
- pBQ->b0.f32 = (float)(pConfig->b0 / pConfig->a0);
- pBQ->b1.f32 = (float)(pConfig->b1 / pConfig->a0);
- pBQ->b2.f32 = (float)(pConfig->b2 / pConfig->a0);
- pBQ->a1.f32 = (float)(pConfig->a1 / pConfig->a0);
- pBQ->a2.f32 = (float)(pConfig->a2 / pConfig->a0);
- } else {
- pBQ->b0.s32 = ma_biquad_float_to_fp(pConfig->b0 / pConfig->a0);
- pBQ->b1.s32 = ma_biquad_float_to_fp(pConfig->b1 / pConfig->a0);
- pBQ->b2.s32 = ma_biquad_float_to_fp(pConfig->b2 / pConfig->a0);
- pBQ->a1.s32 = ma_biquad_float_to_fp(pConfig->a1 / pConfig->a0);
- pBQ->a2.s32 = ma_biquad_float_to_fp(pConfig->a2 / pConfig->a0);
+ if (pSize != NULL) {
+ *pSize = bytesRead;
}
+ MA_ASSERT(ppData != NULL);
+ *ppData = pData;
+
return MA_SUCCESS;
}
-static MA_INLINE void ma_biquad_process_pcm_frame_f32__direct_form_2_transposed(ma_biquad* pBQ, float* pY, const float* pX)
+MA_API ma_result ma_vfs_open_and_read_file(ma_vfs* pVFS, const char* pFilePath, void** ppData, size_t* pSize, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_uint32 c;
- const ma_uint32 channels = pBQ->channels;
- const float b0 = pBQ->b0.f32;
- const float b1 = pBQ->b1.f32;
- const float b2 = pBQ->b2.f32;
- const float a1 = pBQ->a1.f32;
- const float a2 = pBQ->a2.f32;
+ return ma_vfs_open_and_read_file_ex(pVFS, pFilePath, NULL, ppData, pSize, pAllocationCallbacks);
+}
- MA_ASSUME(channels >= MA_MIN_CHANNELS && channels <= MA_MAX_CHANNELS);
- for (c = 0; c < channels; c += 1) {
- float r1 = pBQ->r1[c].f32;
- float r2 = pBQ->r2[c].f32;
- float x = pX[c];
- float y;
+MA_API ma_result ma_vfs_open_and_read_file_w(ma_vfs* pVFS, const wchar_t* pFilePath, void** ppData, size_t* pSize, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ return ma_vfs_open_and_read_file_ex(pVFS, NULL, pFilePath, ppData, pSize, pAllocationCallbacks);
+}
- y = b0*x + r1;
- r1 = b1*x - a1*y + r2;
- r2 = b2*x - a2*y;
- pY[c] = y;
- pBQ->r1[c].f32 = r1;
- pBQ->r2[c].f32 = r2;
+#if defined(MA_WIN32) && defined(MA_WIN32_DESKTOP) && !defined(MA_NO_WIN32_FILEIO)
+static void ma_default_vfs__get_open_settings_win32(ma_uint32 openMode, DWORD* pDesiredAccess, DWORD* pShareMode, DWORD* pCreationDisposition)
+{
+ *pDesiredAccess = 0;
+ if ((openMode & MA_OPEN_MODE_READ) != 0) {
+ *pDesiredAccess |= GENERIC_READ;
+ }
+ if ((openMode & MA_OPEN_MODE_WRITE) != 0) {
+ *pDesiredAccess |= GENERIC_WRITE;
+ }
+
+ *pShareMode = 0;
+ if ((openMode & MA_OPEN_MODE_READ) != 0) {
+ *pShareMode |= FILE_SHARE_READ;
+ }
+
+ if ((openMode & MA_OPEN_MODE_WRITE) != 0) {
+ *pCreationDisposition = CREATE_ALWAYS; /* Opening in write mode. Truncate. */
+ } else {
+ *pCreationDisposition = OPEN_EXISTING; /* Opening in read mode. File must exist. */
}
}
-static MA_INLINE void ma_biquad_process_pcm_frame_f32(ma_biquad* pBQ, float* pY, const float* pX)
+static ma_result ma_default_vfs_open__win32(ma_vfs* pVFS, const char* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
{
- ma_biquad_process_pcm_frame_f32__direct_form_2_transposed(pBQ, pY, pX);
+ HANDLE hFile;
+ DWORD dwDesiredAccess;
+ DWORD dwShareMode;
+ DWORD dwCreationDisposition;
+
+ (void)pVFS;
+
+ ma_default_vfs__get_open_settings_win32(openMode, &dwDesiredAccess, &dwShareMode, &dwCreationDisposition);
+
+ hFile = CreateFileA(pFilePath, dwDesiredAccess, dwShareMode, NULL, dwCreationDisposition, FILE_ATTRIBUTE_NORMAL, NULL);
+ if (hFile == INVALID_HANDLE_VALUE) {
+ return ma_result_from_GetLastError(GetLastError());
+ }
+
+ *pFile = hFile;
+ return MA_SUCCESS;
}
-static MA_INLINE void ma_biquad_process_pcm_frame_s16__direct_form_2_transposed(ma_biquad* pBQ, ma_int16* pY, const ma_int16* pX)
+static ma_result ma_default_vfs_open_w__win32(ma_vfs* pVFS, const wchar_t* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
{
- ma_uint32 c;
- const ma_uint32 channels = pBQ->channels;
- const ma_int32 b0 = pBQ->b0.s32;
- const ma_int32 b1 = pBQ->b1.s32;
- const ma_int32 b2 = pBQ->b2.s32;
- const ma_int32 a1 = pBQ->a1.s32;
- const ma_int32 a2 = pBQ->a2.s32;
+ HANDLE hFile;
+ DWORD dwDesiredAccess;
+ DWORD dwShareMode;
+ DWORD dwCreationDisposition;
- MA_ASSUME(channels >= MA_MIN_CHANNELS && channels <= MA_MAX_CHANNELS);
- for (c = 0; c < channels; c += 1) {
- ma_int32 r1 = pBQ->r1[c].s32;
- ma_int32 r2 = pBQ->r2[c].s32;
- ma_int32 x = pX[c];
- ma_int32 y;
+ (void)pVFS;
- y = (b0*x + r1) >> MA_BIQUAD_FIXED_POINT_SHIFT;
- r1 = (b1*x - a1*y + r2);
- r2 = (b2*x - a2*y);
+ ma_default_vfs__get_open_settings_win32(openMode, &dwDesiredAccess, &dwShareMode, &dwCreationDisposition);
+
+ hFile = CreateFileW(pFilePath, dwDesiredAccess, dwShareMode, NULL, dwCreationDisposition, FILE_ATTRIBUTE_NORMAL, NULL);
+ if (hFile == INVALID_HANDLE_VALUE) {
+ return ma_result_from_GetLastError(GetLastError());
+ }
+
+ *pFile = hFile;
+ return MA_SUCCESS;
+}
+
+static ma_result ma_default_vfs_close__win32(ma_vfs* pVFS, ma_vfs_file file)
+{
+ (void)pVFS;
- pY[c] = (ma_int16)ma_clamp(y, -32768, 32767);
- pBQ->r1[c].s32 = r1;
- pBQ->r2[c].s32 = r2;
+ if (CloseHandle((HANDLE)file) == 0) {
+ return ma_result_from_GetLastError(GetLastError());
}
+
+ return MA_SUCCESS;
}
-static MA_INLINE void ma_biquad_process_pcm_frame_s16(ma_biquad* pBQ, ma_int16* pY, const ma_int16* pX)
-{
- ma_biquad_process_pcm_frame_s16__direct_form_2_transposed(pBQ, pY, pX);
+
+static ma_result ma_default_vfs_read__win32(ma_vfs* pVFS, ma_vfs_file file, void* pDst, size_t sizeInBytes, size_t* pBytesRead)
+{
+ ma_result result = MA_SUCCESS;
+ size_t totalBytesRead;
+
+ (void)pVFS;
+
+ totalBytesRead = 0;
+ while (totalBytesRead < sizeInBytes) {
+ size_t bytesRemaining;
+ DWORD bytesToRead;
+ DWORD bytesRead;
+ BOOL readResult;
+
+ bytesRemaining = sizeInBytes - totalBytesRead;
+ if (bytesRemaining >= 0xFFFFFFFF) {
+ bytesToRead = 0xFFFFFFFF;
+ } else {
+ bytesToRead = (DWORD)bytesRemaining;
+ }
+
+ readResult = ReadFile((HANDLE)file, ma_offset_ptr(pDst, totalBytesRead), bytesToRead, &bytesRead, NULL);
+ if (readResult == 1 && bytesRead == 0) {
+ result = MA_AT_END;
+ break; /* EOF */
+ }
+
+ totalBytesRead += bytesRead;
+
+ if (bytesRead < bytesToRead) {
+ break; /* EOF */
+ }
+
+ if (readResult == 0) {
+ result = ma_result_from_GetLastError(GetLastError());
+ break;
+ }
+ }
+
+ if (pBytesRead != NULL) {
+ *pBytesRead = totalBytesRead;
+ }
+
+ return result;
}
-MA_API ma_result ma_biquad_process_pcm_frames(ma_biquad* pBQ, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+static ma_result ma_default_vfs_write__win32(ma_vfs* pVFS, ma_vfs_file file, const void* pSrc, size_t sizeInBytes, size_t* pBytesWritten)
{
- ma_uint32 n;
-
- if (pBQ == NULL || pFramesOut == NULL || pFramesIn == NULL) {
- return MA_INVALID_ARGS;
- }
+ ma_result result = MA_SUCCESS;
+ size_t totalBytesWritten;
- /* Note that the logic below needs to support in-place filtering. That is, it must support the case where pFramesOut and pFramesIn are the same. */
+ (void)pVFS;
- if (pBQ->format == ma_format_f32) {
- /* */ float* pY = ( float*)pFramesOut;
- const float* pX = (const float*)pFramesIn;
+ totalBytesWritten = 0;
+ while (totalBytesWritten < sizeInBytes) {
+ size_t bytesRemaining;
+ DWORD bytesToWrite;
+ DWORD bytesWritten;
+ BOOL writeResult;
- for (n = 0; n < frameCount; n += 1) {
- ma_biquad_process_pcm_frame_f32__direct_form_2_transposed(pBQ, pY, pX);
- pY += pBQ->channels;
- pX += pBQ->channels;
+ bytesRemaining = sizeInBytes - totalBytesWritten;
+ if (bytesRemaining >= 0xFFFFFFFF) {
+ bytesToWrite = 0xFFFFFFFF;
+ } else {
+ bytesToWrite = (DWORD)bytesRemaining;
}
- } else if (pBQ->format == ma_format_s16) {
- /* */ ma_int16* pY = ( ma_int16*)pFramesOut;
- const ma_int16* pX = (const ma_int16*)pFramesIn;
- for (n = 0; n < frameCount; n += 1) {
- ma_biquad_process_pcm_frame_s16__direct_form_2_transposed(pBQ, pY, pX);
- pY += pBQ->channels;
- pX += pBQ->channels;
+ writeResult = WriteFile((HANDLE)file, ma_offset_ptr(pSrc, totalBytesWritten), bytesToWrite, &bytesWritten, NULL);
+ totalBytesWritten += bytesWritten;
+
+ if (writeResult == 0) {
+ result = ma_result_from_GetLastError(GetLastError());
+ break;
}
- } else {
- MA_ASSERT(MA_FALSE);
- return MA_INVALID_ARGS; /* Format not supported. Should never hit this because it's checked in ma_biquad_init() and ma_biquad_reinit(). */
}
- return MA_SUCCESS;
-}
-
-MA_API ma_uint32 ma_biquad_get_latency(const ma_biquad* pBQ)
-{
- if (pBQ == NULL) {
- return 0;
+ if (pBytesWritten != NULL) {
+ *pBytesWritten = totalBytesWritten;
}
- return 2;
+ return result;
}
-/**************************************************************************************************************************************************************
-
-Low-Pass Filter
-
-**************************************************************************************************************************************************************/
-MA_API ma_lpf1_config ma_lpf1_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency)
+static ma_result ma_default_vfs_seek__win32(ma_vfs* pVFS, ma_vfs_file file, ma_int64 offset, ma_seek_origin origin)
{
- ma_lpf1_config config;
+ LARGE_INTEGER liDistanceToMove;
+ DWORD dwMoveMethod;
+ BOOL result;
- MA_ZERO_OBJECT(&config);
- config.format = format;
- config.channels = channels;
- config.sampleRate = sampleRate;
- config.cutoffFrequency = cutoffFrequency;
- config.q = 0.5;
+ (void)pVFS;
- return config;
-}
+ liDistanceToMove.QuadPart = offset;
-MA_API ma_lpf2_config ma_lpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q)
-{
- ma_lpf2_config config;
+ /* */ if (origin == ma_seek_origin_current) {
+ dwMoveMethod = FILE_CURRENT;
+ } else if (origin == ma_seek_origin_end) {
+ dwMoveMethod = FILE_END;
+ } else {
+ dwMoveMethod = FILE_BEGIN;
+ }
- MA_ZERO_OBJECT(&config);
- config.format = format;
- config.channels = channels;
- config.sampleRate = sampleRate;
- config.cutoffFrequency = cutoffFrequency;
- config.q = q;
+#if (defined(_MSC_VER) && _MSC_VER <= 1200) || defined(__DMC__)
+ /* No SetFilePointerEx() so restrict to 31 bits. */
+ if (origin > 0x7FFFFFFF) {
+ return MA_OUT_OF_RANGE;
+ }
- /* Q cannot be 0 or else it'll result in a division by 0. In this case just default to 0.707107. */
- if (config.q == 0) {
- config.q = 0.707107;
+ result = SetFilePointer((HANDLE)file, (LONG)liDistanceToMove.QuadPart, NULL, dwMoveMethod);
+#else
+ result = SetFilePointerEx((HANDLE)file, liDistanceToMove, NULL, dwMoveMethod);
+#endif
+ if (result == 0) {
+ return ma_result_from_GetLastError(GetLastError());
}
- return config;
+ return MA_SUCCESS;
}
-
-MA_API ma_result ma_lpf1_init(const ma_lpf1_config* pConfig, ma_lpf1* pLPF)
+static ma_result ma_default_vfs_tell__win32(ma_vfs* pVFS, ma_vfs_file file, ma_int64* pCursor)
{
- if (pLPF == NULL) {
- return MA_INVALID_ARGS;
- }
+ LARGE_INTEGER liZero;
+ LARGE_INTEGER liTell;
+ BOOL result;
+#if (defined(_MSC_VER) && _MSC_VER <= 1200) || defined(__DMC__)
+ LONG tell;
+#endif
- MA_ZERO_OBJECT(pLPF);
+ (void)pVFS;
- if (pConfig == NULL) {
- return MA_INVALID_ARGS;
+ liZero.QuadPart = 0;
+
+#if (defined(_MSC_VER) && _MSC_VER <= 1200) || defined(__DMC__)
+ result = SetFilePointer((HANDLE)file, (LONG)liZero.QuadPart, &tell, FILE_CURRENT);
+ liTell.QuadPart = tell;
+#else
+ result = SetFilePointerEx((HANDLE)file, liZero, &liTell, FILE_CURRENT);
+#endif
+ if (result == 0) {
+ return ma_result_from_GetLastError(GetLastError());
}
- if (pConfig->channels < MA_MIN_CHANNELS || pConfig->channels > MA_MAX_CHANNELS) {
- return MA_INVALID_ARGS;
+ if (pCursor != NULL) {
+ *pCursor = liTell.QuadPart;
}
- return ma_lpf1_reinit(pConfig, pLPF);
+ return MA_SUCCESS;
}
-MA_API ma_result ma_lpf1_reinit(const ma_lpf1_config* pConfig, ma_lpf1* pLPF)
+static ma_result ma_default_vfs_info__win32(ma_vfs* pVFS, ma_vfs_file file, ma_file_info* pInfo)
{
- double a;
+ BY_HANDLE_FILE_INFORMATION fi;
+ BOOL result;
- if (pLPF == NULL || pConfig == NULL) {
- return MA_INVALID_ARGS;
- }
+ (void)pVFS;
- /* Only supporting f32 and s16. */
- if (pConfig->format != ma_format_f32 && pConfig->format != ma_format_s16) {
- return MA_INVALID_ARGS;
+ result = GetFileInformationByHandle((HANDLE)file, &fi);
+ if (result == 0) {
+ return ma_result_from_GetLastError(GetLastError());
}
- /* The format cannot be changed after initialization. */
- if (pLPF->format != ma_format_unknown && pLPF->format != pConfig->format) {
- return MA_INVALID_OPERATION;
- }
+ pInfo->sizeInBytes = ((ma_uint64)fi.nFileSizeHigh << 32) | ((ma_uint64)fi.nFileSizeLow);
- /* The channel count cannot be changed after initialization. */
- if (pLPF->channels != 0 && pLPF->channels != pConfig->channels) {
- return MA_INVALID_OPERATION;
- }
+ return MA_SUCCESS;
+}
+#else
+static ma_result ma_default_vfs_open__stdio(ma_vfs* pVFS, const char* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
+{
+ ma_result result;
+ FILE* pFileStd;
+ const char* pOpenModeStr;
- pLPF->format = pConfig->format;
- pLPF->channels = pConfig->channels;
+ MA_ASSERT(pFilePath != NULL);
+ MA_ASSERT(openMode != 0);
+ MA_ASSERT(pFile != NULL);
- a = ma_expd(-2 * MA_PI_D * pConfig->cutoffFrequency / pConfig->sampleRate);
- if (pConfig->format == ma_format_f32) {
- pLPF->a.f32 = (float)a;
+ (void)pVFS;
+
+ if ((openMode & MA_OPEN_MODE_READ) != 0) {
+ if ((openMode & MA_OPEN_MODE_WRITE) != 0) {
+ pOpenModeStr = "r+";
+ } else {
+ pOpenModeStr = "rb";
+ }
} else {
- pLPF->a.s32 = ma_biquad_float_to_fp(a);
+ pOpenModeStr = "wb";
+ }
+
+ result = ma_fopen(&pFileStd, pFilePath, pOpenModeStr);
+ if (result != MA_SUCCESS) {
+ return result;
}
+ *pFile = pFileStd;
+
return MA_SUCCESS;
}
-static MA_INLINE void ma_lpf1_process_pcm_frame_f32(ma_lpf1* pLPF, float* pY, const float* pX)
+static ma_result ma_default_vfs_open_w__stdio(ma_vfs* pVFS, const wchar_t* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
{
- ma_uint32 c;
- const ma_uint32 channels = pLPF->channels;
- const float a = pLPF->a.f32;
- const float b = 1 - a;
+ ma_result result;
+ FILE* pFileStd;
+ const wchar_t* pOpenModeStr;
- MA_ASSUME(channels >= MA_MIN_CHANNELS && channels <= MA_MAX_CHANNELS);
- for (c = 0; c < channels; c += 1) {
- float r1 = pLPF->r1[c].f32;
- float x = pX[c];
- float y;
+ MA_ASSERT(pFilePath != NULL);
+ MA_ASSERT(openMode != 0);
+ MA_ASSERT(pFile != NULL);
- y = b*x + a*r1;
+ (void)pVFS;
- pY[c] = y;
- pLPF->r1[c].f32 = y;
+ if ((openMode & MA_OPEN_MODE_READ) != 0) {
+ if ((openMode & MA_OPEN_MODE_WRITE) != 0) {
+ pOpenModeStr = L"r+";
+ } else {
+ pOpenModeStr = L"rb";
+ }
+ } else {
+ pOpenModeStr = L"wb";
+ }
+
+ result = ma_wfopen(&pFileStd, pFilePath, pOpenModeStr, (pVFS != NULL) ? &((ma_default_vfs*)pVFS)->allocationCallbacks : NULL);
+ if (result != MA_SUCCESS) {
+ return result;
}
+
+ *pFile = pFileStd;
+
+ return MA_SUCCESS;
}
-static MA_INLINE void ma_lpf1_process_pcm_frame_s16(ma_lpf1* pLPF, ma_int16* pY, const ma_int16* pX)
+static ma_result ma_default_vfs_close__stdio(ma_vfs* pVFS, ma_vfs_file file)
{
- ma_uint32 c;
- const ma_uint32 channels = pLPF->channels;
- const ma_int32 a = pLPF->a.s32;
- const ma_int32 b = ((1 << MA_BIQUAD_FIXED_POINT_SHIFT) - a);
+ MA_ASSERT(file != NULL);
- MA_ASSUME(channels >= MA_MIN_CHANNELS && channels <= MA_MAX_CHANNELS);
- for (c = 0; c < channels; c += 1) {
- ma_int32 r1 = pLPF->r1[c].s32;
- ma_int32 x = pX[c];
- ma_int32 y;
+ (void)pVFS;
- y = (b*x + a*r1) >> MA_BIQUAD_FIXED_POINT_SHIFT;
+ fclose((FILE*)file);
- pY[c] = (ma_int16)y;
- pLPF->r1[c].s32 = (ma_int32)y;
- }
+ return MA_SUCCESS;
}
-MA_API ma_result ma_lpf1_process_pcm_frames(ma_lpf1* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+static ma_result ma_default_vfs_read__stdio(ma_vfs* pVFS, ma_vfs_file file, void* pDst, size_t sizeInBytes, size_t* pBytesRead)
{
- ma_uint32 n;
+ size_t result;
- if (pLPF == NULL || pFramesOut == NULL || pFramesIn == NULL) {
- return MA_INVALID_ARGS;
- }
+ MA_ASSERT(file != NULL);
+ MA_ASSERT(pDst != NULL);
- /* Note that the logic below needs to support in-place filtering. That is, it must support the case where pFramesOut and pFramesIn are the same. */
+ (void)pVFS;
- if (pLPF->format == ma_format_f32) {
- /* */ float* pY = ( float*)pFramesOut;
- const float* pX = (const float*)pFramesIn;
+ result = fread(pDst, 1, sizeInBytes, (FILE*)file);
- for (n = 0; n < frameCount; n += 1) {
- ma_lpf1_process_pcm_frame_f32(pLPF, pY, pX);
- pY += pLPF->channels;
- pX += pLPF->channels;
- }
- } else if (pLPF->format == ma_format_s16) {
- /* */ ma_int16* pY = ( ma_int16*)pFramesOut;
- const ma_int16* pX = (const ma_int16*)pFramesIn;
+ if (pBytesRead != NULL) {
+ *pBytesRead = result;
+ }
- for (n = 0; n < frameCount; n += 1) {
- ma_lpf1_process_pcm_frame_s16(pLPF, pY, pX);
- pY += pLPF->channels;
- pX += pLPF->channels;
+ if (result != sizeInBytes) {
+ if (result == 0 && feof((FILE*)file)) {
+ return MA_AT_END;
+ } else {
+ return ma_result_from_errno(ferror((FILE*)file));
}
- } else {
- MA_ASSERT(MA_FALSE);
- return MA_INVALID_ARGS; /* Format not supported. Should never hit this because it's checked in ma_biquad_init() and ma_biquad_reinit(). */
}
return MA_SUCCESS;
}
-MA_API ma_uint32 ma_lpf1_get_latency(const ma_lpf1* pLPF)
+static ma_result ma_default_vfs_write__stdio(ma_vfs* pVFS, ma_vfs_file file, const void* pSrc, size_t sizeInBytes, size_t* pBytesWritten)
{
- if (pLPF == NULL) {
- return 0;
+ size_t result;
+
+ MA_ASSERT(file != NULL);
+ MA_ASSERT(pSrc != NULL);
+
+ (void)pVFS;
+
+ result = fwrite(pSrc, 1, sizeInBytes, (FILE*)file);
+
+ if (pBytesWritten != NULL) {
+ *pBytesWritten = result;
}
- return 1;
-}
+ if (result != sizeInBytes) {
+ return ma_result_from_errno(ferror((FILE*)file));
+ }
+ return MA_SUCCESS;
+}
-static MA_INLINE ma_biquad_config ma_lpf2__get_biquad_config(const ma_lpf2_config* pConfig)
+static ma_result ma_default_vfs_seek__stdio(ma_vfs* pVFS, ma_vfs_file file, ma_int64 offset, ma_seek_origin origin)
{
- ma_biquad_config bqConfig;
- double q;
- double w;
- double s;
- double c;
- double a;
+ int result;
+ int whence;
- MA_ASSERT(pConfig != NULL);
+ MA_ASSERT(file != NULL);
- q = pConfig->q;
- w = 2 * MA_PI_D * pConfig->cutoffFrequency / pConfig->sampleRate;
- s = ma_sind(w);
- c = ma_cosd(w);
- a = s / (2*q);
+ (void)pVFS;
- bqConfig.b0 = (1 - c) / 2;
- bqConfig.b1 = 1 - c;
- bqConfig.b2 = (1 - c) / 2;
- bqConfig.a0 = 1 + a;
- bqConfig.a1 = -2 * c;
- bqConfig.a2 = 1 - a;
+ if (origin == ma_seek_origin_start) {
+ whence = SEEK_SET;
+ } else if (origin == ma_seek_origin_end) {
+ whence = SEEK_END;
+ } else {
+ whence = SEEK_CUR;
+ }
- bqConfig.format = pConfig->format;
- bqConfig.channels = pConfig->channels;
+#if defined(_WIN32)
+ #if defined(_MSC_VER) && _MSC_VER > 1200
+ result = _fseeki64((FILE*)file, offset, whence);
+ #else
+ /* No _fseeki64() so restrict to 31 bits. */
+ if (origin > 0x7FFFFFFF) {
+ return MA_OUT_OF_RANGE;
+ }
- return bqConfig;
+ result = fseek((FILE*)file, (int)offset, whence);
+ #endif
+#else
+ result = fseek((FILE*)file, (long int)offset, whence);
+#endif
+ if (result != 0) {
+ return MA_ERROR;
+ }
+
+ return MA_SUCCESS;
}
-MA_API ma_result ma_lpf2_init(const ma_lpf2_config* pConfig, ma_lpf2* pLPF)
+static ma_result ma_default_vfs_tell__stdio(ma_vfs* pVFS, ma_vfs_file file, ma_int64* pCursor)
{
- ma_result result;
- ma_biquad_config bqConfig;
+ ma_int64 result;
- if (pLPF == NULL) {
- return MA_INVALID_ARGS;
- }
+ MA_ASSERT(file != NULL);
+ MA_ASSERT(pCursor != NULL);
- MA_ZERO_OBJECT(pLPF);
+ (void)pVFS;
- if (pConfig == NULL) {
- return MA_INVALID_ARGS;
- }
+#if defined(_WIN32)
+ #if defined(_MSC_VER) && _MSC_VER > 1200
+ result = _ftelli64((FILE*)file);
+ #else
+ result = ftell((FILE*)file);
+ #endif
+#else
+ result = ftell((FILE*)file);
+#endif
- bqConfig = ma_lpf2__get_biquad_config(pConfig);
- result = ma_biquad_init(&bqConfig, &pLPF->bq);
- if (result != MA_SUCCESS) {
- return result;
- }
+ *pCursor = result;
return MA_SUCCESS;
}
-MA_API ma_result ma_lpf2_reinit(const ma_lpf2_config* pConfig, ma_lpf2* pLPF)
+#if !defined(_MSC_VER) && !((defined(_POSIX_C_SOURCE) && _POSIX_C_SOURCE >= 1) || defined(_XOPEN_SOURCE) || defined(_POSIX_SOURCE)) && !defined(MA_BSD)
+int fileno(FILE *stream);
+#endif
+
+static ma_result ma_default_vfs_info__stdio(ma_vfs* pVFS, ma_vfs_file file, ma_file_info* pInfo)
{
- ma_result result;
- ma_biquad_config bqConfig;
+ int fd;
+ struct stat info;
- if (pLPF == NULL || pConfig == NULL) {
- return MA_INVALID_ARGS;
- }
+ MA_ASSERT(file != NULL);
+ MA_ASSERT(pInfo != NULL);
- bqConfig = ma_lpf2__get_biquad_config(pConfig);
- result = ma_biquad_reinit(&bqConfig, &pLPF->bq);
- if (result != MA_SUCCESS) {
- return result;
+ (void)pVFS;
+
+#if defined(_MSC_VER)
+ fd = _fileno((FILE*)file);
+#else
+ fd = fileno((FILE*)file);
+#endif
+
+ if (fstat(fd, &info) != 0) {
+ return ma_result_from_errno(errno);
}
+ pInfo->sizeInBytes = info.st_size;
+
return MA_SUCCESS;
}
+#endif
-static MA_INLINE void ma_lpf2_process_pcm_frame_s16(ma_lpf2* pLPF, ma_int16* pFrameOut, const ma_int16* pFrameIn)
-{
- ma_biquad_process_pcm_frame_s16(&pLPF->bq, pFrameOut, pFrameIn);
-}
-static MA_INLINE void ma_lpf2_process_pcm_frame_f32(ma_lpf2* pLPF, float* pFrameOut, const float* pFrameIn)
+static ma_result ma_default_vfs_open(ma_vfs* pVFS, const char* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
{
- ma_biquad_process_pcm_frame_f32(&pLPF->bq, pFrameOut, pFrameIn);
-}
+ if (pFile == NULL) {
+ return MA_INVALID_ARGS;
+ }
-MA_API ma_result ma_lpf2_process_pcm_frames(ma_lpf2* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
-{
- if (pLPF == NULL) {
+ *pFile = NULL;
+
+ if (pFilePath == NULL || openMode == 0) {
return MA_INVALID_ARGS;
}
- return ma_biquad_process_pcm_frames(&pLPF->bq, pFramesOut, pFramesIn, frameCount);
+#if defined(MA_WIN32) && defined(MA_WIN32_DESKTOP) && !defined(MA_NO_WIN32_FILEIO)
+ return ma_default_vfs_open__win32(pVFS, pFilePath, openMode, pFile);
+#else
+ return ma_default_vfs_open__stdio(pVFS, pFilePath, openMode, pFile);
+#endif
}
-MA_API ma_uint32 ma_lpf2_get_latency(const ma_lpf2* pLPF)
+static ma_result ma_default_vfs_open_w(ma_vfs* pVFS, const wchar_t* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
{
- if (pLPF == NULL) {
- return 0;
+ if (pFile == NULL) {
+ return MA_INVALID_ARGS;
}
- return ma_biquad_get_latency(&pLPF->bq);
-}
+ *pFile = NULL;
+ if (pFilePath == NULL || openMode == 0) {
+ return MA_INVALID_ARGS;
+ }
-MA_API ma_lpf_config ma_lpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order)
-{
- ma_lpf_config config;
+#if defined(MA_WIN32) && defined(MA_WIN32_DESKTOP) && !defined(MA_NO_WIN32_FILEIO)
+ return ma_default_vfs_open_w__win32(pVFS, pFilePath, openMode, pFile);
+#else
+ return ma_default_vfs_open_w__stdio(pVFS, pFilePath, openMode, pFile);
+#endif
+}
- MA_ZERO_OBJECT(&config);
- config.format = format;
- config.channels = channels;
- config.sampleRate = sampleRate;
- config.cutoffFrequency = cutoffFrequency;
- config.order = ma_min(order, MA_MAX_FILTER_ORDER);
+static ma_result ma_default_vfs_close(ma_vfs* pVFS, ma_vfs_file file)
+{
+ if (file == NULL) {
+ return MA_INVALID_ARGS;
+ }
- return config;
+#if defined(MA_WIN32) && defined(MA_WIN32_DESKTOP) && !defined(MA_NO_WIN32_FILEIO)
+ return ma_default_vfs_close__win32(pVFS, file);
+#else
+ return ma_default_vfs_close__stdio(pVFS, file);
+#endif
}
-static ma_result ma_lpf_reinit__internal(const ma_lpf_config* pConfig, ma_lpf* pLPF, ma_bool32 isNew)
+static ma_result ma_default_vfs_read(ma_vfs* pVFS, ma_vfs_file file, void* pDst, size_t sizeInBytes, size_t* pBytesRead)
{
- ma_result result;
- ma_uint32 lpf1Count;
- ma_uint32 lpf2Count;
- ma_uint32 ilpf1;
- ma_uint32 ilpf2;
-
- if (pLPF == NULL || pConfig == NULL) {
- return MA_INVALID_ARGS;
+ if (pBytesRead != NULL) {
+ *pBytesRead = 0;
}
- /* Only supporting f32 and s16. */
- if (pConfig->format != ma_format_f32 && pConfig->format != ma_format_s16) {
+ if (file == NULL || pDst == NULL) {
return MA_INVALID_ARGS;
}
- /* The format cannot be changed after initialization. */
- if (pLPF->format != ma_format_unknown && pLPF->format != pConfig->format) {
- return MA_INVALID_OPERATION;
- }
+#if defined(MA_WIN32) && defined(MA_WIN32_DESKTOP) && !defined(MA_NO_WIN32_FILEIO)
+ return ma_default_vfs_read__win32(pVFS, file, pDst, sizeInBytes, pBytesRead);
+#else
+ return ma_default_vfs_read__stdio(pVFS, file, pDst, sizeInBytes, pBytesRead);
+#endif
+}
- /* The channel count cannot be changed after initialization. */
- if (pLPF->channels != 0 && pLPF->channels != pConfig->channels) {
- return MA_INVALID_OPERATION;
+static ma_result ma_default_vfs_write(ma_vfs* pVFS, ma_vfs_file file, const void* pSrc, size_t sizeInBytes, size_t* pBytesWritten)
+{
+ if (pBytesWritten != NULL) {
+ *pBytesWritten = 0;
}
- if (pConfig->order > MA_MAX_FILTER_ORDER) {
+ if (file == NULL || pSrc == NULL) {
return MA_INVALID_ARGS;
}
- lpf1Count = pConfig->order % 2;
- lpf2Count = pConfig->order / 2;
-
- MA_ASSERT(lpf1Count <= ma_countof(pLPF->lpf1));
- MA_ASSERT(lpf2Count <= ma_countof(pLPF->lpf2));
+#if defined(MA_WIN32) && defined(MA_WIN32_DESKTOP) && !defined(MA_NO_WIN32_FILEIO)
+ return ma_default_vfs_write__win32(pVFS, file, pSrc, sizeInBytes, pBytesWritten);
+#else
+ return ma_default_vfs_write__stdio(pVFS, file, pSrc, sizeInBytes, pBytesWritten);
+#endif
+}
- /* The filter order can't change between reinits. */
- if (!isNew) {
- if (pLPF->lpf1Count != lpf1Count || pLPF->lpf2Count != lpf2Count) {
- return MA_INVALID_OPERATION;
- }
+static ma_result ma_default_vfs_seek(ma_vfs* pVFS, ma_vfs_file file, ma_int64 offset, ma_seek_origin origin)
+{
+ if (file == NULL) {
+ return MA_INVALID_ARGS;
}
- for (ilpf1 = 0; ilpf1 < lpf1Count; ilpf1 += 1) {
- ma_lpf1_config lpf1Config = ma_lpf1_config_init(pConfig->format, pConfig->channels, pConfig->sampleRate, pConfig->cutoffFrequency);
-
- if (isNew) {
- result = ma_lpf1_init(&lpf1Config, &pLPF->lpf1[ilpf1]);
- } else {
- result = ma_lpf1_reinit(&lpf1Config, &pLPF->lpf1[ilpf1]);
- }
+#if defined(MA_WIN32) && defined(MA_WIN32_DESKTOP) && !defined(MA_NO_WIN32_FILEIO)
+ return ma_default_vfs_seek__win32(pVFS, file, offset, origin);
+#else
+ return ma_default_vfs_seek__stdio(pVFS, file, offset, origin);
+#endif
+}
- if (result != MA_SUCCESS) {
- return result;
- }
+static ma_result ma_default_vfs_tell(ma_vfs* pVFS, ma_vfs_file file, ma_int64* pCursor)
+{
+ if (pCursor == NULL) {
+ return MA_INVALID_ARGS;
}
- for (ilpf2 = 0; ilpf2 < lpf2Count; ilpf2 += 1) {
- ma_lpf2_config lpf2Config;
- double q;
- double a;
-
- /* Tempting to use 0.707107, but won't result in a Butterworth filter if the order is > 2. */
- if (lpf1Count == 1) {
- a = (1 + ilpf2*1) * (MA_PI_D/(pConfig->order*1)); /* Odd order. */
- } else {
- a = (1 + ilpf2*2) * (MA_PI_D/(pConfig->order*2)); /* Even order. */
- }
- q = 1 / (2*ma_cosd(a));
-
- lpf2Config = ma_lpf2_config_init(pConfig->format, pConfig->channels, pConfig->sampleRate, pConfig->cutoffFrequency, q);
-
- if (isNew) {
- result = ma_lpf2_init(&lpf2Config, &pLPF->lpf2[ilpf2]);
- } else {
- result = ma_lpf2_reinit(&lpf2Config, &pLPF->lpf2[ilpf2]);
- }
+ *pCursor = 0;
- if (result != MA_SUCCESS) {
- return result;
- }
+ if (file == NULL) {
+ return MA_INVALID_ARGS;
}
- pLPF->lpf1Count = lpf1Count;
- pLPF->lpf2Count = lpf2Count;
- pLPF->format = pConfig->format;
- pLPF->channels = pConfig->channels;
- pLPF->sampleRate = pConfig->sampleRate;
-
- return MA_SUCCESS;
+#if defined(MA_WIN32) && defined(MA_WIN32_DESKTOP) && !defined(MA_NO_WIN32_FILEIO)
+ return ma_default_vfs_tell__win32(pVFS, file, pCursor);
+#else
+ return ma_default_vfs_tell__stdio(pVFS, file, pCursor);
+#endif
}
-MA_API ma_result ma_lpf_init(const ma_lpf_config* pConfig, ma_lpf* pLPF)
+static ma_result ma_default_vfs_info(ma_vfs* pVFS, ma_vfs_file file, ma_file_info* pInfo)
{
- if (pLPF == NULL) {
+ if (pInfo == NULL) {
return MA_INVALID_ARGS;
}
- MA_ZERO_OBJECT(pLPF);
+ MA_ZERO_OBJECT(pInfo);
- if (pConfig == NULL) {
+ if (file == NULL) {
return MA_INVALID_ARGS;
}
- return ma_lpf_reinit__internal(pConfig, pLPF, /*isNew*/MA_TRUE);
+#if defined(MA_WIN32) && defined(MA_WIN32_DESKTOP) && !defined(MA_NO_WIN32_FILEIO)
+ return ma_default_vfs_info__win32(pVFS, file, pInfo);
+#else
+ return ma_default_vfs_info__stdio(pVFS, file, pInfo);
+#endif
}
-MA_API ma_result ma_lpf_reinit(const ma_lpf_config* pConfig, ma_lpf* pLPF)
-{
- return ma_lpf_reinit__internal(pConfig, pLPF, /*isNew*/MA_FALSE);
-}
-static MA_INLINE void ma_lpf_process_pcm_frame_f32(ma_lpf* pLPF, float* pY, const void* pX)
+MA_API ma_result ma_default_vfs_init(ma_default_vfs* pVFS, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_uint32 ilpf1;
- ma_uint32 ilpf2;
+ if (pVFS == NULL) {
+ return MA_INVALID_ARGS;
+ }
- MA_ASSERT(pLPF->format == ma_format_f32);
+ pVFS->cb.onOpen = ma_default_vfs_open;
+ pVFS->cb.onOpenW = ma_default_vfs_open_w;
+ pVFS->cb.onClose = ma_default_vfs_close;
+ pVFS->cb.onRead = ma_default_vfs_read;
+ pVFS->cb.onWrite = ma_default_vfs_write;
+ pVFS->cb.onSeek = ma_default_vfs_seek;
+ pVFS->cb.onTell = ma_default_vfs_tell;
+ pVFS->cb.onInfo = ma_default_vfs_info;
+ ma_allocation_callbacks_init_copy(&pVFS->allocationCallbacks, pAllocationCallbacks);
- MA_COPY_MEMORY(pY, pX, ma_get_bytes_per_frame(pLPF->format, pLPF->channels));
+ return MA_SUCCESS;
+}
- for (ilpf1 = 0; ilpf1 < pLPF->lpf1Count; ilpf1 += 1) {
- ma_lpf1_process_pcm_frame_f32(&pLPF->lpf1[ilpf1], pY, pY);
- }
- for (ilpf2 = 0; ilpf2 < pLPF->lpf2Count; ilpf2 += 1) {
- ma_lpf2_process_pcm_frame_f32(&pLPF->lpf2[ilpf2], pY, pY);
+MA_API ma_result ma_vfs_or_default_open(ma_vfs* pVFS, const char* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
+{
+ if (pVFS != NULL) {
+ return ma_vfs_open(pVFS, pFilePath, openMode, pFile);
+ } else {
+ return ma_default_vfs_open(pVFS, pFilePath, openMode, pFile);
}
}
-static MA_INLINE void ma_lpf_process_pcm_frame_s16(ma_lpf* pLPF, ma_int16* pY, const ma_int16* pX)
+MA_API ma_result ma_vfs_or_default_open_w(ma_vfs* pVFS, const wchar_t* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
{
- ma_uint32 ilpf1;
- ma_uint32 ilpf2;
-
- MA_ASSERT(pLPF->format == ma_format_s16);
-
- MA_COPY_MEMORY(pY, pX, ma_get_bytes_per_frame(pLPF->format, pLPF->channels));
-
- for (ilpf1 = 0; ilpf1 < pLPF->lpf1Count; ilpf1 += 1) {
- ma_lpf1_process_pcm_frame_s16(&pLPF->lpf1[ilpf1], pY, pY);
+ if (pVFS != NULL) {
+ return ma_vfs_open_w(pVFS, pFilePath, openMode, pFile);
+ } else {
+ return ma_default_vfs_open_w(pVFS, pFilePath, openMode, pFile);
}
+}
- for (ilpf2 = 0; ilpf2 < pLPF->lpf2Count; ilpf2 += 1) {
- ma_lpf2_process_pcm_frame_s16(&pLPF->lpf2[ilpf2], pY, pY);
+MA_API ma_result ma_vfs_or_default_close(ma_vfs* pVFS, ma_vfs_file file)
+{
+ if (pVFS != NULL) {
+ return ma_vfs_close(pVFS, file);
+ } else {
+ return ma_default_vfs_close(pVFS, file);
}
}
-MA_API ma_result ma_lpf_process_pcm_frames(ma_lpf* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+MA_API ma_result ma_vfs_or_default_read(ma_vfs* pVFS, ma_vfs_file file, void* pDst, size_t sizeInBytes, size_t* pBytesRead)
{
- ma_result result;
- ma_uint32 ilpf1;
- ma_uint32 ilpf2;
-
- if (pLPF == NULL) {
- return MA_INVALID_ARGS;
+ if (pVFS != NULL) {
+ return ma_vfs_read(pVFS, file, pDst, sizeInBytes, pBytesRead);
+ } else {
+ return ma_default_vfs_read(pVFS, file, pDst, sizeInBytes, pBytesRead);
}
+}
- /* Faster path for in-place. */
- if (pFramesOut == pFramesIn) {
- for (ilpf1 = 0; ilpf1 < pLPF->lpf1Count; ilpf1 += 1) {
- result = ma_lpf1_process_pcm_frames(&pLPF->lpf1[ilpf1], pFramesOut, pFramesOut, frameCount);
- if (result != MA_SUCCESS) {
- return result;
- }
- }
-
- for (ilpf2 = 0; ilpf2 < pLPF->lpf2Count; ilpf2 += 1) {
- result = ma_lpf2_process_pcm_frames(&pLPF->lpf2[ilpf2], pFramesOut, pFramesOut, frameCount);
- if (result != MA_SUCCESS) {
- return result;
- }
- }
+MA_API ma_result ma_vfs_or_default_write(ma_vfs* pVFS, ma_vfs_file file, const void* pSrc, size_t sizeInBytes, size_t* pBytesWritten)
+{
+ if (pVFS != NULL) {
+ return ma_vfs_write(pVFS, file, pSrc, sizeInBytes, pBytesWritten);
+ } else {
+ return ma_default_vfs_write(pVFS, file, pSrc, sizeInBytes, pBytesWritten);
}
+}
- /* Slightly slower path for copying. */
- if (pFramesOut != pFramesIn) {
- ma_uint32 iFrame;
-
- /* */ if (pLPF->format == ma_format_f32) {
- /* */ float* pFramesOutF32 = ( float*)pFramesOut;
- const float* pFramesInF32 = (const float*)pFramesIn;
-
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- ma_lpf_process_pcm_frame_f32(pLPF, pFramesOutF32, pFramesInF32);
- pFramesOutF32 += pLPF->channels;
- pFramesInF32 += pLPF->channels;
- }
- } else if (pLPF->format == ma_format_s16) {
- /* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut;
- const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn;
-
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- ma_lpf_process_pcm_frame_s16(pLPF, pFramesOutS16, pFramesInS16);
- pFramesOutS16 += pLPF->channels;
- pFramesInS16 += pLPF->channels;
- }
- } else {
- MA_ASSERT(MA_FALSE);
- return MA_INVALID_OPERATION; /* Should never hit this. */
- }
+MA_API ma_result ma_vfs_or_default_seek(ma_vfs* pVFS, ma_vfs_file file, ma_int64 offset, ma_seek_origin origin)
+{
+ if (pVFS != NULL) {
+ return ma_vfs_seek(pVFS, file, offset, origin);
+ } else {
+ return ma_default_vfs_seek(pVFS, file, offset, origin);
}
-
- return MA_SUCCESS;
}
-MA_API ma_uint32 ma_lpf_get_latency(const ma_lpf* pLPF)
+MA_API ma_result ma_vfs_or_default_tell(ma_vfs* pVFS, ma_vfs_file file, ma_int64* pCursor)
{
- if (pLPF == NULL) {
- return 0;
+ if (pVFS != NULL) {
+ return ma_vfs_tell(pVFS, file, pCursor);
+ } else {
+ return ma_default_vfs_tell(pVFS, file, pCursor);
}
+}
- return pLPF->lpf2Count*2 + pLPF->lpf1Count;
+MA_API ma_result ma_vfs_or_default_info(ma_vfs* pVFS, ma_vfs_file file, ma_file_info* pInfo)
+{
+ if (pVFS != NULL) {
+ return ma_vfs_info(pVFS, file, pInfo);
+ } else {
+ return ma_default_vfs_info(pVFS, file, pInfo);
+ }
}
+
/**************************************************************************************************************************************************************
-High-Pass Filtering
+Decoding and Encoding Headers. These are auto-generated from a tool.
**************************************************************************************************************************************************************/
-MA_API ma_hpf1_config ma_hpf1_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency)
+#if !defined(MA_NO_WAV) && (!defined(MA_NO_DECODING) || !defined(MA_NO_ENCODING))
+/* dr_wav_h begin */
+#ifndef dr_wav_h
+#define dr_wav_h
+#ifdef __cplusplus
+extern "C" {
+#endif
+#define DRWAV_STRINGIFY(x) #x
+#define DRWAV_XSTRINGIFY(x) DRWAV_STRINGIFY(x)
+#define DRWAV_VERSION_MAJOR 0
+#define DRWAV_VERSION_MINOR 13
+#define DRWAV_VERSION_REVISION 5
+#define DRWAV_VERSION_STRING DRWAV_XSTRINGIFY(DRWAV_VERSION_MAJOR) "." DRWAV_XSTRINGIFY(DRWAV_VERSION_MINOR) "." DRWAV_XSTRINGIFY(DRWAV_VERSION_REVISION)
+#include
+typedef signed char drwav_int8;
+typedef unsigned char drwav_uint8;
+typedef signed short drwav_int16;
+typedef unsigned short drwav_uint16;
+typedef signed int drwav_int32;
+typedef unsigned int drwav_uint32;
+#if defined(_MSC_VER) && !defined(__clang__)
+ typedef signed __int64 drwav_int64;
+ typedef unsigned __int64 drwav_uint64;
+#else
+ #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
+ #pragma GCC diagnostic push
+ #pragma GCC diagnostic ignored "-Wlong-long"
+ #if defined(__clang__)
+ #pragma GCC diagnostic ignored "-Wc++11-long-long"
+ #endif
+ #endif
+ typedef signed long long drwav_int64;
+ typedef unsigned long long drwav_uint64;
+ #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
+ #pragma GCC diagnostic pop
+ #endif
+#endif
+#if defined(__LP64__) || defined(_WIN64) || (defined(__x86_64__) && !defined(__ILP32__)) || defined(_M_X64) || defined(__ia64) || defined (_M_IA64) || defined(__aarch64__) || defined(_M_ARM64) || defined(__powerpc64__)
+ typedef drwav_uint64 drwav_uintptr;
+#else
+ typedef drwav_uint32 drwav_uintptr;
+#endif
+typedef drwav_uint8 drwav_bool8;
+typedef drwav_uint32 drwav_bool32;
+#define DRWAV_TRUE 1
+#define DRWAV_FALSE 0
+#if !defined(DRWAV_API)
+ #if defined(DRWAV_DLL)
+ #if defined(_WIN32)
+ #define DRWAV_DLL_IMPORT __declspec(dllimport)
+ #define DRWAV_DLL_EXPORT __declspec(dllexport)
+ #define DRWAV_DLL_PRIVATE static
+ #else
+ #if defined(__GNUC__) && __GNUC__ >= 4
+ #define DRWAV_DLL_IMPORT __attribute__((visibility("default")))
+ #define DRWAV_DLL_EXPORT __attribute__((visibility("default")))
+ #define DRWAV_DLL_PRIVATE __attribute__((visibility("hidden")))
+ #else
+ #define DRWAV_DLL_IMPORT
+ #define DRWAV_DLL_EXPORT
+ #define DRWAV_DLL_PRIVATE static
+ #endif
+ #endif
+ #if defined(DR_WAV_IMPLEMENTATION) || defined(DRWAV_IMPLEMENTATION)
+ #define DRWAV_API DRWAV_DLL_EXPORT
+ #else
+ #define DRWAV_API DRWAV_DLL_IMPORT
+ #endif
+ #define DRWAV_PRIVATE DRWAV_DLL_PRIVATE
+ #else
+ #define DRWAV_API extern
+ #define DRWAV_PRIVATE static
+ #endif
+#endif
+typedef drwav_int32 drwav_result;
+#define DRWAV_SUCCESS 0
+#define DRWAV_ERROR -1
+#define DRWAV_INVALID_ARGS -2
+#define DRWAV_INVALID_OPERATION -3
+#define DRWAV_OUT_OF_MEMORY -4
+#define DRWAV_OUT_OF_RANGE -5
+#define DRWAV_ACCESS_DENIED -6
+#define DRWAV_DOES_NOT_EXIST -7
+#define DRWAV_ALREADY_EXISTS -8
+#define DRWAV_TOO_MANY_OPEN_FILES -9
+#define DRWAV_INVALID_FILE -10
+#define DRWAV_TOO_BIG -11
+#define DRWAV_PATH_TOO_LONG -12
+#define DRWAV_NAME_TOO_LONG -13
+#define DRWAV_NOT_DIRECTORY -14
+#define DRWAV_IS_DIRECTORY -15
+#define DRWAV_DIRECTORY_NOT_EMPTY -16
+#define DRWAV_END_OF_FILE -17
+#define DRWAV_NO_SPACE -18
+#define DRWAV_BUSY -19
+#define DRWAV_IO_ERROR -20
+#define DRWAV_INTERRUPT -21
+#define DRWAV_UNAVAILABLE -22
+#define DRWAV_ALREADY_IN_USE -23
+#define DRWAV_BAD_ADDRESS -24
+#define DRWAV_BAD_SEEK -25
+#define DRWAV_BAD_PIPE -26
+#define DRWAV_DEADLOCK -27
+#define DRWAV_TOO_MANY_LINKS -28
+#define DRWAV_NOT_IMPLEMENTED -29
+#define DRWAV_NO_MESSAGE -30
+#define DRWAV_BAD_MESSAGE -31
+#define DRWAV_NO_DATA_AVAILABLE -32
+#define DRWAV_INVALID_DATA -33
+#define DRWAV_TIMEOUT -34
+#define DRWAV_NO_NETWORK -35
+#define DRWAV_NOT_UNIQUE -36
+#define DRWAV_NOT_SOCKET -37
+#define DRWAV_NO_ADDRESS -38
+#define DRWAV_BAD_PROTOCOL -39
+#define DRWAV_PROTOCOL_UNAVAILABLE -40
+#define DRWAV_PROTOCOL_NOT_SUPPORTED -41
+#define DRWAV_PROTOCOL_FAMILY_NOT_SUPPORTED -42
+#define DRWAV_ADDRESS_FAMILY_NOT_SUPPORTED -43
+#define DRWAV_SOCKET_NOT_SUPPORTED -44
+#define DRWAV_CONNECTION_RESET -45
+#define DRWAV_ALREADY_CONNECTED -46
+#define DRWAV_NOT_CONNECTED -47
+#define DRWAV_CONNECTION_REFUSED -48
+#define DRWAV_NO_HOST -49
+#define DRWAV_IN_PROGRESS -50
+#define DRWAV_CANCELLED -51
+#define DRWAV_MEMORY_ALREADY_MAPPED -52
+#define DRWAV_AT_END -53
+#define DR_WAVE_FORMAT_PCM 0x1
+#define DR_WAVE_FORMAT_ADPCM 0x2
+#define DR_WAVE_FORMAT_IEEE_FLOAT 0x3
+#define DR_WAVE_FORMAT_ALAW 0x6
+#define DR_WAVE_FORMAT_MULAW 0x7
+#define DR_WAVE_FORMAT_DVI_ADPCM 0x11
+#define DR_WAVE_FORMAT_EXTENSIBLE 0xFFFE
+#define DRWAV_SEQUENTIAL 0x00000001
+DRWAV_API void drwav_version(drwav_uint32* pMajor, drwav_uint32* pMinor, drwav_uint32* pRevision);
+DRWAV_API const char* drwav_version_string(void);
+typedef enum
{
- ma_hpf1_config config;
-
- MA_ZERO_OBJECT(&config);
- config.format = format;
- config.channels = channels;
- config.sampleRate = sampleRate;
- config.cutoffFrequency = cutoffFrequency;
-
- return config;
-}
-
-MA_API ma_hpf2_config ma_hpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q)
+ drwav_seek_origin_start,
+ drwav_seek_origin_current
+} drwav_seek_origin;
+typedef enum
+{
+ drwav_container_riff,
+ drwav_container_w64,
+ drwav_container_rf64
+} drwav_container;
+typedef struct
+{
+ union
+ {
+ drwav_uint8 fourcc[4];
+ drwav_uint8 guid[16];
+ } id;
+ drwav_uint64 sizeInBytes;
+ unsigned int paddingSize;
+} drwav_chunk_header;
+typedef struct
+{
+ drwav_uint16 formatTag;
+ drwav_uint16 channels;
+ drwav_uint32 sampleRate;
+ drwav_uint32 avgBytesPerSec;
+ drwav_uint16 blockAlign;
+ drwav_uint16 bitsPerSample;
+ drwav_uint16 extendedSize;
+ drwav_uint16 validBitsPerSample;
+ drwav_uint32 channelMask;
+ drwav_uint8 subFormat[16];
+} drwav_fmt;
+DRWAV_API drwav_uint16 drwav_fmt_get_format(const drwav_fmt* pFMT);
+typedef size_t (* drwav_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead);
+typedef size_t (* drwav_write_proc)(void* pUserData, const void* pData, size_t bytesToWrite);
+typedef drwav_bool32 (* drwav_seek_proc)(void* pUserData, int offset, drwav_seek_origin origin);
+typedef drwav_uint64 (* drwav_chunk_proc)(void* pChunkUserData, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pReadSeekUserData, const drwav_chunk_header* pChunkHeader, drwav_container container, const drwav_fmt* pFMT);
+typedef struct
+{
+ void* pUserData;
+ void* (* onMalloc)(size_t sz, void* pUserData);
+ void* (* onRealloc)(void* p, size_t sz, void* pUserData);
+ void (* onFree)(void* p, void* pUserData);
+} drwav_allocation_callbacks;
+typedef struct
+{
+ const drwav_uint8* data;
+ size_t dataSize;
+ size_t currentReadPos;
+} drwav__memory_stream;
+typedef struct
+{
+ void** ppData;
+ size_t* pDataSize;
+ size_t dataSize;
+ size_t dataCapacity;
+ size_t currentWritePos;
+} drwav__memory_stream_write;
+typedef struct
+{
+ drwav_container container;
+ drwav_uint32 format;
+ drwav_uint32 channels;
+ drwav_uint32 sampleRate;
+ drwav_uint32 bitsPerSample;
+} drwav_data_format;
+typedef enum
+{
+ drwav_metadata_type_none = 0,
+ drwav_metadata_type_unknown = 1 << 0,
+ drwav_metadata_type_smpl = 1 << 1,
+ drwav_metadata_type_inst = 1 << 2,
+ drwav_metadata_type_cue = 1 << 3,
+ drwav_metadata_type_acid = 1 << 4,
+ drwav_metadata_type_bext = 1 << 5,
+ drwav_metadata_type_list_label = 1 << 6,
+ drwav_metadata_type_list_note = 1 << 7,
+ drwav_metadata_type_list_labelled_cue_region = 1 << 8,
+ drwav_metadata_type_list_info_software = 1 << 9,
+ drwav_metadata_type_list_info_copyright = 1 << 10,
+ drwav_metadata_type_list_info_title = 1 << 11,
+ drwav_metadata_type_list_info_artist = 1 << 12,
+ drwav_metadata_type_list_info_comment = 1 << 13,
+ drwav_metadata_type_list_info_date = 1 << 14,
+ drwav_metadata_type_list_info_genre = 1 << 15,
+ drwav_metadata_type_list_info_album = 1 << 16,
+ drwav_metadata_type_list_info_tracknumber = 1 << 17,
+ drwav_metadata_type_list_all_info_strings = drwav_metadata_type_list_info_software
+ | drwav_metadata_type_list_info_copyright
+ | drwav_metadata_type_list_info_title
+ | drwav_metadata_type_list_info_artist
+ | drwav_metadata_type_list_info_comment
+ | drwav_metadata_type_list_info_date
+ | drwav_metadata_type_list_info_genre
+ | drwav_metadata_type_list_info_album
+ | drwav_metadata_type_list_info_tracknumber,
+ drwav_metadata_type_list_all_adtl = drwav_metadata_type_list_label
+ | drwav_metadata_type_list_note
+ | drwav_metadata_type_list_labelled_cue_region,
+ drwav_metadata_type_all = -2,
+ drwav_metadata_type_all_including_unknown = -1
+} drwav_metadata_type;
+typedef enum
+{
+ drwav_smpl_loop_type_forward = 0,
+ drwav_smpl_loop_type_pingpong = 1,
+ drwav_smpl_loop_type_backward = 2
+} drwav_smpl_loop_type;
+typedef struct
+{
+ drwav_uint32 cuePointId;
+ drwav_uint32 type;
+ drwav_uint32 firstSampleByteOffset;
+ drwav_uint32 lastSampleByteOffset;
+ drwav_uint32 sampleFraction;
+ drwav_uint32 playCount;
+} drwav_smpl_loop;
+typedef struct
+{
+ drwav_uint32 manufacturerId;
+ drwav_uint32 productId;
+ drwav_uint32 samplePeriodNanoseconds;
+ drwav_uint32 midiUnityNote;
+ drwav_uint32 midiPitchFraction;
+ drwav_uint32 smpteFormat;
+ drwav_uint32 smpteOffset;
+ drwav_uint32 sampleLoopCount;
+ drwav_uint32 samplerSpecificDataSizeInBytes;
+ drwav_smpl_loop* pLoops;
+ drwav_uint8* pSamplerSpecificData;
+} drwav_smpl;
+typedef struct
{
- ma_hpf2_config config;
-
- MA_ZERO_OBJECT(&config);
- config.format = format;
- config.channels = channels;
- config.sampleRate = sampleRate;
- config.cutoffFrequency = cutoffFrequency;
- config.q = q;
-
- /* Q cannot be 0 or else it'll result in a division by 0. In this case just default to 0.707107. */
- if (config.q == 0) {
- config.q = 0.707107;
- }
-
- return config;
-}
-
-
-MA_API ma_result ma_hpf1_init(const ma_hpf1_config* pConfig, ma_hpf1* pHPF)
+ drwav_int8 midiUnityNote;
+ drwav_int8 fineTuneCents;
+ drwav_int8 gainDecibels;
+ drwav_int8 lowNote;
+ drwav_int8 highNote;
+ drwav_int8 lowVelocity;
+ drwav_int8 highVelocity;
+} drwav_inst;
+typedef struct
{
- if (pHPF == NULL) {
- return MA_INVALID_ARGS;
- }
-
- MA_ZERO_OBJECT(pHPF);
-
- if (pConfig == NULL) {
- return MA_INVALID_ARGS;
- }
-
- if (pConfig->channels < MA_MIN_CHANNELS || pConfig->channels > MA_MAX_CHANNELS) {
- return MA_INVALID_ARGS;
- }
-
- return ma_hpf1_reinit(pConfig, pHPF);
-}
-
-MA_API ma_result ma_hpf1_reinit(const ma_hpf1_config* pConfig, ma_hpf1* pHPF)
+ drwav_uint32 id;
+ drwav_uint32 playOrderPosition;
+ drwav_uint8 dataChunkId[4];
+ drwav_uint32 chunkStart;
+ drwav_uint32 blockStart;
+ drwav_uint32 sampleByteOffset;
+} drwav_cue_point;
+typedef struct
{
- double a;
-
- if (pHPF == NULL || pConfig == NULL) {
- return MA_INVALID_ARGS;
- }
-
- /* Only supporting f32 and s16. */
- if (pConfig->format != ma_format_f32 && pConfig->format != ma_format_s16) {
- return MA_INVALID_ARGS;
- }
-
- /* The format cannot be changed after initialization. */
- if (pHPF->format != ma_format_unknown && pHPF->format != pConfig->format) {
- return MA_INVALID_OPERATION;
- }
-
- /* The channel count cannot be changed after initialization. */
- if (pHPF->channels != 0 && pHPF->channels != pConfig->channels) {
- return MA_INVALID_OPERATION;
- }
-
- pHPF->format = pConfig->format;
- pHPF->channels = pConfig->channels;
-
- a = ma_expd(-2 * MA_PI_D * pConfig->cutoffFrequency / pConfig->sampleRate);
- if (pConfig->format == ma_format_f32) {
- pHPF->a.f32 = (float)a;
- } else {
- pHPF->a.s32 = ma_biquad_float_to_fp(a);
- }
-
- return MA_SUCCESS;
-}
-
-static MA_INLINE void ma_hpf1_process_pcm_frame_f32(ma_hpf1* pHPF, float* pY, const float* pX)
+ drwav_uint32 cuePointCount;
+ drwav_cue_point *pCuePoints;
+} drwav_cue;
+typedef enum
{
- ma_uint32 c;
- const ma_uint32 channels = pHPF->channels;
- const float a = 1 - pHPF->a.f32;
- const float b = 1 - a;
-
- MA_ASSUME(channels >= MA_MIN_CHANNELS && channels <= MA_MAX_CHANNELS);
- for (c = 0; c < channels; c += 1) {
- float r1 = pHPF->r1[c].f32;
- float x = pX[c];
- float y;
-
- y = b*x - a*r1;
-
- pY[c] = y;
- pHPF->r1[c].f32 = y;
- }
-}
-
-static MA_INLINE void ma_hpf1_process_pcm_frame_s16(ma_hpf1* pHPF, ma_int16* pY, const ma_int16* pX)
+ drwav_acid_flag_one_shot = 1,
+ drwav_acid_flag_root_note_set = 2,
+ drwav_acid_flag_stretch = 4,
+ drwav_acid_flag_disk_based = 8,
+ drwav_acid_flag_acidizer = 16
+} drwav_acid_flag;
+typedef struct
{
- ma_uint32 c;
- const ma_uint32 channels = pHPF->channels;
- const ma_int32 a = ((1 << MA_BIQUAD_FIXED_POINT_SHIFT) - pHPF->a.s32);
- const ma_int32 b = ((1 << MA_BIQUAD_FIXED_POINT_SHIFT) - a);
-
- MA_ASSUME(channels >= MA_MIN_CHANNELS && channels <= MA_MAX_CHANNELS);
- for (c = 0; c < channels; c += 1) {
- ma_int32 r1 = pHPF->r1[c].s32;
- ma_int32 x = pX[c];
- ma_int32 y;
-
- y = (b*x - a*r1) >> MA_BIQUAD_FIXED_POINT_SHIFT;
-
- pY[c] = (ma_int16)y;
- pHPF->r1[c].s32 = (ma_int32)y;
- }
-}
-
-MA_API ma_result ma_hpf1_process_pcm_frames(ma_hpf1* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+ drwav_uint32 flags;
+ drwav_uint16 midiUnityNote;
+ drwav_uint16 reserved1;
+ float reserved2;
+ drwav_uint32 numBeats;
+ drwav_uint16 meterDenominator;
+ drwav_uint16 meterNumerator;
+ float tempo;
+} drwav_acid;
+typedef struct
{
- ma_uint32 n;
-
- if (pHPF == NULL || pFramesOut == NULL || pFramesIn == NULL) {
- return MA_INVALID_ARGS;
- }
-
- /* Note that the logic below needs to support in-place filtering. That is, it must support the case where pFramesOut and pFramesIn are the same. */
-
- if (pHPF->format == ma_format_f32) {
- /* */ float* pY = ( float*)pFramesOut;
- const float* pX = (const float*)pFramesIn;
-
- for (n = 0; n < frameCount; n += 1) {
- ma_hpf1_process_pcm_frame_f32(pHPF, pY, pX);
- pY += pHPF->channels;
- pX += pHPF->channels;
- }
- } else if (pHPF->format == ma_format_s16) {
- /* */ ma_int16* pY = ( ma_int16*)pFramesOut;
- const ma_int16* pX = (const ma_int16*)pFramesIn;
-
- for (n = 0; n < frameCount; n += 1) {
- ma_hpf1_process_pcm_frame_s16(pHPF, pY, pX);
- pY += pHPF->channels;
- pX += pHPF->channels;
- }
- } else {
- MA_ASSERT(MA_FALSE);
- return MA_INVALID_ARGS; /* Format not supported. Should never hit this because it's checked in ma_biquad_init() and ma_biquad_reinit(). */
- }
-
- return MA_SUCCESS;
-}
-
-MA_API ma_uint32 ma_hpf1_get_latency(const ma_hpf1* pHPF)
+ drwav_uint32 cuePointId;
+ drwav_uint32 stringLength;
+ char* pString;
+} drwav_list_label_or_note;
+typedef struct
{
- if (pHPF == NULL) {
- return 0;
- }
-
- return 1;
-}
-
-
-static MA_INLINE ma_biquad_config ma_hpf2__get_biquad_config(const ma_hpf2_config* pConfig)
+ char* pDescription;
+ char* pOriginatorName;
+ char* pOriginatorReference;
+ char pOriginationDate[10];
+ char pOriginationTime[8];
+ drwav_uint64 timeReference;
+ drwav_uint16 version;
+ char* pCodingHistory;
+ drwav_uint32 codingHistorySize;
+ drwav_uint8* pUMID;
+ drwav_uint16 loudnessValue;
+ drwav_uint16 loudnessRange;
+ drwav_uint16 maxTruePeakLevel;
+ drwav_uint16 maxMomentaryLoudness;
+ drwav_uint16 maxShortTermLoudness;
+} drwav_bext;
+typedef struct
{
- ma_biquad_config bqConfig;
- double q;
- double w;
- double s;
- double c;
- double a;
-
- MA_ASSERT(pConfig != NULL);
-
- q = pConfig->q;
- w = 2 * MA_PI_D * pConfig->cutoffFrequency / pConfig->sampleRate;
- s = ma_sind(w);
- c = ma_cosd(w);
- a = s / (2*q);
-
- bqConfig.b0 = (1 + c) / 2;
- bqConfig.b1 = -(1 + c);
- bqConfig.b2 = (1 + c) / 2;
- bqConfig.a0 = 1 + a;
- bqConfig.a1 = -2 * c;
- bqConfig.a2 = 1 - a;
-
- bqConfig.format = pConfig->format;
- bqConfig.channels = pConfig->channels;
-
- return bqConfig;
-}
-
-MA_API ma_result ma_hpf2_init(const ma_hpf2_config* pConfig, ma_hpf2* pHPF)
+ drwav_uint32 stringLength;
+ char* pString;
+} drwav_list_info_text;
+typedef struct
{
- ma_result result;
- ma_biquad_config bqConfig;
-
- if (pHPF == NULL) {
- return MA_INVALID_ARGS;
- }
-
- MA_ZERO_OBJECT(pHPF);
-
- if (pConfig == NULL) {
- return MA_INVALID_ARGS;
- }
-
- bqConfig = ma_hpf2__get_biquad_config(pConfig);
- result = ma_biquad_init(&bqConfig, &pHPF->bq);
- if (result != MA_SUCCESS) {
- return result;
- }
-
- return MA_SUCCESS;
-}
-
-MA_API ma_result ma_hpf2_reinit(const ma_hpf2_config* pConfig, ma_hpf2* pHPF)
+ drwav_uint32 cuePointId;
+ drwav_uint32 sampleLength;
+ drwav_uint8 purposeId[4];
+ drwav_uint16 country;
+ drwav_uint16 language;
+ drwav_uint16 dialect;
+ drwav_uint16 codePage;
+ drwav_uint32 stringLength;
+ char* pString;
+} drwav_list_labelled_cue_region;
+typedef enum
{
- ma_result result;
- ma_biquad_config bqConfig;
-
- if (pHPF == NULL || pConfig == NULL) {
- return MA_INVALID_ARGS;
- }
-
- bqConfig = ma_hpf2__get_biquad_config(pConfig);
- result = ma_biquad_reinit(&bqConfig, &pHPF->bq);
- if (result != MA_SUCCESS) {
- return result;
- }
-
- return MA_SUCCESS;
-}
-
-static MA_INLINE void ma_hpf2_process_pcm_frame_s16(ma_hpf2* pHPF, ma_int16* pFrameOut, const ma_int16* pFrameIn)
+ drwav_metadata_location_invalid,
+ drwav_metadata_location_top_level,
+ drwav_metadata_location_inside_info_list,
+ drwav_metadata_location_inside_adtl_list
+} drwav_metadata_location;
+typedef struct
{
- ma_biquad_process_pcm_frame_s16(&pHPF->bq, pFrameOut, pFrameIn);
-}
-
-static MA_INLINE void ma_hpf2_process_pcm_frame_f32(ma_hpf2* pHPF, float* pFrameOut, const float* pFrameIn)
+ drwav_uint8 id[4];
+ drwav_metadata_location chunkLocation;
+ drwav_uint32 dataSizeInBytes;
+ drwav_uint8* pData;
+} drwav_unknown_metadata;
+typedef struct
{
- ma_biquad_process_pcm_frame_f32(&pHPF->bq, pFrameOut, pFrameIn);
+ drwav_metadata_type type;
+ union
+ {
+ drwav_cue cue;
+ drwav_smpl smpl;
+ drwav_acid acid;
+ drwav_inst inst;
+ drwav_bext bext;
+ drwav_list_label_or_note labelOrNote;
+ drwav_list_labelled_cue_region labelledCueRegion;
+ drwav_list_info_text infoText;
+ drwav_unknown_metadata unknown;
+ } data;
+} drwav_metadata;
+typedef struct
+{
+ drwav_read_proc onRead;
+ drwav_write_proc onWrite;
+ drwav_seek_proc onSeek;
+ void* pUserData;
+ drwav_allocation_callbacks allocationCallbacks;
+ drwav_container container;
+ drwav_fmt fmt;
+ drwav_uint32 sampleRate;
+ drwav_uint16 channels;
+ drwav_uint16 bitsPerSample;
+ drwav_uint16 translatedFormatTag;
+ drwav_uint64 totalPCMFrameCount;
+ drwav_uint64 dataChunkDataSize;
+ drwav_uint64 dataChunkDataPos;
+ drwav_uint64 bytesRemaining;
+ drwav_uint64 readCursorInPCMFrames;
+ drwav_uint64 dataChunkDataSizeTargetWrite;
+ drwav_bool32 isSequentialWrite;
+ drwav_metadata_type allowedMetadataTypes;
+ drwav_metadata* pMetadata;
+ drwav_uint32 metadataCount;
+ drwav__memory_stream memoryStream;
+ drwav__memory_stream_write memoryStreamWrite;
+ struct
+ {
+ drwav_uint32 bytesRemainingInBlock;
+ drwav_uint16 predictor[2];
+ drwav_int32 delta[2];
+ drwav_int32 cachedFrames[4];
+ drwav_uint32 cachedFrameCount;
+ drwav_int32 prevFrames[2][2];
+ } msadpcm;
+ struct
+ {
+ drwav_uint32 bytesRemainingInBlock;
+ drwav_int32 predictor[2];
+ drwav_int32 stepIndex[2];
+ drwav_int32 cachedFrames[16];
+ drwav_uint32 cachedFrameCount;
+ } ima;
+} drwav;
+DRWAV_API drwav_bool32 drwav_init(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_ex(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_chunk_proc onChunk, void* pReadSeekUserData, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_with_metadata(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_write(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_write_sequential(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_write_sequential_pcm_frames(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_write_with_metadata(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks, drwav_metadata* pMetadata, drwav_uint32 metadataCount);
+DRWAV_API drwav_uint64 drwav_target_write_size_bytes(const drwav_data_format* pFormat, drwav_uint64 totalFrameCount, drwav_metadata* pMetadata, drwav_uint32 metadataCount);
+DRWAV_API drwav_metadata* drwav_take_ownership_of_metadata(drwav* pWav);
+DRWAV_API drwav_result drwav_uninit(drwav* pWav);
+DRWAV_API size_t drwav_read_raw(drwav* pWav, size_t bytesToRead, void* pBufferOut);
+DRWAV_API drwav_uint64 drwav_read_pcm_frames(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut);
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_le(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut);
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_be(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut);
+DRWAV_API drwav_bool32 drwav_seek_to_pcm_frame(drwav* pWav, drwav_uint64 targetFrameIndex);
+DRWAV_API drwav_result drwav_get_cursor_in_pcm_frames(drwav* pWav, drwav_uint64* pCursor);
+DRWAV_API drwav_result drwav_get_length_in_pcm_frames(drwav* pWav, drwav_uint64* pLength);
+DRWAV_API size_t drwav_write_raw(drwav* pWav, size_t bytesToWrite, const void* pData);
+DRWAV_API drwav_uint64 drwav_write_pcm_frames(drwav* pWav, drwav_uint64 framesToWrite, const void* pData);
+DRWAV_API drwav_uint64 drwav_write_pcm_frames_le(drwav* pWav, drwav_uint64 framesToWrite, const void* pData);
+DRWAV_API drwav_uint64 drwav_write_pcm_frames_be(drwav* pWav, drwav_uint64 framesToWrite, const void* pData);
+#ifndef DR_WAV_NO_CONVERSION_API
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut);
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16le(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut);
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16be(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut);
+DRWAV_API void drwav_u8_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount);
+DRWAV_API void drwav_s24_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount);
+DRWAV_API void drwav_s32_to_s16(drwav_int16* pOut, const drwav_int32* pIn, size_t sampleCount);
+DRWAV_API void drwav_f32_to_s16(drwav_int16* pOut, const float* pIn, size_t sampleCount);
+DRWAV_API void drwav_f64_to_s16(drwav_int16* pOut, const double* pIn, size_t sampleCount);
+DRWAV_API void drwav_alaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount);
+DRWAV_API void drwav_mulaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount);
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut);
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32le(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut);
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32be(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut);
+DRWAV_API void drwav_u8_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount);
+DRWAV_API void drwav_s16_to_f32(float* pOut, const drwav_int16* pIn, size_t sampleCount);
+DRWAV_API void drwav_s24_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount);
+DRWAV_API void drwav_s32_to_f32(float* pOut, const drwav_int32* pIn, size_t sampleCount);
+DRWAV_API void drwav_f64_to_f32(float* pOut, const double* pIn, size_t sampleCount);
+DRWAV_API void drwav_alaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount);
+DRWAV_API void drwav_mulaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount);
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut);
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32le(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut);
+DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32be(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut);
+DRWAV_API void drwav_u8_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount);
+DRWAV_API void drwav_s16_to_s32(drwav_int32* pOut, const drwav_int16* pIn, size_t sampleCount);
+DRWAV_API void drwav_s24_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount);
+DRWAV_API void drwav_f32_to_s32(drwav_int32* pOut, const float* pIn, size_t sampleCount);
+DRWAV_API void drwav_f64_to_s32(drwav_int32* pOut, const double* pIn, size_t sampleCount);
+DRWAV_API void drwav_alaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount);
+DRWAV_API void drwav_mulaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount);
+#endif
+#ifndef DR_WAV_NO_STDIO
+DRWAV_API drwav_bool32 drwav_init_file(drwav* pWav, const char* filename, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_ex(drwav* pWav, const char* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_w(drwav* pWav, const wchar_t* filename, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_ex_w(drwav* pWav, const wchar_t* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_with_metadata(drwav* pWav, const char* filename, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_with_metadata_w(drwav* pWav, const wchar_t* filename, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_write(drwav* pWav, const char* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_write_sequential(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_write_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_write_sequential_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks);
+#endif
+DRWAV_API drwav_bool32 drwav_init_memory(drwav* pWav, const void* data, size_t dataSize, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_memory_ex(drwav* pWav, const void* data, size_t dataSize, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_memory_with_metadata(drwav* pWav, const void* data, size_t dataSize, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_memory_write(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_memory_write_sequential(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_bool32 drwav_init_memory_write_sequential_pcm_frames(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks);
+#ifndef DR_WAV_NO_CONVERSION_API
+DRWAV_API drwav_int16* drwav_open_and_read_pcm_frames_s16(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API float* drwav_open_and_read_pcm_frames_f32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_int32* drwav_open_and_read_pcm_frames_s32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+#ifndef DR_WAV_NO_STDIO
+DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+#endif
+DRWAV_API drwav_int16* drwav_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API float* drwav_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_int32* drwav_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
+#endif
+DRWAV_API void drwav_free(void* p, const drwav_allocation_callbacks* pAllocationCallbacks);
+DRWAV_API drwav_uint16 drwav_bytes_to_u16(const drwav_uint8* data);
+DRWAV_API drwav_int16 drwav_bytes_to_s16(const drwav_uint8* data);
+DRWAV_API drwav_uint32 drwav_bytes_to_u32(const drwav_uint8* data);
+DRWAV_API drwav_int32 drwav_bytes_to_s32(const drwav_uint8* data);
+DRWAV_API drwav_uint64 drwav_bytes_to_u64(const drwav_uint8* data);
+DRWAV_API drwav_int64 drwav_bytes_to_s64(const drwav_uint8* data);
+DRWAV_API float drwav_bytes_to_f32(const drwav_uint8* data);
+DRWAV_API drwav_bool32 drwav_guid_equal(const drwav_uint8 a[16], const drwav_uint8 b[16]);
+DRWAV_API drwav_bool32 drwav_fourcc_equal(const drwav_uint8* a, const char* b);
+#ifdef __cplusplus
}
+#endif
+#endif
+/* dr_wav_h end */
+#endif /* MA_NO_WAV */
-MA_API ma_result ma_hpf2_process_pcm_frames(ma_hpf2* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+#if !defined(MA_NO_FLAC) && !defined(MA_NO_DECODING)
+/* dr_flac_h begin */
+#ifndef dr_flac_h
+#define dr_flac_h
+#ifdef __cplusplus
+extern "C" {
+#endif
+#define DRFLAC_STRINGIFY(x) #x
+#define DRFLAC_XSTRINGIFY(x) DRFLAC_STRINGIFY(x)
+#define DRFLAC_VERSION_MAJOR 0
+#define DRFLAC_VERSION_MINOR 12
+#define DRFLAC_VERSION_REVISION 37
+#define DRFLAC_VERSION_STRING DRFLAC_XSTRINGIFY(DRFLAC_VERSION_MAJOR) "." DRFLAC_XSTRINGIFY(DRFLAC_VERSION_MINOR) "." DRFLAC_XSTRINGIFY(DRFLAC_VERSION_REVISION)
+#include
+typedef signed char drflac_int8;
+typedef unsigned char drflac_uint8;
+typedef signed short drflac_int16;
+typedef unsigned short drflac_uint16;
+typedef signed int drflac_int32;
+typedef unsigned int drflac_uint32;
+#if defined(_MSC_VER) && !defined(__clang__)
+ typedef signed __int64 drflac_int64;
+ typedef unsigned __int64 drflac_uint64;
+#else
+ #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
+ #pragma GCC diagnostic push
+ #pragma GCC diagnostic ignored "-Wlong-long"
+ #if defined(__clang__)
+ #pragma GCC diagnostic ignored "-Wc++11-long-long"
+ #endif
+ #endif
+ typedef signed long long drflac_int64;
+ typedef unsigned long long drflac_uint64;
+ #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
+ #pragma GCC diagnostic pop
+ #endif
+#endif
+#if defined(__LP64__) || defined(_WIN64) || (defined(__x86_64__) && !defined(__ILP32__)) || defined(_M_X64) || defined(__ia64) || defined(_M_IA64) || defined(__aarch64__) || defined(_M_ARM64) || defined(__powerpc64__)
+ typedef drflac_uint64 drflac_uintptr;
+#else
+ typedef drflac_uint32 drflac_uintptr;
+#endif
+typedef drflac_uint8 drflac_bool8;
+typedef drflac_uint32 drflac_bool32;
+#define DRFLAC_TRUE 1
+#define DRFLAC_FALSE 0
+#if !defined(DRFLAC_API)
+ #if defined(DRFLAC_DLL)
+ #if defined(_WIN32)
+ #define DRFLAC_DLL_IMPORT __declspec(dllimport)
+ #define DRFLAC_DLL_EXPORT __declspec(dllexport)
+ #define DRFLAC_DLL_PRIVATE static
+ #else
+ #if defined(__GNUC__) && __GNUC__ >= 4
+ #define DRFLAC_DLL_IMPORT __attribute__((visibility("default")))
+ #define DRFLAC_DLL_EXPORT __attribute__((visibility("default")))
+ #define DRFLAC_DLL_PRIVATE __attribute__((visibility("hidden")))
+ #else
+ #define DRFLAC_DLL_IMPORT
+ #define DRFLAC_DLL_EXPORT
+ #define DRFLAC_DLL_PRIVATE static
+ #endif
+ #endif
+ #if defined(DR_FLAC_IMPLEMENTATION) || defined(DRFLAC_IMPLEMENTATION)
+ #define DRFLAC_API DRFLAC_DLL_EXPORT
+ #else
+ #define DRFLAC_API DRFLAC_DLL_IMPORT
+ #endif
+ #define DRFLAC_PRIVATE DRFLAC_DLL_PRIVATE
+ #else
+ #define DRFLAC_API extern
+ #define DRFLAC_PRIVATE static
+ #endif
+#endif
+#if defined(_MSC_VER) && _MSC_VER >= 1700
+ #define DRFLAC_DEPRECATED __declspec(deprecated)
+#elif (defined(__GNUC__) && __GNUC__ >= 4)
+ #define DRFLAC_DEPRECATED __attribute__((deprecated))
+#elif defined(__has_feature)
+ #if __has_feature(attribute_deprecated)
+ #define DRFLAC_DEPRECATED __attribute__((deprecated))
+ #else
+ #define DRFLAC_DEPRECATED
+ #endif
+#else
+ #define DRFLAC_DEPRECATED
+#endif
+DRFLAC_API void drflac_version(drflac_uint32* pMajor, drflac_uint32* pMinor, drflac_uint32* pRevision);
+DRFLAC_API const char* drflac_version_string(void);
+#ifndef DR_FLAC_BUFFER_SIZE
+#define DR_FLAC_BUFFER_SIZE 4096
+#endif
+#if defined(_WIN64) || defined(_LP64) || defined(__LP64__)
+#define DRFLAC_64BIT
+#endif
+#ifdef DRFLAC_64BIT
+typedef drflac_uint64 drflac_cache_t;
+#else
+typedef drflac_uint32 drflac_cache_t;
+#endif
+#define DRFLAC_METADATA_BLOCK_TYPE_STREAMINFO 0
+#define DRFLAC_METADATA_BLOCK_TYPE_PADDING 1
+#define DRFLAC_METADATA_BLOCK_TYPE_APPLICATION 2
+#define DRFLAC_METADATA_BLOCK_TYPE_SEEKTABLE 3
+#define DRFLAC_METADATA_BLOCK_TYPE_VORBIS_COMMENT 4
+#define DRFLAC_METADATA_BLOCK_TYPE_CUESHEET 5
+#define DRFLAC_METADATA_BLOCK_TYPE_PICTURE 6
+#define DRFLAC_METADATA_BLOCK_TYPE_INVALID 127
+#define DRFLAC_PICTURE_TYPE_OTHER 0
+#define DRFLAC_PICTURE_TYPE_FILE_ICON 1
+#define DRFLAC_PICTURE_TYPE_OTHER_FILE_ICON 2
+#define DRFLAC_PICTURE_TYPE_COVER_FRONT 3
+#define DRFLAC_PICTURE_TYPE_COVER_BACK 4
+#define DRFLAC_PICTURE_TYPE_LEAFLET_PAGE 5
+#define DRFLAC_PICTURE_TYPE_MEDIA 6
+#define DRFLAC_PICTURE_TYPE_LEAD_ARTIST 7
+#define DRFLAC_PICTURE_TYPE_ARTIST 8
+#define DRFLAC_PICTURE_TYPE_CONDUCTOR 9
+#define DRFLAC_PICTURE_TYPE_BAND 10
+#define DRFLAC_PICTURE_TYPE_COMPOSER 11
+#define DRFLAC_PICTURE_TYPE_LYRICIST 12
+#define DRFLAC_PICTURE_TYPE_RECORDING_LOCATION 13
+#define DRFLAC_PICTURE_TYPE_DURING_RECORDING 14
+#define DRFLAC_PICTURE_TYPE_DURING_PERFORMANCE 15
+#define DRFLAC_PICTURE_TYPE_SCREEN_CAPTURE 16
+#define DRFLAC_PICTURE_TYPE_BRIGHT_COLORED_FISH 17
+#define DRFLAC_PICTURE_TYPE_ILLUSTRATION 18
+#define DRFLAC_PICTURE_TYPE_BAND_LOGOTYPE 19
+#define DRFLAC_PICTURE_TYPE_PUBLISHER_LOGOTYPE 20
+typedef enum
{
- if (pHPF == NULL) {
- return MA_INVALID_ARGS;
- }
-
- return ma_biquad_process_pcm_frames(&pHPF->bq, pFramesOut, pFramesIn, frameCount);
-}
-
-MA_API ma_uint32 ma_hpf2_get_latency(const ma_hpf2* pHPF)
+ drflac_container_native,
+ drflac_container_ogg,
+ drflac_container_unknown
+} drflac_container;
+typedef enum
{
- if (pHPF == NULL) {
- return 0;
- }
-
- return ma_biquad_get_latency(&pHPF->bq);
-}
-
-
-MA_API ma_hpf_config ma_hpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order)
+ drflac_seek_origin_start,
+ drflac_seek_origin_current
+} drflac_seek_origin;
+#pragma pack(2)
+typedef struct
{
- ma_hpf_config config;
-
- MA_ZERO_OBJECT(&config);
- config.format = format;
- config.channels = channels;
- config.sampleRate = sampleRate;
- config.cutoffFrequency = cutoffFrequency;
- config.order = ma_min(order, MA_MAX_FILTER_ORDER);
-
- return config;
-}
-
-static ma_result ma_hpf_reinit__internal(const ma_hpf_config* pConfig, ma_hpf* pHPF, ma_bool32 isNew)
+ drflac_uint64 firstPCMFrame;
+ drflac_uint64 flacFrameOffset;
+ drflac_uint16 pcmFrameCount;
+} drflac_seekpoint;
+#pragma pack()
+typedef struct
{
- ma_result result;
- ma_uint32 hpf1Count;
- ma_uint32 hpf2Count;
- ma_uint32 ihpf1;
- ma_uint32 ihpf2;
-
- if (pHPF == NULL || pConfig == NULL) {
- return MA_INVALID_ARGS;
- }
-
- /* Only supporting f32 and s16. */
- if (pConfig->format != ma_format_f32 && pConfig->format != ma_format_s16) {
- return MA_INVALID_ARGS;
- }
-
- /* The format cannot be changed after initialization. */
- if (pHPF->format != ma_format_unknown && pHPF->format != pConfig->format) {
- return MA_INVALID_OPERATION;
- }
-
- /* The channel count cannot be changed after initialization. */
- if (pHPF->channels != 0 && pHPF->channels != pConfig->channels) {
- return MA_INVALID_OPERATION;
- }
-
- if (pConfig->order > MA_MAX_FILTER_ORDER) {
- return MA_INVALID_ARGS;
- }
-
- hpf1Count = pConfig->order % 2;
- hpf2Count = pConfig->order / 2;
-
- MA_ASSERT(hpf1Count <= ma_countof(pHPF->hpf1));
- MA_ASSERT(hpf2Count <= ma_countof(pHPF->hpf2));
-
- /* The filter order can't change between reinits. */
- if (!isNew) {
- if (pHPF->hpf1Count != hpf1Count || pHPF->hpf2Count != hpf2Count) {
- return MA_INVALID_OPERATION;
- }
- }
-
- for (ihpf1 = 0; ihpf1 < hpf1Count; ihpf1 += 1) {
- ma_hpf1_config hpf1Config = ma_hpf1_config_init(pConfig->format, pConfig->channels, pConfig->sampleRate, pConfig->cutoffFrequency);
-
- if (isNew) {
- result = ma_hpf1_init(&hpf1Config, &pHPF->hpf1[ihpf1]);
- } else {
- result = ma_hpf1_reinit(&hpf1Config, &pHPF->hpf1[ihpf1]);
- }
-
- if (result != MA_SUCCESS) {
- return result;
- }
- }
-
- for (ihpf2 = 0; ihpf2 < hpf2Count; ihpf2 += 1) {
- ma_hpf2_config hpf2Config;
- double q;
- double a;
-
- /* Tempting to use 0.707107, but won't result in a Butterworth filter if the order is > 2. */
- if (hpf1Count == 1) {
- a = (1 + ihpf2*1) * (MA_PI_D/(pConfig->order*1)); /* Odd order. */
- } else {
- a = (1 + ihpf2*2) * (MA_PI_D/(pConfig->order*2)); /* Even order. */
- }
- q = 1 / (2*ma_cosd(a));
-
- hpf2Config = ma_hpf2_config_init(pConfig->format, pConfig->channels, pConfig->sampleRate, pConfig->cutoffFrequency, q);
-
- if (isNew) {
- result = ma_hpf2_init(&hpf2Config, &pHPF->hpf2[ihpf2]);
- } else {
- result = ma_hpf2_reinit(&hpf2Config, &pHPF->hpf2[ihpf2]);
- }
-
- if (result != MA_SUCCESS) {
- return result;
- }
- }
-
- pHPF->hpf1Count = hpf1Count;
- pHPF->hpf2Count = hpf2Count;
- pHPF->format = pConfig->format;
- pHPF->channels = pConfig->channels;
- pHPF->sampleRate = pConfig->sampleRate;
-
- return MA_SUCCESS;
-}
-
-MA_API ma_result ma_hpf_init(const ma_hpf_config* pConfig, ma_hpf* pHPF)
+ drflac_uint16 minBlockSizeInPCMFrames;
+ drflac_uint16 maxBlockSizeInPCMFrames;
+ drflac_uint32 minFrameSizeInPCMFrames;
+ drflac_uint32 maxFrameSizeInPCMFrames;
+ drflac_uint32 sampleRate;
+ drflac_uint8 channels;
+ drflac_uint8 bitsPerSample;
+ drflac_uint64 totalPCMFrameCount;
+ drflac_uint8 md5[16];
+} drflac_streaminfo;
+typedef struct
{
- if (pHPF == NULL) {
- return MA_INVALID_ARGS;
- }
-
- MA_ZERO_OBJECT(pHPF);
-
- if (pConfig == NULL) {
- return MA_INVALID_ARGS;
- }
-
- return ma_hpf_reinit__internal(pConfig, pHPF, /*isNew*/MA_TRUE);
-}
-
-MA_API ma_result ma_hpf_reinit(const ma_hpf_config* pConfig, ma_hpf* pHPF)
+ drflac_uint32 type;
+ const void* pRawData;
+ drflac_uint32 rawDataSize;
+ union
+ {
+ drflac_streaminfo streaminfo;
+ struct
+ {
+ int unused;
+ } padding;
+ struct
+ {
+ drflac_uint32 id;
+ const void* pData;
+ drflac_uint32 dataSize;
+ } application;
+ struct
+ {
+ drflac_uint32 seekpointCount;
+ const drflac_seekpoint* pSeekpoints;
+ } seektable;
+ struct
+ {
+ drflac_uint32 vendorLength;
+ const char* vendor;
+ drflac_uint32 commentCount;
+ const void* pComments;
+ } vorbis_comment;
+ struct
+ {
+ char catalog[128];
+ drflac_uint64 leadInSampleCount;
+ drflac_bool32 isCD;
+ drflac_uint8 trackCount;
+ const void* pTrackData;
+ } cuesheet;
+ struct
+ {
+ drflac_uint32 type;
+ drflac_uint32 mimeLength;
+ const char* mime;
+ drflac_uint32 descriptionLength;
+ const char* description;
+ drflac_uint32 width;
+ drflac_uint32 height;
+ drflac_uint32 colorDepth;
+ drflac_uint32 indexColorCount;
+ drflac_uint32 pictureDataSize;
+ const drflac_uint8* pPictureData;
+ } picture;
+ } data;
+} drflac_metadata;
+typedef size_t (* drflac_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead);
+typedef drflac_bool32 (* drflac_seek_proc)(void* pUserData, int offset, drflac_seek_origin origin);
+typedef void (* drflac_meta_proc)(void* pUserData, drflac_metadata* pMetadata);
+typedef struct
{
- return ma_hpf_reinit__internal(pConfig, pHPF, /*isNew*/MA_FALSE);
-}
-
-MA_API ma_result ma_hpf_process_pcm_frames(ma_hpf* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+ void* pUserData;
+ void* (* onMalloc)(size_t sz, void* pUserData);
+ void* (* onRealloc)(void* p, size_t sz, void* pUserData);
+ void (* onFree)(void* p, void* pUserData);
+} drflac_allocation_callbacks;
+typedef struct
{
- ma_result result;
- ma_uint32 ihpf1;
- ma_uint32 ihpf2;
-
- if (pHPF == NULL) {
- return MA_INVALID_ARGS;
- }
-
- /* Faster path for in-place. */
- if (pFramesOut == pFramesIn) {
- for (ihpf1 = 0; ihpf1 < pHPF->hpf1Count; ihpf1 += 1) {
- result = ma_hpf1_process_pcm_frames(&pHPF->hpf1[ihpf1], pFramesOut, pFramesOut, frameCount);
- if (result != MA_SUCCESS) {
- return result;
- }
- }
-
- for (ihpf2 = 0; ihpf2 < pHPF->hpf2Count; ihpf2 += 1) {
- result = ma_hpf2_process_pcm_frames(&pHPF->hpf2[ihpf2], pFramesOut, pFramesOut, frameCount);
- if (result != MA_SUCCESS) {
- return result;
- }
- }
- }
-
- /* Slightly slower path for copying. */
- if (pFramesOut != pFramesIn) {
- ma_uint32 iFrame;
-
- /* */ if (pHPF->format == ma_format_f32) {
- /* */ float* pFramesOutF32 = ( float*)pFramesOut;
- const float* pFramesInF32 = (const float*)pFramesIn;
-
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- MA_COPY_MEMORY(pFramesOutF32, pFramesInF32, ma_get_bytes_per_frame(pHPF->format, pHPF->channels));
-
- for (ihpf1 = 0; ihpf1 < pHPF->hpf1Count; ihpf1 += 1) {
- ma_hpf1_process_pcm_frame_f32(&pHPF->hpf1[ihpf1], pFramesOutF32, pFramesOutF32);
- }
-
- for (ihpf2 = 0; ihpf2 < pHPF->hpf2Count; ihpf2 += 1) {
- ma_hpf2_process_pcm_frame_f32(&pHPF->hpf2[ihpf2], pFramesOutF32, pFramesOutF32);
- }
-
- pFramesOutF32 += pHPF->channels;
- pFramesInF32 += pHPF->channels;
- }
- } else if (pHPF->format == ma_format_s16) {
- /* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut;
- const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn;
-
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- MA_COPY_MEMORY(pFramesOutS16, pFramesInS16, ma_get_bytes_per_frame(pHPF->format, pHPF->channels));
-
- for (ihpf1 = 0; ihpf1 < pHPF->hpf1Count; ihpf1 += 1) {
- ma_hpf1_process_pcm_frame_s16(&pHPF->hpf1[ihpf1], pFramesOutS16, pFramesOutS16);
- }
-
- for (ihpf2 = 0; ihpf2 < pHPF->hpf2Count; ihpf2 += 1) {
- ma_hpf2_process_pcm_frame_s16(&pHPF->hpf2[ihpf2], pFramesOutS16, pFramesOutS16);
- }
-
- pFramesOutS16 += pHPF->channels;
- pFramesInS16 += pHPF->channels;
- }
- } else {
- MA_ASSERT(MA_FALSE);
- return MA_INVALID_OPERATION; /* Should never hit this. */
- }
- }
-
- return MA_SUCCESS;
-}
-
-MA_API ma_uint32 ma_hpf_get_latency(const ma_hpf* pHPF)
+ const drflac_uint8* data;
+ size_t dataSize;
+ size_t currentReadPos;
+} drflac__memory_stream;
+typedef struct
{
- if (pHPF == NULL) {
- return 0;
- }
-
- return pHPF->hpf2Count*2 + pHPF->hpf1Count;
-}
-
-
-/**************************************************************************************************************************************************************
-
-Band-Pass Filtering
-
-**************************************************************************************************************************************************************/
-MA_API ma_bpf2_config ma_bpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q)
+ drflac_read_proc onRead;
+ drflac_seek_proc onSeek;
+ void* pUserData;
+ size_t unalignedByteCount;
+ drflac_cache_t unalignedCache;
+ drflac_uint32 nextL2Line;
+ drflac_uint32 consumedBits;
+ drflac_cache_t cacheL2[DR_FLAC_BUFFER_SIZE/sizeof(drflac_cache_t)];
+ drflac_cache_t cache;
+ drflac_uint16 crc16;
+ drflac_cache_t crc16Cache;
+ drflac_uint32 crc16CacheIgnoredBytes;
+} drflac_bs;
+typedef struct
{
- ma_bpf2_config config;
-
- MA_ZERO_OBJECT(&config);
- config.format = format;
- config.channels = channels;
- config.sampleRate = sampleRate;
- config.cutoffFrequency = cutoffFrequency;
- config.q = q;
-
- /* Q cannot be 0 or else it'll result in a division by 0. In this case just default to 0.707107. */
- if (config.q == 0) {
- config.q = 0.707107;
- }
-
- return config;
-}
-
-
-static MA_INLINE ma_biquad_config ma_bpf2__get_biquad_config(const ma_bpf2_config* pConfig)
+ drflac_uint8 subframeType;
+ drflac_uint8 wastedBitsPerSample;
+ drflac_uint8 lpcOrder;
+ drflac_int32* pSamplesS32;
+} drflac_subframe;
+typedef struct
{
- ma_biquad_config bqConfig;
- double q;
- double w;
- double s;
- double c;
- double a;
-
- MA_ASSERT(pConfig != NULL);
-
- q = pConfig->q;
- w = 2 * MA_PI_D * pConfig->cutoffFrequency / pConfig->sampleRate;
- s = ma_sind(w);
- c = ma_cosd(w);
- a = s / (2*q);
-
- bqConfig.b0 = q * a;
- bqConfig.b1 = 0;
- bqConfig.b2 = -q * a;
- bqConfig.a0 = 1 + a;
- bqConfig.a1 = -2 * c;
- bqConfig.a2 = 1 - a;
-
- bqConfig.format = pConfig->format;
- bqConfig.channels = pConfig->channels;
-
- return bqConfig;
-}
-
-MA_API ma_result ma_bpf2_init(const ma_bpf2_config* pConfig, ma_bpf2* pBPF)
+ drflac_uint64 pcmFrameNumber;
+ drflac_uint32 flacFrameNumber;
+ drflac_uint32 sampleRate;
+ drflac_uint16 blockSizeInPCMFrames;
+ drflac_uint8 channelAssignment;
+ drflac_uint8 bitsPerSample;
+ drflac_uint8 crc8;
+} drflac_frame_header;
+typedef struct
{
- ma_result result;
- ma_biquad_config bqConfig;
-
- if (pBPF == NULL) {
- return MA_INVALID_ARGS;
- }
-
- MA_ZERO_OBJECT(pBPF);
-
- if (pConfig == NULL) {
- return MA_INVALID_ARGS;
- }
-
- bqConfig = ma_bpf2__get_biquad_config(pConfig);
- result = ma_biquad_init(&bqConfig, &pBPF->bq);
- if (result != MA_SUCCESS) {
- return result;
- }
-
- return MA_SUCCESS;
-}
-
-MA_API ma_result ma_bpf2_reinit(const ma_bpf2_config* pConfig, ma_bpf2* pBPF)
+ drflac_frame_header header;
+ drflac_uint32 pcmFramesRemaining;
+ drflac_subframe subframes[8];
+} drflac_frame;
+typedef struct
{
- ma_result result;
- ma_biquad_config bqConfig;
-
- if (pBPF == NULL || pConfig == NULL) {
- return MA_INVALID_ARGS;
- }
-
- bqConfig = ma_bpf2__get_biquad_config(pConfig);
- result = ma_biquad_reinit(&bqConfig, &pBPF->bq);
- if (result != MA_SUCCESS) {
- return result;
- }
-
- return MA_SUCCESS;
-}
-
-static MA_INLINE void ma_bpf2_process_pcm_frame_s16(ma_bpf2* pBPF, ma_int16* pFrameOut, const ma_int16* pFrameIn)
+ drflac_meta_proc onMeta;
+ void* pUserDataMD;
+ drflac_allocation_callbacks allocationCallbacks;
+ drflac_uint32 sampleRate;
+ drflac_uint8 channels;
+ drflac_uint8 bitsPerSample;
+ drflac_uint16 maxBlockSizeInPCMFrames;
+ drflac_uint64 totalPCMFrameCount;
+ drflac_container container;
+ drflac_uint32 seekpointCount;
+ drflac_frame currentFLACFrame;
+ drflac_uint64 currentPCMFrame;
+ drflac_uint64 firstFLACFramePosInBytes;
+ drflac__memory_stream memoryStream;
+ drflac_int32* pDecodedSamples;
+ drflac_seekpoint* pSeekpoints;
+ void* _oggbs;
+ drflac_bool32 _noSeekTableSeek : 1;
+ drflac_bool32 _noBinarySearchSeek : 1;
+ drflac_bool32 _noBruteForceSeek : 1;
+ drflac_bs bs;
+ drflac_uint8 pExtraData[1];
+} drflac;
+DRFLAC_API drflac* drflac_open(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
+DRFLAC_API drflac* drflac_open_relaxed(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_container container, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
+DRFLAC_API drflac* drflac_open_with_metadata(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
+DRFLAC_API drflac* drflac_open_with_metadata_relaxed(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, drflac_container container, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
+DRFLAC_API void drflac_close(drflac* pFlac);
+DRFLAC_API drflac_uint64 drflac_read_pcm_frames_s32(drflac* pFlac, drflac_uint64 framesToRead, drflac_int32* pBufferOut);
+DRFLAC_API drflac_uint64 drflac_read_pcm_frames_s16(drflac* pFlac, drflac_uint64 framesToRead, drflac_int16* pBufferOut);
+DRFLAC_API drflac_uint64 drflac_read_pcm_frames_f32(drflac* pFlac, drflac_uint64 framesToRead, float* pBufferOut);
+DRFLAC_API drflac_bool32 drflac_seek_to_pcm_frame(drflac* pFlac, drflac_uint64 pcmFrameIndex);
+#ifndef DR_FLAC_NO_STDIO
+DRFLAC_API drflac* drflac_open_file(const char* pFileName, const drflac_allocation_callbacks* pAllocationCallbacks);
+DRFLAC_API drflac* drflac_open_file_w(const wchar_t* pFileName, const drflac_allocation_callbacks* pAllocationCallbacks);
+DRFLAC_API drflac* drflac_open_file_with_metadata(const char* pFileName, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
+DRFLAC_API drflac* drflac_open_file_with_metadata_w(const wchar_t* pFileName, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
+#endif
+DRFLAC_API drflac* drflac_open_memory(const void* pData, size_t dataSize, const drflac_allocation_callbacks* pAllocationCallbacks);
+DRFLAC_API drflac* drflac_open_memory_with_metadata(const void* pData, size_t dataSize, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
+DRFLAC_API drflac_int32* drflac_open_and_read_pcm_frames_s32(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
+DRFLAC_API drflac_int16* drflac_open_and_read_pcm_frames_s16(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
+DRFLAC_API float* drflac_open_and_read_pcm_frames_f32(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
+#ifndef DR_FLAC_NO_STDIO
+DRFLAC_API drflac_int32* drflac_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
+DRFLAC_API drflac_int16* drflac_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
+DRFLAC_API float* drflac_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
+#endif
+DRFLAC_API drflac_int32* drflac_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
+DRFLAC_API drflac_int16* drflac_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
+DRFLAC_API float* drflac_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
+DRFLAC_API void drflac_free(void* p, const drflac_allocation_callbacks* pAllocationCallbacks);
+typedef struct
{
- ma_biquad_process_pcm_frame_s16(&pBPF->bq, pFrameOut, pFrameIn);
-}
-
-static MA_INLINE void ma_bpf2_process_pcm_frame_f32(ma_bpf2* pBPF, float* pFrameOut, const float* pFrameIn)
+ drflac_uint32 countRemaining;
+ const char* pRunningData;
+} drflac_vorbis_comment_iterator;
+DRFLAC_API void drflac_init_vorbis_comment_iterator(drflac_vorbis_comment_iterator* pIter, drflac_uint32 commentCount, const void* pComments);
+DRFLAC_API const char* drflac_next_vorbis_comment(drflac_vorbis_comment_iterator* pIter, drflac_uint32* pCommentLengthOut);
+typedef struct
{
- ma_biquad_process_pcm_frame_f32(&pBPF->bq, pFrameOut, pFrameIn);
-}
-
-MA_API ma_result ma_bpf2_process_pcm_frames(ma_bpf2* pBPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+ drflac_uint32 countRemaining;
+ const char* pRunningData;
+} drflac_cuesheet_track_iterator;
+#pragma pack(4)
+typedef struct
{
- if (pBPF == NULL) {
- return MA_INVALID_ARGS;
- }
-
- return ma_biquad_process_pcm_frames(&pBPF->bq, pFramesOut, pFramesIn, frameCount);
-}
-
-MA_API ma_uint32 ma_bpf2_get_latency(const ma_bpf2* pBPF)
+ drflac_uint64 offset;
+ drflac_uint8 index;
+ drflac_uint8 reserved[3];
+} drflac_cuesheet_track_index;
+#pragma pack()
+typedef struct
{
- if (pBPF == NULL) {
- return 0;
- }
-
- return ma_biquad_get_latency(&pBPF->bq);
+ drflac_uint64 offset;
+ drflac_uint8 trackNumber;
+ char ISRC[12];
+ drflac_bool8 isAudio;
+ drflac_bool8 preEmphasis;
+ drflac_uint8 indexCount;
+ const drflac_cuesheet_track_index* pIndexPoints;
+} drflac_cuesheet_track;
+DRFLAC_API void drflac_init_cuesheet_track_iterator(drflac_cuesheet_track_iterator* pIter, drflac_uint32 trackCount, const void* pTrackData);
+DRFLAC_API drflac_bool32 drflac_next_cuesheet_track(drflac_cuesheet_track_iterator* pIter, drflac_cuesheet_track* pCuesheetTrack);
+#ifdef __cplusplus
}
+#endif
+#endif
+/* dr_flac_h end */
+#endif /* MA_NO_FLAC */
-
-MA_API ma_bpf_config ma_bpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order)
+#if !defined(MA_NO_MP3) && !defined(MA_NO_DECODING)
+/* dr_mp3_h begin */
+#ifndef dr_mp3_h
+#define dr_mp3_h
+#ifdef __cplusplus
+extern "C" {
+#endif
+#define DRMP3_STRINGIFY(x) #x
+#define DRMP3_XSTRINGIFY(x) DRMP3_STRINGIFY(x)
+#define DRMP3_VERSION_MAJOR 0
+#define DRMP3_VERSION_MINOR 6
+#define DRMP3_VERSION_REVISION 32
+#define DRMP3_VERSION_STRING DRMP3_XSTRINGIFY(DRMP3_VERSION_MAJOR) "." DRMP3_XSTRINGIFY(DRMP3_VERSION_MINOR) "." DRMP3_XSTRINGIFY(DRMP3_VERSION_REVISION)
+#include
+typedef signed char drmp3_int8;
+typedef unsigned char drmp3_uint8;
+typedef signed short drmp3_int16;
+typedef unsigned short drmp3_uint16;
+typedef signed int drmp3_int32;
+typedef unsigned int drmp3_uint32;
+#if defined(_MSC_VER) && !defined(__clang__)
+ typedef signed __int64 drmp3_int64;
+ typedef unsigned __int64 drmp3_uint64;
+#else
+ #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
+ #pragma GCC diagnostic push
+ #pragma GCC diagnostic ignored "-Wlong-long"
+ #if defined(__clang__)
+ #pragma GCC diagnostic ignored "-Wc++11-long-long"
+ #endif
+ #endif
+ typedef signed long long drmp3_int64;
+ typedef unsigned long long drmp3_uint64;
+ #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
+ #pragma GCC diagnostic pop
+ #endif
+#endif
+#if defined(__LP64__) || defined(_WIN64) || (defined(__x86_64__) && !defined(__ILP32__)) || defined(_M_X64) || defined(__ia64) || defined (_M_IA64) || defined(__aarch64__) || defined(_M_ARM64) || defined(__powerpc64__)
+ typedef drmp3_uint64 drmp3_uintptr;
+#else
+ typedef drmp3_uint32 drmp3_uintptr;
+#endif
+typedef drmp3_uint8 drmp3_bool8;
+typedef drmp3_uint32 drmp3_bool32;
+#define DRMP3_TRUE 1
+#define DRMP3_FALSE 0
+#if !defined(DRMP3_API)
+ #if defined(DRMP3_DLL)
+ #if defined(_WIN32)
+ #define DRMP3_DLL_IMPORT __declspec(dllimport)
+ #define DRMP3_DLL_EXPORT __declspec(dllexport)
+ #define DRMP3_DLL_PRIVATE static
+ #else
+ #if defined(__GNUC__) && __GNUC__ >= 4
+ #define DRMP3_DLL_IMPORT __attribute__((visibility("default")))
+ #define DRMP3_DLL_EXPORT __attribute__((visibility("default")))
+ #define DRMP3_DLL_PRIVATE __attribute__((visibility("hidden")))
+ #else
+ #define DRMP3_DLL_IMPORT
+ #define DRMP3_DLL_EXPORT
+ #define DRMP3_DLL_PRIVATE static
+ #endif
+ #endif
+ #if defined(DR_MP3_IMPLEMENTATION) || defined(DRMP3_IMPLEMENTATION)
+ #define DRMP3_API DRMP3_DLL_EXPORT
+ #else
+ #define DRMP3_API DRMP3_DLL_IMPORT
+ #endif
+ #define DRMP3_PRIVATE DRMP3_DLL_PRIVATE
+ #else
+ #define DRMP3_API extern
+ #define DRMP3_PRIVATE static
+ #endif
+#endif
+typedef drmp3_int32 drmp3_result;
+#define DRMP3_SUCCESS 0
+#define DRMP3_ERROR -1
+#define DRMP3_INVALID_ARGS -2
+#define DRMP3_INVALID_OPERATION -3
+#define DRMP3_OUT_OF_MEMORY -4
+#define DRMP3_OUT_OF_RANGE -5
+#define DRMP3_ACCESS_DENIED -6
+#define DRMP3_DOES_NOT_EXIST -7
+#define DRMP3_ALREADY_EXISTS -8
+#define DRMP3_TOO_MANY_OPEN_FILES -9
+#define DRMP3_INVALID_FILE -10
+#define DRMP3_TOO_BIG -11
+#define DRMP3_PATH_TOO_LONG -12
+#define DRMP3_NAME_TOO_LONG -13
+#define DRMP3_NOT_DIRECTORY -14
+#define DRMP3_IS_DIRECTORY -15
+#define DRMP3_DIRECTORY_NOT_EMPTY -16
+#define DRMP3_END_OF_FILE -17
+#define DRMP3_NO_SPACE -18
+#define DRMP3_BUSY -19
+#define DRMP3_IO_ERROR -20
+#define DRMP3_INTERRUPT -21
+#define DRMP3_UNAVAILABLE -22
+#define DRMP3_ALREADY_IN_USE -23
+#define DRMP3_BAD_ADDRESS -24
+#define DRMP3_BAD_SEEK -25
+#define DRMP3_BAD_PIPE -26
+#define DRMP3_DEADLOCK -27
+#define DRMP3_TOO_MANY_LINKS -28
+#define DRMP3_NOT_IMPLEMENTED -29
+#define DRMP3_NO_MESSAGE -30
+#define DRMP3_BAD_MESSAGE -31
+#define DRMP3_NO_DATA_AVAILABLE -32
+#define DRMP3_INVALID_DATA -33
+#define DRMP3_TIMEOUT -34
+#define DRMP3_NO_NETWORK -35
+#define DRMP3_NOT_UNIQUE -36
+#define DRMP3_NOT_SOCKET -37
+#define DRMP3_NO_ADDRESS -38
+#define DRMP3_BAD_PROTOCOL -39
+#define DRMP3_PROTOCOL_UNAVAILABLE -40
+#define DRMP3_PROTOCOL_NOT_SUPPORTED -41
+#define DRMP3_PROTOCOL_FAMILY_NOT_SUPPORTED -42
+#define DRMP3_ADDRESS_FAMILY_NOT_SUPPORTED -43
+#define DRMP3_SOCKET_NOT_SUPPORTED -44
+#define DRMP3_CONNECTION_RESET -45
+#define DRMP3_ALREADY_CONNECTED -46
+#define DRMP3_NOT_CONNECTED -47
+#define DRMP3_CONNECTION_REFUSED -48
+#define DRMP3_NO_HOST -49
+#define DRMP3_IN_PROGRESS -50
+#define DRMP3_CANCELLED -51
+#define DRMP3_MEMORY_ALREADY_MAPPED -52
+#define DRMP3_AT_END -53
+#define DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME 1152
+#define DRMP3_MAX_SAMPLES_PER_FRAME (DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME*2)
+#ifdef _MSC_VER
+ #define DRMP3_INLINE __forceinline
+#elif defined(__GNUC__)
+ #if defined(__STRICT_ANSI__)
+ #define DRMP3_INLINE __inline__ __attribute__((always_inline))
+ #else
+ #define DRMP3_INLINE inline __attribute__((always_inline))
+ #endif
+#elif defined(__WATCOMC__)
+ #define DRMP3_INLINE __inline
+#else
+ #define DRMP3_INLINE
+#endif
+DRMP3_API void drmp3_version(drmp3_uint32* pMajor, drmp3_uint32* pMinor, drmp3_uint32* pRevision);
+DRMP3_API const char* drmp3_version_string(void);
+typedef struct
{
- ma_bpf_config config;
-
- MA_ZERO_OBJECT(&config);
- config.format = format;
- config.channels = channels;
- config.sampleRate = sampleRate;
- config.cutoffFrequency = cutoffFrequency;
- config.order = ma_min(order, MA_MAX_FILTER_ORDER);
-
- return config;
-}
-
-static ma_result ma_bpf_reinit__internal(const ma_bpf_config* pConfig, ma_bpf* pBPF, ma_bool32 isNew)
+ int frame_bytes, channels, hz, layer, bitrate_kbps;
+} drmp3dec_frame_info;
+typedef struct
{
- ma_result result;
- ma_uint32 bpf2Count;
- ma_uint32 ibpf2;
-
- if (pBPF == NULL || pConfig == NULL) {
- return MA_INVALID_ARGS;
- }
-
- /* Only supporting f32 and s16. */
- if (pConfig->format != ma_format_f32 && pConfig->format != ma_format_s16) {
- return MA_INVALID_ARGS;
- }
-
- /* The format cannot be changed after initialization. */
- if (pBPF->format != ma_format_unknown && pBPF->format != pConfig->format) {
- return MA_INVALID_OPERATION;
- }
-
- /* The channel count cannot be changed after initialization. */
- if (pBPF->channels != 0 && pBPF->channels != pConfig->channels) {
- return MA_INVALID_OPERATION;
- }
-
- if (pConfig->order > MA_MAX_FILTER_ORDER) {
- return MA_INVALID_ARGS;
- }
-
- /* We must have an even number of order. */
- if ((pConfig->order & 0x1) != 0) {
- return MA_INVALID_ARGS;
- }
-
- bpf2Count = pConfig->order / 2;
-
- MA_ASSERT(bpf2Count <= ma_countof(pBPF->bpf2));
-
- /* The filter order can't change between reinits. */
- if (!isNew) {
- if (pBPF->bpf2Count != bpf2Count) {
- return MA_INVALID_OPERATION;
- }
- }
-
- for (ibpf2 = 0; ibpf2 < bpf2Count; ibpf2 += 1) {
- ma_bpf2_config bpf2Config;
- double q;
-
- /* TODO: Calculate Q to make this a proper Butterworth filter. */
- q = 0.707107;
-
- bpf2Config = ma_bpf2_config_init(pConfig->format, pConfig->channels, pConfig->sampleRate, pConfig->cutoffFrequency, q);
-
- if (isNew) {
- result = ma_bpf2_init(&bpf2Config, &pBPF->bpf2[ibpf2]);
- } else {
- result = ma_bpf2_reinit(&bpf2Config, &pBPF->bpf2[ibpf2]);
- }
-
- if (result != MA_SUCCESS) {
- return result;
- }
- }
-
- pBPF->bpf2Count = bpf2Count;
- pBPF->format = pConfig->format;
- pBPF->channels = pConfig->channels;
-
- return MA_SUCCESS;
-}
-
-MA_API ma_result ma_bpf_init(const ma_bpf_config* pConfig, ma_bpf* pBPF)
+ float mdct_overlap[2][9*32], qmf_state[15*2*32];
+ int reserv, free_format_bytes;
+ drmp3_uint8 header[4], reserv_buf[511];
+} drmp3dec;
+DRMP3_API void drmp3dec_init(drmp3dec *dec);
+DRMP3_API int drmp3dec_decode_frame(drmp3dec *dec, const drmp3_uint8 *mp3, int mp3_bytes, void *pcm, drmp3dec_frame_info *info);
+DRMP3_API void drmp3dec_f32_to_s16(const float *in, drmp3_int16 *out, size_t num_samples);
+typedef enum
{
- if (pBPF == NULL) {
- return MA_INVALID_ARGS;
- }
-
- MA_ZERO_OBJECT(pBPF);
-
- if (pConfig == NULL) {
- return MA_INVALID_ARGS;
- }
-
- return ma_bpf_reinit__internal(pConfig, pBPF, /*isNew*/MA_TRUE);
-}
-
-MA_API ma_result ma_bpf_reinit(const ma_bpf_config* pConfig, ma_bpf* pBPF)
+ drmp3_seek_origin_start,
+ drmp3_seek_origin_current
+} drmp3_seek_origin;
+typedef struct
{
- return ma_bpf_reinit__internal(pConfig, pBPF, /*isNew*/MA_FALSE);
-}
-
-MA_API ma_result ma_bpf_process_pcm_frames(ma_bpf* pBPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+ drmp3_uint64 seekPosInBytes;
+ drmp3_uint64 pcmFrameIndex;
+ drmp3_uint16 mp3FramesToDiscard;
+ drmp3_uint16 pcmFramesToDiscard;
+} drmp3_seek_point;
+typedef size_t (* drmp3_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead);
+typedef drmp3_bool32 (* drmp3_seek_proc)(void* pUserData, int offset, drmp3_seek_origin origin);
+typedef struct
{
- ma_result result;
- ma_uint32 ibpf2;
-
- if (pBPF == NULL) {
- return MA_INVALID_ARGS;
- }
-
- /* Faster path for in-place. */
- if (pFramesOut == pFramesIn) {
- for (ibpf2 = 0; ibpf2 < pBPF->bpf2Count; ibpf2 += 1) {
- result = ma_bpf2_process_pcm_frames(&pBPF->bpf2[ibpf2], pFramesOut, pFramesOut, frameCount);
- if (result != MA_SUCCESS) {
- return result;
- }
- }
- }
-
- /* Slightly slower path for copying. */
- if (pFramesOut != pFramesIn) {
- ma_uint32 iFrame;
-
- /* */ if (pBPF->format == ma_format_f32) {
- /* */ float* pFramesOutF32 = ( float*)pFramesOut;
- const float* pFramesInF32 = (const float*)pFramesIn;
-
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- MA_COPY_MEMORY(pFramesOutF32, pFramesInF32, ma_get_bytes_per_frame(pBPF->format, pBPF->channels));
-
- for (ibpf2 = 0; ibpf2 < pBPF->bpf2Count; ibpf2 += 1) {
- ma_bpf2_process_pcm_frame_f32(&pBPF->bpf2[ibpf2], pFramesOutF32, pFramesOutF32);
- }
-
- pFramesOutF32 += pBPF->channels;
- pFramesInF32 += pBPF->channels;
- }
- } else if (pBPF->format == ma_format_s16) {
- /* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut;
- const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn;
-
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- MA_COPY_MEMORY(pFramesOutS16, pFramesInS16, ma_get_bytes_per_frame(pBPF->format, pBPF->channels));
-
- for (ibpf2 = 0; ibpf2 < pBPF->bpf2Count; ibpf2 += 1) {
- ma_bpf2_process_pcm_frame_s16(&pBPF->bpf2[ibpf2], pFramesOutS16, pFramesOutS16);
- }
-
- pFramesOutS16 += pBPF->channels;
- pFramesInS16 += pBPF->channels;
- }
- } else {
- MA_ASSERT(MA_FALSE);
- return MA_INVALID_OPERATION; /* Should never hit this. */
- }
- }
-
- return MA_SUCCESS;
-}
-
-MA_API ma_uint32 ma_bpf_get_latency(const ma_bpf* pBPF)
+ void* pUserData;
+ void* (* onMalloc)(size_t sz, void* pUserData);
+ void* (* onRealloc)(void* p, size_t sz, void* pUserData);
+ void (* onFree)(void* p, void* pUserData);
+} drmp3_allocation_callbacks;
+typedef struct
{
- if (pBPF == NULL) {
- return 0;
- }
-
- return pBPF->bpf2Count*2;
-}
-
-
-/**************************************************************************************************************************************************************
-
-Notching Filter
-
-**************************************************************************************************************************************************************/
-MA_API ma_notch2_config ma_notch2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double q, double frequency)
+ drmp3_uint32 channels;
+ drmp3_uint32 sampleRate;
+} drmp3_config;
+typedef struct
{
- ma_notch2_config config;
-
- MA_ZERO_OBJECT(&config);
- config.format = format;
- config.channels = channels;
- config.sampleRate = sampleRate;
- config.q = q;
- config.frequency = frequency;
-
- if (config.q == 0) {
- config.q = 0.707107;
- }
-
- return config;
+ drmp3dec decoder;
+ drmp3dec_frame_info frameInfo;
+ drmp3_uint32 channels;
+ drmp3_uint32 sampleRate;
+ drmp3_read_proc onRead;
+ drmp3_seek_proc onSeek;
+ void* pUserData;
+ drmp3_allocation_callbacks allocationCallbacks;
+ drmp3_uint32 mp3FrameChannels;
+ drmp3_uint32 mp3FrameSampleRate;
+ drmp3_uint32 pcmFramesConsumedInMP3Frame;
+ drmp3_uint32 pcmFramesRemainingInMP3Frame;
+ drmp3_uint8 pcmFrames[sizeof(float)*DRMP3_MAX_SAMPLES_PER_FRAME];
+ drmp3_uint64 currentPCMFrame;
+ drmp3_uint64 streamCursor;
+ drmp3_seek_point* pSeekPoints;
+ drmp3_uint32 seekPointCount;
+ size_t dataSize;
+ size_t dataCapacity;
+ size_t dataConsumed;
+ drmp3_uint8* pData;
+ drmp3_bool32 atEnd : 1;
+ struct
+ {
+ const drmp3_uint8* pData;
+ size_t dataSize;
+ size_t currentReadPos;
+ } memory;
+} drmp3;
+DRMP3_API drmp3_bool32 drmp3_init(drmp3* pMP3, drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, const drmp3_allocation_callbacks* pAllocationCallbacks);
+DRMP3_API drmp3_bool32 drmp3_init_memory(drmp3* pMP3, const void* pData, size_t dataSize, const drmp3_allocation_callbacks* pAllocationCallbacks);
+#ifndef DR_MP3_NO_STDIO
+DRMP3_API drmp3_bool32 drmp3_init_file(drmp3* pMP3, const char* pFilePath, const drmp3_allocation_callbacks* pAllocationCallbacks);
+DRMP3_API drmp3_bool32 drmp3_init_file_w(drmp3* pMP3, const wchar_t* pFilePath, const drmp3_allocation_callbacks* pAllocationCallbacks);
+#endif
+DRMP3_API void drmp3_uninit(drmp3* pMP3);
+DRMP3_API drmp3_uint64 drmp3_read_pcm_frames_f32(drmp3* pMP3, drmp3_uint64 framesToRead, float* pBufferOut);
+DRMP3_API drmp3_uint64 drmp3_read_pcm_frames_s16(drmp3* pMP3, drmp3_uint64 framesToRead, drmp3_int16* pBufferOut);
+DRMP3_API drmp3_bool32 drmp3_seek_to_pcm_frame(drmp3* pMP3, drmp3_uint64 frameIndex);
+DRMP3_API drmp3_uint64 drmp3_get_pcm_frame_count(drmp3* pMP3);
+DRMP3_API drmp3_uint64 drmp3_get_mp3_frame_count(drmp3* pMP3);
+DRMP3_API drmp3_bool32 drmp3_get_mp3_and_pcm_frame_count(drmp3* pMP3, drmp3_uint64* pMP3FrameCount, drmp3_uint64* pPCMFrameCount);
+DRMP3_API drmp3_bool32 drmp3_calculate_seek_points(drmp3* pMP3, drmp3_uint32* pSeekPointCount, drmp3_seek_point* pSeekPoints);
+DRMP3_API drmp3_bool32 drmp3_bind_seek_table(drmp3* pMP3, drmp3_uint32 seekPointCount, drmp3_seek_point* pSeekPoints);
+DRMP3_API float* drmp3_open_and_read_pcm_frames_f32(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks);
+DRMP3_API drmp3_int16* drmp3_open_and_read_pcm_frames_s16(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks);
+DRMP3_API float* drmp3_open_memory_and_read_pcm_frames_f32(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks);
+DRMP3_API drmp3_int16* drmp3_open_memory_and_read_pcm_frames_s16(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks);
+#ifndef DR_MP3_NO_STDIO
+DRMP3_API float* drmp3_open_file_and_read_pcm_frames_f32(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks);
+DRMP3_API drmp3_int16* drmp3_open_file_and_read_pcm_frames_s16(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks);
+#endif
+DRMP3_API void* drmp3_malloc(size_t sz, const drmp3_allocation_callbacks* pAllocationCallbacks);
+DRMP3_API void drmp3_free(void* p, const drmp3_allocation_callbacks* pAllocationCallbacks);
+#ifdef __cplusplus
}
+#endif
+#endif
+/* dr_mp3_h end */
+#endif /* MA_NO_MP3 */
-static MA_INLINE ma_biquad_config ma_notch2__get_biquad_config(const ma_notch2_config* pConfig)
-{
- ma_biquad_config bqConfig;
- double q;
- double w;
- double s;
- double c;
- double a;
-
- MA_ASSERT(pConfig != NULL);
+/**************************************************************************************************************************************************************
- q = pConfig->q;
- w = 2 * MA_PI_D * pConfig->frequency / pConfig->sampleRate;
- s = ma_sind(w);
- c = ma_cosd(w);
- a = s / (2*q);
+Decoding
- bqConfig.b0 = 1;
- bqConfig.b1 = -2 * c;
- bqConfig.b2 = 1;
- bqConfig.a0 = 1 + a;
- bqConfig.a1 = -2 * c;
- bqConfig.a2 = 1 - a;
+**************************************************************************************************************************************************************/
+#ifndef MA_NO_DECODING
- bqConfig.format = pConfig->format;
- bqConfig.channels = pConfig->channels;
+static ma_result ma_decoder_read_bytes(ma_decoder* pDecoder, void* pBufferOut, size_t bytesToRead, size_t* pBytesRead)
+{
+ MA_ASSERT(pDecoder != NULL);
- return bqConfig;
+ return pDecoder->onRead(pDecoder, pBufferOut, bytesToRead, pBytesRead);
}
-MA_API ma_result ma_notch2_init(const ma_notch2_config* pConfig, ma_notch2* pFilter)
+static ma_result ma_decoder_seek_bytes(ma_decoder* pDecoder, ma_int64 byteOffset, ma_seek_origin origin)
{
- ma_result result;
- ma_biquad_config bqConfig;
-
- if (pFilter == NULL) {
- return MA_INVALID_ARGS;
- }
+ MA_ASSERT(pDecoder != NULL);
- MA_ZERO_OBJECT(pFilter);
+ return pDecoder->onSeek(pDecoder, byteOffset, origin);
+}
- if (pConfig == NULL) {
- return MA_INVALID_ARGS;
- }
+static ma_result ma_decoder_tell_bytes(ma_decoder* pDecoder, ma_int64* pCursor)
+{
+ MA_ASSERT(pDecoder != NULL);
- bqConfig = ma_notch2__get_biquad_config(pConfig);
- result = ma_biquad_init(&bqConfig, &pFilter->bq);
- if (result != MA_SUCCESS) {
- return result;
+ if (pDecoder->onTell == NULL) {
+ return MA_NOT_IMPLEMENTED;
}
- return MA_SUCCESS;
+ return pDecoder->onTell(pDecoder, pCursor);
}
-MA_API ma_result ma_notch2_reinit(const ma_notch2_config* pConfig, ma_notch2* pFilter)
-{
- ma_result result;
- ma_biquad_config bqConfig;
- if (pFilter == NULL || pConfig == NULL) {
- return MA_INVALID_ARGS;
- }
+MA_API ma_decoding_backend_config ma_decoding_backend_config_init(ma_format preferredFormat, ma_uint32 seekPointCount)
+{
+ ma_decoding_backend_config config;
- bqConfig = ma_notch2__get_biquad_config(pConfig);
- result = ma_biquad_reinit(&bqConfig, &pFilter->bq);
- if (result != MA_SUCCESS) {
- return result;
- }
+ MA_ZERO_OBJECT(&config);
+ config.preferredFormat = preferredFormat;
+ config.seekPointCount = seekPointCount;
- return MA_SUCCESS;
+ return config;
}
-static MA_INLINE void ma_notch2_process_pcm_frame_s16(ma_notch2* pFilter, ma_int16* pFrameOut, const ma_int16* pFrameIn)
-{
- ma_biquad_process_pcm_frame_s16(&pFilter->bq, pFrameOut, pFrameIn);
-}
-static MA_INLINE void ma_notch2_process_pcm_frame_f32(ma_notch2* pFilter, float* pFrameOut, const float* pFrameIn)
+MA_API ma_decoder_config ma_decoder_config_init(ma_format outputFormat, ma_uint32 outputChannels, ma_uint32 outputSampleRate)
{
- ma_biquad_process_pcm_frame_f32(&pFilter->bq, pFrameOut, pFrameIn);
-}
+ ma_decoder_config config;
+ MA_ZERO_OBJECT(&config);
+ config.format = outputFormat;
+ config.channels = outputChannels;
+ config.sampleRate = outputSampleRate;
+ config.resampling = ma_resampler_config_init(ma_format_unknown, 0, 0, 0, ma_resample_algorithm_linear); /* Format/channels/rate doesn't matter here. */
+ config.encodingFormat = ma_encoding_format_unknown;
-MA_API ma_result ma_notch2_process_pcm_frames(ma_notch2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
-{
- if (pFilter == NULL) {
- return MA_INVALID_ARGS;
- }
+ /* Note that we are intentionally leaving the channel map empty here which will cause the default channel map to be used. */
- return ma_biquad_process_pcm_frames(&pFilter->bq, pFramesOut, pFramesIn, frameCount);
+ return config;
}
-MA_API ma_uint32 ma_notch2_get_latency(const ma_notch2* pFilter)
+MA_API ma_decoder_config ma_decoder_config_init_default()
{
- if (pFilter == NULL) {
- return 0;
- }
-
- return ma_biquad_get_latency(&pFilter->bq);
+ return ma_decoder_config_init(ma_format_unknown, 0, 0);
}
-
-
-/**************************************************************************************************************************************************************
-
-Peaking EQ Filter
-
-**************************************************************************************************************************************************************/
-MA_API ma_peak2_config ma_peak2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double q, double frequency)
+MA_API ma_decoder_config ma_decoder_config_init_copy(const ma_decoder_config* pConfig)
{
- ma_peak2_config config;
-
- MA_ZERO_OBJECT(&config);
- config.format = format;
- config.channels = channels;
- config.sampleRate = sampleRate;
- config.gainDB = gainDB;
- config.q = q;
- config.frequency = frequency;
-
- if (config.q == 0) {
- config.q = 0.707107;
+ ma_decoder_config config;
+ if (pConfig != NULL) {
+ config = *pConfig;
+ } else {
+ MA_ZERO_OBJECT(&config);
}
return config;
}
-
-static MA_INLINE ma_biquad_config ma_peak2__get_biquad_config(const ma_peak2_config* pConfig)
+static ma_result ma_decoder__init_data_converter(ma_decoder* pDecoder, const ma_decoder_config* pConfig)
{
- ma_biquad_config bqConfig;
- double q;
- double w;
- double s;
- double c;
- double a;
- double A;
+ ma_result result;
+ ma_data_converter_config converterConfig;
+ ma_format internalFormat;
+ ma_uint32 internalChannels;
+ ma_uint32 internalSampleRate;
+ ma_channel internalChannelMap[MA_MAX_CHANNELS];
- MA_ASSERT(pConfig != NULL);
+ MA_ASSERT(pDecoder != NULL);
+ MA_ASSERT(pConfig != NULL);
- q = pConfig->q;
- w = 2 * MA_PI_D * pConfig->frequency / pConfig->sampleRate;
- s = ma_sind(w);
- c = ma_cosd(w);
- a = s / (2*q);
- A = ma_powd(10, (pConfig->gainDB / 40));
+ result = ma_data_source_get_data_format(pDecoder->pBackend, &internalFormat, &internalChannels, &internalSampleRate, internalChannelMap, ma_countof(internalChannelMap));
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to retrieve the internal data format. */
+ }
- bqConfig.b0 = 1 + (a * A);
- bqConfig.b1 = -2 * c;
- bqConfig.b2 = 1 - (a * A);
- bqConfig.a0 = 1 + (a / A);
- bqConfig.a1 = -2 * c;
- bqConfig.a2 = 1 - (a / A);
- bqConfig.format = pConfig->format;
- bqConfig.channels = pConfig->channels;
+ /* Make sure we're not asking for too many channels. */
+ if (pConfig->channels > MA_MAX_CHANNELS) {
+ return MA_INVALID_ARGS;
+ }
- return bqConfig;
-}
+ /* The internal channels should have already been validated at a higher level, but we'll do it again explicitly here for safety. */
+ if (internalChannels > MA_MAX_CHANNELS) {
+ return MA_INVALID_ARGS;
+ }
-MA_API ma_result ma_peak2_init(const ma_peak2_config* pConfig, ma_peak2* pFilter)
-{
- ma_result result;
- ma_biquad_config bqConfig;
- if (pFilter == NULL) {
- return MA_INVALID_ARGS;
+ /* Output format. */
+ if (pConfig->format == ma_format_unknown) {
+ pDecoder->outputFormat = internalFormat;
+ } else {
+ pDecoder->outputFormat = pConfig->format;
}
- MA_ZERO_OBJECT(pFilter);
+ if (pConfig->channels == 0) {
+ pDecoder->outputChannels = internalChannels;
+ } else {
+ pDecoder->outputChannels = pConfig->channels;
+ }
- if (pConfig == NULL) {
- return MA_INVALID_ARGS;
+ if (pConfig->sampleRate == 0) {
+ pDecoder->outputSampleRate = internalSampleRate;
+ } else {
+ pDecoder->outputSampleRate = pConfig->sampleRate;
}
- bqConfig = ma_peak2__get_biquad_config(pConfig);
- result = ma_biquad_init(&bqConfig, &pFilter->bq);
+ converterConfig = ma_data_converter_config_init(
+ internalFormat, pDecoder->outputFormat,
+ internalChannels, pDecoder->outputChannels,
+ internalSampleRate, pDecoder->outputSampleRate
+ );
+ converterConfig.pChannelMapIn = internalChannelMap;
+ converterConfig.pChannelMapOut = pConfig->pChannelMap;
+ converterConfig.channelMixMode = pConfig->channelMixMode;
+ converterConfig.ditherMode = pConfig->ditherMode;
+ converterConfig.allowDynamicSampleRate = MA_FALSE; /* Never allow dynamic sample rate conversion. Setting this to true will disable passthrough optimizations. */
+ converterConfig.resampling = pConfig->resampling;
+
+ result = ma_data_converter_init(&converterConfig, &pDecoder->allocationCallbacks, &pDecoder->converter);
if (result != MA_SUCCESS) {
return result;
}
- return MA_SUCCESS;
-}
+ /*
+ Now that we have the decoder we need to determine whether or not we need a heap-allocated cache. We'll
+ need this if the data converter does not support calculation of the required input frame count. To
+ determine support for this we'll just run a test.
+ */
+ {
+ ma_uint64 unused;
-MA_API ma_result ma_peak2_reinit(const ma_peak2_config* pConfig, ma_peak2* pFilter)
-{
- ma_result result;
- ma_biquad_config bqConfig;
+ result = ma_data_converter_get_required_input_frame_count(&pDecoder->converter, 1, &unused);
+ if (result != MA_SUCCESS) {
+ /*
+ We were unable to calculate the required input frame count which means we'll need to use
+ a heap-allocated cache.
+ */
+ ma_uint64 inputCacheCapSizeInBytes;
- if (pFilter == NULL || pConfig == NULL) {
- return MA_INVALID_ARGS;
- }
+ pDecoder->inputCacheCap = MA_DATA_CONVERTER_STACK_BUFFER_SIZE / ma_get_bytes_per_frame(internalFormat, internalChannels);
- bqConfig = ma_peak2__get_biquad_config(pConfig);
- result = ma_biquad_reinit(&bqConfig, &pFilter->bq);
- if (result != MA_SUCCESS) {
- return result;
+ /* Not strictly necessary, but keeping here for safety in case we change the default value of pDecoder->inputCacheCap. */
+ inputCacheCapSizeInBytes = pDecoder->inputCacheCap * ma_get_bytes_per_frame(internalFormat, internalChannels);
+ if (inputCacheCapSizeInBytes > MA_SIZE_MAX) {
+ ma_data_converter_uninit(&pDecoder->converter, &pDecoder->allocationCallbacks);
+ return MA_OUT_OF_MEMORY;
+ }
+
+ pDecoder->pInputCache = ma_malloc((size_t)inputCacheCapSizeInBytes, &pDecoder->allocationCallbacks); /* Safe cast to size_t. */
+ if (pDecoder->pInputCache == NULL) {
+ ma_data_converter_uninit(&pDecoder->converter, &pDecoder->allocationCallbacks);
+ return MA_OUT_OF_MEMORY;
+ }
+ }
}
return MA_SUCCESS;
}
-static MA_INLINE void ma_peak2_process_pcm_frame_s16(ma_peak2* pFilter, ma_int16* pFrameOut, const ma_int16* pFrameIn)
-{
- ma_biquad_process_pcm_frame_s16(&pFilter->bq, pFrameOut, pFrameIn);
-}
-static MA_INLINE void ma_peak2_process_pcm_frame_f32(ma_peak2* pFilter, float* pFrameOut, const float* pFrameIn)
-{
- ma_biquad_process_pcm_frame_f32(&pFilter->bq, pFrameOut, pFrameIn);
-}
-MA_API ma_result ma_peak2_process_pcm_frames(ma_peak2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+static ma_result ma_decoder_internal_on_read__custom(void* pUserData, void* pBufferOut, size_t bytesToRead, size_t* pBytesRead)
{
- if (pFilter == NULL) {
- return MA_INVALID_ARGS;
- }
+ ma_decoder* pDecoder = (ma_decoder*)pUserData;
+ MA_ASSERT(pDecoder != NULL);
- return ma_biquad_process_pcm_frames(&pFilter->bq, pFramesOut, pFramesIn, frameCount);
+ return ma_decoder_read_bytes(pDecoder, pBufferOut, bytesToRead, pBytesRead);
}
-MA_API ma_uint32 ma_peak2_get_latency(const ma_peak2* pFilter)
+static ma_result ma_decoder_internal_on_seek__custom(void* pUserData, ma_int64 offset, ma_seek_origin origin)
{
- if (pFilter == NULL) {
- return 0;
- }
+ ma_decoder* pDecoder = (ma_decoder*)pUserData;
+ MA_ASSERT(pDecoder != NULL);
- return ma_biquad_get_latency(&pFilter->bq);
+ return ma_decoder_seek_bytes(pDecoder, offset, origin);
}
-
-/**************************************************************************************************************************************************************
-
-Low Shelf Filter
-
-**************************************************************************************************************************************************************/
-MA_API ma_loshelf2_config ma_loshelf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double shelfSlope, double frequency)
+static ma_result ma_decoder_internal_on_tell__custom(void* pUserData, ma_int64* pCursor)
{
- ma_loshelf2_config config;
-
- MA_ZERO_OBJECT(&config);
- config.format = format;
- config.channels = channels;
- config.sampleRate = sampleRate;
- config.gainDB = gainDB;
- config.shelfSlope = shelfSlope;
- config.frequency = frequency;
+ ma_decoder* pDecoder = (ma_decoder*)pUserData;
+ MA_ASSERT(pDecoder != NULL);
- return config;
+ return ma_decoder_tell_bytes(pDecoder, pCursor);
}
-static MA_INLINE ma_biquad_config ma_loshelf2__get_biquad_config(const ma_loshelf2_config* pConfig)
+static ma_result ma_decoder_init_from_vtable(const ma_decoding_backend_vtable* pVTable, void* pVTableUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
- ma_biquad_config bqConfig;
- double w;
- double s;
- double c;
- double A;
- double S;
- double a;
- double sqrtA;
+ ma_result result;
+ ma_decoding_backend_config backendConfig;
+ ma_data_source* pBackend;
- MA_ASSERT(pConfig != NULL);
+ MA_ASSERT(pVTable != NULL);
+ MA_ASSERT(pConfig != NULL);
+ MA_ASSERT(pDecoder != NULL);
- w = 2 * MA_PI_D * pConfig->frequency / pConfig->sampleRate;
- s = ma_sind(w);
- c = ma_cosd(w);
- A = ma_powd(10, (pConfig->gainDB / 40));
- S = pConfig->shelfSlope;
- a = s/2 * ma_sqrtd((A + 1/A) * (1/S - 1) + 2);
- sqrtA = 2*ma_sqrtd(A)*a;
+ if (pVTable->onInit == NULL) {
+ return MA_NOT_IMPLEMENTED;
+ }
- bqConfig.b0 = A * ((A + 1) - (A - 1)*c + sqrtA);
- bqConfig.b1 = 2 * A * ((A - 1) - (A + 1)*c);
- bqConfig.b2 = A * ((A + 1) - (A - 1)*c - sqrtA);
- bqConfig.a0 = (A + 1) + (A - 1)*c + sqrtA;
- bqConfig.a1 = -2 * ((A - 1) + (A + 1)*c);
- bqConfig.a2 = (A + 1) + (A - 1)*c - sqrtA;
+ backendConfig = ma_decoding_backend_config_init(pConfig->format, pConfig->seekPointCount);
- bqConfig.format = pConfig->format;
- bqConfig.channels = pConfig->channels;
+ result = pVTable->onInit(pVTableUserData, ma_decoder_internal_on_read__custom, ma_decoder_internal_on_seek__custom, ma_decoder_internal_on_tell__custom, pDecoder, &backendConfig, &pDecoder->allocationCallbacks, &pBackend);
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to initialize the backend from this vtable. */
+ }
- return bqConfig;
+ /* Getting here means we were able to initialize the backend so we can now initialize the decoder. */
+ pDecoder->pBackend = pBackend;
+ pDecoder->pBackendVTable = pVTable;
+ pDecoder->pBackendUserData = pConfig->pCustomBackendUserData;
+
+ return MA_SUCCESS;
}
-MA_API ma_result ma_loshelf2_init(const ma_loshelf2_config* pConfig, ma_loshelf2* pFilter)
-{
- ma_result result;
- ma_biquad_config bqConfig;
- if (pFilter == NULL) {
- return MA_INVALID_ARGS;
- }
- MA_ZERO_OBJECT(pFilter);
+static ma_result ma_decoder_init_custom__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+{
+ ma_result result = MA_NO_BACKEND;
+ size_t ivtable;
- if (pConfig == NULL) {
- return MA_INVALID_ARGS;
+ MA_ASSERT(pConfig != NULL);
+ MA_ASSERT(pDecoder != NULL);
+
+ if (pConfig->ppCustomBackendVTables == NULL) {
+ return MA_NO_BACKEND;
}
- bqConfig = ma_loshelf2__get_biquad_config(pConfig);
- result = ma_biquad_init(&bqConfig, &pFilter->bq);
- if (result != MA_SUCCESS) {
- return result;
+ /* The order each backend is listed is what defines the priority. */
+ for (ivtable = 0; ivtable < pConfig->customBackendCount; ivtable += 1) {
+ const ma_decoding_backend_vtable* pVTable = pConfig->ppCustomBackendVTables[ivtable];
+ if (pVTable != NULL && pVTable->onInit != NULL) {
+ result = ma_decoder_init_from_vtable(pVTable, pConfig->pCustomBackendUserData, pConfig, pDecoder);
+ if (result == MA_SUCCESS) {
+ return MA_SUCCESS;
+ } else {
+ /* Initialization failed. Move on to the next one, but seek back to the start first so the next vtable starts from the first byte of the file. */
+ result = ma_decoder_seek_bytes(pDecoder, 0, ma_seek_origin_start);
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to seek back to the start. */
+ }
+ }
+ } else {
+ /* No vtable. */
+ }
}
- return MA_SUCCESS;
+ /* Getting here means we couldn't find a backend. */
+ return MA_NO_BACKEND;
}
-MA_API ma_result ma_loshelf2_reinit(const ma_loshelf2_config* pConfig, ma_loshelf2* pFilter)
+
+/* WAV */
+#ifdef dr_wav_h
+#define MA_HAS_WAV
+
+typedef struct
{
- ma_result result;
- ma_biquad_config bqConfig;
+ ma_data_source_base ds;
+ ma_read_proc onRead;
+ ma_seek_proc onSeek;
+ ma_tell_proc onTell;
+ void* pReadSeekTellUserData;
+ ma_format format; /* Can be f32, s16 or s32. */
+#if !defined(MA_NO_WAV)
+ drwav dr;
+#endif
+} ma_wav;
- if (pFilter == NULL || pConfig == NULL) {
- return MA_INVALID_ARGS;
- }
+MA_API ma_result ma_wav_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav);
+MA_API ma_result ma_wav_init_file(const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav);
+MA_API ma_result ma_wav_init_file_w(const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav);
+MA_API ma_result ma_wav_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav);
+MA_API void ma_wav_uninit(ma_wav* pWav, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_wav_read_pcm_frames(ma_wav* pWav, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
+MA_API ma_result ma_wav_seek_to_pcm_frame(ma_wav* pWav, ma_uint64 frameIndex);
+MA_API ma_result ma_wav_get_data_format(ma_wav* pWav, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
+MA_API ma_result ma_wav_get_cursor_in_pcm_frames(ma_wav* pWav, ma_uint64* pCursor);
+MA_API ma_result ma_wav_get_length_in_pcm_frames(ma_wav* pWav, ma_uint64* pLength);
- bqConfig = ma_loshelf2__get_biquad_config(pConfig);
- result = ma_biquad_reinit(&bqConfig, &pFilter->bq);
- if (result != MA_SUCCESS) {
- return result;
- }
- return MA_SUCCESS;
+static ma_result ma_wav_ds_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+{
+ return ma_wav_read_pcm_frames((ma_wav*)pDataSource, pFramesOut, frameCount, pFramesRead);
}
-static MA_INLINE void ma_loshelf2_process_pcm_frame_s16(ma_loshelf2* pFilter, ma_int16* pFrameOut, const ma_int16* pFrameIn)
+static ma_result ma_wav_ds_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
- ma_biquad_process_pcm_frame_s16(&pFilter->bq, pFrameOut, pFrameIn);
+ return ma_wav_seek_to_pcm_frame((ma_wav*)pDataSource, frameIndex);
}
-static MA_INLINE void ma_loshelf2_process_pcm_frame_f32(ma_loshelf2* pFilter, float* pFrameOut, const float* pFrameIn)
+static ma_result ma_wav_ds_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
- ma_biquad_process_pcm_frame_f32(&pFilter->bq, pFrameOut, pFrameIn);
+ return ma_wav_get_data_format((ma_wav*)pDataSource, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
}
-MA_API ma_result ma_loshelf2_process_pcm_frames(ma_loshelf2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+static ma_result ma_wav_ds_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
{
- if (pFilter == NULL) {
- return MA_INVALID_ARGS;
- }
+ return ma_wav_get_cursor_in_pcm_frames((ma_wav*)pDataSource, pCursor);
+}
- return ma_biquad_process_pcm_frames(&pFilter->bq, pFramesOut, pFramesIn, frameCount);
+static ma_result ma_wav_ds_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
+{
+ return ma_wav_get_length_in_pcm_frames((ma_wav*)pDataSource, pLength);
}
-MA_API ma_uint32 ma_loshelf2_get_latency(const ma_loshelf2* pFilter)
+static ma_data_source_vtable g_ma_wav_ds_vtable =
{
- if (pFilter == NULL) {
- return 0;
- }
+ ma_wav_ds_read,
+ ma_wav_ds_seek,
+ ma_wav_ds_get_data_format,
+ ma_wav_ds_get_cursor,
+ ma_wav_ds_get_length,
+ NULL, /* onSetLooping */
+ 0
+};
- return ma_biquad_get_latency(&pFilter->bq);
-}
+#if !defined(MA_NO_WAV)
+static drwav_allocation_callbacks drwav_allocation_callbacks_from_miniaudio(const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ drwav_allocation_callbacks callbacks;
-/**************************************************************************************************************************************************************
+ if (pAllocationCallbacks != NULL) {
+ callbacks.onMalloc = pAllocationCallbacks->onMalloc;
+ callbacks.onRealloc = pAllocationCallbacks->onRealloc;
+ callbacks.onFree = pAllocationCallbacks->onFree;
+ callbacks.pUserData = pAllocationCallbacks->pUserData;
+ } else {
+ callbacks.onMalloc = ma__malloc_default;
+ callbacks.onRealloc = ma__realloc_default;
+ callbacks.onFree = ma__free_default;
+ callbacks.pUserData = NULL;
+ }
-High Shelf Filter
+ return callbacks;
+}
-**************************************************************************************************************************************************************/
-MA_API ma_hishelf2_config ma_hishelf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double shelfSlope, double frequency)
+static size_t ma_wav_dr_callback__read(void* pUserData, void* pBufferOut, size_t bytesToRead)
{
- ma_hishelf2_config config;
+ ma_wav* pWav = (ma_wav*)pUserData;
+ ma_result result;
+ size_t bytesRead;
- MA_ZERO_OBJECT(&config);
- config.format = format;
- config.channels = channels;
- config.sampleRate = sampleRate;
- config.gainDB = gainDB;
- config.shelfSlope = shelfSlope;
- config.frequency = frequency;
+ MA_ASSERT(pWav != NULL);
- return config;
-}
+ result = pWav->onRead(pWav->pReadSeekTellUserData, pBufferOut, bytesToRead, &bytesRead);
+ (void)result;
+ return bytesRead;
+}
-static MA_INLINE ma_biquad_config ma_hishelf2__get_biquad_config(const ma_hishelf2_config* pConfig)
+static drwav_bool32 ma_wav_dr_callback__seek(void* pUserData, int offset, drwav_seek_origin origin)
{
- ma_biquad_config bqConfig;
- double w;
- double s;
- double c;
- double A;
- double S;
- double a;
- double sqrtA;
-
- MA_ASSERT(pConfig != NULL);
+ ma_wav* pWav = (ma_wav*)pUserData;
+ ma_result result;
+ ma_seek_origin maSeekOrigin;
- w = 2 * MA_PI_D * pConfig->frequency / pConfig->sampleRate;
- s = ma_sind(w);
- c = ma_cosd(w);
- A = ma_powd(10, (pConfig->gainDB / 40));
- S = pConfig->shelfSlope;
- a = s/2 * ma_sqrtd((A + 1/A) * (1/S - 1) + 2);
- sqrtA = 2*ma_sqrtd(A)*a;
+ MA_ASSERT(pWav != NULL);
- bqConfig.b0 = A * ((A + 1) + (A - 1)*c + sqrtA);
- bqConfig.b1 = -2 * A * ((A - 1) + (A + 1)*c);
- bqConfig.b2 = A * ((A + 1) + (A - 1)*c - sqrtA);
- bqConfig.a0 = (A + 1) - (A - 1)*c + sqrtA;
- bqConfig.a1 = 2 * ((A - 1) - (A + 1)*c);
- bqConfig.a2 = (A + 1) - (A - 1)*c - sqrtA;
+ maSeekOrigin = ma_seek_origin_start;
+ if (origin == drwav_seek_origin_current) {
+ maSeekOrigin = ma_seek_origin_current;
+ }
- bqConfig.format = pConfig->format;
- bqConfig.channels = pConfig->channels;
+ result = pWav->onSeek(pWav->pReadSeekTellUserData, offset, maSeekOrigin);
+ if (result != MA_SUCCESS) {
+ return MA_FALSE;
+ }
- return bqConfig;
+ return MA_TRUE;
}
+#endif
-MA_API ma_result ma_hishelf2_init(const ma_hishelf2_config* pConfig, ma_hishelf2* pFilter)
+static ma_result ma_wav_init_internal(const ma_decoding_backend_config* pConfig, ma_wav* pWav)
{
ma_result result;
- ma_biquad_config bqConfig;
+ ma_data_source_config dataSourceConfig;
- if (pFilter == NULL) {
+ if (pWav == NULL) {
return MA_INVALID_ARGS;
}
- MA_ZERO_OBJECT(pFilter);
+ MA_ZERO_OBJECT(pWav);
+ pWav->format = ma_format_unknown; /* Use closest match to source file by default. */
- if (pConfig == NULL) {
- return MA_INVALID_ARGS;
+ if (pConfig != NULL && (pConfig->preferredFormat == ma_format_f32 || pConfig->preferredFormat == ma_format_s16 || pConfig->preferredFormat == ma_format_s32)) {
+ pWav->format = pConfig->preferredFormat;
+ } else {
+ /* Getting here means something other than f32 and s16 was specified. Just leave this unset to use the default format. */
}
- bqConfig = ma_hishelf2__get_biquad_config(pConfig);
- result = ma_biquad_init(&bqConfig, &pFilter->bq);
+ dataSourceConfig = ma_data_source_config_init();
+ dataSourceConfig.vtable = &g_ma_wav_ds_vtable;
+
+ result = ma_data_source_init(&dataSourceConfig, &pWav->ds);
if (result != MA_SUCCESS) {
- return result;
+ return result; /* Failed to initialize the base data source. */
}
return MA_SUCCESS;
}
-MA_API ma_result ma_hishelf2_reinit(const ma_hishelf2_config* pConfig, ma_hishelf2* pFilter)
+MA_API ma_result ma_wav_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav)
{
ma_result result;
- ma_biquad_config bqConfig;
-
- if (pFilter == NULL || pConfig == NULL) {
- return MA_INVALID_ARGS;
- }
- bqConfig = ma_hishelf2__get_biquad_config(pConfig);
- result = ma_biquad_reinit(&bqConfig, &pFilter->bq);
+ result = ma_wav_init_internal(pConfig, pWav);
if (result != MA_SUCCESS) {
return result;
}
- return MA_SUCCESS;
-}
-
-static MA_INLINE void ma_hishelf2_process_pcm_frame_s16(ma_hishelf2* pFilter, ma_int16* pFrameOut, const ma_int16* pFrameIn)
-{
- ma_biquad_process_pcm_frame_s16(&pFilter->bq, pFrameOut, pFrameIn);
-}
-
-static MA_INLINE void ma_hishelf2_process_pcm_frame_f32(ma_hishelf2* pFilter, float* pFrameOut, const float* pFrameIn)
-{
- ma_biquad_process_pcm_frame_f32(&pFilter->bq, pFrameOut, pFrameIn);
-}
-
-MA_API ma_result ma_hishelf2_process_pcm_frames(ma_hishelf2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
-{
- if (pFilter == NULL) {
- return MA_INVALID_ARGS;
+ if (onRead == NULL || onSeek == NULL) {
+ return MA_INVALID_ARGS; /* onRead and onSeek are mandatory. */
}
- return ma_biquad_process_pcm_frames(&pFilter->bq, pFramesOut, pFramesIn, frameCount);
-}
-
-MA_API ma_uint32 ma_hishelf2_get_latency(const ma_hishelf2* pFilter)
-{
- if (pFilter == NULL) {
- return 0;
- }
+ pWav->onRead = onRead;
+ pWav->onSeek = onSeek;
+ pWav->onTell = onTell;
+ pWav->pReadSeekTellUserData = pReadSeekTellUserData;
- return ma_biquad_get_latency(&pFilter->bq);
-}
+ #if !defined(MA_NO_WAV)
+ {
+ drwav_allocation_callbacks wavAllocationCallbacks = drwav_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
+ drwav_bool32 wavResult;
+ wavResult = drwav_init(&pWav->dr, ma_wav_dr_callback__read, ma_wav_dr_callback__seek, pWav, &wavAllocationCallbacks);
+ if (wavResult != MA_TRUE) {
+ return MA_INVALID_FILE;
+ }
+ /*
+ If an explicit format was not specified, try picking the closest match based on the internal
+ format. The format needs to be supported by miniaudio.
+ */
+ if (pWav->format == ma_format_unknown) {
+ switch (pWav->dr.translatedFormatTag)
+ {
+ case DR_WAVE_FORMAT_PCM:
+ {
+ if (pWav->dr.bitsPerSample == 8) {
+ pWav->format = ma_format_u8;
+ } else if (pWav->dr.bitsPerSample == 16) {
+ pWav->format = ma_format_s16;
+ } else if (pWav->dr.bitsPerSample == 24) {
+ pWav->format = ma_format_s24;
+ } else if (pWav->dr.bitsPerSample == 32) {
+ pWav->format = ma_format_s32;
+ }
+ } break;
-/**************************************************************************************************************************************************************
+ case DR_WAVE_FORMAT_IEEE_FLOAT:
+ {
+ if (pWav->dr.bitsPerSample == 32) {
+ pWav->format = ma_format_f32;
+ }
+ } break;
-Resampling
+ default: break;
+ }
-**************************************************************************************************************************************************************/
-MA_API ma_linear_resampler_config ma_linear_resampler_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut)
-{
- ma_linear_resampler_config config;
- MA_ZERO_OBJECT(&config);
- config.format = format;
- config.channels = channels;
- config.sampleRateIn = sampleRateIn;
- config.sampleRateOut = sampleRateOut;
- config.lpfOrder = ma_min(MA_DEFAULT_RESAMPLER_LPF_ORDER, MA_MAX_FILTER_ORDER);
- config.lpfNyquistFactor = 1;
+ /* Fall back to f32 if we couldn't find anything. */
+ if (pWav->format == ma_format_unknown) {
+ pWav->format = ma_format_f32;
+ }
+ }
- return config;
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ /* wav is disabled. */
+ (void)pAllocationCallbacks;
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
}
-static void ma_linear_resampler_adjust_timer_for_new_rate(ma_linear_resampler* pResampler, ma_uint32 oldSampleRateOut, ma_uint32 newSampleRateOut)
+MA_API ma_result ma_wav_init_file(const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav)
{
- /*
- So what's happening here? Basically we need to adjust the fractional component of the time advance based on the new rate. The old time advance will
- be based on the old sample rate, but we are needing to adjust it to that it's based on the new sample rate.
- */
- ma_uint32 oldRateTimeWhole = pResampler->inTimeFrac / oldSampleRateOut; /* <-- This should almost never be anything other than 0, but leaving it here to make this more general and robust just in case. */
- ma_uint32 oldRateTimeFract = pResampler->inTimeFrac % oldSampleRateOut;
+ ma_result result;
- pResampler->inTimeFrac =
- (oldRateTimeWhole * newSampleRateOut) +
- ((oldRateTimeFract * newSampleRateOut) / oldSampleRateOut);
+ result = ma_wav_init_internal(pConfig, pWav);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- /* Make sure the fractional part is less than the output sample rate. */
- pResampler->inTimeInt += pResampler->inTimeFrac / pResampler->config.sampleRateOut;
- pResampler->inTimeFrac = pResampler->inTimeFrac % pResampler->config.sampleRateOut;
+ #if !defined(MA_NO_WAV)
+ {
+ drwav_allocation_callbacks wavAllocationCallbacks = drwav_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
+ drwav_bool32 wavResult;
+
+ wavResult = drwav_init_file(&pWav->dr, pFilePath, &wavAllocationCallbacks);
+ if (wavResult != MA_TRUE) {
+ return MA_INVALID_FILE;
+ }
+
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ /* wav is disabled. */
+ (void)pFilePath;
+ (void)pAllocationCallbacks;
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
}
-static ma_result ma_linear_resampler_set_rate_internal(ma_linear_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut, ma_bool32 isResamplerAlreadyInitialized)
+MA_API ma_result ma_wav_init_file_w(const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav)
{
ma_result result;
- ma_uint32 gcf;
- ma_uint32 lpfSampleRate;
- double lpfCutoffFrequency;
- ma_lpf_config lpfConfig;
- ma_uint32 oldSampleRateOut; /* Required for adjusting time advance down the bottom. */
- if (pResampler == NULL) {
- return MA_INVALID_ARGS;
+ result = ma_wav_init_internal(pConfig, pWav);
+ if (result != MA_SUCCESS) {
+ return result;
}
- if (sampleRateIn == 0 || sampleRateOut == 0) {
- return MA_INVALID_ARGS;
- }
+ #if !defined(MA_NO_WAV)
+ {
+ drwav_allocation_callbacks wavAllocationCallbacks = drwav_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
+ drwav_bool32 wavResult;
- oldSampleRateOut = pResampler->config.sampleRateOut;
+ wavResult = drwav_init_file_w(&pWav->dr, pFilePath, &wavAllocationCallbacks);
+ if (wavResult != MA_TRUE) {
+ return MA_INVALID_FILE;
+ }
- pResampler->config.sampleRateIn = sampleRateIn;
- pResampler->config.sampleRateOut = sampleRateOut;
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ /* wav is disabled. */
+ (void)pFilePath;
+ (void)pAllocationCallbacks;
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
+}
- /* Simplify the sample rate. */
- gcf = ma_gcf_u32(pResampler->config.sampleRateIn, pResampler->config.sampleRateOut);
- pResampler->config.sampleRateIn /= gcf;
- pResampler->config.sampleRateOut /= gcf;
+MA_API ma_result ma_wav_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav)
+{
+ ma_result result;
- /* Always initialize the low-pass filter, even when the order is 0. */
- if (pResampler->config.lpfOrder > MA_MAX_FILTER_ORDER) {
- return MA_INVALID_ARGS;
+ result = ma_wav_init_internal(pConfig, pWav);
+ if (result != MA_SUCCESS) {
+ return result;
}
- lpfSampleRate = (ma_uint32)(ma_max(pResampler->config.sampleRateIn, pResampler->config.sampleRateOut));
- lpfCutoffFrequency = ( double)(ma_min(pResampler->config.sampleRateIn, pResampler->config.sampleRateOut) * 0.5 * pResampler->config.lpfNyquistFactor);
+ #if !defined(MA_NO_WAV)
+ {
+ drwav_allocation_callbacks wavAllocationCallbacks = drwav_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
+ drwav_bool32 wavResult;
- lpfConfig = ma_lpf_config_init(pResampler->config.format, pResampler->config.channels, lpfSampleRate, lpfCutoffFrequency, pResampler->config.lpfOrder);
+ wavResult = drwav_init_memory(&pWav->dr, pData, dataSize, &wavAllocationCallbacks);
+ if (wavResult != MA_TRUE) {
+ return MA_INVALID_FILE;
+ }
- /*
- If the resampler is alreay initialized we don't want to do a fresh initialization of the low-pass filter because it will result in the cached frames
- getting cleared. Instead we re-initialize the filter which will maintain any cached frames.
- */
- if (isResamplerAlreadyInitialized) {
- result = ma_lpf_reinit(&lpfConfig, &pResampler->lpf);
- } else {
- result = ma_lpf_init(&lpfConfig, &pResampler->lpf);
+ return MA_SUCCESS;
}
-
- if (result != MA_SUCCESS) {
- return result;
+ #else
+ {
+ /* wav is disabled. */
+ (void)pData;
+ (void)dataSize;
+ (void)pAllocationCallbacks;
+ return MA_NOT_IMPLEMENTED;
}
+ #endif
+}
+MA_API void ma_wav_uninit(ma_wav* pWav, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pWav == NULL) {
+ return;
+ }
- pResampler->inAdvanceInt = pResampler->config.sampleRateIn / pResampler->config.sampleRateOut;
- pResampler->inAdvanceFrac = pResampler->config.sampleRateIn % pResampler->config.sampleRateOut;
+ (void)pAllocationCallbacks;
- /* Our timer was based on the old rate. We need to adjust it so that it's based on the new rate. */
- ma_linear_resampler_adjust_timer_for_new_rate(pResampler, oldSampleRateOut, pResampler->config.sampleRateOut);
+ #if !defined(MA_NO_WAV)
+ {
+ drwav_uninit(&pWav->dr);
+ }
+ #else
+ {
+ /* wav is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
+ }
+ #endif
- return MA_SUCCESS;
+ ma_data_source_uninit(&pWav->ds);
}
-MA_API ma_result ma_linear_resampler_init(const ma_linear_resampler_config* pConfig, ma_linear_resampler* pResampler)
+MA_API ma_result ma_wav_read_pcm_frames(ma_wav* pWav, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
- ma_result result;
-
- if (pResampler == NULL) {
- return MA_INVALID_ARGS;
+ if (pFramesRead != NULL) {
+ *pFramesRead = 0;
}
- MA_ZERO_OBJECT(pResampler);
-
- if (pConfig == NULL) {
+ if (frameCount == 0) {
return MA_INVALID_ARGS;
}
- if (pConfig->channels < MA_MIN_CHANNELS || pConfig->channels > MA_MAX_CHANNELS) {
+ if (pWav == NULL) {
return MA_INVALID_ARGS;
}
- pResampler->config = *pConfig;
-
- /* Setting the rate will set up the filter and time advances for us. */
- result = ma_linear_resampler_set_rate_internal(pResampler, pConfig->sampleRateIn, pConfig->sampleRateOut, /* isResamplerAlreadyInitialized = */ MA_FALSE);
- if (result != MA_SUCCESS) {
- return result;
- }
+ #if !defined(MA_NO_WAV)
+ {
+ /* We always use floating point format. */
+ ma_result result = MA_SUCCESS; /* Must be initialized to MA_SUCCESS. */
+ ma_uint64 totalFramesRead = 0;
+ ma_format format;
- pResampler->inTimeInt = 1; /* Set this to one to force an input sample to always be loaded for the first output frame. */
- pResampler->inTimeFrac = 0;
+ ma_wav_get_data_format(pWav, &format, NULL, NULL, NULL, 0);
- return MA_SUCCESS;
-}
+ switch (format)
+ {
+ case ma_format_f32:
+ {
+ totalFramesRead = drwav_read_pcm_frames_f32(&pWav->dr, frameCount, (float*)pFramesOut);
+ } break;
-MA_API void ma_linear_resampler_uninit(ma_linear_resampler* pResampler)
-{
- if (pResampler == NULL) {
- return;
- }
-}
+ case ma_format_s16:
+ {
+ totalFramesRead = drwav_read_pcm_frames_s16(&pWav->dr, frameCount, (drwav_int16*)pFramesOut);
+ } break;
-static MA_INLINE ma_int16 ma_linear_resampler_mix_s16(ma_int16 x, ma_int16 y, ma_int32 a, const ma_int32 shift)
-{
- ma_int32 b;
- ma_int32 c;
- ma_int32 r;
+ case ma_format_s32:
+ {
+ totalFramesRead = drwav_read_pcm_frames_s32(&pWav->dr, frameCount, (drwav_int32*)pFramesOut);
+ } break;
- MA_ASSERT(a <= (1<dr, frameCount, pFramesOut);
+ } break;
+ }
- b = x * ((1<> shift);
-}
+ if (pFramesRead != NULL) {
+ *pFramesRead = totalFramesRead;
+ }
-static void ma_linear_resampler_interpolate_frame_s16(ma_linear_resampler* pResampler, ma_int16* MA_RESTRICT pFrameOut)
-{
- ma_uint32 c;
- ma_uint32 a;
- const ma_uint32 channels = pResampler->config.channels;
- const ma_uint32 shift = 12;
+ if (result == MA_SUCCESS && totalFramesRead == 0) {
+ result = MA_AT_END;
+ }
- MA_ASSERT(pResampler != NULL);
- MA_ASSERT(pFrameOut != NULL);
+ return result;
+ }
+ #else
+ {
+ /* wav is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
- a = (pResampler->inTimeFrac << shift) / pResampler->config.sampleRateOut;
+ (void)pFramesOut;
+ (void)frameCount;
+ (void)pFramesRead;
- MA_ASSUME(channels >= MA_MIN_CHANNELS && channels <= MA_MAX_CHANNELS);
- for (c = 0; c < channels; c += 1) {
- ma_int16 s = ma_linear_resampler_mix_s16(pResampler->x0.s16[c], pResampler->x1.s16[c], a, shift);
- pFrameOut[c] = s;
+ return MA_NOT_IMPLEMENTED;
}
+ #endif
}
-
-static void ma_linear_resampler_interpolate_frame_f32(ma_linear_resampler* pResampler, float* MA_RESTRICT pFrameOut)
+MA_API ma_result ma_wav_seek_to_pcm_frame(ma_wav* pWav, ma_uint64 frameIndex)
{
- ma_uint32 c;
- float a;
- const ma_uint32 channels = pResampler->config.channels;
+ if (pWav == NULL) {
+ return MA_INVALID_ARGS;
+ }
- MA_ASSERT(pResampler != NULL);
- MA_ASSERT(pFrameOut != NULL);
+ #if !defined(MA_NO_WAV)
+ {
+ drwav_bool32 wavResult;
- a = (float)pResampler->inTimeFrac / pResampler->config.sampleRateOut;
+ wavResult = drwav_seek_to_pcm_frame(&pWav->dr, frameIndex);
+ if (wavResult != DRWAV_TRUE) {
+ return MA_ERROR;
+ }
- MA_ASSUME(channels >= MA_MIN_CHANNELS && channels <= MA_MAX_CHANNELS);
- for (c = 0; c < channels; c += 1) {
- float s = ma_mix_f32_fast(pResampler->x0.f32[c], pResampler->x1.f32[c], a);
- pFrameOut[c] = s;
+ return MA_SUCCESS;
}
-}
-
-static ma_result ma_linear_resampler_process_pcm_frames_s16_downsample(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
-{
- const ma_int16* pFramesInS16;
- /* */ ma_int16* pFramesOutS16;
- ma_uint64 frameCountIn;
- ma_uint64 frameCountOut;
- ma_uint64 framesProcessedIn;
- ma_uint64 framesProcessedOut;
+ #else
+ {
+ /* wav is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
- MA_ASSERT(pResampler != NULL);
- MA_ASSERT(pFrameCountIn != NULL);
- MA_ASSERT(pFrameCountOut != NULL);
+ (void)frameIndex;
- pFramesInS16 = (const ma_int16*)pFramesIn;
- pFramesOutS16 = ( ma_int16*)pFramesOut;
- frameCountIn = *pFrameCountIn;
- frameCountOut = *pFrameCountOut;
- framesProcessedIn = 0;
- framesProcessedOut = 0;
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
+}
- while (framesProcessedOut < frameCountOut) {
- /* Before interpolating we need to load the buffers. When doing this we need to ensure we run every input sample through the filter. */
- while (pResampler->inTimeInt > 0 && frameCountIn > framesProcessedIn) {
- ma_uint32 iChannel;
+MA_API ma_result ma_wav_get_data_format(ma_wav* pWav, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
+{
+ /* Defaults for safety. */
+ if (pFormat != NULL) {
+ *pFormat = ma_format_unknown;
+ }
+ if (pChannels != NULL) {
+ *pChannels = 0;
+ }
+ if (pSampleRate != NULL) {
+ *pSampleRate = 0;
+ }
+ if (pChannelMap != NULL) {
+ MA_ZERO_MEMORY(pChannelMap, sizeof(*pChannelMap) * channelMapCap);
+ }
- if (pFramesInS16 != NULL) {
- for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
- pResampler->x0.s16[iChannel] = pResampler->x1.s16[iChannel];
- pResampler->x1.s16[iChannel] = pFramesInS16[iChannel];
- }
- pFramesInS16 += pResampler->config.channels;
- } else {
- for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
- pResampler->x0.s16[iChannel] = pResampler->x1.s16[iChannel];
- pResampler->x1.s16[iChannel] = 0;
- }
- }
+ if (pWav == NULL) {
+ return MA_INVALID_OPERATION;
+ }
- /* Filter. */
- ma_lpf_process_pcm_frame_s16(&pResampler->lpf, pResampler->x1.s16, pResampler->x1.s16);
+ if (pFormat != NULL) {
+ *pFormat = pWav->format;
+ }
- framesProcessedIn += 1;
- pResampler->inTimeInt -= 1;
+ #if !defined(MA_NO_WAV)
+ {
+ if (pChannels != NULL) {
+ *pChannels = pWav->dr.channels;
}
- if (pResampler->inTimeInt > 0) {
- break; /* Ran out of input data. */
+ if (pSampleRate != NULL) {
+ *pSampleRate = pWav->dr.sampleRate;
}
- /* Getting here means the frames have been loaded and filtered and we can generate the next output frame. */
- if (pFramesOutS16 != NULL) {
- MA_ASSERT(pResampler->inTimeInt == 0);
- ma_linear_resampler_interpolate_frame_s16(pResampler, pFramesOutS16);
-
- pFramesOutS16 += pResampler->config.channels;
+ if (pChannelMap != NULL) {
+ ma_channel_map_init_standard(ma_standard_channel_map_microsoft, pChannelMap, channelMapCap, pWav->dr.channels);
}
- framesProcessedOut += 1;
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ /* wav is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
+}
- /* Advance time forward. */
- pResampler->inTimeInt += pResampler->inAdvanceInt;
- pResampler->inTimeFrac += pResampler->inAdvanceFrac;
- if (pResampler->inTimeFrac >= pResampler->config.sampleRateOut) {
- pResampler->inTimeFrac -= pResampler->config.sampleRateOut;
- pResampler->inTimeInt += 1;
- }
+MA_API ma_result ma_wav_get_cursor_in_pcm_frames(ma_wav* pWav, ma_uint64* pCursor)
+{
+ if (pCursor == NULL) {
+ return MA_INVALID_ARGS;
}
- *pFrameCountIn = framesProcessedIn;
- *pFrameCountOut = framesProcessedOut;
+ *pCursor = 0; /* Safety. */
- return MA_SUCCESS;
-}
+ if (pWav == NULL) {
+ return MA_INVALID_ARGS;
+ }
-static ma_result ma_linear_resampler_process_pcm_frames_s16_upsample(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
-{
- const ma_int16* pFramesInS16;
- /* */ ma_int16* pFramesOutS16;
- ma_uint64 frameCountIn;
- ma_uint64 frameCountOut;
- ma_uint64 framesProcessedIn;
- ma_uint64 framesProcessedOut;
+ #if !defined(MA_NO_WAV)
+ {
+ drwav_result wavResult = drwav_get_cursor_in_pcm_frames(&pWav->dr, pCursor);
+ if (wavResult != DRWAV_SUCCESS) {
+ return (ma_result)wavResult; /* dr_wav result codes map to miniaudio's. */
+ }
- MA_ASSERT(pResampler != NULL);
- MA_ASSERT(pFrameCountIn != NULL);
- MA_ASSERT(pFrameCountOut != NULL);
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ /* wav is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
+}
- pFramesInS16 = (const ma_int16*)pFramesIn;
- pFramesOutS16 = ( ma_int16*)pFramesOut;
- frameCountIn = *pFrameCountIn;
- frameCountOut = *pFrameCountOut;
- framesProcessedIn = 0;
- framesProcessedOut = 0;
+MA_API ma_result ma_wav_get_length_in_pcm_frames(ma_wav* pWav, ma_uint64* pLength)
+{
+ if (pLength == NULL) {
+ return MA_INVALID_ARGS;
+ }
- while (framesProcessedOut < frameCountOut) {
- /* Before interpolating we need to load the buffers. */
- while (pResampler->inTimeInt > 0 && frameCountIn > framesProcessedIn) {
- ma_uint32 iChannel;
+ *pLength = 0; /* Safety. */
- if (pFramesInS16 != NULL) {
- for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
- pResampler->x0.s16[iChannel] = pResampler->x1.s16[iChannel];
- pResampler->x1.s16[iChannel] = pFramesInS16[iChannel];
- }
- pFramesInS16 += pResampler->config.channels;
- } else {
- for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
- pResampler->x0.s16[iChannel] = pResampler->x1.s16[iChannel];
- pResampler->x1.s16[iChannel] = 0;
- }
- }
+ if (pWav == NULL) {
+ return MA_INVALID_ARGS;
+ }
- framesProcessedIn += 1;
- pResampler->inTimeInt -= 1;
+ #if !defined(MA_NO_WAV)
+ {
+ drwav_result wavResult = drwav_get_length_in_pcm_frames(&pWav->dr, pLength);
+ if (wavResult != DRWAV_SUCCESS) {
+ return (ma_result)wavResult; /* dr_wav result codes map to miniaudio's. */
}
- if (pResampler->inTimeInt > 0) {
- break; /* Ran out of input data. */
- }
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ /* wav is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
+}
- /* Getting here means the frames have been loaded and we can generate the next output frame. */
- if (pFramesOutS16 != NULL) {
- MA_ASSERT(pResampler->inTimeInt == 0);
- ma_linear_resampler_interpolate_frame_s16(pResampler, pFramesOutS16);
- /* Filter. */
- ma_lpf_process_pcm_frame_s16(&pResampler->lpf, pFramesOutS16, pFramesOutS16);
+static ma_result ma_decoding_backend_init__wav(void* pUserData, ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
+{
+ ma_result result;
+ ma_wav* pWav;
- pFramesOutS16 += pResampler->config.channels;
- }
+ (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
- framesProcessedOut += 1;
+ /* For now we're just allocating the decoder backend on the heap. */
+ pWav = (ma_wav*)ma_malloc(sizeof(*pWav), pAllocationCallbacks);
+ if (pWav == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
- /* Advance time forward. */
- pResampler->inTimeInt += pResampler->inAdvanceInt;
- pResampler->inTimeFrac += pResampler->inAdvanceFrac;
- if (pResampler->inTimeFrac >= pResampler->config.sampleRateOut) {
- pResampler->inTimeFrac -= pResampler->config.sampleRateOut;
- pResampler->inTimeInt += 1;
- }
+ result = ma_wav_init(onRead, onSeek, onTell, pReadSeekTellUserData, pConfig, pAllocationCallbacks, pWav);
+ if (result != MA_SUCCESS) {
+ ma_free(pWav, pAllocationCallbacks);
+ return result;
}
- *pFrameCountIn = framesProcessedIn;
- *pFrameCountOut = framesProcessedOut;
+ *ppBackend = pWav;
return MA_SUCCESS;
}
-static ma_result ma_linear_resampler_process_pcm_frames_s16(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+static ma_result ma_decoding_backend_init_file__wav(void* pUserData, const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
{
- MA_ASSERT(pResampler != NULL);
+ ma_result result;
+ ma_wav* pWav;
- if (pResampler->config.sampleRateIn > pResampler->config.sampleRateOut) {
- return ma_linear_resampler_process_pcm_frames_s16_downsample(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
- } else {
- return ma_linear_resampler_process_pcm_frames_s16_upsample(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
+ (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
+
+ /* For now we're just allocating the decoder backend on the heap. */
+ pWav = (ma_wav*)ma_malloc(sizeof(*pWav), pAllocationCallbacks);
+ if (pWav == NULL) {
+ return MA_OUT_OF_MEMORY;
}
-}
+ result = ma_wav_init_file(pFilePath, pConfig, pAllocationCallbacks, pWav);
+ if (result != MA_SUCCESS) {
+ ma_free(pWav, pAllocationCallbacks);
+ return result;
+ }
-static ma_result ma_linear_resampler_process_pcm_frames_f32_downsample(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
-{
- const float* pFramesInF32;
- /* */ float* pFramesOutF32;
- ma_uint64 frameCountIn;
- ma_uint64 frameCountOut;
- ma_uint64 framesProcessedIn;
- ma_uint64 framesProcessedOut;
+ *ppBackend = pWav;
- MA_ASSERT(pResampler != NULL);
- MA_ASSERT(pFrameCountIn != NULL);
- MA_ASSERT(pFrameCountOut != NULL);
+ return MA_SUCCESS;
+}
- pFramesInF32 = (const float*)pFramesIn;
- pFramesOutF32 = ( float*)pFramesOut;
- frameCountIn = *pFrameCountIn;
- frameCountOut = *pFrameCountOut;
- framesProcessedIn = 0;
- framesProcessedOut = 0;
+static ma_result ma_decoding_backend_init_file_w__wav(void* pUserData, const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
+{
+ ma_result result;
+ ma_wav* pWav;
- while (framesProcessedOut < frameCountOut) {
- /* Before interpolating we need to load the buffers. When doing this we need to ensure we run every input sample through the filter. */
- while (pResampler->inTimeInt > 0 && frameCountIn > framesProcessedIn) {
- ma_uint32 iChannel;
+ (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
- if (pFramesInF32 != NULL) {
- for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
- pResampler->x0.f32[iChannel] = pResampler->x1.f32[iChannel];
- pResampler->x1.f32[iChannel] = pFramesInF32[iChannel];
- }
- pFramesInF32 += pResampler->config.channels;
- } else {
- for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
- pResampler->x0.f32[iChannel] = pResampler->x1.f32[iChannel];
- pResampler->x1.f32[iChannel] = 0;
- }
- }
+ /* For now we're just allocating the decoder backend on the heap. */
+ pWav = (ma_wav*)ma_malloc(sizeof(*pWav), pAllocationCallbacks);
+ if (pWav == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
- /* Filter. */
- ma_lpf_process_pcm_frame_f32(&pResampler->lpf, pResampler->x1.f32, pResampler->x1.f32);
+ result = ma_wav_init_file_w(pFilePath, pConfig, pAllocationCallbacks, pWav);
+ if (result != MA_SUCCESS) {
+ ma_free(pWav, pAllocationCallbacks);
+ return result;
+ }
- framesProcessedIn += 1;
- pResampler->inTimeInt -= 1;
- }
+ *ppBackend = pWav;
- if (pResampler->inTimeInt > 0) {
- break; /* Ran out of input data. */
- }
+ return MA_SUCCESS;
+}
- /* Getting here means the frames have been loaded and filtered and we can generate the next output frame. */
- if (pFramesOutF32 != NULL) {
- MA_ASSERT(pResampler->inTimeInt == 0);
- ma_linear_resampler_interpolate_frame_f32(pResampler, pFramesOutF32);
+static ma_result ma_decoding_backend_init_memory__wav(void* pUserData, const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
+{
+ ma_result result;
+ ma_wav* pWav;
- pFramesOutF32 += pResampler->config.channels;
- }
+ (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
- framesProcessedOut += 1;
+ /* For now we're just allocating the decoder backend on the heap. */
+ pWav = (ma_wav*)ma_malloc(sizeof(*pWav), pAllocationCallbacks);
+ if (pWav == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
- /* Advance time forward. */
- pResampler->inTimeInt += pResampler->inAdvanceInt;
- pResampler->inTimeFrac += pResampler->inAdvanceFrac;
- if (pResampler->inTimeFrac >= pResampler->config.sampleRateOut) {
- pResampler->inTimeFrac -= pResampler->config.sampleRateOut;
- pResampler->inTimeInt += 1;
- }
+ result = ma_wav_init_memory(pData, dataSize, pConfig, pAllocationCallbacks, pWav);
+ if (result != MA_SUCCESS) {
+ ma_free(pWav, pAllocationCallbacks);
+ return result;
}
- *pFrameCountIn = framesProcessedIn;
- *pFrameCountOut = framesProcessedOut;
+ *ppBackend = pWav;
return MA_SUCCESS;
}
-static ma_result ma_linear_resampler_process_pcm_frames_f32_upsample(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+static void ma_decoding_backend_uninit__wav(void* pUserData, ma_data_source* pBackend, const ma_allocation_callbacks* pAllocationCallbacks)
{
- const float* pFramesInF32;
- /* */ float* pFramesOutF32;
- ma_uint64 frameCountIn;
- ma_uint64 frameCountOut;
- ma_uint64 framesProcessedIn;
- ma_uint64 framesProcessedOut;
-
- MA_ASSERT(pResampler != NULL);
- MA_ASSERT(pFrameCountIn != NULL);
- MA_ASSERT(pFrameCountOut != NULL);
-
- pFramesInF32 = (const float*)pFramesIn;
- pFramesOutF32 = ( float*)pFramesOut;
- frameCountIn = *pFrameCountIn;
- frameCountOut = *pFrameCountOut;
- framesProcessedIn = 0;
- framesProcessedOut = 0;
+ ma_wav* pWav = (ma_wav*)pBackend;
- while (framesProcessedOut < frameCountOut) {
- /* Before interpolating we need to load the buffers. */
- while (pResampler->inTimeInt > 0 && frameCountIn > framesProcessedIn) {
- ma_uint32 iChannel;
+ (void)pUserData;
- if (pFramesInF32 != NULL) {
- for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
- pResampler->x0.f32[iChannel] = pResampler->x1.f32[iChannel];
- pResampler->x1.f32[iChannel] = pFramesInF32[iChannel];
- }
- pFramesInF32 += pResampler->config.channels;
- } else {
- for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
- pResampler->x0.f32[iChannel] = pResampler->x1.f32[iChannel];
- pResampler->x1.f32[iChannel] = 0;
- }
- }
+ ma_wav_uninit(pWav, pAllocationCallbacks);
+ ma_free(pWav, pAllocationCallbacks);
+}
- framesProcessedIn += 1;
- pResampler->inTimeInt -= 1;
- }
+static ma_decoding_backend_vtable g_ma_decoding_backend_vtable_wav =
+{
+ ma_decoding_backend_init__wav,
+ ma_decoding_backend_init_file__wav,
+ ma_decoding_backend_init_file_w__wav,
+ ma_decoding_backend_init_memory__wav,
+ ma_decoding_backend_uninit__wav
+};
- if (pResampler->inTimeInt > 0) {
- break; /* Ran out of input data. */
- }
+static ma_result ma_decoder_init_wav__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+{
+ return ma_decoder_init_from_vtable(&g_ma_decoding_backend_vtable_wav, NULL, pConfig, pDecoder);
+}
+#endif /* dr_wav_h */
- /* Getting here means the frames have been loaded and we can generate the next output frame. */
- if (pFramesOutF32 != NULL) {
- MA_ASSERT(pResampler->inTimeInt == 0);
- ma_linear_resampler_interpolate_frame_f32(pResampler, pFramesOutF32);
+/* FLAC */
+#ifdef dr_flac_h
+#define MA_HAS_FLAC
- /* Filter. */
- ma_lpf_process_pcm_frame_f32(&pResampler->lpf, pFramesOutF32, pFramesOutF32);
+typedef struct
+{
+ ma_data_source_base ds;
+ ma_read_proc onRead;
+ ma_seek_proc onSeek;
+ ma_tell_proc onTell;
+ void* pReadSeekTellUserData;
+ ma_format format; /* Can be f32, s16 or s32. */
+#if !defined(MA_NO_FLAC)
+ drflac* dr;
+#endif
+} ma_flac;
- pFramesOutF32 += pResampler->config.channels;
- }
+MA_API ma_result ma_flac_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac);
+MA_API ma_result ma_flac_init_file(const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac);
+MA_API ma_result ma_flac_init_file_w(const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac);
+MA_API ma_result ma_flac_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac);
+MA_API void ma_flac_uninit(ma_flac* pFlac, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_flac_read_pcm_frames(ma_flac* pFlac, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
+MA_API ma_result ma_flac_seek_to_pcm_frame(ma_flac* pFlac, ma_uint64 frameIndex);
+MA_API ma_result ma_flac_get_data_format(ma_flac* pFlac, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
+MA_API ma_result ma_flac_get_cursor_in_pcm_frames(ma_flac* pFlac, ma_uint64* pCursor);
+MA_API ma_result ma_flac_get_length_in_pcm_frames(ma_flac* pFlac, ma_uint64* pLength);
- framesProcessedOut += 1;
- /* Advance time forward. */
- pResampler->inTimeInt += pResampler->inAdvanceInt;
- pResampler->inTimeFrac += pResampler->inAdvanceFrac;
- if (pResampler->inTimeFrac >= pResampler->config.sampleRateOut) {
- pResampler->inTimeFrac -= pResampler->config.sampleRateOut;
- pResampler->inTimeInt += 1;
- }
- }
+static ma_result ma_flac_ds_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+{
+ return ma_flac_read_pcm_frames((ma_flac*)pDataSource, pFramesOut, frameCount, pFramesRead);
+}
- *pFrameCountIn = framesProcessedIn;
- *pFrameCountOut = framesProcessedOut;
+static ma_result ma_flac_ds_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
+{
+ return ma_flac_seek_to_pcm_frame((ma_flac*)pDataSource, frameIndex);
+}
- return MA_SUCCESS;
+static ma_result ma_flac_ds_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
+{
+ return ma_flac_get_data_format((ma_flac*)pDataSource, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
}
-static ma_result ma_linear_resampler_process_pcm_frames_f32(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+static ma_result ma_flac_ds_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
{
- MA_ASSERT(pResampler != NULL);
+ return ma_flac_get_cursor_in_pcm_frames((ma_flac*)pDataSource, pCursor);
+}
- if (pResampler->config.sampleRateIn > pResampler->config.sampleRateOut) {
- return ma_linear_resampler_process_pcm_frames_f32_downsample(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
- } else {
- return ma_linear_resampler_process_pcm_frames_f32_upsample(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
- }
+static ma_result ma_flac_ds_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
+{
+ return ma_flac_get_length_in_pcm_frames((ma_flac*)pDataSource, pLength);
}
+static ma_data_source_vtable g_ma_flac_ds_vtable =
+{
+ ma_flac_ds_read,
+ ma_flac_ds_seek,
+ ma_flac_ds_get_data_format,
+ ma_flac_ds_get_cursor,
+ ma_flac_ds_get_length,
+ NULL, /* onSetLooping */
+ 0
+};
+
-MA_API ma_result ma_linear_resampler_process_pcm_frames(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+#if !defined(MA_NO_FLAC)
+static drflac_allocation_callbacks drflac_allocation_callbacks_from_miniaudio(const ma_allocation_callbacks* pAllocationCallbacks)
{
- if (pResampler == NULL) {
- return MA_INVALID_ARGS;
- }
+ drflac_allocation_callbacks callbacks;
- /* */ if (pResampler->config.format == ma_format_s16) {
- return ma_linear_resampler_process_pcm_frames_s16(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
- } else if (pResampler->config.format == ma_format_f32) {
- return ma_linear_resampler_process_pcm_frames_f32(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
+ if (pAllocationCallbacks != NULL) {
+ callbacks.onMalloc = pAllocationCallbacks->onMalloc;
+ callbacks.onRealloc = pAllocationCallbacks->onRealloc;
+ callbacks.onFree = pAllocationCallbacks->onFree;
+ callbacks.pUserData = pAllocationCallbacks->pUserData;
} else {
- /* Should never get here. Getting here means the format is not supported and you didn't check the return value of ma_linear_resampler_init(). */
- MA_ASSERT(MA_FALSE);
- return MA_INVALID_ARGS;
+ callbacks.onMalloc = ma__malloc_default;
+ callbacks.onRealloc = ma__realloc_default;
+ callbacks.onFree = ma__free_default;
+ callbacks.pUserData = NULL;
}
-}
+ return callbacks;
+}
-MA_API ma_result ma_linear_resampler_set_rate(ma_linear_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut)
+static size_t ma_flac_dr_callback__read(void* pUserData, void* pBufferOut, size_t bytesToRead)
{
- return ma_linear_resampler_set_rate_internal(pResampler, sampleRateIn, sampleRateOut, /* isResamplerAlreadyInitialized = */ MA_TRUE);
+ ma_flac* pFlac = (ma_flac*)pUserData;
+ ma_result result;
+ size_t bytesRead;
+
+ MA_ASSERT(pFlac != NULL);
+
+ result = pFlac->onRead(pFlac->pReadSeekTellUserData, pBufferOut, bytesToRead, &bytesRead);
+ (void)result;
+
+ return bytesRead;
}
-MA_API ma_result ma_linear_resampler_set_rate_ratio(ma_linear_resampler* pResampler, float ratioInOut)
+static drflac_bool32 ma_flac_dr_callback__seek(void* pUserData, int offset, drflac_seek_origin origin)
{
- ma_uint32 n;
- ma_uint32 d;
+ ma_flac* pFlac = (ma_flac*)pUserData;
+ ma_result result;
+ ma_seek_origin maSeekOrigin;
- d = 1000;
- n = (ma_uint32)(ratioInOut * d);
+ MA_ASSERT(pFlac != NULL);
- if (n == 0) {
- return MA_INVALID_ARGS; /* Ratio too small. */
+ maSeekOrigin = ma_seek_origin_start;
+ if (origin == drflac_seek_origin_current) {
+ maSeekOrigin = ma_seek_origin_current;
}
- MA_ASSERT(n != 0);
+ result = pFlac->onSeek(pFlac->pReadSeekTellUserData, offset, maSeekOrigin);
+ if (result != MA_SUCCESS) {
+ return MA_FALSE;
+ }
- return ma_linear_resampler_set_rate(pResampler, n, d);
+ return MA_TRUE;
}
+#endif
-
-MA_API ma_uint64 ma_linear_resampler_get_required_input_frame_count(const ma_linear_resampler* pResampler, ma_uint64 outputFrameCount)
+static ma_result ma_flac_init_internal(const ma_decoding_backend_config* pConfig, ma_flac* pFlac)
{
- ma_uint64 inputFrameCount;
+ ma_result result;
+ ma_data_source_config dataSourceConfig;
- if (pResampler == NULL) {
- return 0;
+ if (pFlac == NULL) {
+ return MA_INVALID_ARGS;
}
- if (outputFrameCount == 0) {
- return 0;
+ MA_ZERO_OBJECT(pFlac);
+ pFlac->format = ma_format_f32; /* f32 by default. */
+
+ if (pConfig != NULL && (pConfig->preferredFormat == ma_format_f32 || pConfig->preferredFormat == ma_format_s16 || pConfig->preferredFormat == ma_format_s32)) {
+ pFlac->format = pConfig->preferredFormat;
+ } else {
+ /* Getting here means something other than f32 and s16 was specified. Just leave this unset to use the default format. */
}
- /* Any whole input frames are consumed before the first output frame is generated. */
- inputFrameCount = pResampler->inTimeInt;
- outputFrameCount -= 1;
+ dataSourceConfig = ma_data_source_config_init();
+ dataSourceConfig.vtable = &g_ma_flac_ds_vtable;
- /* The rest of the output frames can be calculated in constant time. */
- inputFrameCount += outputFrameCount * pResampler->inAdvanceInt;
- inputFrameCount += (pResampler->inTimeFrac + (outputFrameCount * pResampler->inAdvanceFrac)) / pResampler->config.sampleRateOut;
+ result = ma_data_source_init(&dataSourceConfig, &pFlac->ds);
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to initialize the base data source. */
+ }
- return inputFrameCount;
+ return MA_SUCCESS;
}
-MA_API ma_uint64 ma_linear_resampler_get_expected_output_frame_count(const ma_linear_resampler* pResampler, ma_uint64 inputFrameCount)
+MA_API ma_result ma_flac_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac)
{
- ma_uint64 outputFrameCount;
- ma_uint64 preliminaryInputFrameCountFromFrac;
- ma_uint64 preliminaryInputFrameCount;
+ ma_result result;
- if (pResampler == NULL) {
- return 0;
+ result = ma_flac_init_internal(pConfig, pFlac);
+ if (result != MA_SUCCESS) {
+ return result;
}
- /*
- The first step is to get a preliminary output frame count. This will either be exactly equal to what we need, or less by 1. We need to
- determine how many input frames will be consumed by this value. If it's greater than our original input frame count it means we won't
- be able to generate an extra frame because we will have run out of input data. Otherwise we will have enough input for the generation
- of an extra output frame. This add-by-one logic is necessary due to how the data loading logic works when processing frames.
- */
- outputFrameCount = (inputFrameCount * pResampler->config.sampleRateOut) / pResampler->config.sampleRateIn;
+ if (onRead == NULL || onSeek == NULL) {
+ return MA_INVALID_ARGS; /* onRead and onSeek are mandatory. */
+ }
- /*
- We need to determine how many *whole* input frames will have been processed to generate our preliminary output frame count. This is
- used in the logic below to determine whether or not we need to add an extra output frame.
- */
- preliminaryInputFrameCountFromFrac = (pResampler->inTimeFrac + outputFrameCount*pResampler->inAdvanceFrac) / pResampler->config.sampleRateOut;
- preliminaryInputFrameCount = (pResampler->inTimeInt + outputFrameCount*pResampler->inAdvanceInt ) + preliminaryInputFrameCountFromFrac;
+ pFlac->onRead = onRead;
+ pFlac->onSeek = onSeek;
+ pFlac->onTell = onTell;
+ pFlac->pReadSeekTellUserData = pReadSeekTellUserData;
- /*
- If the total number of *whole* input frames that would be required to generate our preliminary output frame count is greather than
- the amount of whole input frames we have available as input we need to *not* add an extra output frame as there won't be enough data
- to actually process. Otherwise we need to add the extra output frame.
- */
- if (preliminaryInputFrameCount <= inputFrameCount) {
- outputFrameCount += 1;
- }
+ #if !defined(MA_NO_FLAC)
+ {
+ drflac_allocation_callbacks flacAllocationCallbacks = drflac_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
- return outputFrameCount;
-}
+ pFlac->dr = drflac_open(ma_flac_dr_callback__read, ma_flac_dr_callback__seek, pFlac, &flacAllocationCallbacks);
+ if (pFlac->dr == NULL) {
+ return MA_INVALID_FILE;
+ }
-MA_API ma_uint64 ma_linear_resampler_get_input_latency(const ma_linear_resampler* pResampler)
-{
- if (pResampler == NULL) {
- return 0;
+ return MA_SUCCESS;
}
-
- return 1 + ma_lpf_get_latency(&pResampler->lpf);
+ #else
+ {
+ /* flac is disabled. */
+ (void)pAllocationCallbacks;
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
}
-MA_API ma_uint64 ma_linear_resampler_get_output_latency(const ma_linear_resampler* pResampler)
+MA_API ma_result ma_flac_init_file(const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac)
{
- if (pResampler == NULL) {
- return 0;
- }
+ ma_result result;
- return ma_linear_resampler_get_input_latency(pResampler) * pResampler->config.sampleRateOut / pResampler->config.sampleRateIn;
-}
+ result = ma_flac_init_internal(pConfig, pFlac);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ #if !defined(MA_NO_FLAC)
+ {
+ drflac_allocation_callbacks flacAllocationCallbacks = drflac_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
-#if defined(ma_speex_resampler_h)
-#define MA_HAS_SPEEX_RESAMPLER
+ pFlac->dr = drflac_open_file(pFilePath, &flacAllocationCallbacks);
+ if (pFlac->dr == NULL) {
+ return MA_INVALID_FILE;
+ }
-static ma_result ma_result_from_speex_err(int err)
-{
- switch (err)
+ return MA_SUCCESS;
+ }
+ #else
{
- case RESAMPLER_ERR_SUCCESS: return MA_SUCCESS;
- case RESAMPLER_ERR_ALLOC_FAILED: return MA_OUT_OF_MEMORY;
- case RESAMPLER_ERR_BAD_STATE: return MA_ERROR;
- case RESAMPLER_ERR_INVALID_ARG: return MA_INVALID_ARGS;
- case RESAMPLER_ERR_PTR_OVERLAP: return MA_INVALID_ARGS;
- case RESAMPLER_ERR_OVERFLOW: return MA_ERROR;
- default: return MA_ERROR;
+ /* flac is disabled. */
+ (void)pFilePath;
+ (void)pAllocationCallbacks;
+ return MA_NOT_IMPLEMENTED;
}
+ #endif
}
-#endif /* ma_speex_resampler_h */
-MA_API ma_resampler_config ma_resampler_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut, ma_resample_algorithm algorithm)
+MA_API ma_result ma_flac_init_file_w(const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac)
{
- ma_resampler_config config;
+ ma_result result;
- MA_ZERO_OBJECT(&config);
- config.format = format;
- config.channels = channels;
- config.sampleRateIn = sampleRateIn;
- config.sampleRateOut = sampleRateOut;
- config.algorithm = algorithm;
+ result = ma_flac_init_internal(pConfig, pFlac);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- /* Linear. */
- config.linear.lpfOrder = ma_min(MA_DEFAULT_RESAMPLER_LPF_ORDER, MA_MAX_FILTER_ORDER);
- config.linear.lpfNyquistFactor = 1;
+ #if !defined(MA_NO_FLAC)
+ {
+ drflac_allocation_callbacks flacAllocationCallbacks = drflac_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
- /* Speex. */
- config.speex.quality = 3; /* Cannot leave this as 0 as that is actually a valid value for Speex resampling quality. */
+ pFlac->dr = drflac_open_file_w(pFilePath, &flacAllocationCallbacks);
+ if (pFlac->dr == NULL) {
+ return MA_INVALID_FILE;
+ }
- return config;
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ /* flac is disabled. */
+ (void)pFilePath;
+ (void)pAllocationCallbacks;
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
}
-MA_API ma_result ma_resampler_init(const ma_resampler_config* pConfig, ma_resampler* pResampler)
+MA_API ma_result ma_flac_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac)
{
ma_result result;
- if (pResampler == NULL) {
- return MA_INVALID_ARGS;
+ result = ma_flac_init_internal(pConfig, pFlac);
+ if (result != MA_SUCCESS) {
+ return result;
}
- MA_ZERO_OBJECT(pResampler);
+ #if !defined(MA_NO_FLAC)
+ {
+ drflac_allocation_callbacks flacAllocationCallbacks = drflac_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
- if (pConfig == NULL) {
- return MA_INVALID_ARGS;
- }
+ pFlac->dr = drflac_open_memory(pData, dataSize, &flacAllocationCallbacks);
+ if (pFlac->dr == NULL) {
+ return MA_INVALID_FILE;
+ }
- if (pConfig->format != ma_format_f32 && pConfig->format != ma_format_s16) {
- return MA_INVALID_ARGS;
+ return MA_SUCCESS;
}
-
- pResampler->config = *pConfig;
-
- switch (pConfig->algorithm)
+ #else
{
- case ma_resample_algorithm_linear:
- {
- ma_linear_resampler_config linearConfig;
- linearConfig = ma_linear_resampler_config_init(pConfig->format, pConfig->channels, pConfig->sampleRateIn, pConfig->sampleRateOut);
- linearConfig.lpfOrder = pConfig->linear.lpfOrder;
- linearConfig.lpfNyquistFactor = pConfig->linear.lpfNyquistFactor;
+ /* flac is disabled. */
+ (void)pData;
+ (void)dataSize;
+ (void)pAllocationCallbacks;
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
+}
- result = ma_linear_resampler_init(&linearConfig, &pResampler->state.linear);
- if (result != MA_SUCCESS) {
- return result;
- }
- } break;
+MA_API void ma_flac_uninit(ma_flac* pFlac, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pFlac == NULL) {
+ return;
+ }
- case ma_resample_algorithm_speex:
- {
- #if defined(MA_HAS_SPEEX_RESAMPLER)
- int speexErr;
- pResampler->state.speex.pSpeexResamplerState = speex_resampler_init(pConfig->channels, pConfig->sampleRateIn, pConfig->sampleRateOut, pConfig->speex.quality, &speexErr);
- if (pResampler->state.speex.pSpeexResamplerState == NULL) {
- return ma_result_from_speex_err(speexErr);
- }
- #else
- /* Speex resampler not available. */
- return MA_NO_BACKEND;
- #endif
- } break;
+ (void)pAllocationCallbacks;
- default: return MA_INVALID_ARGS;
+ #if !defined(MA_NO_FLAC)
+ {
+ drflac_close(pFlac->dr);
+ }
+ #else
+ {
+ /* flac is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
}
+ #endif
- return MA_SUCCESS;
+ ma_data_source_uninit(&pFlac->ds);
}
-MA_API void ma_resampler_uninit(ma_resampler* pResampler)
+MA_API ma_result ma_flac_read_pcm_frames(ma_flac* pFlac, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
- if (pResampler == NULL) {
- return;
+ if (pFramesRead != NULL) {
+ *pFramesRead = 0;
}
- if (pResampler->config.algorithm == ma_resample_algorithm_linear) {
- ma_linear_resampler_uninit(&pResampler->state.linear);
+ if (frameCount == 0) {
+ return MA_INVALID_ARGS;
}
-#if defined(MA_HAS_SPEEX_RESAMPLER)
- if (pResampler->config.algorithm == ma_resample_algorithm_speex) {
- speex_resampler_destroy((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState);
+ if (pFlac == NULL) {
+ return MA_INVALID_ARGS;
}
-#endif
-}
-static ma_result ma_resampler_process_pcm_frames__read__linear(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
-{
- return ma_linear_resampler_process_pcm_frames(&pResampler->state.linear, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
-}
+ #if !defined(MA_NO_FLAC)
+ {
+ /* We always use floating point format. */
+ ma_result result = MA_SUCCESS; /* Must be initialized to MA_SUCCESS. */
+ ma_uint64 totalFramesRead = 0;
+ ma_format format;
-#if defined(MA_HAS_SPEEX_RESAMPLER)
-static ma_result ma_resampler_process_pcm_frames__read__speex(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
-{
- int speexErr;
- ma_uint64 frameCountOut;
- ma_uint64 frameCountIn;
- ma_uint64 framesProcessedOut;
- ma_uint64 framesProcessedIn;
- unsigned int framesPerIteration = UINT_MAX;
+ ma_flac_get_data_format(pFlac, &format, NULL, NULL, NULL, 0);
- MA_ASSERT(pResampler != NULL);
- MA_ASSERT(pFramesOut != NULL);
- MA_ASSERT(pFrameCountOut != NULL);
- MA_ASSERT(pFrameCountIn != NULL);
+ switch (format)
+ {
+ case ma_format_f32:
+ {
+ totalFramesRead = drflac_read_pcm_frames_f32(pFlac->dr, frameCount, (float*)pFramesOut);
+ } break;
- /*
- Reading from the Speex resampler requires a bit of dancing around for a few reasons. The first thing is that it's frame counts
- are in unsigned int's whereas ours is in ma_uint64. We therefore need to run the conversion in a loop. The other, more complicated
- problem, is that we need to keep track of the input time, similar to what we do with the linear resampler. The reason we need to
- do this is for ma_resampler_get_required_input_frame_count() and ma_resampler_get_expected_output_frame_count().
- */
- frameCountOut = *pFrameCountOut;
- frameCountIn = *pFrameCountIn;
- framesProcessedOut = 0;
- framesProcessedIn = 0;
+ case ma_format_s16:
+ {
+ totalFramesRead = drflac_read_pcm_frames_s16(pFlac->dr, frameCount, (drflac_int16*)pFramesOut);
+ } break;
- while (framesProcessedOut < frameCountOut && framesProcessedIn < frameCountIn) {
- unsigned int frameCountInThisIteration;
- unsigned int frameCountOutThisIteration;
- const void* pFramesInThisIteration;
- void* pFramesOutThisIteration;
+ case ma_format_s32:
+ {
+ totalFramesRead = drflac_read_pcm_frames_s32(pFlac->dr, frameCount, (drflac_int32*)pFramesOut);
+ } break;
- frameCountInThisIteration = framesPerIteration;
- if ((ma_uint64)frameCountInThisIteration > (frameCountIn - framesProcessedIn)) {
- frameCountInThisIteration = (unsigned int)(frameCountIn - framesProcessedIn);
+ case ma_format_u8:
+ case ma_format_s24:
+ case ma_format_unknown:
+ default:
+ {
+ return MA_INVALID_OPERATION;
+ };
}
- frameCountOutThisIteration = framesPerIteration;
- if ((ma_uint64)frameCountOutThisIteration > (frameCountOut - framesProcessedOut)) {
- frameCountOutThisIteration = (unsigned int)(frameCountOut - framesProcessedOut);
+ /* In the future we'll update dr_flac to return MA_AT_END for us. */
+ if (totalFramesRead == 0) {
+ result = MA_AT_END;
}
- pFramesInThisIteration = ma_offset_ptr(pFramesIn, framesProcessedIn * ma_get_bytes_per_frame(pResampler->config.format, pResampler->config.channels));
- pFramesOutThisIteration = ma_offset_ptr(pFramesOut, framesProcessedOut * ma_get_bytes_per_frame(pResampler->config.format, pResampler->config.channels));
-
- if (pResampler->config.format == ma_format_f32) {
- speexErr = speex_resampler_process_interleaved_float((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState, (const float*)pFramesInThisIteration, &frameCountInThisIteration, (float*)pFramesOutThisIteration, &frameCountOutThisIteration);
- } else if (pResampler->config.format == ma_format_s16) {
- speexErr = speex_resampler_process_interleaved_int((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState, (const spx_int16_t*)pFramesInThisIteration, &frameCountInThisIteration, (spx_int16_t*)pFramesOutThisIteration, &frameCountOutThisIteration);
- } else {
- /* Format not supported. Should never get here. */
- MA_ASSERT(MA_FALSE);
- return MA_INVALID_OPERATION;
+ if (pFramesRead != NULL) {
+ *pFramesRead = totalFramesRead;
}
- if (speexErr != RESAMPLER_ERR_SUCCESS) {
- return ma_result_from_speex_err(speexErr);
+ if (result == MA_SUCCESS && totalFramesRead == 0) {
+ result = MA_AT_END;
}
- framesProcessedIn += frameCountInThisIteration;
- framesProcessedOut += frameCountOutThisIteration;
+ return result;
}
+ #else
+ {
+ /* flac is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
- *pFrameCountOut = framesProcessedOut;
- *pFrameCountIn = framesProcessedIn;
+ (void)pFramesOut;
+ (void)frameCount;
+ (void)pFramesRead;
- return MA_SUCCESS;
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
}
-#endif
-static ma_result ma_resampler_process_pcm_frames__read(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+MA_API ma_result ma_flac_seek_to_pcm_frame(ma_flac* pFlac, ma_uint64 frameIndex)
{
- MA_ASSERT(pResampler != NULL);
- MA_ASSERT(pFramesOut != NULL);
-
- /* pFramesOut is not NULL, which means we must have a capacity. */
- if (pFrameCountOut == NULL) {
- return MA_INVALID_ARGS;
- }
-
- /* It doesn't make sense to not have any input frames to process. */
- if (pFrameCountIn == NULL || pFramesIn == NULL) {
+ if (pFlac == NULL) {
return MA_INVALID_ARGS;
}
- switch (pResampler->config.algorithm)
+ #if !defined(MA_NO_FLAC)
{
- case ma_resample_algorithm_linear:
- {
- return ma_resampler_process_pcm_frames__read__linear(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
- }
+ drflac_bool32 flacResult;
- case ma_resample_algorithm_speex:
- {
- #if defined(MA_HAS_SPEEX_RESAMPLER)
- return ma_resampler_process_pcm_frames__read__speex(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
- #else
- break;
- #endif
+ flacResult = drflac_seek_to_pcm_frame(pFlac->dr, frameIndex);
+ if (flacResult != DRFLAC_TRUE) {
+ return MA_ERROR;
}
- default: break;
+ return MA_SUCCESS;
}
+ #else
+ {
+ /* flac is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
- /* Should never get here. */
- MA_ASSERT(MA_FALSE);
- return MA_INVALID_ARGS;
-}
-
-
-static ma_result ma_resampler_process_pcm_frames__seek__linear(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, ma_uint64* pFrameCountOut)
-{
- MA_ASSERT(pResampler != NULL);
+ (void)frameIndex;
- /* Seeking is supported natively by the linear resampler. */
- return ma_linear_resampler_process_pcm_frames(&pResampler->state.linear, pFramesIn, pFrameCountIn, NULL, pFrameCountOut);
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
}
-#if defined(MA_HAS_SPEEX_RESAMPLER)
-static ma_result ma_resampler_process_pcm_frames__seek__speex(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, ma_uint64* pFrameCountOut)
+MA_API ma_result ma_flac_get_data_format(ma_flac* pFlac, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
- /* The generic seek method is implemented in on top of ma_resampler_process_pcm_frames__read() by just processing into a dummy buffer. */
- float devnull[4096];
- ma_uint64 totalOutputFramesToProcess;
- ma_uint64 totalOutputFramesProcessed;
- ma_uint64 totalInputFramesProcessed;
- ma_uint32 bpf;
- ma_result result;
-
- MA_ASSERT(pResampler != NULL);
-
- totalOutputFramesProcessed = 0;
- totalInputFramesProcessed = 0;
- bpf = ma_get_bytes_per_frame(pResampler->config.format, pResampler->config.channels);
-
- if (pFrameCountOut != NULL) {
- /* Seek by output frames. */
- totalOutputFramesToProcess = *pFrameCountOut;
- } else {
- /* Seek by input frames. */
- MA_ASSERT(pFrameCountIn != NULL);
- totalOutputFramesToProcess = ma_resampler_get_expected_output_frame_count(pResampler, *pFrameCountIn);
+ /* Defaults for safety. */
+ if (pFormat != NULL) {
+ *pFormat = ma_format_unknown;
+ }
+ if (pChannels != NULL) {
+ *pChannels = 0;
+ }
+ if (pSampleRate != NULL) {
+ *pSampleRate = 0;
+ }
+ if (pChannelMap != NULL) {
+ MA_ZERO_MEMORY(pChannelMap, sizeof(*pChannelMap) * channelMapCap);
}
- if (pFramesIn != NULL) {
- /* Process input data. */
- MA_ASSERT(pFrameCountIn != NULL);
- while (totalOutputFramesProcessed < totalOutputFramesToProcess && totalInputFramesProcessed < *pFrameCountIn) {
- ma_uint64 inputFramesToProcessThisIteration = (*pFrameCountIn - totalInputFramesProcessed);
- ma_uint64 outputFramesToProcessThisIteration = (totalOutputFramesToProcess - totalOutputFramesProcessed);
- if (outputFramesToProcessThisIteration > sizeof(devnull) / bpf) {
- outputFramesToProcessThisIteration = sizeof(devnull) / bpf;
- }
+ if (pFlac == NULL) {
+ return MA_INVALID_OPERATION;
+ }
- result = ma_resampler_process_pcm_frames__read(pResampler, ma_offset_ptr(pFramesIn, totalInputFramesProcessed*bpf), &inputFramesToProcessThisIteration, ma_offset_ptr(devnull, totalOutputFramesProcessed*bpf), &outputFramesToProcessThisIteration);
- if (result != MA_SUCCESS) {
- return result;
- }
+ if (pFormat != NULL) {
+ *pFormat = pFlac->format;
+ }
- totalOutputFramesProcessed += outputFramesToProcessThisIteration;
- totalInputFramesProcessed += inputFramesToProcessThisIteration;
+ #if !defined(MA_NO_FLAC)
+ {
+ if (pChannels != NULL) {
+ *pChannels = pFlac->dr->channels;
}
- } else {
- /* Don't process input data - just update timing and filter state as if zeroes were passed in. */
- while (totalOutputFramesProcessed < totalOutputFramesToProcess) {
- ma_uint64 inputFramesToProcessThisIteration = 16384;
- ma_uint64 outputFramesToProcessThisIteration = (totalOutputFramesToProcess - totalOutputFramesProcessed);
- if (outputFramesToProcessThisIteration > sizeof(devnull) / bpf) {
- outputFramesToProcessThisIteration = sizeof(devnull) / bpf;
- }
-
- result = ma_resampler_process_pcm_frames__read(pResampler, NULL, &inputFramesToProcessThisIteration, ma_offset_ptr(devnull, totalOutputFramesProcessed*bpf), &outputFramesToProcessThisIteration);
- if (result != MA_SUCCESS) {
- return result;
- }
- totalOutputFramesProcessed += outputFramesToProcessThisIteration;
- totalInputFramesProcessed += inputFramesToProcessThisIteration;
+ if (pSampleRate != NULL) {
+ *pSampleRate = pFlac->dr->sampleRate;
}
- }
+ if (pChannelMap != NULL) {
+ ma_channel_map_init_standard(ma_standard_channel_map_microsoft, pChannelMap, channelMapCap, pFlac->dr->channels);
+ }
- if (pFrameCountIn != NULL) {
- *pFrameCountIn = totalInputFramesProcessed;
- }
- if (pFrameCountOut != NULL) {
- *pFrameCountOut = totalOutputFramesProcessed;
+ return MA_SUCCESS;
}
-
- return MA_SUCCESS;
-}
-#endif
-
-static ma_result ma_resampler_process_pcm_frames__seek(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, ma_uint64* pFrameCountOut)
-{
- MA_ASSERT(pResampler != NULL);
-
- switch (pResampler->config.algorithm)
+ #else
{
- case ma_resample_algorithm_linear:
- {
- return ma_resampler_process_pcm_frames__seek__linear(pResampler, pFramesIn, pFrameCountIn, pFrameCountOut);
- } break;
-
- case ma_resample_algorithm_speex:
- {
- #if defined(MA_HAS_SPEEX_RESAMPLER)
- return ma_resampler_process_pcm_frames__seek__speex(pResampler, pFramesIn, pFrameCountIn, pFrameCountOut);
- #else
- break;
- #endif
- };
-
- default: break;
+ /* flac is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
+ return MA_NOT_IMPLEMENTED;
}
-
- /* Should never hit this. */
- MA_ASSERT(MA_FALSE);
- return MA_INVALID_ARGS;
+ #endif
}
-
-MA_API ma_result ma_resampler_process_pcm_frames(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+MA_API ma_result ma_flac_get_cursor_in_pcm_frames(ma_flac* pFlac, ma_uint64* pCursor)
{
- if (pResampler == NULL) {
+ if (pCursor == NULL) {
return MA_INVALID_ARGS;
}
- if (pFrameCountOut == NULL && pFrameCountIn == NULL) {
+ *pCursor = 0; /* Safety. */
+
+ if (pFlac == NULL) {
return MA_INVALID_ARGS;
}
- if (pFramesOut != NULL) {
- /* Reading. */
- return ma_resampler_process_pcm_frames__read(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
- } else {
- /* Seeking. */
- return ma_resampler_process_pcm_frames__seek(pResampler, pFramesIn, pFrameCountIn, pFrameCountOut);
+ #if !defined(MA_NO_FLAC)
+ {
+ *pCursor = pFlac->dr->currentPCMFrame;
+
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ /* flac is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
+ return MA_NOT_IMPLEMENTED;
}
+ #endif
}
-MA_API ma_result ma_resampler_set_rate(ma_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut)
+MA_API ma_result ma_flac_get_length_in_pcm_frames(ma_flac* pFlac, ma_uint64* pLength)
{
- if (pResampler == NULL) {
+ if (pLength == NULL) {
return MA_INVALID_ARGS;
}
- if (sampleRateIn == 0 || sampleRateOut == 0) {
+ *pLength = 0; /* Safety. */
+
+ if (pFlac == NULL) {
return MA_INVALID_ARGS;
}
- pResampler->config.sampleRateIn = sampleRateIn;
- pResampler->config.sampleRateOut = sampleRateOut;
-
- switch (pResampler->config.algorithm)
+ #if !defined(MA_NO_FLAC)
{
- case ma_resample_algorithm_linear:
- {
- return ma_linear_resampler_set_rate(&pResampler->state.linear, sampleRateIn, sampleRateOut);
- } break;
-
- case ma_resample_algorithm_speex:
- {
- #if defined(MA_HAS_SPEEX_RESAMPLER)
- return ma_result_from_speex_err(speex_resampler_set_rate((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState, sampleRateIn, sampleRateOut));
- #else
- break;
- #endif
- };
+ *pLength = pFlac->dr->totalPCMFrameCount;
- default: break;
+ return MA_SUCCESS;
}
-
- /* Should never get here. */
- MA_ASSERT(MA_FALSE);
- return MA_INVALID_OPERATION;
+ #else
+ {
+ /* flac is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
}
-MA_API ma_result ma_resampler_set_rate_ratio(ma_resampler* pResampler, float ratio)
+
+static ma_result ma_decoding_backend_init__flac(void* pUserData, ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
{
- if (pResampler == NULL) {
- return MA_INVALID_ARGS;
- }
+ ma_result result;
+ ma_flac* pFlac;
- if (pResampler->config.algorithm == ma_resample_algorithm_linear) {
- return ma_linear_resampler_set_rate_ratio(&pResampler->state.linear, ratio);
- } else {
- /* Getting here means the backend does not have native support for setting the rate as a ratio so we just do it generically. */
- ma_uint32 n;
- ma_uint32 d;
+ (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
- d = 1000;
- n = (ma_uint32)(ratio * d);
+ /* For now we're just allocating the decoder backend on the heap. */
+ pFlac = (ma_flac*)ma_malloc(sizeof(*pFlac), pAllocationCallbacks);
+ if (pFlac == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
- if (n == 0) {
- return MA_INVALID_ARGS; /* Ratio too small. */
- }
+ result = ma_flac_init(onRead, onSeek, onTell, pReadSeekTellUserData, pConfig, pAllocationCallbacks, pFlac);
+ if (result != MA_SUCCESS) {
+ ma_free(pFlac, pAllocationCallbacks);
+ return result;
+ }
- MA_ASSERT(n != 0);
+ *ppBackend = pFlac;
- return ma_resampler_set_rate(pResampler, n, d);
- }
+ return MA_SUCCESS;
}
-MA_API ma_uint64 ma_resampler_get_required_input_frame_count(const ma_resampler* pResampler, ma_uint64 outputFrameCount)
+static ma_result ma_decoding_backend_init_file__flac(void* pUserData, const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
{
- if (pResampler == NULL) {
- return 0;
- }
-
- if (outputFrameCount == 0) {
- return 0;
- }
-
- switch (pResampler->config.algorithm)
- {
- case ma_resample_algorithm_linear:
- {
- return ma_linear_resampler_get_required_input_frame_count(&pResampler->state.linear, outputFrameCount);
- }
+ ma_result result;
+ ma_flac* pFlac;
- case ma_resample_algorithm_speex:
- {
- #if defined(MA_HAS_SPEEX_RESAMPLER)
- spx_uint64_t count;
- int speexErr = ma_speex_resampler_get_required_input_frame_count((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState, outputFrameCount, &count);
- if (speexErr != RESAMPLER_ERR_SUCCESS) {
- return 0;
- }
+ (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
- return (ma_uint64)count;
- #else
- break;
- #endif
- }
+ /* For now we're just allocating the decoder backend on the heap. */
+ pFlac = (ma_flac*)ma_malloc(sizeof(*pFlac), pAllocationCallbacks);
+ if (pFlac == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
- default: break;
+ result = ma_flac_init_file(pFilePath, pConfig, pAllocationCallbacks, pFlac);
+ if (result != MA_SUCCESS) {
+ ma_free(pFlac, pAllocationCallbacks);
+ return result;
}
- /* Should never get here. */
- MA_ASSERT(MA_FALSE);
- return 0;
+ *ppBackend = pFlac;
+
+ return MA_SUCCESS;
}
-MA_API ma_uint64 ma_resampler_get_expected_output_frame_count(const ma_resampler* pResampler, ma_uint64 inputFrameCount)
+static ma_result ma_decoding_backend_init_file_w__flac(void* pUserData, const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
{
- if (pResampler == NULL) {
- return 0; /* Invalid args. */
+ ma_result result;
+ ma_flac* pFlac;
+
+ (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
+
+ /* For now we're just allocating the decoder backend on the heap. */
+ pFlac = (ma_flac*)ma_malloc(sizeof(*pFlac), pAllocationCallbacks);
+ if (pFlac == NULL) {
+ return MA_OUT_OF_MEMORY;
}
- if (inputFrameCount == 0) {
- return 0;
+ result = ma_flac_init_file_w(pFilePath, pConfig, pAllocationCallbacks, pFlac);
+ if (result != MA_SUCCESS) {
+ ma_free(pFlac, pAllocationCallbacks);
+ return result;
}
- switch (pResampler->config.algorithm)
- {
- case ma_resample_algorithm_linear:
- {
- return ma_linear_resampler_get_expected_output_frame_count(&pResampler->state.linear, inputFrameCount);
- }
+ *ppBackend = pFlac;
- case ma_resample_algorithm_speex:
- {
- #if defined(MA_HAS_SPEEX_RESAMPLER)
- spx_uint64_t count;
- int speexErr = ma_speex_resampler_get_expected_output_frame_count((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState, inputFrameCount, &count);
- if (speexErr != RESAMPLER_ERR_SUCCESS) {
- return 0;
- }
+ return MA_SUCCESS;
+}
- return (ma_uint64)count;
- #else
- break;
- #endif
- }
+static ma_result ma_decoding_backend_init_memory__flac(void* pUserData, const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
+{
+ ma_result result;
+ ma_flac* pFlac;
- default: break;
+ (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
+
+ /* For now we're just allocating the decoder backend on the heap. */
+ pFlac = (ma_flac*)ma_malloc(sizeof(*pFlac), pAllocationCallbacks);
+ if (pFlac == NULL) {
+ return MA_OUT_OF_MEMORY;
}
- /* Should never get here. */
- MA_ASSERT(MA_FALSE);
- return 0;
+ result = ma_flac_init_memory(pData, dataSize, pConfig, pAllocationCallbacks, pFlac);
+ if (result != MA_SUCCESS) {
+ ma_free(pFlac, pAllocationCallbacks);
+ return result;
+ }
+
+ *ppBackend = pFlac;
+
+ return MA_SUCCESS;
}
-MA_API ma_uint64 ma_resampler_get_input_latency(const ma_resampler* pResampler)
+static void ma_decoding_backend_uninit__flac(void* pUserData, ma_data_source* pBackend, const ma_allocation_callbacks* pAllocationCallbacks)
{
- if (pResampler == NULL) {
- return 0;
- }
+ ma_flac* pFlac = (ma_flac*)pBackend;
- switch (pResampler->config.algorithm)
- {
- case ma_resample_algorithm_linear:
- {
- return ma_linear_resampler_get_input_latency(&pResampler->state.linear);
- }
+ (void)pUserData;
- case ma_resample_algorithm_speex:
- {
- #if defined(MA_HAS_SPEEX_RESAMPLER)
- return (ma_uint64)ma_speex_resampler_get_input_latency((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState);
- #else
- break;
- #endif
- }
+ ma_flac_uninit(pFlac, pAllocationCallbacks);
+ ma_free(pFlac, pAllocationCallbacks);
+}
- default: break;
- }
+static ma_decoding_backend_vtable g_ma_decoding_backend_vtable_flac =
+{
+ ma_decoding_backend_init__flac,
+ ma_decoding_backend_init_file__flac,
+ ma_decoding_backend_init_file_w__flac,
+ ma_decoding_backend_init_memory__flac,
+ ma_decoding_backend_uninit__flac
+};
- /* Should never get here. */
- MA_ASSERT(MA_FALSE);
- return 0;
+static ma_result ma_decoder_init_flac__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+{
+ return ma_decoder_init_from_vtable(&g_ma_decoding_backend_vtable_flac, NULL, pConfig, pDecoder);
}
+#endif /* dr_flac_h */
+
+/* MP3 */
+#ifdef dr_mp3_h
+#define MA_HAS_MP3
-MA_API ma_uint64 ma_resampler_get_output_latency(const ma_resampler* pResampler)
+typedef struct
{
- if (pResampler == NULL) {
- return 0;
- }
-
- switch (pResampler->config.algorithm)
- {
- case ma_resample_algorithm_linear:
- {
- return ma_linear_resampler_get_output_latency(&pResampler->state.linear);
- }
+ ma_data_source_base ds;
+ ma_read_proc onRead;
+ ma_seek_proc onSeek;
+ ma_tell_proc onTell;
+ void* pReadSeekTellUserData;
+ ma_format format; /* Can be f32 or s16. */
+#if !defined(MA_NO_MP3)
+ drmp3 dr;
+ drmp3_uint32 seekPointCount;
+ drmp3_seek_point* pSeekPoints; /* Only used if seek table generation is used. */
+#endif
+} ma_mp3;
- case ma_resample_algorithm_speex:
- {
- #if defined(MA_HAS_SPEEX_RESAMPLER)
- return (ma_uint64)ma_speex_resampler_get_output_latency((SpeexResamplerState*)pResampler->state.speex.pSpeexResamplerState);
- #else
- break;
- #endif
- }
+MA_API ma_result ma_mp3_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3);
+MA_API ma_result ma_mp3_init_file(const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3);
+MA_API ma_result ma_mp3_init_file_w(const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3);
+MA_API ma_result ma_mp3_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3);
+MA_API void ma_mp3_uninit(ma_mp3* pMP3, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_mp3_read_pcm_frames(ma_mp3* pMP3, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
+MA_API ma_result ma_mp3_seek_to_pcm_frame(ma_mp3* pMP3, ma_uint64 frameIndex);
+MA_API ma_result ma_mp3_get_data_format(ma_mp3* pMP3, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
+MA_API ma_result ma_mp3_get_cursor_in_pcm_frames(ma_mp3* pMP3, ma_uint64* pCursor);
+MA_API ma_result ma_mp3_get_length_in_pcm_frames(ma_mp3* pMP3, ma_uint64* pLength);
- default: break;
- }
- /* Should never get here. */
- MA_ASSERT(MA_FALSE);
- return 0;
+static ma_result ma_mp3_ds_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+{
+ return ma_mp3_read_pcm_frames((ma_mp3*)pDataSource, pFramesOut, frameCount, pFramesRead);
}
-/**************************************************************************************************************************************************************
+static ma_result ma_mp3_ds_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
+{
+ return ma_mp3_seek_to_pcm_frame((ma_mp3*)pDataSource, frameIndex);
+}
-Channel Conversion
+static ma_result ma_mp3_ds_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
+{
+ return ma_mp3_get_data_format((ma_mp3*)pDataSource, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
+}
-**************************************************************************************************************************************************************/
-#ifndef MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT
-#define MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT 12
-#endif
+static ma_result ma_mp3_ds_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
+{
+ return ma_mp3_get_cursor_in_pcm_frames((ma_mp3*)pDataSource, pCursor);
+}
-#define MA_PLANE_LEFT 0
-#define MA_PLANE_RIGHT 1
-#define MA_PLANE_FRONT 2
-#define MA_PLANE_BACK 3
-#define MA_PLANE_BOTTOM 4
-#define MA_PLANE_TOP 5
+static ma_result ma_mp3_ds_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
+{
+ return ma_mp3_get_length_in_pcm_frames((ma_mp3*)pDataSource, pLength);
+}
-static float g_maChannelPlaneRatios[MA_CHANNEL_POSITION_COUNT][6] = {
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_NONE */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_MONO */
- { 0.5f, 0.0f, 0.5f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_LEFT */
- { 0.0f, 0.5f, 0.5f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_RIGHT */
- { 0.0f, 0.0f, 1.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_CENTER */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_LFE */
- { 0.5f, 0.0f, 0.0f, 0.5f, 0.0f, 0.0f}, /* MA_CHANNEL_BACK_LEFT */
- { 0.0f, 0.5f, 0.0f, 0.5f, 0.0f, 0.0f}, /* MA_CHANNEL_BACK_RIGHT */
- { 0.25f, 0.0f, 0.75f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_LEFT_CENTER */
- { 0.0f, 0.25f, 0.75f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_RIGHT_CENTER */
- { 0.0f, 0.0f, 0.0f, 1.0f, 0.0f, 0.0f}, /* MA_CHANNEL_BACK_CENTER */
- { 1.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_SIDE_LEFT */
- { 0.0f, 1.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_SIDE_RIGHT */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 1.0f}, /* MA_CHANNEL_TOP_CENTER */
- { 0.33f, 0.0f, 0.33f, 0.0f, 0.0f, 0.34f}, /* MA_CHANNEL_TOP_FRONT_LEFT */
- { 0.0f, 0.0f, 0.5f, 0.0f, 0.0f, 0.5f}, /* MA_CHANNEL_TOP_FRONT_CENTER */
- { 0.0f, 0.33f, 0.33f, 0.0f, 0.0f, 0.34f}, /* MA_CHANNEL_TOP_FRONT_RIGHT */
- { 0.33f, 0.0f, 0.0f, 0.33f, 0.0f, 0.34f}, /* MA_CHANNEL_TOP_BACK_LEFT */
- { 0.0f, 0.0f, 0.0f, 0.5f, 0.0f, 0.5f}, /* MA_CHANNEL_TOP_BACK_CENTER */
- { 0.0f, 0.33f, 0.0f, 0.33f, 0.0f, 0.34f}, /* MA_CHANNEL_TOP_BACK_RIGHT */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_0 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_1 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_2 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_3 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_4 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_5 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_6 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_7 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_8 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_9 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_10 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_11 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_12 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_13 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_14 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_15 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_16 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_17 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_18 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_19 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_20 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_21 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_22 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_23 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_24 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_25 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_26 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_27 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_28 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_29 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_30 */
- { 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_AUX_31 */
+static ma_data_source_vtable g_ma_mp3_ds_vtable =
+{
+ ma_mp3_ds_read,
+ ma_mp3_ds_seek,
+ ma_mp3_ds_get_data_format,
+ ma_mp3_ds_get_cursor,
+ ma_mp3_ds_get_length,
+ NULL, /* onSetLooping */
+ 0
};
-static float ma_calculate_channel_position_rectangular_weight(ma_channel channelPositionA, ma_channel channelPositionB)
-{
- /*
- Imagine the following simplified example: You have a single input speaker which is the front/left speaker which you want to convert to
- the following output configuration:
- - front/left
- - side/left
- - back/left
+#if !defined(MA_NO_MP3)
+static drmp3_allocation_callbacks drmp3_allocation_callbacks_from_miniaudio(const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ drmp3_allocation_callbacks callbacks;
- The front/left output is easy - it the same speaker position so it receives the full contribution of the front/left input. The amount
- of contribution to apply to the side/left and back/left speakers, however, is a bit more complicated.
+ if (pAllocationCallbacks != NULL) {
+ callbacks.onMalloc = pAllocationCallbacks->onMalloc;
+ callbacks.onRealloc = pAllocationCallbacks->onRealloc;
+ callbacks.onFree = pAllocationCallbacks->onFree;
+ callbacks.pUserData = pAllocationCallbacks->pUserData;
+ } else {
+ callbacks.onMalloc = ma__malloc_default;
+ callbacks.onRealloc = ma__realloc_default;
+ callbacks.onFree = ma__free_default;
+ callbacks.pUserData = NULL;
+ }
- Imagine the front/left speaker as emitting audio from two planes - the front plane and the left plane. You can think of the front/left
- speaker emitting half of it's total volume from the front, and the other half from the left. Since part of it's volume is being emitted
- from the left side, and the side/left and back/left channels also emit audio from the left plane, one would expect that they would
- receive some amount of contribution from front/left speaker. The amount of contribution depends on how many planes are shared between
- the two speakers. Note that in the examples below I've added a top/front/left speaker as an example just to show how the math works
- across 3 spatial dimensions.
+ return callbacks;
+}
- The first thing to do is figure out how each speaker's volume is spread over each of plane:
- - front/left: 2 planes (front and left) = 1/2 = half it's total volume on each plane
- - side/left: 1 plane (left only) = 1/1 = entire volume from left plane
- - back/left: 2 planes (back and left) = 1/2 = half it's total volume on each plane
- - top/front/left: 3 planes (top, front and left) = 1/3 = one third it's total volume on each plane
+static size_t ma_mp3_dr_callback__read(void* pUserData, void* pBufferOut, size_t bytesToRead)
+{
+ ma_mp3* pMP3 = (ma_mp3*)pUserData;
+ ma_result result;
+ size_t bytesRead;
- The amount of volume each channel contributes to each of it's planes is what controls how much it is willing to given and take to other
- channels on the same plane. The volume that is willing to the given by one channel is multiplied by the volume that is willing to be
- taken by the other to produce the final contribution.
- */
+ MA_ASSERT(pMP3 != NULL);
- /* Contribution = Sum(Volume to Give * Volume to Take) */
- float contribution =
- g_maChannelPlaneRatios[channelPositionA][0] * g_maChannelPlaneRatios[channelPositionB][0] +
- g_maChannelPlaneRatios[channelPositionA][1] * g_maChannelPlaneRatios[channelPositionB][1] +
- g_maChannelPlaneRatios[channelPositionA][2] * g_maChannelPlaneRatios[channelPositionB][2] +
- g_maChannelPlaneRatios[channelPositionA][3] * g_maChannelPlaneRatios[channelPositionB][3] +
- g_maChannelPlaneRatios[channelPositionA][4] * g_maChannelPlaneRatios[channelPositionB][4] +
- g_maChannelPlaneRatios[channelPositionA][5] * g_maChannelPlaneRatios[channelPositionB][5];
+ result = pMP3->onRead(pMP3->pReadSeekTellUserData, pBufferOut, bytesToRead, &bytesRead);
+ (void)result;
- return contribution;
+ return bytesRead;
}
-MA_API ma_channel_converter_config ma_channel_converter_config_init(ma_format format, ma_uint32 channelsIn, const ma_channel* pChannelMapIn, ma_uint32 channelsOut, const ma_channel* pChannelMapOut, ma_channel_mix_mode mixingMode)
+static drmp3_bool32 ma_mp3_dr_callback__seek(void* pUserData, int offset, drmp3_seek_origin origin)
{
- ma_channel_converter_config config;
+ ma_mp3* pMP3 = (ma_mp3*)pUserData;
+ ma_result result;
+ ma_seek_origin maSeekOrigin;
- /* Channel counts need to be clamped. */
- channelsIn = ma_min(channelsIn, ma_countof(config.channelMapIn));
- channelsOut = ma_min(channelsOut, ma_countof(config.channelMapOut));
+ MA_ASSERT(pMP3 != NULL);
- MA_ZERO_OBJECT(&config);
- config.format = format;
- config.channelsIn = channelsIn;
- config.channelsOut = channelsOut;
- ma_channel_map_copy_or_default(config.channelMapIn, pChannelMapIn, channelsIn);
- ma_channel_map_copy_or_default(config.channelMapOut, pChannelMapOut, channelsOut);
- config.mixingMode = mixingMode;
+ maSeekOrigin = ma_seek_origin_start;
+ if (origin == drmp3_seek_origin_current) {
+ maSeekOrigin = ma_seek_origin_current;
+ }
- return config;
-}
+ result = pMP3->onSeek(pMP3->pReadSeekTellUserData, offset, maSeekOrigin);
+ if (result != MA_SUCCESS) {
+ return MA_FALSE;
+ }
-static ma_int32 ma_channel_converter_float_to_fixed(float x)
-{
- return (ma_int32)(x * (1<= MA_CHANNEL_AUX_0 && channelPosition <= MA_CHANNEL_AUX_31) {
- return MA_FALSE;
+ MA_ZERO_OBJECT(pMP3);
+ pMP3->format = ma_format_f32; /* f32 by default. */
+
+ if (pConfig != NULL && (pConfig->preferredFormat == ma_format_f32 || pConfig->preferredFormat == ma_format_s16)) {
+ pMP3->format = pConfig->preferredFormat;
+ } else {
+ /* Getting here means something other than f32 and s16 was specified. Just leave this unset to use the default format. */
}
- for (i = 0; i < 6; ++i) { /* Each side of a cube. */
- if (g_maChannelPlaneRatios[channelPosition][i] != 0) {
- return MA_TRUE;
- }
+ dataSourceConfig = ma_data_source_config_init();
+ dataSourceConfig.vtable = &g_ma_mp3_ds_vtable;
+
+ result = ma_data_source_init(&dataSourceConfig, &pMP3->ds);
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to initialize the base data source. */
}
- return MA_FALSE;
+ return MA_SUCCESS;
}
-MA_API ma_result ma_channel_converter_init(const ma_channel_converter_config* pConfig, ma_channel_converter* pConverter)
+static ma_result ma_mp3_generate_seek_table(ma_mp3* pMP3, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_uint32 iChannelIn;
- ma_uint32 iChannelOut;
-
- if (pConverter == NULL) {
- return MA_INVALID_ARGS;
- }
+ drmp3_bool32 mp3Result;
+ drmp3_uint32 seekPointCount = 0;
+ drmp3_seek_point* pSeekPoints = NULL;
- MA_ZERO_OBJECT(pConverter);
+ MA_ASSERT(pMP3 != NULL);
+ MA_ASSERT(pConfig != NULL);
- if (pConfig == NULL) {
- return MA_INVALID_ARGS;
+ seekPointCount = pConfig->seekPointCount;
+ if (seekPointCount > 0) {
+ pSeekPoints = (drmp3_seek_point*)ma_malloc(sizeof(*pMP3->pSeekPoints) * seekPointCount, pAllocationCallbacks);
+ if (pSeekPoints == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
}
- /* Basic validation for channel counts. */
- if (pConfig->channelsIn < MA_MIN_CHANNELS || pConfig->channelsIn > MA_MAX_CHANNELS ||
- pConfig->channelsOut < MA_MIN_CHANNELS || pConfig->channelsOut > MA_MAX_CHANNELS) {
- return MA_INVALID_ARGS;
+ mp3Result = drmp3_calculate_seek_points(&pMP3->dr, &seekPointCount, pSeekPoints);
+ if (mp3Result != MA_TRUE) {
+ return MA_ERROR;
}
- if (!ma_channel_map_valid(pConfig->channelsIn, pConfig->channelMapIn)) {
- return MA_INVALID_ARGS; /* Invalid input channel map. */
- }
- if (!ma_channel_map_valid(pConfig->channelsOut, pConfig->channelMapOut)) {
- return MA_INVALID_ARGS; /* Invalid output channel map. */
+ mp3Result = drmp3_bind_seek_table(&pMP3->dr, seekPointCount, pSeekPoints);
+ if (mp3Result != MA_TRUE) {
+ ma_free(pSeekPoints, pAllocationCallbacks);
+ return MA_ERROR;
}
- pConverter->format = pConfig->format;
- pConverter->channelsIn = pConfig->channelsIn;
- pConverter->channelsOut = pConfig->channelsOut;
- ma_channel_map_copy_or_default(pConverter->channelMapIn, pConfig->channelMapIn, pConfig->channelsIn);
- ma_channel_map_copy_or_default(pConverter->channelMapOut, pConfig->channelMapOut, pConfig->channelsOut);
- pConverter->mixingMode = pConfig->mixingMode;
+ pMP3->seekPointCount = seekPointCount;
+ pMP3->pSeekPoints = pSeekPoints;
- for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; iChannelIn += 1) {
- for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
- if (pConverter->format == ma_format_f32) {
- pConverter->weights.f32[iChannelIn][iChannelOut] = pConfig->weights[iChannelIn][iChannelOut];
- } else {
- pConverter->weights.s16[iChannelIn][iChannelOut] = ma_channel_converter_float_to_fixed(pConfig->weights[iChannelIn][iChannelOut]);
- }
- }
- }
+ return MA_SUCCESS;
+}
+MA_API ma_result ma_mp3_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3)
+{
+ ma_result result;
+ result = ma_mp3_init_internal(pConfig, pMP3);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- /* If the input and output channels and channel maps are the same we should use a passthrough. */
- if (pConverter->channelsIn == pConverter->channelsOut) {
- if (ma_channel_map_equal(pConverter->channelsIn, pConverter->channelMapIn, pConverter->channelMapOut)) {
- pConverter->isPassthrough = MA_TRUE;
- }
- if (ma_channel_map_blank(pConverter->channelsIn, pConverter->channelMapIn) || ma_channel_map_blank(pConverter->channelsOut, pConverter->channelMapOut)) {
- pConverter->isPassthrough = MA_TRUE;
- }
+ if (onRead == NULL || onSeek == NULL) {
+ return MA_INVALID_ARGS; /* onRead and onSeek are mandatory. */
}
+ pMP3->onRead = onRead;
+ pMP3->onSeek = onSeek;
+ pMP3->onTell = onTell;
+ pMP3->pReadSeekTellUserData = pReadSeekTellUserData;
- /*
- We can use a simple case for expanding the mono channel. This will used when expanding a mono input into any output so long
- as no LFE is present in the output.
- */
- if (!pConverter->isPassthrough) {
- if (pConverter->channelsIn == 1 && pConverter->channelMapIn[0] == MA_CHANNEL_MONO) {
- /* Optimal case if no LFE is in the output channel map. */
- pConverter->isSimpleMonoExpansion = MA_TRUE;
- if (ma_channel_map_contains_channel_position(pConverter->channelsOut, pConverter->channelMapOut, MA_CHANNEL_LFE)) {
- pConverter->isSimpleMonoExpansion = MA_FALSE;
- }
- }
- }
+ #if !defined(MA_NO_MP3)
+ {
+ drmp3_allocation_callbacks mp3AllocationCallbacks = drmp3_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
+ drmp3_bool32 mp3Result;
- /* Another optimized case is stereo to mono. */
- if (!pConverter->isPassthrough) {
- if (pConverter->channelsOut == 1 && pConverter->channelMapOut[0] == MA_CHANNEL_MONO && pConverter->channelsIn == 2) {
- /* Optimal case if no LFE is in the input channel map. */
- pConverter->isStereoToMono = MA_TRUE;
- if (ma_channel_map_contains_channel_position(pConverter->channelsIn, pConverter->channelMapIn, MA_CHANNEL_LFE)) {
- pConverter->isStereoToMono = MA_FALSE;
- }
+ mp3Result = drmp3_init(&pMP3->dr, ma_mp3_dr_callback__read, ma_mp3_dr_callback__seek, pMP3, &mp3AllocationCallbacks);
+ if (mp3Result != MA_TRUE) {
+ return MA_INVALID_FILE;
}
- }
+ ma_mp3_generate_seek_table(pMP3, pConfig, pAllocationCallbacks);
- /*
- Here is where we do a bit of pre-processing to know how each channel should be combined to make up the output. Rules:
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ /* mp3 is disabled. */
+ (void)pAllocationCallbacks;
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
+}
- 1) If it's a passthrough, do nothing - it's just a simple memcpy().
- 2) If the channel counts are the same and every channel position in the input map is present in the output map, use a
- simple shuffle. An example might be different 5.1 channel layouts.
- 3) Otherwise channels are blended based on spatial locality.
- */
- if (!pConverter->isPassthrough) {
- if (pConverter->channelsIn == pConverter->channelsOut) {
- ma_bool32 areAllChannelPositionsPresent = MA_TRUE;
- for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
- ma_bool32 isInputChannelPositionInOutput = MA_FALSE;
- for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
- if (pConverter->channelMapIn[iChannelIn] == pConverter->channelMapOut[iChannelOut]) {
- isInputChannelPositionInOutput = MA_TRUE;
- break;
- }
- }
+MA_API ma_result ma_mp3_init_file(const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3)
+{
+ ma_result result;
- if (!isInputChannelPositionInOutput) {
- areAllChannelPositionsPresent = MA_FALSE;
- break;
- }
- }
+ result = ma_mp3_init_internal(pConfig, pMP3);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- if (areAllChannelPositionsPresent) {
- pConverter->isSimpleShuffle = MA_TRUE;
+ #if !defined(MA_NO_MP3)
+ {
+ drmp3_allocation_callbacks mp3AllocationCallbacks = drmp3_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
+ drmp3_bool32 mp3Result;
- /*
- All the router will be doing is rearranging channels which means all we need to do is use a shuffling table which is just
- a mapping between the index of the input channel to the index of the output channel.
- */
- for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
- for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
- if (pConverter->channelMapIn[iChannelIn] == pConverter->channelMapOut[iChannelOut]) {
- pConverter->shuffleTable[iChannelIn] = (ma_uint8)iChannelOut;
- break;
- }
- }
- }
- }
+ mp3Result = drmp3_init_file(&pMP3->dr, pFilePath, &mp3AllocationCallbacks);
+ if (mp3Result != MA_TRUE) {
+ return MA_INVALID_FILE;
}
- }
+ ma_mp3_generate_seek_table(pMP3, pConfig, pAllocationCallbacks);
- /*
- Here is where weights are calculated. Note that we calculate the weights at all times, even when using a passthrough and simple
- shuffling. We use different algorithms for calculating weights depending on our mixing mode.
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ /* mp3 is disabled. */
+ (void)pFilePath;
+ (void)pAllocationCallbacks;
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
+}
- In simple mode we don't do any blending (except for converting between mono, which is done in a later step). Instead we just
- map 1:1 matching channels. In this mode, if no channels in the input channel map correspond to anything in the output channel
- map, nothing will be heard!
- */
+MA_API ma_result ma_mp3_init_file_w(const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3)
+{
+ ma_result result;
- /* In all cases we need to make sure all channels that are present in both channel maps have a 1:1 mapping. */
- for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
- ma_channel channelPosIn = pConverter->channelMapIn[iChannelIn];
+ result = ma_mp3_init_internal(pConfig, pMP3);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
- ma_channel channelPosOut = pConverter->channelMapOut[iChannelOut];
+ #if !defined(MA_NO_MP3)
+ {
+ drmp3_allocation_callbacks mp3AllocationCallbacks = drmp3_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
+ drmp3_bool32 mp3Result;
- if (channelPosIn == channelPosOut) {
- if (pConverter->format == ma_format_f32) {
- pConverter->weights.f32[iChannelIn][iChannelOut] = 1;
- } else {
- pConverter->weights.s16[iChannelIn][iChannelOut] = (1 << MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT);
- }
- }
+ mp3Result = drmp3_init_file_w(&pMP3->dr, pFilePath, &mp3AllocationCallbacks);
+ if (mp3Result != MA_TRUE) {
+ return MA_INVALID_FILE;
}
- }
- /*
- The mono channel is accumulated on all other channels, except LFE. Make sure in this loop we exclude output mono channels since
- they were handled in the pass above.
- */
- for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
- ma_channel channelPosIn = pConverter->channelMapIn[iChannelIn];
+ ma_mp3_generate_seek_table(pMP3, pConfig, pAllocationCallbacks);
+
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ /* mp3 is disabled. */
+ (void)pFilePath;
+ (void)pAllocationCallbacks;
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
+}
- if (channelPosIn == MA_CHANNEL_MONO) {
- for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
- ma_channel channelPosOut = pConverter->channelMapOut[iChannelOut];
+MA_API ma_result ma_mp3_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3)
+{
+ ma_result result;
- if (channelPosOut != MA_CHANNEL_NONE && channelPosOut != MA_CHANNEL_MONO && channelPosOut != MA_CHANNEL_LFE) {
- if (pConverter->format == ma_format_f32) {
- pConverter->weights.f32[iChannelIn][iChannelOut] = 1;
- } else {
- pConverter->weights.s16[iChannelIn][iChannelOut] = (1 << MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT);
- }
- }
- }
- }
+ result = ma_mp3_init_internal(pConfig, pMP3);
+ if (result != MA_SUCCESS) {
+ return result;
}
- /* The output mono channel is the average of all non-none, non-mono and non-lfe input channels. */
+ #if !defined(MA_NO_MP3)
{
- ma_uint32 len = 0;
- for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
- ma_channel channelPosIn = pConverter->channelMapIn[iChannelIn];
+ drmp3_allocation_callbacks mp3AllocationCallbacks = drmp3_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
+ drmp3_bool32 mp3Result;
- if (channelPosIn != MA_CHANNEL_NONE && channelPosIn != MA_CHANNEL_MONO && channelPosIn != MA_CHANNEL_LFE) {
- len += 1;
- }
+ mp3Result = drmp3_init_memory(&pMP3->dr, pData, dataSize, &mp3AllocationCallbacks);
+ if (mp3Result != MA_TRUE) {
+ return MA_INVALID_FILE;
}
- if (len > 0) {
- float monoWeight = 1.0f / len;
+ ma_mp3_generate_seek_table(pMP3, pConfig, pAllocationCallbacks);
- for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
- ma_channel channelPosOut = pConverter->channelMapOut[iChannelOut];
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ /* mp3 is disabled. */
+ (void)pData;
+ (void)dataSize;
+ (void)pAllocationCallbacks;
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
+}
- if (channelPosOut == MA_CHANNEL_MONO) {
- for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
- ma_channel channelPosIn = pConverter->channelMapIn[iChannelIn];
+MA_API void ma_mp3_uninit(ma_mp3* pMP3, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pMP3 == NULL) {
+ return;
+ }
- if (channelPosIn != MA_CHANNEL_NONE && channelPosIn != MA_CHANNEL_MONO && channelPosIn != MA_CHANNEL_LFE) {
- if (pConverter->format == ma_format_f32) {
- pConverter->weights.f32[iChannelIn][iChannelOut] = monoWeight;
- } else {
- pConverter->weights.s16[iChannelIn][iChannelOut] = ma_channel_converter_float_to_fixed(monoWeight);
- }
- }
- }
- }
- }
- }
+ #if !defined(MA_NO_MP3)
+ {
+ drmp3_uninit(&pMP3->dr);
}
+ #else
+ {
+ /* mp3 is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
+ }
+ #endif
+ /* Seek points need to be freed after the MP3 decoder has been uninitialized to ensure they're no longer being referenced. */
+ ma_free(pMP3->pSeekPoints, pAllocationCallbacks);
- /* Input and output channels that are not present on the other side need to be blended in based on spatial locality. */
- switch (pConverter->mixingMode)
- {
- case ma_channel_mix_mode_rectangular:
- {
- /* Unmapped input channels. */
- for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
- ma_channel channelPosIn = pConverter->channelMapIn[iChannelIn];
-
- if (ma_is_spatial_channel_position(channelPosIn)) {
- if (!ma_channel_map_contains_channel_position(pConverter->channelsOut, pConverter->channelMapOut, channelPosIn)) {
- for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
- ma_channel channelPosOut = pConverter->channelMapOut[iChannelOut];
-
- if (ma_is_spatial_channel_position(channelPosOut)) {
- float weight = 0;
- if (pConverter->mixingMode == ma_channel_mix_mode_rectangular) {
- weight = ma_calculate_channel_position_rectangular_weight(channelPosIn, channelPosOut);
- }
+ ma_data_source_uninit(&pMP3->ds);
+}
- /* Only apply the weight if we haven't already got some contribution from the respective channels. */
- if (pConverter->format == ma_format_f32) {
- if (pConverter->weights.f32[iChannelIn][iChannelOut] == 0) {
- pConverter->weights.f32[iChannelIn][iChannelOut] = weight;
- }
- } else {
- if (pConverter->weights.s16[iChannelIn][iChannelOut] == 0) {
- pConverter->weights.s16[iChannelIn][iChannelOut] = ma_channel_converter_float_to_fixed(weight);
- }
- }
- }
- }
- }
- }
- }
+MA_API ma_result ma_mp3_read_pcm_frames(ma_mp3* pMP3, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+{
+ if (pFramesRead != NULL) {
+ *pFramesRead = 0;
+ }
- /* Unmapped output channels. */
- for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
- ma_channel channelPosOut = pConverter->channelMapOut[iChannelOut];
+ if (frameCount == 0) {
+ return MA_INVALID_ARGS;
+ }
- if (ma_is_spatial_channel_position(channelPosOut)) {
- if (!ma_channel_map_contains_channel_position(pConverter->channelsIn, pConverter->channelMapIn, channelPosOut)) {
- for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
- ma_channel channelPosIn = pConverter->channelMapIn[iChannelIn];
+ if (pMP3 == NULL) {
+ return MA_INVALID_ARGS;
+ }
- if (ma_is_spatial_channel_position(channelPosIn)) {
- float weight = 0;
- if (pConverter->mixingMode == ma_channel_mix_mode_rectangular) {
- weight = ma_calculate_channel_position_rectangular_weight(channelPosIn, channelPosOut);
- }
+ #if !defined(MA_NO_MP3)
+ {
+ /* We always use floating point format. */
+ ma_result result = MA_SUCCESS; /* Must be initialized to MA_SUCCESS. */
+ ma_uint64 totalFramesRead = 0;
+ ma_format format;
- /* Only apply the weight if we haven't already got some contribution from the respective channels. */
- if (pConverter->format == ma_format_f32) {
- if (pConverter->weights.f32[iChannelIn][iChannelOut] == 0) {
- pConverter->weights.f32[iChannelIn][iChannelOut] = weight;
- }
- } else {
- if (pConverter->weights.s16[iChannelIn][iChannelOut] == 0) {
- pConverter->weights.s16[iChannelIn][iChannelOut] = ma_channel_converter_float_to_fixed(weight);
- }
- }
- }
- }
- }
- }
- }
- } break;
+ ma_mp3_get_data_format(pMP3, &format, NULL, NULL, NULL, 0);
- case ma_channel_mix_mode_simple:
+ switch (format)
{
- /* In simple mode, excess channels need to be silenced or dropped. */
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < ma_min(pConverter->channelsIn, pConverter->channelsOut); iChannel += 1) {
- if (pConverter->format == ma_format_f32) {
- if (pConverter->weights.f32[iChannel][iChannel] == 0) {
- pConverter->weights.f32[iChannel][iChannel] = 1;
- }
- } else {
- if (pConverter->weights.s16[iChannel][iChannel] == 0) {
- pConverter->weights.s16[iChannel][iChannel] = ma_channel_converter_float_to_fixed(1);
- }
- }
- }
- } break;
+ case ma_format_f32:
+ {
+ totalFramesRead = drmp3_read_pcm_frames_f32(&pMP3->dr, frameCount, (float*)pFramesOut);
+ } break;
- case ma_channel_mix_mode_custom_weights:
- default:
- {
- /* Fallthrough. */
- } break;
- }
+ case ma_format_s16:
+ {
+ totalFramesRead = drmp3_read_pcm_frames_s16(&pMP3->dr, frameCount, (drmp3_int16*)pFramesOut);
+ } break;
+
+ case ma_format_u8:
+ case ma_format_s24:
+ case ma_format_s32:
+ case ma_format_unknown:
+ default:
+ {
+ return MA_INVALID_OPERATION;
+ };
+ }
+ /* In the future we'll update dr_mp3 to return MA_AT_END for us. */
+ if (totalFramesRead == 0) {
+ result = MA_AT_END;
+ }
- return MA_SUCCESS;
-}
+ if (pFramesRead != NULL) {
+ *pFramesRead = totalFramesRead;
+ }
-MA_API void ma_channel_converter_uninit(ma_channel_converter* pConverter)
-{
- if (pConverter == NULL) {
- return;
+ return result;
}
-}
+ #else
+ {
+ /* mp3 is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
-static ma_result ma_channel_converter_process_pcm_frames__passthrough(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
-{
- MA_ASSERT(pConverter != NULL);
- MA_ASSERT(pFramesOut != NULL);
- MA_ASSERT(pFramesIn != NULL);
+ (void)pFramesOut;
+ (void)frameCount;
+ (void)pFramesRead;
- ma_copy_memory_64(pFramesOut, pFramesIn, frameCount * ma_get_bytes_per_frame(pConverter->format, pConverter->channelsOut));
- return MA_SUCCESS;
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
}
-static ma_result ma_channel_converter_process_pcm_frames__simple_shuffle(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+MA_API ma_result ma_mp3_seek_to_pcm_frame(ma_mp3* pMP3, ma_uint64 frameIndex)
{
- ma_uint32 iFrame;
- ma_uint32 iChannelIn;
-
- MA_ASSERT(pConverter != NULL);
- MA_ASSERT(pFramesOut != NULL);
- MA_ASSERT(pFramesIn != NULL);
- MA_ASSERT(pConverter->channelsIn == pConverter->channelsOut);
+ if (pMP3 == NULL) {
+ return MA_INVALID_ARGS;
+ }
- switch (pConverter->format)
+ #if !defined(MA_NO_MP3)
{
- case ma_format_u8:
- {
- /* */ ma_uint8* pFramesOutU8 = ( ma_uint8*)pFramesOut;
- const ma_uint8* pFramesInU8 = (const ma_uint8*)pFramesIn;
-
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
- pFramesOutU8[pConverter->shuffleTable[iChannelIn]] = pFramesInU8[iChannelIn];
- }
-
- pFramesOutU8 += pConverter->channelsOut;
- pFramesInU8 += pConverter->channelsIn;
- }
- } break;
-
- case ma_format_s16:
- {
- /* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut;
- const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn;
-
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
- pFramesOutS16[pConverter->shuffleTable[iChannelIn]] = pFramesInS16[iChannelIn];
- }
+ drmp3_bool32 mp3Result;
- pFramesOutS16 += pConverter->channelsOut;
- pFramesInS16 += pConverter->channelsIn;
- }
- } break;
+ mp3Result = drmp3_seek_to_pcm_frame(&pMP3->dr, frameIndex);
+ if (mp3Result != DRMP3_TRUE) {
+ return MA_ERROR;
+ }
- case ma_format_s24:
- {
- /* */ ma_uint8* pFramesOutS24 = ( ma_uint8*)pFramesOut;
- const ma_uint8* pFramesInS24 = (const ma_uint8*)pFramesIn;
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ /* mp3 is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
- ma_uint32 iChannelOut = pConverter->shuffleTable[iChannelIn];
- pFramesOutS24[iChannelOut*3 + 0] = pFramesInS24[iChannelIn*3 + 0];
- pFramesOutS24[iChannelOut*3 + 1] = pFramesInS24[iChannelIn*3 + 1];
- pFramesOutS24[iChannelOut*3 + 2] = pFramesInS24[iChannelIn*3 + 2];
- }
+ (void)frameIndex;
- pFramesOutS24 += pConverter->channelsOut*3;
- pFramesInS24 += pConverter->channelsIn*3;
- }
- } break;
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
+}
- case ma_format_s32:
- {
- /* */ ma_int32* pFramesOutS32 = ( ma_int32*)pFramesOut;
- const ma_int32* pFramesInS32 = (const ma_int32*)pFramesIn;
+MA_API ma_result ma_mp3_get_data_format(ma_mp3* pMP3, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
+{
+ /* Defaults for safety. */
+ if (pFormat != NULL) {
+ *pFormat = ma_format_unknown;
+ }
+ if (pChannels != NULL) {
+ *pChannels = 0;
+ }
+ if (pSampleRate != NULL) {
+ *pSampleRate = 0;
+ }
+ if (pChannelMap != NULL) {
+ MA_ZERO_MEMORY(pChannelMap, sizeof(*pChannelMap) * channelMapCap);
+ }
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
- pFramesOutS32[pConverter->shuffleTable[iChannelIn]] = pFramesInS32[iChannelIn];
- }
+ if (pMP3 == NULL) {
+ return MA_INVALID_OPERATION;
+ }
- pFramesOutS32 += pConverter->channelsOut;
- pFramesInS32 += pConverter->channelsIn;
- }
- } break;
+ if (pFormat != NULL) {
+ *pFormat = pMP3->format;
+ }
- case ma_format_f32:
- {
- /* */ float* pFramesOutF32 = ( float*)pFramesOut;
- const float* pFramesInF32 = (const float*)pFramesIn;
+ #if !defined(MA_NO_MP3)
+ {
+ if (pChannels != NULL) {
+ *pChannels = pMP3->dr.channels;
+ }
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
- pFramesOutF32[pConverter->shuffleTable[iChannelIn]] = pFramesInF32[iChannelIn];
- }
+ if (pSampleRate != NULL) {
+ *pSampleRate = pMP3->dr.sampleRate;
+ }
- pFramesOutF32 += pConverter->channelsOut;
- pFramesInF32 += pConverter->channelsIn;
- }
- } break;
+ if (pChannelMap != NULL) {
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, pMP3->dr.channels);
+ }
- default: return MA_INVALID_OPERATION; /* Unknown format. */
+ return MA_SUCCESS;
}
-
- return MA_SUCCESS;
+ #else
+ {
+ /* mp3 is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
}
-static ma_result ma_channel_converter_process_pcm_frames__simple_mono_expansion(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+MA_API ma_result ma_mp3_get_cursor_in_pcm_frames(ma_mp3* pMP3, ma_uint64* pCursor)
{
- ma_uint64 iFrame;
+ if (pCursor == NULL) {
+ return MA_INVALID_ARGS;
+ }
- MA_ASSERT(pConverter != NULL);
- MA_ASSERT(pFramesOut != NULL);
- MA_ASSERT(pFramesIn != NULL);
- MA_ASSERT(pConverter->channelsIn == 1);
+ *pCursor = 0; /* Safety. */
- switch (pConverter->format)
+ if (pMP3 == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ #if !defined(MA_NO_MP3)
{
- case ma_format_u8:
- {
- /* */ ma_uint8* pFramesOutU8 = ( ma_uint8*)pFramesOut;
- const ma_uint8* pFramesInU8 = (const ma_uint8*)pFramesIn;
+ *pCursor = pMP3->dr.currentPCMFrame;
- for (iFrame = 0; iFrame < frameCount; ++iFrame) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < pConverter->channelsOut; iChannel += 1) {
- pFramesOutU8[iFrame*pConverter->channelsOut + iChannel] = pFramesInU8[iFrame];
- }
- }
- } break;
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ /* mp3 is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
+}
- case ma_format_s16:
- {
- /* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut;
- const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn;
+MA_API ma_result ma_mp3_get_length_in_pcm_frames(ma_mp3* pMP3, ma_uint64* pLength)
+{
+ if (pLength == NULL) {
+ return MA_INVALID_ARGS;
+ }
- if (pConverter->channelsOut == 2) {
- for (iFrame = 0; iFrame < frameCount; ++iFrame) {
- pFramesOutS16[iFrame*2 + 0] = pFramesInS16[iFrame];
- pFramesOutS16[iFrame*2 + 1] = pFramesInS16[iFrame];
- }
- } else {
- for (iFrame = 0; iFrame < frameCount; ++iFrame) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < pConverter->channelsOut; iChannel += 1) {
- pFramesOutS16[iFrame*pConverter->channelsOut + iChannel] = pFramesInS16[iFrame];
- }
- }
- }
- } break;
+ *pLength = 0; /* Safety. */
- case ma_format_s24:
- {
- /* */ ma_uint8* pFramesOutS24 = ( ma_uint8*)pFramesOut;
- const ma_uint8* pFramesInS24 = (const ma_uint8*)pFramesIn;
+ if (pMP3 == NULL) {
+ return MA_INVALID_ARGS;
+ }
- for (iFrame = 0; iFrame < frameCount; ++iFrame) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < pConverter->channelsOut; iChannel += 1) {
- ma_uint64 iSampleOut = iFrame*pConverter->channelsOut + iChannel;
- ma_uint64 iSampleIn = iFrame;
- pFramesOutS24[iSampleOut*3 + 0] = pFramesInS24[iSampleIn*3 + 0];
- pFramesOutS24[iSampleOut*3 + 1] = pFramesInS24[iSampleIn*3 + 1];
- pFramesOutS24[iSampleOut*3 + 2] = pFramesInS24[iSampleIn*3 + 2];
- }
- }
- } break;
+ #if !defined(MA_NO_MP3)
+ {
+ *pLength = drmp3_get_pcm_frame_count(&pMP3->dr);
- case ma_format_s32:
- {
- /* */ ma_int32* pFramesOutS32 = ( ma_int32*)pFramesOut;
- const ma_int32* pFramesInS32 = (const ma_int32*)pFramesIn;
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ /* mp3 is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
+}
- for (iFrame = 0; iFrame < frameCount; ++iFrame) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < pConverter->channelsOut; iChannel += 1) {
- pFramesOutS32[iFrame*pConverter->channelsOut + iChannel] = pFramesInS32[iFrame];
- }
- }
- } break;
- case ma_format_f32:
- {
- /* */ float* pFramesOutF32 = ( float*)pFramesOut;
- const float* pFramesInF32 = (const float*)pFramesIn;
+static ma_result ma_decoding_backend_init__mp3(void* pUserData, ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
+{
+ ma_result result;
+ ma_mp3* pMP3;
- if (pConverter->channelsOut == 2) {
- for (iFrame = 0; iFrame < frameCount; ++iFrame) {
- pFramesOutF32[iFrame*2 + 0] = pFramesInF32[iFrame];
- pFramesOutF32[iFrame*2 + 1] = pFramesInF32[iFrame];
- }
- } else {
- for (iFrame = 0; iFrame < frameCount; ++iFrame) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < pConverter->channelsOut; iChannel += 1) {
- pFramesOutF32[iFrame*pConverter->channelsOut + iChannel] = pFramesInF32[iFrame];
- }
- }
- }
- } break;
+ (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
- default: return MA_INVALID_OPERATION; /* Unknown format. */
+ /* For now we're just allocating the decoder backend on the heap. */
+ pMP3 = (ma_mp3*)ma_malloc(sizeof(*pMP3), pAllocationCallbacks);
+ if (pMP3 == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+
+ result = ma_mp3_init(onRead, onSeek, onTell, pReadSeekTellUserData, pConfig, pAllocationCallbacks, pMP3);
+ if (result != MA_SUCCESS) {
+ ma_free(pMP3, pAllocationCallbacks);
+ return result;
}
+ *ppBackend = pMP3;
+
return MA_SUCCESS;
}
-static ma_result ma_channel_converter_process_pcm_frames__stereo_to_mono(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+static ma_result ma_decoding_backend_init_file__mp3(void* pUserData, const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
{
- ma_uint64 iFrame;
-
- MA_ASSERT(pConverter != NULL);
- MA_ASSERT(pFramesOut != NULL);
- MA_ASSERT(pFramesIn != NULL);
- MA_ASSERT(pConverter->channelsIn == 2);
- MA_ASSERT(pConverter->channelsOut == 1);
-
- switch (pConverter->format)
- {
- case ma_format_u8:
- {
- /* */ ma_uint8* pFramesOutU8 = ( ma_uint8*)pFramesOut;
- const ma_uint8* pFramesInU8 = (const ma_uint8*)pFramesIn;
-
- for (iFrame = 0; iFrame < frameCount; ++iFrame) {
- pFramesOutU8[iFrame] = ma_clip_u8((ma_int16)((ma_pcm_sample_u8_to_s16_no_scale(pFramesInU8[iFrame*2+0]) + ma_pcm_sample_u8_to_s16_no_scale(pFramesInU8[iFrame*2+1])) / 2));
- }
- } break;
+ ma_result result;
+ ma_mp3* pMP3;
- case ma_format_s16:
- {
- /* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut;
- const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn;
+ (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
- for (iFrame = 0; iFrame < frameCount; ++iFrame) {
- pFramesOutS16[iFrame] = (ma_int16)(((ma_int32)pFramesInS16[iFrame*2+0] + (ma_int32)pFramesInS16[iFrame*2+1]) / 2);
- }
- } break;
+ /* For now we're just allocating the decoder backend on the heap. */
+ pMP3 = (ma_mp3*)ma_malloc(sizeof(*pMP3), pAllocationCallbacks);
+ if (pMP3 == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
- case ma_format_s24:
- {
- /* */ ma_uint8* pFramesOutS24 = ( ma_uint8*)pFramesOut;
- const ma_uint8* pFramesInS24 = (const ma_uint8*)pFramesIn;
+ result = ma_mp3_init_file(pFilePath, pConfig, pAllocationCallbacks, pMP3);
+ if (result != MA_SUCCESS) {
+ ma_free(pMP3, pAllocationCallbacks);
+ return result;
+ }
- for (iFrame = 0; iFrame < frameCount; ++iFrame) {
- ma_int64 s24_0 = ma_pcm_sample_s24_to_s32_no_scale(&pFramesInS24[(iFrame*2+0)*3]);
- ma_int64 s24_1 = ma_pcm_sample_s24_to_s32_no_scale(&pFramesInS24[(iFrame*2+1)*3]);
- ma_pcm_sample_s32_to_s24_no_scale((s24_0 + s24_1) / 2, &pFramesOutS24[iFrame*3]);
- }
- } break;
+ *ppBackend = pMP3;
- case ma_format_s32:
- {
- /* */ ma_int32* pFramesOutS32 = ( ma_int32*)pFramesOut;
- const ma_int32* pFramesInS32 = (const ma_int32*)pFramesIn;
+ return MA_SUCCESS;
+}
- for (iFrame = 0; iFrame < frameCount; ++iFrame) {
- pFramesOutS32[iFrame] = (ma_int16)(((ma_int32)pFramesInS32[iFrame*2+0] + (ma_int32)pFramesInS32[iFrame*2+1]) / 2);
- }
- } break;
+static ma_result ma_decoding_backend_init_file_w__mp3(void* pUserData, const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
+{
+ ma_result result;
+ ma_mp3* pMP3;
- case ma_format_f32:
- {
- /* */ float* pFramesOutF32 = ( float*)pFramesOut;
- const float* pFramesInF32 = (const float*)pFramesIn;
+ (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
- for (iFrame = 0; iFrame < frameCount; ++iFrame) {
- pFramesOutF32[iFrame] = (pFramesInF32[iFrame*2+0] + pFramesInF32[iFrame*2+1]) * 0.5f;
- }
- } break;
+ /* For now we're just allocating the decoder backend on the heap. */
+ pMP3 = (ma_mp3*)ma_malloc(sizeof(*pMP3), pAllocationCallbacks);
+ if (pMP3 == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
- default: return MA_INVALID_OPERATION; /* Unknown format. */
+ result = ma_mp3_init_file_w(pFilePath, pConfig, pAllocationCallbacks, pMP3);
+ if (result != MA_SUCCESS) {
+ ma_free(pMP3, pAllocationCallbacks);
+ return result;
}
+ *ppBackend = pMP3;
+
return MA_SUCCESS;
}
-static ma_result ma_channel_converter_process_pcm_frames__weights(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+static ma_result ma_decoding_backend_init_memory__mp3(void* pUserData, const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
{
- ma_uint32 iFrame;
- ma_uint32 iChannelIn;
- ma_uint32 iChannelOut;
-
- MA_ASSERT(pConverter != NULL);
- MA_ASSERT(pFramesOut != NULL);
- MA_ASSERT(pFramesIn != NULL);
+ ma_result result;
+ ma_mp3* pMP3;
- /* This is the more complicated case. Each of the output channels is accumulated with 0 or more input channels. */
+ (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
- /* Clear. */
- ma_zero_memory_64(pFramesOut, frameCount * ma_get_bytes_per_frame(pConverter->format, pConverter->channelsOut));
+ /* For now we're just allocating the decoder backend on the heap. */
+ pMP3 = (ma_mp3*)ma_malloc(sizeof(*pMP3), pAllocationCallbacks);
+ if (pMP3 == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
- /* Accumulate. */
- switch (pConverter->format)
- {
- case ma_format_u8:
- {
- /* */ ma_uint8* pFramesOutU8 = ( ma_uint8*)pFramesOut;
- const ma_uint8* pFramesInU8 = (const ma_uint8*)pFramesIn;
+ result = ma_mp3_init_memory(pData, dataSize, pConfig, pAllocationCallbacks, pMP3);
+ if (result != MA_SUCCESS) {
+ ma_free(pMP3, pAllocationCallbacks);
+ return result;
+ }
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
- for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
- ma_int16 u8_O = ma_pcm_sample_u8_to_s16_no_scale(pFramesOutU8[iFrame*pConverter->channelsOut + iChannelOut]);
- ma_int16 u8_I = ma_pcm_sample_u8_to_s16_no_scale(pFramesInU8 [iFrame*pConverter->channelsIn + iChannelIn ]);
- ma_int32 s = (ma_int32)ma_clamp(u8_O + ((u8_I * pConverter->weights.s16[iChannelIn][iChannelOut]) >> MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT), -128, 127);
- pFramesOutU8[iFrame*pConverter->channelsOut + iChannelOut] = ma_clip_u8((ma_int16)s);
- }
- }
- }
- } break;
+ *ppBackend = pMP3;
- case ma_format_s16:
- {
- /* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut;
- const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn;
+ return MA_SUCCESS;
+}
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
- for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
- ma_int32 s = pFramesOutS16[iFrame*pConverter->channelsOut + iChannelOut];
- s += (pFramesInS16[iFrame*pConverter->channelsIn + iChannelIn] * pConverter->weights.s16[iChannelIn][iChannelOut]) >> MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT;
+static void ma_decoding_backend_uninit__mp3(void* pUserData, ma_data_source* pBackend, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ ma_mp3* pMP3 = (ma_mp3*)pBackend;
- pFramesOutS16[iFrame*pConverter->channelsOut + iChannelOut] = (ma_int16)ma_clamp(s, -32768, 32767);
- }
- }
- }
- } break;
+ (void)pUserData;
- case ma_format_s24:
- {
- /* */ ma_uint8* pFramesOutS24 = ( ma_uint8*)pFramesOut;
- const ma_uint8* pFramesInS24 = (const ma_uint8*)pFramesIn;
+ ma_mp3_uninit(pMP3, pAllocationCallbacks);
+ ma_free(pMP3, pAllocationCallbacks);
+}
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
- for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
- ma_int64 s24_O = ma_pcm_sample_s24_to_s32_no_scale(&pFramesOutS24[(iFrame*pConverter->channelsOut + iChannelOut)*3]);
- ma_int64 s24_I = ma_pcm_sample_s24_to_s32_no_scale(&pFramesInS24 [(iFrame*pConverter->channelsIn + iChannelIn )*3]);
- ma_int64 s24 = (ma_int32)ma_clamp(s24_O + ((s24_I * pConverter->weights.s16[iChannelIn][iChannelOut]) >> MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT), -8388608, 8388607);
- ma_pcm_sample_s32_to_s24_no_scale(s24, &pFramesOutS24[(iFrame*pConverter->channelsOut + iChannelOut)*3]);
- }
- }
- }
- } break;
+static ma_decoding_backend_vtable g_ma_decoding_backend_vtable_mp3 =
+{
+ ma_decoding_backend_init__mp3,
+ ma_decoding_backend_init_file__mp3,
+ ma_decoding_backend_init_file_w__mp3,
+ ma_decoding_backend_init_memory__mp3,
+ ma_decoding_backend_uninit__mp3
+};
- case ma_format_s32:
- {
- /* */ ma_int32* pFramesOutS32 = ( ma_int32*)pFramesOut;
- const ma_int32* pFramesInS32 = (const ma_int32*)pFramesIn;
+static ma_result ma_decoder_init_mp3__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+{
+ return ma_decoder_init_from_vtable(&g_ma_decoding_backend_vtable_mp3, NULL, pConfig, pDecoder);
+}
+#endif /* dr_mp3_h */
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
- for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
- ma_int64 s = pFramesOutS32[iFrame*pConverter->channelsOut + iChannelOut];
- s += ((ma_int64)pFramesInS32[iFrame*pConverter->channelsIn + iChannelIn] * pConverter->weights.s16[iChannelIn][iChannelOut]) >> MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT;
+/* Vorbis */
+#ifdef STB_VORBIS_INCLUDE_STB_VORBIS_H
+#define MA_HAS_VORBIS
- pFramesOutS32[iFrame*pConverter->channelsOut + iChannelOut] = ma_clip_s32(s);
- }
- }
- }
- } break;
+/* The size in bytes of each chunk of data to read from the Vorbis stream. */
+#define MA_VORBIS_DATA_CHUNK_SIZE 4096
- case ma_format_f32:
- {
- /* */ float* pFramesOutF32 = ( float*)pFramesOut;
- const float* pFramesInF32 = (const float*)pFramesIn;
+typedef struct
+{
+ ma_data_source_base ds;
+ ma_read_proc onRead;
+ ma_seek_proc onSeek;
+ ma_tell_proc onTell;
+ void* pReadSeekTellUserData;
+ ma_allocation_callbacks allocationCallbacks; /* Store the allocation callbacks within the structure because we may need to dynamically expand a buffer in ma_stbvorbis_read_pcm_frames() when using push mode. */
+ ma_format format; /* Only f32 is allowed with stb_vorbis. */
+ ma_uint32 channels;
+ ma_uint32 sampleRate;
+ ma_uint64 cursor;
+#if !defined(MA_NO_VORBIS)
+ stb_vorbis* stb;
+ ma_bool32 usingPushMode;
+ struct
+ {
+ ma_uint8* pData;
+ size_t dataSize;
+ size_t dataCapacity;
+ ma_uint32 framesConsumed; /* The number of frames consumed in ppPacketData. */
+ ma_uint32 framesRemaining; /* The number of frames remaining in ppPacketData. */
+ float** ppPacketData;
+ } push;
+#endif
+} ma_stbvorbis;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
- for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
- pFramesOutF32[iFrame*pConverter->channelsOut + iChannelOut] += pFramesInF32[iFrame*pConverter->channelsIn + iChannelIn] * pConverter->weights.f32[iChannelIn][iChannelOut];
- }
- }
- }
- } break;
+MA_API ma_result ma_stbvorbis_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_stbvorbis* pVorbis);
+MA_API ma_result ma_stbvorbis_init_file(const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_stbvorbis* pVorbis);
+MA_API ma_result ma_stbvorbis_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_stbvorbis* pVorbis);
+MA_API void ma_stbvorbis_uninit(ma_stbvorbis* pVorbis, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_stbvorbis_read_pcm_frames(ma_stbvorbis* pVorbis, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
+MA_API ma_result ma_stbvorbis_seek_to_pcm_frame(ma_stbvorbis* pVorbis, ma_uint64 frameIndex);
+MA_API ma_result ma_stbvorbis_get_data_format(ma_stbvorbis* pVorbis, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
+MA_API ma_result ma_stbvorbis_get_cursor_in_pcm_frames(ma_stbvorbis* pVorbis, ma_uint64* pCursor);
+MA_API ma_result ma_stbvorbis_get_length_in_pcm_frames(ma_stbvorbis* pVorbis, ma_uint64* pLength);
- default: return MA_INVALID_OPERATION; /* Unknown format. */
- }
- return MA_SUCCESS;
+static ma_result ma_stbvorbis_ds_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+{
+ return ma_stbvorbis_read_pcm_frames((ma_stbvorbis*)pDataSource, pFramesOut, frameCount, pFramesRead);
}
-MA_API ma_result ma_channel_converter_process_pcm_frames(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
+static ma_result ma_stbvorbis_ds_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
- if (pConverter == NULL) {
- return MA_INVALID_ARGS;
- }
-
- if (pFramesOut == NULL) {
- return MA_INVALID_ARGS;
- }
+ return ma_stbvorbis_seek_to_pcm_frame((ma_stbvorbis*)pDataSource, frameIndex);
+}
- if (pFramesIn == NULL) {
- ma_zero_memory_64(pFramesOut, frameCount * ma_get_bytes_per_frame(pConverter->format, pConverter->channelsOut));
- return MA_SUCCESS;
- }
+static ma_result ma_stbvorbis_ds_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
+{
+ return ma_stbvorbis_get_data_format((ma_stbvorbis*)pDataSource, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
+}
- if (pConverter->isPassthrough) {
- return ma_channel_converter_process_pcm_frames__passthrough(pConverter, pFramesOut, pFramesIn, frameCount);
- } else if (pConverter->isSimpleShuffle) {
- return ma_channel_converter_process_pcm_frames__simple_shuffle(pConverter, pFramesOut, pFramesIn, frameCount);
- } else if (pConverter->isSimpleMonoExpansion) {
- return ma_channel_converter_process_pcm_frames__simple_mono_expansion(pConverter, pFramesOut, pFramesIn, frameCount);
- } else if (pConverter->isStereoToMono) {
- return ma_channel_converter_process_pcm_frames__stereo_to_mono(pConverter, pFramesOut, pFramesIn, frameCount);
- } else {
- return ma_channel_converter_process_pcm_frames__weights(pConverter, pFramesOut, pFramesIn, frameCount);
- }
+static ma_result ma_stbvorbis_ds_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
+{
+ return ma_stbvorbis_get_cursor_in_pcm_frames((ma_stbvorbis*)pDataSource, pCursor);
}
+static ma_result ma_stbvorbis_ds_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
+{
+ return ma_stbvorbis_get_length_in_pcm_frames((ma_stbvorbis*)pDataSource, pLength);
+}
-/**************************************************************************************************************************************************************
+static ma_data_source_vtable g_ma_stbvorbis_ds_vtable =
+{
+ ma_stbvorbis_ds_read,
+ ma_stbvorbis_ds_seek,
+ ma_stbvorbis_ds_get_data_format,
+ ma_stbvorbis_ds_get_cursor,
+ ma_stbvorbis_ds_get_length,
+ NULL, /* onSetLooping */
+ 0
+};
-Data Conversion
-**************************************************************************************************************************************************************/
-MA_API ma_data_converter_config ma_data_converter_config_init_default()
+static ma_result ma_stbvorbis_init_internal(const ma_decoding_backend_config* pConfig, ma_stbvorbis* pVorbis)
{
- ma_data_converter_config config;
- MA_ZERO_OBJECT(&config);
+ ma_result result;
+ ma_data_source_config dataSourceConfig;
- config.ditherMode = ma_dither_mode_none;
- config.resampling.algorithm = ma_resample_algorithm_linear;
- config.resampling.allowDynamicSampleRate = MA_FALSE; /* Disable dynamic sample rates by default because dynamic rate adjustments should be quite rare and it allows an optimization for cases when the in and out sample rates are the same. */
+ (void)pConfig;
- /* Linear resampling defaults. */
- config.resampling.linear.lpfOrder = 1;
- config.resampling.linear.lpfNyquistFactor = 1;
+ if (pVorbis == NULL) {
+ return MA_INVALID_ARGS;
+ }
- /* Speex resampling defaults. */
- config.resampling.speex.quality = 3;
+ MA_ZERO_OBJECT(pVorbis);
+ pVorbis->format = ma_format_f32; /* Only supporting f32. */
- return config;
+ dataSourceConfig = ma_data_source_config_init();
+ dataSourceConfig.vtable = &g_ma_stbvorbis_ds_vtable;
+
+ result = ma_data_source_init(&dataSourceConfig, &pVorbis->ds);
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to initialize the base data source. */
+ }
+
+ return MA_SUCCESS;
}
-MA_API ma_data_converter_config ma_data_converter_config_init(ma_format formatIn, ma_format formatOut, ma_uint32 channelsIn, ma_uint32 channelsOut, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut)
+#if !defined(MA_NO_VORBIS)
+static ma_result ma_stbvorbis_post_init(ma_stbvorbis* pVorbis)
{
- ma_data_converter_config config = ma_data_converter_config_init_default();
- config.formatIn = formatIn;
- config.formatOut = formatOut;
- config.channelsIn = ma_min(channelsIn, MA_MAX_CHANNELS);
- config.channelsOut = ma_min(channelsOut, MA_MAX_CHANNELS);
- config.sampleRateIn = sampleRateIn;
- config.sampleRateOut = sampleRateOut;
+ stb_vorbis_info info;
- return config;
+ MA_ASSERT(pVorbis != NULL);
+
+ info = stb_vorbis_get_info(pVorbis->stb);
+
+ pVorbis->channels = info.channels;
+ pVorbis->sampleRate = info.sample_rate;
+
+ return MA_SUCCESS;
}
+#endif
-MA_API ma_result ma_data_converter_init(const ma_data_converter_config* pConfig, ma_data_converter* pConverter)
+MA_API ma_result ma_stbvorbis_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_stbvorbis* pVorbis)
{
ma_result result;
- ma_format midFormat;
- if (pConverter == NULL) {
- return MA_INVALID_ARGS;
+ result = ma_stbvorbis_init_internal(pConfig, pVorbis);
+ if (result != MA_SUCCESS) {
+ return result;
}
- MA_ZERO_OBJECT(pConverter);
-
- if (pConfig == NULL) {
- return MA_INVALID_ARGS;
+ if (onRead == NULL || onSeek == NULL) {
+ return MA_INVALID_ARGS; /* onRead and onSeek are mandatory. */
}
- pConverter->config = *pConfig;
+ pVorbis->onRead = onRead;
+ pVorbis->onSeek = onSeek;
+ pVorbis->onTell = onTell;
+ pVorbis->pReadSeekTellUserData = pReadSeekTellUserData;
+ ma_allocation_callbacks_init_copy(&pVorbis->allocationCallbacks, pAllocationCallbacks);
- /* Basic validation. */
- if (pConfig->channelsIn < MA_MIN_CHANNELS || pConfig->channelsOut < MA_MIN_CHANNELS ||
- pConfig->channelsIn > MA_MAX_CHANNELS || pConfig->channelsOut > MA_MAX_CHANNELS) {
- return MA_INVALID_ARGS;
- }
+ #if !defined(MA_NO_VORBIS)
+ {
+ /*
+ stb_vorbis lacks a callback based API for it's pulling API which means we're stuck with the
+ pushing API. In order for us to be able to successfully initialize the decoder we need to
+ supply it with enough data. We need to keep loading data until we have enough.
+ */
+ stb_vorbis* stb;
+ size_t dataSize = 0;
+ size_t dataCapacity = 0;
+ ma_uint8* pData = NULL; /* <-- Must be initialized to NULL. */
- /*
- We want to avoid as much data conversion as possible. The channel converter and resampler both support s16 and f32 natively. We need to decide
- on the format to use for this stage. We call this the mid format because it's used in the middle stage of the conversion pipeline. If the output
- format is either s16 or f32 we use that one. If that is not the case it will do the same thing for the input format. If it's neither we just
- use f32.
- */
- /* */ if (pConverter->config.formatOut == ma_format_s16 || pConverter->config.formatOut == ma_format_f32) {
- midFormat = pConverter->config.formatOut;
- } else if (pConverter->config.formatIn == ma_format_s16 || pConverter->config.formatIn == ma_format_f32) {
- midFormat = pConverter->config.formatIn;
- } else {
- midFormat = ma_format_f32;
- }
+ for (;;) {
+ int vorbisError;
+ int consumedDataSize; /* <-- Fill by stb_vorbis_open_pushdata(). */
+ size_t bytesRead;
+ ma_uint8* pNewData;
- /* Channel converter. We always initialize this, but we check if it configures itself as a passthrough to determine whether or not it's needed. */
- {
- ma_uint32 iChannelIn;
- ma_uint32 iChannelOut;
- ma_channel_converter_config channelConverterConfig;
+ /* Allocate memory for the new chunk. */
+ dataCapacity += MA_VORBIS_DATA_CHUNK_SIZE;
+ pNewData = (ma_uint8*)ma_realloc(pData, dataCapacity, pAllocationCallbacks);
+ if (pNewData == NULL) {
+ ma_free(pData, pAllocationCallbacks);
+ return MA_OUT_OF_MEMORY;
+ }
+
+ pData = pNewData;
+
+ /* Read in the next chunk. */
+ result = pVorbis->onRead(pVorbis->pReadSeekTellUserData, ma_offset_ptr(pData, dataSize), (dataCapacity - dataSize), &bytesRead);
+ dataSize += bytesRead;
+
+ if (result != MA_SUCCESS) {
+ ma_free(pData, pAllocationCallbacks);
+ return result;
+ }
- channelConverterConfig = ma_channel_converter_config_init(midFormat, pConverter->config.channelsIn, pConverter->config.channelMapIn, pConverter->config.channelsOut, pConverter->config.channelMapOut, pConverter->config.channelMixMode);
+ /* We have a maximum of 31 bits with stb_vorbis. */
+ if (dataSize > INT_MAX) {
+ ma_free(pData, pAllocationCallbacks);
+ return MA_TOO_BIG;
+ }
- /* Channel weights. */
- for (iChannelIn = 0; iChannelIn < pConverter->config.channelsIn; iChannelIn += 1) {
- for (iChannelOut = 0; iChannelOut < pConverter->config.channelsOut; iChannelOut += 1) {
- channelConverterConfig.weights[iChannelIn][iChannelOut] = pConverter->config.channelWeights[iChannelIn][iChannelOut];
+ stb = stb_vorbis_open_pushdata(pData, (int)dataSize, &consumedDataSize, &vorbisError, NULL);
+ if (stb != NULL) {
+ /*
+ Successfully opened the Vorbis decoder. We might have some leftover unprocessed
+ data so we'll need to move that down to the front.
+ */
+ dataSize -= (size_t)consumedDataSize; /* Consume the data. */
+ MA_MOVE_MEMORY(pData, ma_offset_ptr(pData, consumedDataSize), dataSize);
+ break;
+ } else {
+ /* Failed to open the decoder. */
+ if (vorbisError == VORBIS_need_more_data) {
+ continue;
+ } else {
+ ma_free(pData, pAllocationCallbacks);
+ return MA_ERROR; /* Failed to open the stb_vorbis decoder. */
+ }
}
}
- result = ma_channel_converter_init(&channelConverterConfig, &pConverter->channelConverter);
+ MA_ASSERT(stb != NULL);
+ pVorbis->stb = stb;
+ pVorbis->push.pData = pData;
+ pVorbis->push.dataSize = dataSize;
+ pVorbis->push.dataCapacity = dataCapacity;
+
+ pVorbis->usingPushMode = MA_TRUE;
+
+ result = ma_stbvorbis_post_init(pVorbis);
if (result != MA_SUCCESS) {
+ stb_vorbis_close(pVorbis->stb);
+ ma_free(pData, pAllocationCallbacks);
return result;
}
- /* If the channel converter is not a passthrough we need to enable it. Otherwise we can skip it. */
- if (pConverter->channelConverter.isPassthrough == MA_FALSE) {
- pConverter->hasChannelConverter = MA_TRUE;
- }
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ /* vorbis is disabled. */
+ (void)pAllocationCallbacks;
+ return MA_NOT_IMPLEMENTED;
}
+ #endif
+}
+MA_API ma_result ma_stbvorbis_init_file(const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_stbvorbis* pVorbis)
+{
+ ma_result result;
- /* Always enable dynamic sample rates if the input sample rate is different because we're always going to need a resampler in this case anyway. */
- if (pConverter->config.resampling.allowDynamicSampleRate == MA_FALSE) {
- pConverter->config.resampling.allowDynamicSampleRate = pConverter->config.sampleRateIn != pConverter->config.sampleRateOut;
+ result = ma_stbvorbis_init_internal(pConfig, pVorbis);
+ if (result != MA_SUCCESS) {
+ return result;
}
- /* Resampler. */
- if (pConverter->config.resampling.allowDynamicSampleRate) {
- ma_resampler_config resamplerConfig;
- ma_uint32 resamplerChannels;
+ #if !defined(MA_NO_VORBIS)
+ {
+ (void)pAllocationCallbacks; /* Don't know how to make use of this with stb_vorbis. */
- /* The resampler is the most expensive part of the conversion process, so we need to do it at the stage where the channel count is at it's lowest. */
- if (pConverter->config.channelsIn < pConverter->config.channelsOut) {
- resamplerChannels = pConverter->config.channelsIn;
- } else {
- resamplerChannels = pConverter->config.channelsOut;
+ /* We can use stb_vorbis' pull mode for file based streams. */
+ pVorbis->stb = stb_vorbis_open_filename(pFilePath, NULL, NULL);
+ if (pVorbis->stb == NULL) {
+ return MA_INVALID_FILE;
}
- resamplerConfig = ma_resampler_config_init(midFormat, resamplerChannels, pConverter->config.sampleRateIn, pConverter->config.sampleRateOut, pConverter->config.resampling.algorithm);
- resamplerConfig.linear.lpfOrder = pConverter->config.resampling.linear.lpfOrder;
- resamplerConfig.linear.lpfNyquistFactor = pConverter->config.resampling.linear.lpfNyquistFactor;
- resamplerConfig.speex.quality = pConverter->config.resampling.speex.quality;
+ pVorbis->usingPushMode = MA_FALSE;
- result = ma_resampler_init(&resamplerConfig, &pConverter->resampler);
+ result = ma_stbvorbis_post_init(pVorbis);
if (result != MA_SUCCESS) {
+ stb_vorbis_close(pVorbis->stb);
return result;
}
- pConverter->hasResampler = MA_TRUE;
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ /* vorbis is disabled. */
+ (void)pFilePath;
+ (void)pAllocationCallbacks;
+ return MA_NOT_IMPLEMENTED;
}
+ #endif
+}
+MA_API ma_result ma_stbvorbis_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_stbvorbis* pVorbis)
+{
+ ma_result result;
- /* We can simplify pre- and post-format conversion if we have neither channel conversion nor resampling. */
- if (pConverter->hasChannelConverter == MA_FALSE && pConverter->hasResampler == MA_FALSE) {
- /* We have neither channel conversion nor resampling so we'll only need one of pre- or post-format conversion, or none if the input and output formats are the same. */
- if (pConverter->config.formatIn == pConverter->config.formatOut) {
- /* The formats are the same so we can just pass through. */
- pConverter->hasPreFormatConversion = MA_FALSE;
- pConverter->hasPostFormatConversion = MA_FALSE;
- } else {
- /* The formats are different so we need to do either pre- or post-format conversion. It doesn't matter which. */
- pConverter->hasPreFormatConversion = MA_FALSE;
- pConverter->hasPostFormatConversion = MA_TRUE;
- }
- } else {
- /* We have a channel converter and/or resampler so we'll need channel conversion based on the mid format. */
- if (pConverter->config.formatIn != midFormat) {
- pConverter->hasPreFormatConversion = MA_TRUE;
+ result = ma_stbvorbis_init_internal(pConfig, pVorbis);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ #if !defined(MA_NO_VORBIS)
+ {
+ (void)pAllocationCallbacks;
+
+ /* stb_vorbis uses an int as it's size specifier, restricting it to 32-bit even on 64-bit systems. *sigh*. */
+ if (dataSize > INT_MAX) {
+ return MA_TOO_BIG;
}
- if (pConverter->config.formatOut != midFormat) {
- pConverter->hasPostFormatConversion = MA_TRUE;
+
+ pVorbis->stb = stb_vorbis_open_memory((const unsigned char*)pData, (int)dataSize, NULL, NULL);
+ if (pVorbis->stb == NULL) {
+ return MA_INVALID_FILE;
}
- }
- /* We can enable passthrough optimizations if applicable. Note that we'll only be able to do this if the sample rate is static. */
- if (pConverter->hasPreFormatConversion == MA_FALSE &&
- pConverter->hasPostFormatConversion == MA_FALSE &&
- pConverter->hasChannelConverter == MA_FALSE &&
- pConverter->hasResampler == MA_FALSE) {
- pConverter->isPassthrough = MA_TRUE;
- }
+ pVorbis->usingPushMode = MA_FALSE;
- return MA_SUCCESS;
-}
+ result = ma_stbvorbis_post_init(pVorbis);
+ if (result != MA_SUCCESS) {
+ stb_vorbis_close(pVorbis->stb);
+ return result;
+ }
-MA_API void ma_data_converter_uninit(ma_data_converter* pConverter)
-{
- if (pConverter == NULL) {
- return;
+ return MA_SUCCESS;
}
-
- if (pConverter->hasResampler) {
- ma_resampler_uninit(&pConverter->resampler);
+ #else
+ {
+ /* vorbis is disabled. */
+ (void)pData;
+ (void)dataSize;
+ (void)pAllocationCallbacks;
+ return MA_NOT_IMPLEMENTED;
}
+ #endif
}
-static ma_result ma_data_converter_process_pcm_frames__passthrough(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+MA_API void ma_stbvorbis_uninit(ma_stbvorbis* pVorbis, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_uint64 frameCountIn;
- ma_uint64 frameCountOut;
- ma_uint64 frameCount;
-
- MA_ASSERT(pConverter != NULL);
-
- frameCountIn = 0;
- if (pFrameCountIn != NULL) {
- frameCountIn = *pFrameCountIn;
- }
-
- frameCountOut = 0;
- if (pFrameCountOut != NULL) {
- frameCountOut = *pFrameCountOut;
+ if (pVorbis == NULL) {
+ return;
}
- frameCount = ma_min(frameCountIn, frameCountOut);
+ #if !defined(MA_NO_VORBIS)
+ {
+ stb_vorbis_close(pVorbis->stb);
- if (pFramesOut != NULL) {
- if (pFramesIn != NULL) {
- ma_copy_memory_64(pFramesOut, pFramesIn, frameCount * ma_get_bytes_per_frame(pConverter->config.formatOut, pConverter->config.channelsOut));
- } else {
- ma_zero_memory_64(pFramesOut, frameCount * ma_get_bytes_per_frame(pConverter->config.formatOut, pConverter->config.channelsOut));
+ /* We'll have to clear some memory if we're using push mode. */
+ if (pVorbis->usingPushMode) {
+ ma_free(pVorbis->push.pData, pAllocationCallbacks);
}
}
-
- if (pFrameCountIn != NULL) {
- *pFrameCountIn = frameCount;
- }
- if (pFrameCountOut != NULL) {
- *pFrameCountOut = frameCount;
+ #else
+ {
+ /* vorbis is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
}
+ #endif
- return MA_SUCCESS;
+ ma_data_source_uninit(&pVorbis->ds);
}
-static ma_result ma_data_converter_process_pcm_frames__format_only(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+MA_API ma_result ma_stbvorbis_read_pcm_frames(ma_stbvorbis* pVorbis, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
- ma_uint64 frameCountIn;
- ma_uint64 frameCountOut;
- ma_uint64 frameCount;
-
- MA_ASSERT(pConverter != NULL);
+ if (pFramesRead != NULL) {
+ *pFramesRead = 0;
+ }
- frameCountIn = 0;
- if (pFrameCountIn != NULL) {
- frameCountIn = *pFrameCountIn;
+ if (frameCount == 0) {
+ return MA_INVALID_ARGS;
}
- frameCountOut = 0;
- if (pFrameCountOut != NULL) {
- frameCountOut = *pFrameCountOut;
+ if (pVorbis == NULL) {
+ return MA_INVALID_ARGS;
}
- frameCount = ma_min(frameCountIn, frameCountOut);
+ #if !defined(MA_NO_VORBIS)
+ {
+ /* We always use floating point format. */
+ ma_result result = MA_SUCCESS; /* Must be initialized to MA_SUCCESS. */
+ ma_uint64 totalFramesRead = 0;
+ ma_format format;
+ ma_uint32 channels;
- if (pFramesOut != NULL) {
- if (pFramesIn != NULL) {
- ma_convert_pcm_frames_format(pFramesOut, pConverter->config.formatOut, pFramesIn, pConverter->config.formatIn, frameCount, pConverter->config.channelsIn, pConverter->config.ditherMode);
- } else {
- ma_zero_memory_64(pFramesOut, frameCount * ma_get_bytes_per_frame(pConverter->config.formatOut, pConverter->config.channelsOut));
- }
- }
+ ma_stbvorbis_get_data_format(pVorbis, &format, &channels, NULL, NULL, 0);
- if (pFrameCountIn != NULL) {
- *pFrameCountIn = frameCount;
- }
- if (pFrameCountOut != NULL) {
- *pFrameCountOut = frameCount;
- }
+ if (format == ma_format_f32) {
+ /* We read differently depending on whether or not we're using push mode. */
+ if (pVorbis->usingPushMode) {
+ /* Push mode. This is the complex case. */
+ float* pFramesOutF32 = (float*)pFramesOut;
- return MA_SUCCESS;
-}
+ while (totalFramesRead < frameCount) {
+ /* The first thing to do is read from any already-cached frames. */
+ ma_uint32 framesToReadFromCache = (ma_uint32)ma_min(pVorbis->push.framesRemaining, (frameCount - totalFramesRead)); /* Safe cast because pVorbis->framesRemaining is 32-bit. */
+ /* The output pointer can be null in which case we just treate it as a seek. */
+ if (pFramesOut != NULL) {
+ ma_uint64 iFrame;
+ for (iFrame = 0; iFrame < framesToReadFromCache; iFrame += 1) {
+ ma_uint32 iChannel;
+ for (iChannel = 0; iChannel < pVorbis->channels; iChannel += 1) {
+ pFramesOutF32[iChannel] = pVorbis->push.ppPacketData[iChannel][pVorbis->push.framesConsumed + iFrame];
+ }
-static ma_result ma_data_converter_process_pcm_frames__resample_with_format_conversion(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
-{
- ma_result result = MA_SUCCESS;
- ma_uint64 frameCountIn;
- ma_uint64 frameCountOut;
- ma_uint64 framesProcessedIn;
- ma_uint64 framesProcessedOut;
+ pFramesOutF32 += pVorbis->channels;
+ }
+ }
- MA_ASSERT(pConverter != NULL);
+ /* Update pointers and counters. */
+ pVorbis->push.framesConsumed += framesToReadFromCache;
+ pVorbis->push.framesRemaining -= framesToReadFromCache;
+ totalFramesRead += framesToReadFromCache;
- frameCountIn = 0;
- if (pFrameCountIn != NULL) {
- frameCountIn = *pFrameCountIn;
- }
+ /* Don't bother reading any more frames right now if we've just finished loading. */
+ if (totalFramesRead == frameCount) {
+ break;
+ }
- frameCountOut = 0;
- if (pFrameCountOut != NULL) {
- frameCountOut = *pFrameCountOut;
- }
+ MA_ASSERT(pVorbis->push.framesRemaining == 0);
- framesProcessedIn = 0;
- framesProcessedOut = 0;
+ /* Getting here means we've run out of cached frames. We'll need to load some more. */
+ for (;;) {
+ int samplesRead = 0;
+ int consumedDataSize;
- while (framesProcessedOut < frameCountOut) {
- ma_uint8 pTempBufferOut[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
- const ma_uint32 tempBufferOutCap = sizeof(pTempBufferOut) / ma_get_bytes_per_frame(pConverter->resampler.config.format, pConverter->resampler.config.channels);
- const void* pFramesInThisIteration;
- /* */ void* pFramesOutThisIteration;
- ma_uint64 frameCountInThisIteration;
- ma_uint64 frameCountOutThisIteration;
+ /* We need to case dataSize to an int, so make sure we can do it safely. */
+ if (pVorbis->push.dataSize > INT_MAX) {
+ break; /* Too big. */
+ }
- if (pFramesIn != NULL) {
- pFramesInThisIteration = ma_offset_ptr(pFramesIn, framesProcessedIn * ma_get_bytes_per_frame(pConverter->config.formatIn, pConverter->config.channelsIn));
- } else {
- pFramesInThisIteration = NULL;
- }
+ consumedDataSize = stb_vorbis_decode_frame_pushdata(pVorbis->stb, pVorbis->push.pData, (int)pVorbis->push.dataSize, NULL, &pVorbis->push.ppPacketData, &samplesRead);
+ if (consumedDataSize != 0) {
+ /* Successfully decoded a Vorbis frame. Consume the data. */
+ pVorbis->push.dataSize -= (size_t)consumedDataSize;
+ MA_MOVE_MEMORY(pVorbis->push.pData, ma_offset_ptr(pVorbis->push.pData, consumedDataSize), pVorbis->push.dataSize);
- if (pFramesOut != NULL) {
- pFramesOutThisIteration = ma_offset_ptr(pFramesOut, framesProcessedOut * ma_get_bytes_per_frame(pConverter->config.formatOut, pConverter->config.channelsOut));
- } else {
- pFramesOutThisIteration = NULL;
- }
+ pVorbis->push.framesConsumed = 0;
+ pVorbis->push.framesRemaining = samplesRead;
- /* Do a pre format conversion if necessary. */
- if (pConverter->hasPreFormatConversion) {
- ma_uint8 pTempBufferIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
- const ma_uint32 tempBufferInCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->resampler.config.format, pConverter->resampler.config.channels);
+ break;
+ } else {
+ /* Not enough data. Read more. */
+ size_t bytesRead;
- frameCountInThisIteration = (frameCountIn - framesProcessedIn);
- if (frameCountInThisIteration > tempBufferInCap) {
- frameCountInThisIteration = tempBufferInCap;
- }
+ /* Expand the data buffer if necessary. */
+ if (pVorbis->push.dataCapacity == pVorbis->push.dataSize) {
+ size_t newCap = pVorbis->push.dataCapacity + MA_VORBIS_DATA_CHUNK_SIZE;
+ ma_uint8* pNewData;
- if (pConverter->hasPostFormatConversion) {
- if (frameCountInThisIteration > tempBufferOutCap) {
- frameCountInThisIteration = tempBufferOutCap;
- }
- }
+ pNewData = (ma_uint8*)ma_realloc(pVorbis->push.pData, newCap, &pVorbis->allocationCallbacks);
+ if (pNewData == NULL) {
+ result = MA_OUT_OF_MEMORY;
+ break;
+ }
- if (pFramesInThisIteration != NULL) {
- ma_convert_pcm_frames_format(pTempBufferIn, pConverter->resampler.config.format, pFramesInThisIteration, pConverter->config.formatIn, frameCountInThisIteration, pConverter->config.channelsIn, pConverter->config.ditherMode);
- } else {
- MA_ZERO_MEMORY(pTempBufferIn, sizeof(pTempBufferIn));
- }
+ pVorbis->push.pData = pNewData;
+ pVorbis->push.dataCapacity = newCap;
+ }
- frameCountOutThisIteration = (frameCountOut - framesProcessedOut);
+ /* We should have enough room to load some data. */
+ result = pVorbis->onRead(pVorbis->pReadSeekTellUserData, ma_offset_ptr(pVorbis->push.pData, pVorbis->push.dataSize), (pVorbis->push.dataCapacity - pVorbis->push.dataSize), &bytesRead);
+ pVorbis->push.dataSize += bytesRead;
- if (pConverter->hasPostFormatConversion) {
- /* Both input and output conversion required. Output to the temp buffer. */
- if (frameCountOutThisIteration > tempBufferOutCap) {
- frameCountOutThisIteration = tempBufferOutCap;
+ if (result != MA_SUCCESS) {
+ break; /* Failed to read any data. Get out. */
+ }
+ }
+ }
+
+ /* If we don't have a success code at this point it means we've encounted an error or the end of the file has been reached (probably the latter). */
+ if (result != MA_SUCCESS) {
+ break;
+ }
}
-
- result = ma_resampler_process_pcm_frames(&pConverter->resampler, pTempBufferIn, &frameCountInThisIteration, pTempBufferOut, &frameCountOutThisIteration);
} else {
- /* Only pre-format required. Output straight to the output buffer. */
- result = ma_resampler_process_pcm_frames(&pConverter->resampler, pTempBufferIn, &frameCountInThisIteration, pFramesOutThisIteration, &frameCountOutThisIteration);
- }
+ /* Pull mode. This is the simple case, but we still need to run in a loop because stb_vorbis loves using 32-bit instead of 64-bit. */
+ while (totalFramesRead < frameCount) {
+ ma_uint64 framesRemaining = (frameCount - totalFramesRead);
+ int framesRead;
- if (result != MA_SUCCESS) {
- break;
- }
- } else {
- /* No pre-format required. Just read straight from the input buffer. */
- MA_ASSERT(pConverter->hasPostFormatConversion == MA_TRUE);
+ if (framesRemaining > INT_MAX) {
+ framesRemaining = INT_MAX;
+ }
- frameCountInThisIteration = (frameCountIn - framesProcessedIn);
- frameCountOutThisIteration = (frameCountOut - framesProcessedOut);
- if (frameCountOutThisIteration > tempBufferOutCap) {
- frameCountOutThisIteration = tempBufferOutCap;
- }
+ framesRead = stb_vorbis_get_samples_float_interleaved(pVorbis->stb, channels, (float*)ma_offset_pcm_frames_ptr(pFramesOut, totalFramesRead, format, channels), (int)framesRemaining * channels); /* Safe cast. */
+ totalFramesRead += framesRead;
- result = ma_resampler_process_pcm_frames(&pConverter->resampler, pFramesInThisIteration, &frameCountInThisIteration, pTempBufferOut, &frameCountOutThisIteration);
- if (result != MA_SUCCESS) {
- break;
+ if (framesRead < (int)framesRemaining) {
+ break; /* Nothing left to read. Get out. */
+ }
+ }
}
+ } else {
+ result = MA_INVALID_ARGS;
}
- /* If we are doing a post format conversion we need to do that now. */
- if (pConverter->hasPostFormatConversion) {
- if (pFramesOutThisIteration != NULL) {
- ma_convert_pcm_frames_format(pFramesOutThisIteration, pConverter->config.formatOut, pTempBufferOut, pConverter->resampler.config.format, frameCountOutThisIteration, pConverter->resampler.config.channels, pConverter->config.ditherMode);
- }
- }
+ pVorbis->cursor += totalFramesRead;
- framesProcessedIn += frameCountInThisIteration;
- framesProcessedOut += frameCountOutThisIteration;
+ if (totalFramesRead == 0) {
+ result = MA_AT_END;
+ }
- MA_ASSERT(framesProcessedIn <= frameCountIn);
- MA_ASSERT(framesProcessedOut <= frameCountOut);
+ if (pFramesRead != NULL) {
+ *pFramesRead = totalFramesRead;
+ }
- if (frameCountOutThisIteration == 0) {
- break; /* Consumed all of our input data. */
+ if (result == MA_SUCCESS && totalFramesRead == 0) {
+ result = MA_AT_END;
}
- }
- if (pFrameCountIn != NULL) {
- *pFrameCountIn = framesProcessedIn;
- }
- if (pFrameCountOut != NULL) {
- *pFrameCountOut = framesProcessedOut;
+ return result;
}
+ #else
+ {
+ /* vorbis is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
- return result;
-}
-
-static ma_result ma_data_converter_process_pcm_frames__resample_only(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
-{
- MA_ASSERT(pConverter != NULL);
+ (void)pFramesOut;
+ (void)frameCount;
+ (void)pFramesRead;
- if (pConverter->hasPreFormatConversion == MA_FALSE && pConverter->hasPostFormatConversion == MA_FALSE) {
- /* Neither pre- nor post-format required. This is simple case where only resampling is required. */
- return ma_resampler_process_pcm_frames(&pConverter->resampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
- } else {
- /* Format conversion required. */
- return ma_data_converter_process_pcm_frames__resample_with_format_conversion(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
+ return MA_NOT_IMPLEMENTED;
}
+ #endif
}
-static ma_result ma_data_converter_process_pcm_frames__channels_only(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+MA_API ma_result ma_stbvorbis_seek_to_pcm_frame(ma_stbvorbis* pVorbis, ma_uint64 frameIndex)
{
- ma_result result;
- ma_uint64 frameCountIn;
- ma_uint64 frameCountOut;
- ma_uint64 frameCount;
-
- MA_ASSERT(pConverter != NULL);
-
- frameCountIn = 0;
- if (pFrameCountIn != NULL) {
- frameCountIn = *pFrameCountIn;
- }
-
- frameCountOut = 0;
- if (pFrameCountOut != NULL) {
- frameCountOut = *pFrameCountOut;
+ if (pVorbis == NULL) {
+ return MA_INVALID_ARGS;
}
- frameCount = ma_min(frameCountIn, frameCountOut);
-
- if (pConverter->hasPreFormatConversion == MA_FALSE && pConverter->hasPostFormatConversion == MA_FALSE) {
- /* No format conversion required. */
- result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pFramesOut, pFramesIn, frameCount);
- if (result != MA_SUCCESS) {
- return result;
- }
- } else {
- /* Format conversion required. */
- ma_uint64 framesProcessed = 0;
+ #if !defined(MA_NO_VORBIS)
+ {
+ /* Different seeking methods depending on whether or not we're using push mode. */
+ if (pVorbis->usingPushMode) {
+ /* Push mode. This is the complex case. */
+ ma_result result;
+ float buffer[4096];
- while (framesProcessed < frameCount) {
- ma_uint8 pTempBufferOut[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
- const ma_uint32 tempBufferOutCap = sizeof(pTempBufferOut) / ma_get_bytes_per_frame(pConverter->channelConverter.format, pConverter->channelConverter.channelsOut);
- const void* pFramesInThisIteration;
- /* */ void* pFramesOutThisIteration;
- ma_uint64 frameCountThisIteration;
+ /*
+ This is terribly inefficient because stb_vorbis does not have a good seeking solution with it's push API. Currently this just performs
+ a full decode right from the start of the stream. Later on I'll need to write a layer that goes through all of the Ogg pages until we
+ find the one containing the sample we need. Then we know exactly where to seek for stb_vorbis.
- if (pFramesIn != NULL) {
- pFramesInThisIteration = ma_offset_ptr(pFramesIn, framesProcessed * ma_get_bytes_per_frame(pConverter->config.formatIn, pConverter->config.channelsIn));
- } else {
- pFramesInThisIteration = NULL;
- }
+ TODO: Use seeking logic documented for stb_vorbis_flush_pushdata().
+ */
- if (pFramesOut != NULL) {
- pFramesOutThisIteration = ma_offset_ptr(pFramesOut, framesProcessed * ma_get_bytes_per_frame(pConverter->config.formatOut, pConverter->config.channelsOut));
- } else {
- pFramesOutThisIteration = NULL;
+ /* Seek to the start of the file to begin with. */
+ result = pVorbis->onSeek(pVorbis->pReadSeekTellUserData, 0, ma_seek_origin_start);
+ if (result != MA_SUCCESS) {
+ return result;
}
- /* Do a pre format conversion if necessary. */
- if (pConverter->hasPreFormatConversion) {
- ma_uint8 pTempBufferIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
- const ma_uint32 tempBufferInCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->channelConverter.format, pConverter->channelConverter.channelsIn);
-
- frameCountThisIteration = (frameCount - framesProcessed);
- if (frameCountThisIteration > tempBufferInCap) {
- frameCountThisIteration = tempBufferInCap;
- }
+ stb_vorbis_flush_pushdata(pVorbis->stb);
+ pVorbis->push.framesRemaining = 0;
+ pVorbis->push.dataSize = 0;
- if (pConverter->hasPostFormatConversion) {
- if (frameCountThisIteration > tempBufferOutCap) {
- frameCountThisIteration = tempBufferOutCap;
- }
- }
+ /* Move the cursor back to the start. We'll increment this in the loop below. */
+ pVorbis->cursor = 0;
- if (pFramesInThisIteration != NULL) {
- ma_convert_pcm_frames_format(pTempBufferIn, pConverter->channelConverter.format, pFramesInThisIteration, pConverter->config.formatIn, frameCountThisIteration, pConverter->config.channelsIn, pConverter->config.ditherMode);
- } else {
- MA_ZERO_MEMORY(pTempBufferIn, sizeof(pTempBufferIn));
+ while (pVorbis->cursor < frameIndex) {
+ ma_uint64 framesRead;
+ ma_uint64 framesToRead = ma_countof(buffer)/pVorbis->channels;
+ if (framesToRead > (frameIndex - pVorbis->cursor)) {
+ framesToRead = (frameIndex - pVorbis->cursor);
}
- if (pConverter->hasPostFormatConversion) {
- /* Both input and output conversion required. Output to the temp buffer. */
- result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pTempBufferOut, pTempBufferIn, frameCountThisIteration);
- } else {
- /* Only pre-format required. Output straight to the output buffer. */
- result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pFramesOutThisIteration, pTempBufferIn, frameCountThisIteration);
- }
+ result = ma_stbvorbis_read_pcm_frames(pVorbis, buffer, framesToRead, &framesRead);
+ pVorbis->cursor += framesRead;
if (result != MA_SUCCESS) {
- break;
- }
- } else {
- /* No pre-format required. Just read straight from the input buffer. */
- MA_ASSERT(pConverter->hasPostFormatConversion == MA_TRUE);
-
- frameCountThisIteration = (frameCount - framesProcessed);
- if (frameCountThisIteration > tempBufferOutCap) {
- frameCountThisIteration = tempBufferOutCap;
+ return result;
}
+ }
+ } else {
+ /* Pull mode. This is the simple case. */
+ int vorbisResult;
- result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pTempBufferOut, pFramesInThisIteration, frameCountThisIteration);
- if (result != MA_SUCCESS) {
- break;
- }
+ if (frameIndex > UINT_MAX) {
+ return MA_INVALID_ARGS; /* Trying to seek beyond the 32-bit maximum of stb_vorbis. */
}
- /* If we are doing a post format conversion we need to do that now. */
- if (pConverter->hasPostFormatConversion) {
- if (pFramesOutThisIteration != NULL) {
- ma_convert_pcm_frames_format(pFramesOutThisIteration, pConverter->config.formatOut, pTempBufferOut, pConverter->channelConverter.format, frameCountThisIteration, pConverter->channelConverter.channelsOut, pConverter->config.ditherMode);
- }
+ vorbisResult = stb_vorbis_seek(pVorbis->stb, (unsigned int)frameIndex); /* Safe cast. */
+ if (vorbisResult == 0) {
+ return MA_ERROR; /* See failed. */
}
- framesProcessed += frameCountThisIteration;
+ pVorbis->cursor = frameIndex;
}
- }
- if (pFrameCountIn != NULL) {
- *pFrameCountIn = frameCount;
- }
- if (pFrameCountOut != NULL) {
- *pFrameCountOut = frameCount;
+ return MA_SUCCESS;
}
+ #else
+ {
+ /* vorbis is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
- return MA_SUCCESS;
+ (void)frameIndex;
+
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
}
-static ma_result ma_data_converter_process_pcm_frames__resampling_first(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+MA_API ma_result ma_stbvorbis_get_data_format(ma_stbvorbis* pVorbis, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
- ma_result result;
- ma_uint64 frameCountIn;
- ma_uint64 frameCountOut;
- ma_uint64 framesProcessedIn;
- ma_uint64 framesProcessedOut;
- ma_uint8 pTempBufferIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In resampler format. */
- ma_uint64 tempBufferInCap;
- ma_uint8 pTempBufferMid[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In resampler format, channel converter input format. */
- ma_uint64 tempBufferMidCap;
- ma_uint8 pTempBufferOut[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In channel converter output format. */
- ma_uint64 tempBufferOutCap;
-
- MA_ASSERT(pConverter != NULL);
- MA_ASSERT(pConverter->resampler.config.format == pConverter->channelConverter.format);
- MA_ASSERT(pConverter->resampler.config.channels == pConverter->channelConverter.channelsIn);
- MA_ASSERT(pConverter->resampler.config.channels < pConverter->channelConverter.channelsOut);
-
- frameCountIn = 0;
- if (pFrameCountIn != NULL) {
- frameCountIn = *pFrameCountIn;
+ /* Defaults for safety. */
+ if (pFormat != NULL) {
+ *pFormat = ma_format_unknown;
}
-
- frameCountOut = 0;
- if (pFrameCountOut != NULL) {
- frameCountOut = *pFrameCountOut;
+ if (pChannels != NULL) {
+ *pChannels = 0;
+ }
+ if (pSampleRate != NULL) {
+ *pSampleRate = 0;
+ }
+ if (pChannelMap != NULL) {
+ MA_ZERO_MEMORY(pChannelMap, sizeof(*pChannelMap) * channelMapCap);
}
- framesProcessedIn = 0;
- framesProcessedOut = 0;
-
- tempBufferInCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->resampler.config.format, pConverter->resampler.config.channels);
- tempBufferMidCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->resampler.config.format, pConverter->resampler.config.channels);
- tempBufferOutCap = sizeof(pTempBufferOut) / ma_get_bytes_per_frame(pConverter->channelConverter.format, pConverter->channelConverter.channelsOut);
-
- while (framesProcessedOut < frameCountOut) {
- ma_uint64 frameCountInThisIteration;
- ma_uint64 frameCountOutThisIteration;
- const void* pRunningFramesIn = NULL;
- void* pRunningFramesOut = NULL;
- const void* pResampleBufferIn;
- void* pChannelsBufferOut;
-
- if (pFramesIn != NULL) {
- pRunningFramesIn = ma_offset_ptr(pFramesIn, framesProcessedIn * ma_get_bytes_per_frame(pConverter->config.formatIn, pConverter->config.channelsIn));
- }
- if (pFramesOut != NULL) {
- pRunningFramesOut = ma_offset_ptr(pFramesOut, framesProcessedOut * ma_get_bytes_per_frame(pConverter->config.formatOut, pConverter->config.channelsOut));
- }
-
- /* Run input data through the resampler and output it to the temporary buffer. */
- frameCountInThisIteration = (frameCountIn - framesProcessedIn);
-
- if (pConverter->hasPreFormatConversion) {
- if (frameCountInThisIteration > tempBufferInCap) {
- frameCountInThisIteration = tempBufferInCap;
- }
- }
-
- frameCountOutThisIteration = (frameCountOut - framesProcessedOut);
- if (frameCountOutThisIteration > tempBufferMidCap) {
- frameCountOutThisIteration = tempBufferMidCap;
- }
-
- /* We can't read more frames than can fit in the output buffer. */
- if (pConverter->hasPostFormatConversion) {
- if (frameCountOutThisIteration > tempBufferOutCap) {
- frameCountOutThisIteration = tempBufferOutCap;
- }
- }
-
- /* We need to ensure we don't try to process too many input frames that we run out of room in the output buffer. If this happens we'll end up glitching. */
- {
- ma_uint64 requiredInputFrameCount = ma_resampler_get_required_input_frame_count(&pConverter->resampler, frameCountOutThisIteration);
- if (frameCountInThisIteration > requiredInputFrameCount) {
- frameCountInThisIteration = requiredInputFrameCount;
- }
- }
+ if (pVorbis == NULL) {
+ return MA_INVALID_OPERATION;
+ }
- if (pConverter->hasPreFormatConversion) {
- if (pFramesIn != NULL) {
- ma_convert_pcm_frames_format(pTempBufferIn, pConverter->resampler.config.format, pRunningFramesIn, pConverter->config.formatIn, frameCountInThisIteration, pConverter->config.channelsIn, pConverter->config.ditherMode);
- pResampleBufferIn = pTempBufferIn;
- } else {
- pResampleBufferIn = NULL;
- }
- } else {
- pResampleBufferIn = pRunningFramesIn;
- }
+ if (pFormat != NULL) {
+ *pFormat = pVorbis->format;
+ }
- result = ma_resampler_process_pcm_frames(&pConverter->resampler, pResampleBufferIn, &frameCountInThisIteration, pTempBufferMid, &frameCountOutThisIteration);
- if (result != MA_SUCCESS) {
- return result;
+ #if !defined(MA_NO_VORBIS)
+ {
+ if (pChannels != NULL) {
+ *pChannels = pVorbis->channels;
}
-
- /*
- The input data has been resampled so now we need to run it through the channel converter. The input data is always contained in pTempBufferMid. We only need to do
- this part if we have an output buffer.
- */
- if (pFramesOut != NULL) {
- if (pConverter->hasPostFormatConversion) {
- pChannelsBufferOut = pTempBufferOut;
- } else {
- pChannelsBufferOut = pRunningFramesOut;
- }
-
- result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pChannelsBufferOut, pTempBufferMid, frameCountOutThisIteration);
- if (result != MA_SUCCESS) {
- return result;
- }
-
- /* Finally we do post format conversion. */
- if (pConverter->hasPostFormatConversion) {
- ma_convert_pcm_frames_format(pRunningFramesOut, pConverter->config.formatOut, pChannelsBufferOut, pConverter->channelConverter.format, frameCountOutThisIteration, pConverter->channelConverter.channelsOut, pConverter->config.ditherMode);
- }
+ if (pSampleRate != NULL) {
+ *pSampleRate = pVorbis->sampleRate;
}
-
- framesProcessedIn += frameCountInThisIteration;
- framesProcessedOut += frameCountOutThisIteration;
-
- MA_ASSERT(framesProcessedIn <= frameCountIn);
- MA_ASSERT(framesProcessedOut <= frameCountOut);
-
- if (frameCountOutThisIteration == 0) {
- break; /* Consumed all of our input data. */
+ if (pChannelMap != NULL) {
+ ma_channel_map_init_standard(ma_standard_channel_map_vorbis, pChannelMap, channelMapCap, pVorbis->channels);
}
- }
- if (pFrameCountIn != NULL) {
- *pFrameCountIn = framesProcessedIn;
+ return MA_SUCCESS;
}
- if (pFrameCountOut != NULL) {
- *pFrameCountOut = framesProcessedOut;
+ #else
+ {
+ /* vorbis is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
+ return MA_NOT_IMPLEMENTED;
}
-
- return MA_SUCCESS;
+ #endif
}
-static ma_result ma_data_converter_process_pcm_frames__channels_first(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+MA_API ma_result ma_stbvorbis_get_cursor_in_pcm_frames(ma_stbvorbis* pVorbis, ma_uint64* pCursor)
{
- ma_result result;
- ma_uint64 frameCountIn;
- ma_uint64 frameCountOut;
- ma_uint64 framesProcessedIn;
- ma_uint64 framesProcessedOut;
- ma_uint8 pTempBufferIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In resampler format. */
- ma_uint64 tempBufferInCap;
- ma_uint8 pTempBufferMid[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In resampler format, channel converter input format. */
- ma_uint64 tempBufferMidCap;
- ma_uint8 pTempBufferOut[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In channel converter output format. */
- ma_uint64 tempBufferOutCap;
-
- MA_ASSERT(pConverter != NULL);
- MA_ASSERT(pConverter->resampler.config.format == pConverter->channelConverter.format);
- MA_ASSERT(pConverter->resampler.config.channels == pConverter->channelConverter.channelsOut);
- MA_ASSERT(pConverter->resampler.config.channels < pConverter->channelConverter.channelsIn);
-
- frameCountIn = 0;
- if (pFrameCountIn != NULL) {
- frameCountIn = *pFrameCountIn;
+ if (pCursor == NULL) {
+ return MA_INVALID_ARGS;
}
- frameCountOut = 0;
- if (pFrameCountOut != NULL) {
- frameCountOut = *pFrameCountOut;
- }
+ *pCursor = 0; /* Safety. */
- framesProcessedIn = 0;
- framesProcessedOut = 0;
+ if (pVorbis == NULL) {
+ return MA_INVALID_ARGS;
+ }
- tempBufferInCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->channelConverter.format, pConverter->channelConverter.channelsIn);
- tempBufferMidCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->channelConverter.format, pConverter->channelConverter.channelsOut);
- tempBufferOutCap = sizeof(pTempBufferOut) / ma_get_bytes_per_frame(pConverter->resampler.config.format, pConverter->resampler.config.channels);
+ #if !defined(MA_NO_VORBIS)
+ {
+ *pCursor = pVorbis->cursor;
- while (framesProcessedOut < frameCountOut) {
- ma_uint64 frameCountInThisIteration;
- ma_uint64 frameCountOutThisIteration;
- const void* pRunningFramesIn = NULL;
- void* pRunningFramesOut = NULL;
- const void* pChannelsBufferIn;
- void* pResampleBufferOut;
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ /* vorbis is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
+}
- if (pFramesIn != NULL) {
- pRunningFramesIn = ma_offset_ptr(pFramesIn, framesProcessedIn * ma_get_bytes_per_frame(pConverter->config.formatIn, pConverter->config.channelsIn));
- }
- if (pFramesOut != NULL) {
- pRunningFramesOut = ma_offset_ptr(pFramesOut, framesProcessedOut * ma_get_bytes_per_frame(pConverter->config.formatOut, pConverter->config.channelsOut));
- }
+MA_API ma_result ma_stbvorbis_get_length_in_pcm_frames(ma_stbvorbis* pVorbis, ma_uint64* pLength)
+{
+ if (pLength == NULL) {
+ return MA_INVALID_ARGS;
+ }
- /* Run input data through the channel converter and output it to the temporary buffer. */
- frameCountInThisIteration = (frameCountIn - framesProcessedIn);
+ *pLength = 0; /* Safety. */
- if (pConverter->hasPreFormatConversion) {
- if (frameCountInThisIteration > tempBufferInCap) {
- frameCountInThisIteration = tempBufferInCap;
- }
+ if (pVorbis == NULL) {
+ return MA_INVALID_ARGS;
+ }
- if (pRunningFramesIn != NULL) {
- ma_convert_pcm_frames_format(pTempBufferIn, pConverter->channelConverter.format, pRunningFramesIn, pConverter->config.formatIn, frameCountInThisIteration, pConverter->config.channelsIn, pConverter->config.ditherMode);
- pChannelsBufferIn = pTempBufferIn;
- } else {
- pChannelsBufferIn = NULL;
- }
+ #if !defined(MA_NO_VORBIS)
+ {
+ if (pVorbis->usingPushMode) {
+ *pLength = 0; /* I don't know of a good way to determine this reliably with stb_vorbis and push mode. */
} else {
- pChannelsBufferIn = pRunningFramesIn;
- }
-
- /*
- We can't convert more frames than will fit in the output buffer. We shouldn't actually need to do this check because the channel count is always reduced
- in this case which means we should always have capacity, but I'm leaving it here just for safety for future maintenance.
- */
- if (frameCountInThisIteration > tempBufferMidCap) {
- frameCountInThisIteration = tempBufferMidCap;
- }
-
- /*
- Make sure we don't read any more input frames than we need to fill the output frame count. If we do this we will end up in a situation where we lose some
- input samples and will end up glitching.
- */
- frameCountOutThisIteration = (frameCountOut - framesProcessedOut);
- if (frameCountOutThisIteration > tempBufferMidCap) {
- frameCountOutThisIteration = tempBufferMidCap;
+ *pLength = stb_vorbis_stream_length_in_samples(pVorbis->stb);
}
- if (pConverter->hasPostFormatConversion) {
- ma_uint64 requiredInputFrameCount;
-
- if (frameCountOutThisIteration > tempBufferOutCap) {
- frameCountOutThisIteration = tempBufferOutCap;
- }
+ return MA_SUCCESS;
+ }
+ #else
+ {
+ /* vorbis is disabled. Should never hit this since initialization would have failed. */
+ MA_ASSERT(MA_FALSE);
+ return MA_NOT_IMPLEMENTED;
+ }
+ #endif
+}
- requiredInputFrameCount = ma_resampler_get_required_input_frame_count(&pConverter->resampler, frameCountOutThisIteration);
- if (frameCountInThisIteration > requiredInputFrameCount) {
- frameCountInThisIteration = requiredInputFrameCount;
- }
- }
- result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pTempBufferMid, pChannelsBufferIn, frameCountInThisIteration);
- if (result != MA_SUCCESS) {
- return result;
- }
+static ma_result ma_decoding_backend_init__stbvorbis(void* pUserData, ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
+{
+ ma_result result;
+ ma_stbvorbis* pVorbis;
+ (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
- /* At this point we have converted the channels to the output channel count which we now need to resample. */
- if (pConverter->hasPostFormatConversion) {
- pResampleBufferOut = pTempBufferOut;
- } else {
- pResampleBufferOut = pRunningFramesOut;
- }
+ /* For now we're just allocating the decoder backend on the heap. */
+ pVorbis = (ma_stbvorbis*)ma_malloc(sizeof(*pVorbis), pAllocationCallbacks);
+ if (pVorbis == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
- result = ma_resampler_process_pcm_frames(&pConverter->resampler, pTempBufferMid, &frameCountInThisIteration, pResampleBufferOut, &frameCountOutThisIteration);
- if (result != MA_SUCCESS) {
- return result;
- }
+ result = ma_stbvorbis_init(onRead, onSeek, onTell, pReadSeekTellUserData, pConfig, pAllocationCallbacks, pVorbis);
+ if (result != MA_SUCCESS) {
+ ma_free(pVorbis, pAllocationCallbacks);
+ return result;
+ }
- /* Finally we can do the post format conversion. */
- if (pConverter->hasPostFormatConversion) {
- if (pRunningFramesOut != NULL) {
- ma_convert_pcm_frames_format(pRunningFramesOut, pConverter->config.formatOut, pResampleBufferOut, pConverter->resampler.config.format, frameCountOutThisIteration, pConverter->config.channelsOut, pConverter->config.ditherMode);
- }
- }
+ *ppBackend = pVorbis;
- framesProcessedIn += frameCountInThisIteration;
- framesProcessedOut += frameCountOutThisIteration;
+ return MA_SUCCESS;
+}
- MA_ASSERT(framesProcessedIn <= frameCountIn);
- MA_ASSERT(framesProcessedOut <= frameCountOut);
+static ma_result ma_decoding_backend_init_file__stbvorbis(void* pUserData, const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
+{
+ ma_result result;
+ ma_stbvorbis* pVorbis;
- if (frameCountOutThisIteration == 0) {
- break; /* Consumed all of our input data. */
- }
- }
+ (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
- if (pFrameCountIn != NULL) {
- *pFrameCountIn = framesProcessedIn;
+ /* For now we're just allocating the decoder backend on the heap. */
+ pVorbis = (ma_stbvorbis*)ma_malloc(sizeof(*pVorbis), pAllocationCallbacks);
+ if (pVorbis == NULL) {
+ return MA_OUT_OF_MEMORY;
}
- if (pFrameCountOut != NULL) {
- *pFrameCountOut = framesProcessedOut;
+
+ result = ma_stbvorbis_init_file(pFilePath, pConfig, pAllocationCallbacks, pVorbis);
+ if (result != MA_SUCCESS) {
+ ma_free(pVorbis, pAllocationCallbacks);
+ return result;
}
+ *ppBackend = pVorbis;
+
return MA_SUCCESS;
}
-MA_API ma_result ma_data_converter_process_pcm_frames(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
+static ma_result ma_decoding_backend_init_memory__stbvorbis(void* pUserData, const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
{
- if (pConverter == NULL) {
- return MA_INVALID_ARGS;
+ ma_result result;
+ ma_stbvorbis* pVorbis;
+
+ (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
+
+ /* For now we're just allocating the decoder backend on the heap. */
+ pVorbis = (ma_stbvorbis*)ma_malloc(sizeof(*pVorbis), pAllocationCallbacks);
+ if (pVorbis == NULL) {
+ return MA_OUT_OF_MEMORY;
}
- if (pConverter->isPassthrough) {
- return ma_data_converter_process_pcm_frames__passthrough(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
+ result = ma_stbvorbis_init_memory(pData, dataSize, pConfig, pAllocationCallbacks, pVorbis);
+ if (result != MA_SUCCESS) {
+ ma_free(pVorbis, pAllocationCallbacks);
+ return result;
}
- /*
- Here is where the real work is done. Getting here means we're not using a passthrough and we need to move the data through each of the relevant stages. The order
- of our stages depends on the input and output channel count. If the input channels is less than the output channels we want to do sample rate conversion first so
- that it has less work (resampling is the most expensive part of format conversion).
- */
- if (pConverter->config.channelsIn < pConverter->config.channelsOut) {
- /* Do resampling first, if necessary. */
- MA_ASSERT(pConverter->hasChannelConverter == MA_TRUE);
+ *ppBackend = pVorbis;
- if (pConverter->hasResampler) {
- /* Resampling first. */
- return ma_data_converter_process_pcm_frames__resampling_first(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
- } else {
- /* Resampling not required. */
- return ma_data_converter_process_pcm_frames__channels_only(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
- }
- } else {
- /* Do channel conversion first, if necessary. */
- if (pConverter->hasChannelConverter) {
- if (pConverter->hasResampler) {
- /* Channel routing first. */
- return ma_data_converter_process_pcm_frames__channels_first(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
- } else {
- /* Resampling not required. */
- return ma_data_converter_process_pcm_frames__channels_only(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
- }
- } else {
- /* Channel routing not required. */
- if (pConverter->hasResampler) {
- /* Resampling only. */
- return ma_data_converter_process_pcm_frames__resample_only(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
- } else {
- /* No channel routing nor resampling required. Just format conversion. */
- return ma_data_converter_process_pcm_frames__format_only(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
- }
- }
- }
+ return MA_SUCCESS;
}
-MA_API ma_result ma_data_converter_set_rate(ma_data_converter* pConverter, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut)
+static void ma_decoding_backend_uninit__stbvorbis(void* pUserData, ma_data_source* pBackend, const ma_allocation_callbacks* pAllocationCallbacks)
{
- if (pConverter == NULL) {
- return MA_INVALID_ARGS;
- }
+ ma_stbvorbis* pVorbis = (ma_stbvorbis*)pBackend;
- if (pConverter->hasResampler == MA_FALSE) {
- return MA_INVALID_OPERATION; /* Dynamic resampling not enabled. */
- }
+ (void)pUserData;
- return ma_resampler_set_rate(&pConverter->resampler, sampleRateIn, sampleRateOut);
+ ma_stbvorbis_uninit(pVorbis, pAllocationCallbacks);
+ ma_free(pVorbis, pAllocationCallbacks);
}
-MA_API ma_result ma_data_converter_set_rate_ratio(ma_data_converter* pConverter, float ratioInOut)
+static ma_decoding_backend_vtable g_ma_decoding_backend_vtable_stbvorbis =
{
- if (pConverter == NULL) {
- return MA_INVALID_ARGS;
- }
-
- if (pConverter->hasResampler == MA_FALSE) {
- return MA_INVALID_OPERATION; /* Dynamic resampling not enabled. */
- }
+ ma_decoding_backend_init__stbvorbis,
+ ma_decoding_backend_init_file__stbvorbis,
+ NULL, /* onInitFileW() */
+ ma_decoding_backend_init_memory__stbvorbis,
+ ma_decoding_backend_uninit__stbvorbis
+};
- return ma_resampler_set_rate_ratio(&pConverter->resampler, ratioInOut);
+static ma_result ma_decoder_init_vorbis__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+{
+ return ma_decoder_init_from_vtable(&g_ma_decoding_backend_vtable_stbvorbis, NULL, pConfig, pDecoder);
}
+#endif /* STB_VORBIS_INCLUDE_STB_VORBIS_H */
+
-MA_API ma_uint64 ma_data_converter_get_required_input_frame_count(const ma_data_converter* pConverter, ma_uint64 outputFrameCount)
+
+static ma_result ma_decoder__init_allocation_callbacks(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
- if (pConverter == NULL) {
- return 0;
- }
+ MA_ASSERT(pDecoder != NULL);
- if (pConverter->hasResampler) {
- return ma_resampler_get_required_input_frame_count(&pConverter->resampler, outputFrameCount);
+ if (pConfig != NULL) {
+ return ma_allocation_callbacks_init_copy(&pDecoder->allocationCallbacks, &pConfig->allocationCallbacks);
} else {
- return outputFrameCount; /* 1:1 */
+ pDecoder->allocationCallbacks = ma_allocation_callbacks_init_default();
+ return MA_SUCCESS;
}
}
-MA_API ma_uint64 ma_data_converter_get_expected_output_frame_count(const ma_data_converter* pConverter, ma_uint64 inputFrameCount)
+static ma_result ma_decoder__data_source_on_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
- if (pConverter == NULL) {
- return 0;
- }
+ return ma_decoder_read_pcm_frames((ma_decoder*)pDataSource, pFramesOut, frameCount, pFramesRead);
+}
- if (pConverter->hasResampler) {
- return ma_resampler_get_expected_output_frame_count(&pConverter->resampler, inputFrameCount);
- } else {
- return inputFrameCount; /* 1:1 */
- }
+static ma_result ma_decoder__data_source_on_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
+{
+ return ma_decoder_seek_to_pcm_frame((ma_decoder*)pDataSource, frameIndex);
}
-MA_API ma_uint64 ma_data_converter_get_input_latency(const ma_data_converter* pConverter)
+static ma_result ma_decoder__data_source_on_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
- if (pConverter == NULL) {
- return 0;
- }
+ return ma_decoder_get_data_format((ma_decoder*)pDataSource, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
+}
- if (pConverter->hasResampler) {
- return ma_resampler_get_input_latency(&pConverter->resampler);
- }
+static ma_result ma_decoder__data_source_on_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
+{
+ return ma_decoder_get_cursor_in_pcm_frames((ma_decoder*)pDataSource, pCursor);
+}
- return 0; /* No latency without a resampler. */
+static ma_result ma_decoder__data_source_on_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
+{
+ return ma_decoder_get_length_in_pcm_frames((ma_decoder*)pDataSource, pLength);
}
-MA_API ma_uint64 ma_data_converter_get_output_latency(const ma_data_converter* pConverter)
+static ma_data_source_vtable g_ma_decoder_data_source_vtable =
{
- if (pConverter == NULL) {
- return 0;
+ ma_decoder__data_source_on_read,
+ ma_decoder__data_source_on_seek,
+ ma_decoder__data_source_on_get_data_format,
+ ma_decoder__data_source_on_get_cursor,
+ ma_decoder__data_source_on_get_length,
+ NULL, /* onSetLooping */
+ 0
+};
+
+static ma_result ma_decoder__preinit(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, ma_decoder_tell_proc onTell, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+{
+ ma_result result;
+ ma_data_source_config dataSourceConfig;
+
+ MA_ASSERT(pConfig != NULL);
+
+ if (pDecoder == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pConverter->hasResampler) {
- return ma_resampler_get_output_latency(&pConverter->resampler);
+ MA_ZERO_OBJECT(pDecoder);
+
+ if (onRead == NULL || onSeek == NULL) {
+ return MA_INVALID_ARGS;
}
- return 0; /* No latency without a resampler. */
-}
+ dataSourceConfig = ma_data_source_config_init();
+ dataSourceConfig.vtable = &g_ma_decoder_data_source_vtable;
+ result = ma_data_source_init(&dataSourceConfig, &pDecoder->ds);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ pDecoder->onRead = onRead;
+ pDecoder->onSeek = onSeek;
+ pDecoder->onTell = onTell;
+ pDecoder->pUserData = pUserData;
-/**************************************************************************************************************************************************************
+ result = ma_decoder__init_allocation_callbacks(pConfig, pDecoder);
+ if (result != MA_SUCCESS) {
+ ma_data_source_uninit(&pDecoder->ds);
+ return result;
+ }
-Channel Maps
+ return MA_SUCCESS;
+}
-**************************************************************************************************************************************************************/
-MA_API ma_channel ma_channel_map_get_default_channel(ma_uint32 channelCount, ma_uint32 channelIndex)
+static ma_result ma_decoder__postinit(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
- if (channelCount == 0 || channelIndex >= channelCount) {
- return MA_CHANNEL_NONE;
- }
+ ma_result result;
- /* This is the Microsoft channel map. Based off the speaker configurations mentioned here: https://docs.microsoft.com/en-us/windows-hardware/drivers/ddi/content/ksmedia/ns-ksmedia-ksaudio_channel_config */
- switch (channelCount)
- {
- case 0: return MA_CHANNEL_NONE;
+ result = ma_decoder__init_data_converter(pDecoder, pConfig);
- case 1:
- {
- return MA_CHANNEL_MONO;
- } break;
+ /* If we failed post initialization we need to uninitialize the decoder before returning to prevent a memory leak. */
+ if (result != MA_SUCCESS) {
+ ma_decoder_uninit(pDecoder);
+ return result;
+ }
- case 2:
- {
- switch (channelIndex) {
- case 0: return MA_CHANNEL_FRONT_LEFT;
- case 1: return MA_CHANNEL_FRONT_RIGHT;
- }
- } break;
+ return result;
+}
- case 3: /* No defined, but best guess. */
- {
- switch (channelIndex) {
- case 0: return MA_CHANNEL_FRONT_LEFT;
- case 1: return MA_CHANNEL_FRONT_RIGHT;
- case 2: return MA_CHANNEL_FRONT_CENTER;
- }
- } break;
- case 4:
- {
- switch (channelIndex) {
- #ifndef MA_USE_QUAD_MICROSOFT_CHANNEL_MAP
- /* Surround. Using the Surround profile has the advantage of the 3rd channel (MA_CHANNEL_FRONT_CENTER) mapping nicely with higher channel counts. */
- case 0: return MA_CHANNEL_FRONT_LEFT;
- case 1: return MA_CHANNEL_FRONT_RIGHT;
- case 2: return MA_CHANNEL_FRONT_CENTER;
- case 3: return MA_CHANNEL_BACK_CENTER;
- #else
- /* Quad. */
- case 0: return MA_CHANNEL_FRONT_LEFT;
- case 1: return MA_CHANNEL_FRONT_RIGHT;
- case 2: return MA_CHANNEL_BACK_LEFT;
- case 3: return MA_CHANNEL_BACK_RIGHT;
- #endif
- }
- } break;
+static ma_result ma_decoder_init__internal(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+{
+ ma_result result = MA_NO_BACKEND;
- case 5: /* Not defined, but best guess. */
- {
- switch (channelIndex) {
- case 0: return MA_CHANNEL_FRONT_LEFT;
- case 1: return MA_CHANNEL_FRONT_RIGHT;
- case 2: return MA_CHANNEL_FRONT_CENTER;
- case 3: return MA_CHANNEL_BACK_LEFT;
- case 4: return MA_CHANNEL_BACK_RIGHT;
- }
- } break;
+ MA_ASSERT(pConfig != NULL);
+ MA_ASSERT(pDecoder != NULL);
- case 6:
- {
- switch (channelIndex) {
- case 0: return MA_CHANNEL_FRONT_LEFT;
- case 1: return MA_CHANNEL_FRONT_RIGHT;
- case 2: return MA_CHANNEL_FRONT_CENTER;
- case 3: return MA_CHANNEL_LFE;
- case 4: return MA_CHANNEL_SIDE_LEFT;
- case 5: return MA_CHANNEL_SIDE_RIGHT;
- }
- } break;
+ /* Silence some warnings in the case that we don't have any decoder backends enabled. */
+ (void)onRead;
+ (void)onSeek;
+ (void)pUserData;
- case 7: /* Not defined, but best guess. */
- {
- switch (channelIndex) {
- case 0: return MA_CHANNEL_FRONT_LEFT;
- case 1: return MA_CHANNEL_FRONT_RIGHT;
- case 2: return MA_CHANNEL_FRONT_CENTER;
- case 3: return MA_CHANNEL_LFE;
- case 4: return MA_CHANNEL_BACK_CENTER;
- case 5: return MA_CHANNEL_SIDE_LEFT;
- case 6: return MA_CHANNEL_SIDE_RIGHT;
- }
- } break;
- case 8:
- default:
- {
- switch (channelIndex) {
- case 0: return MA_CHANNEL_FRONT_LEFT;
- case 1: return MA_CHANNEL_FRONT_RIGHT;
- case 2: return MA_CHANNEL_FRONT_CENTER;
- case 3: return MA_CHANNEL_LFE;
- case 4: return MA_CHANNEL_BACK_LEFT;
- case 5: return MA_CHANNEL_BACK_RIGHT;
- case 6: return MA_CHANNEL_SIDE_LEFT;
- case 7: return MA_CHANNEL_SIDE_RIGHT;
- }
- } break;
- }
+ /* If we've specified a specific encoding type, try that first. */
+ if (pConfig->encodingFormat != ma_encoding_format_unknown) {
+ #ifdef MA_HAS_WAV
+ if (pConfig->encodingFormat == ma_encoding_format_wav) {
+ result = ma_decoder_init_wav__internal(pConfig, pDecoder);
+ }
+ #endif
+ #ifdef MA_HAS_FLAC
+ if (pConfig->encodingFormat == ma_encoding_format_flac) {
+ result = ma_decoder_init_flac__internal(pConfig, pDecoder);
+ }
+ #endif
+ #ifdef MA_HAS_MP3
+ if (pConfig->encodingFormat == ma_encoding_format_mp3) {
+ result = ma_decoder_init_mp3__internal(pConfig, pDecoder);
+ }
+ #endif
+ #ifdef MA_HAS_VORBIS
+ if (pConfig->encodingFormat == ma_encoding_format_vorbis) {
+ result = ma_decoder_init_vorbis__internal(pConfig, pDecoder);
+ }
+ #endif
- if (channelCount > 8) {
- if (channelIndex < 32) { /* We have 32 AUX channels. */
- return (ma_channel)(MA_CHANNEL_AUX_0 + (channelIndex - 8));
+ /* If we weren't able to initialize the decoder, seek back to the start to give the next attempts a clean start. */
+ if (result != MA_SUCCESS) {
+ onSeek(pDecoder, 0, ma_seek_origin_start);
}
}
- /* Getting here means we don't know how to map the channel position so just return MA_CHANNEL_NONE. */
- return MA_CHANNEL_NONE;
-}
+ if (result != MA_SUCCESS) {
+ /* Getting here means we couldn't load a specific decoding backend based on the encoding format. */
-MA_API ma_channel ma_channel_map_get_channel(const ma_channel* pChannelMap, ma_uint32 channelCount, ma_uint32 channelIndex)
-{
- if (pChannelMap == NULL) {
- return ma_channel_map_get_default_channel(channelCount, channelIndex);
- } else {
- if (channelIndex >= channelCount) {
- return MA_CHANNEL_NONE;
+ /*
+ We use trial and error to open a decoder. We prioritize custom decoders so that if they
+ implement the same encoding format they take priority over the built-in decoders.
+ */
+ if (result != MA_SUCCESS) {
+ result = ma_decoder_init_custom__internal(pConfig, pDecoder);
+ if (result != MA_SUCCESS) {
+ onSeek(pDecoder, 0, ma_seek_origin_start);
+ }
}
- return pChannelMap[channelIndex];
- }
-}
+ /*
+ If we get to this point and we still haven't found a decoder, and the caller has requested a
+ specific encoding format, there's no hope for it. Abort.
+ */
+ if (pConfig->encodingFormat != ma_encoding_format_unknown) {
+ return MA_NO_BACKEND;
+ }
+ #ifdef MA_HAS_WAV
+ if (result != MA_SUCCESS) {
+ result = ma_decoder_init_wav__internal(pConfig, pDecoder);
+ if (result != MA_SUCCESS) {
+ onSeek(pDecoder, 0, ma_seek_origin_start);
+ }
+ }
+ #endif
+ #ifdef MA_HAS_FLAC
+ if (result != MA_SUCCESS) {
+ result = ma_decoder_init_flac__internal(pConfig, pDecoder);
+ if (result != MA_SUCCESS) {
+ onSeek(pDecoder, 0, ma_seek_origin_start);
+ }
+ }
+ #endif
+ #ifdef MA_HAS_MP3
+ if (result != MA_SUCCESS) {
+ result = ma_decoder_init_mp3__internal(pConfig, pDecoder);
+ if (result != MA_SUCCESS) {
+ onSeek(pDecoder, 0, ma_seek_origin_start);
+ }
+ }
+ #endif
+ #ifdef MA_HAS_VORBIS
+ if (result != MA_SUCCESS) {
+ result = ma_decoder_init_vorbis__internal(pConfig, pDecoder);
+ if (result != MA_SUCCESS) {
+ onSeek(pDecoder, 0, ma_seek_origin_start);
+ }
+ }
+ #endif
+ }
-MA_API void ma_channel_map_init_blank(ma_uint32 channels, ma_channel* pChannelMap)
-{
- if (pChannelMap == NULL) {
- return;
+ if (result != MA_SUCCESS) {
+ return result;
}
- MA_ZERO_MEMORY(pChannelMap, sizeof(*pChannelMap) * channels);
+ return ma_decoder__postinit(pConfig, pDecoder);
}
-static void ma_get_standard_channel_map_microsoft(ma_uint32 channels, ma_channel* pChannelMap)
+MA_API ma_result ma_decoder_init(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
- /* Based off the speaker configurations mentioned here: https://docs.microsoft.com/en-us/windows-hardware/drivers/ddi/content/ksmedia/ns-ksmedia-ksaudio_channel_config */
- switch (channels)
- {
- case 1:
- {
- pChannelMap[0] = MA_CHANNEL_MONO;
- } break;
+ ma_decoder_config config;
+ ma_result result;
- case 2:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- } break;
+ config = ma_decoder_config_init_copy(pConfig);
- case 3: /* Not defined, but best guess. */
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_FRONT_CENTER;
- } break;
+ result = ma_decoder__preinit(onRead, onSeek, NULL, pUserData, &config, pDecoder);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- case 4:
- {
-#ifndef MA_USE_QUAD_MICROSOFT_CHANNEL_MAP
- /* Surround. Using the Surround profile has the advantage of the 3rd channel (MA_CHANNEL_FRONT_CENTER) mapping nicely with higher channel counts. */
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[3] = MA_CHANNEL_BACK_CENTER;
-#else
- /* Quad. */
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[3] = MA_CHANNEL_BACK_RIGHT;
-#endif
- } break;
+ return ma_decoder_init__internal(onRead, onSeek, pUserData, &config, pDecoder);
+}
- case 5: /* Not defined, but best guess. */
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[3] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[4] = MA_CHANNEL_BACK_RIGHT;
- } break;
- case 6:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[3] = MA_CHANNEL_LFE;
- pChannelMap[4] = MA_CHANNEL_SIDE_LEFT;
- pChannelMap[5] = MA_CHANNEL_SIDE_RIGHT;
- } break;
+static ma_result ma_decoder__on_read_memory(ma_decoder* pDecoder, void* pBufferOut, size_t bytesToRead, size_t* pBytesRead)
+{
+ size_t bytesRemaining;
- case 7: /* Not defined, but best guess. */
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[3] = MA_CHANNEL_LFE;
- pChannelMap[4] = MA_CHANNEL_BACK_CENTER;
- pChannelMap[5] = MA_CHANNEL_SIDE_LEFT;
- pChannelMap[6] = MA_CHANNEL_SIDE_RIGHT;
- } break;
+ MA_ASSERT(pDecoder->data.memory.dataSize >= pDecoder->data.memory.currentReadPos);
- case 8:
- default:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[3] = MA_CHANNEL_LFE;
- pChannelMap[4] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[5] = MA_CHANNEL_BACK_RIGHT;
- pChannelMap[6] = MA_CHANNEL_SIDE_LEFT;
- pChannelMap[7] = MA_CHANNEL_SIDE_RIGHT;
- } break;
+ if (pBytesRead != NULL) {
+ *pBytesRead = 0;
}
- /* Remainder. */
- if (channels > 8) {
- ma_uint32 iChannel;
- for (iChannel = 8; iChannel < channels; ++iChannel) {
- if (iChannel < MA_MAX_CHANNELS) {
- pChannelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-8));
- } else {
- pChannelMap[iChannel] = MA_CHANNEL_NONE;
- }
- }
+ bytesRemaining = pDecoder->data.memory.dataSize - pDecoder->data.memory.currentReadPos;
+ if (bytesToRead > bytesRemaining) {
+ bytesToRead = bytesRemaining;
}
-}
-static void ma_get_standard_channel_map_alsa(ma_uint32 channels, ma_channel* pChannelMap)
-{
- switch (channels)
- {
- case 1:
- {
- pChannelMap[0] = MA_CHANNEL_MONO;
- } break;
+ if (bytesRemaining == 0) {
+ return MA_AT_END;
+ }
- case 2:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- } break;
+ if (bytesToRead > 0) {
+ MA_COPY_MEMORY(pBufferOut, pDecoder->data.memory.pData + pDecoder->data.memory.currentReadPos, bytesToRead);
+ pDecoder->data.memory.currentReadPos += bytesToRead;
+ }
- case 3:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_FRONT_CENTER;
- } break;
+ if (pBytesRead != NULL) {
+ *pBytesRead = bytesToRead;
+ }
- case 4:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[3] = MA_CHANNEL_BACK_RIGHT;
- } break;
+ return MA_SUCCESS;
+}
- case 5:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[3] = MA_CHANNEL_BACK_RIGHT;
- pChannelMap[4] = MA_CHANNEL_FRONT_CENTER;
- } break;
+static ma_result ma_decoder__on_seek_memory(ma_decoder* pDecoder, ma_int64 byteOffset, ma_seek_origin origin)
+{
+ if (byteOffset > 0 && (ma_uint64)byteOffset > MA_SIZE_MAX) {
+ return MA_BAD_SEEK;
+ }
- case 6:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[3] = MA_CHANNEL_BACK_RIGHT;
- pChannelMap[4] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[5] = MA_CHANNEL_LFE;
- } break;
+ if (origin == ma_seek_origin_current) {
+ if (byteOffset > 0) {
+ if (pDecoder->data.memory.currentReadPos + byteOffset > pDecoder->data.memory.dataSize) {
+ byteOffset = (ma_int64)(pDecoder->data.memory.dataSize - pDecoder->data.memory.currentReadPos); /* Trying to seek too far forward. */
+ }
- case 7:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[3] = MA_CHANNEL_BACK_RIGHT;
- pChannelMap[4] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[5] = MA_CHANNEL_LFE;
- pChannelMap[6] = MA_CHANNEL_BACK_CENTER;
- } break;
+ pDecoder->data.memory.currentReadPos += (size_t)byteOffset;
+ } else {
+ if (pDecoder->data.memory.currentReadPos < (size_t)-byteOffset) {
+ byteOffset = -(ma_int64)pDecoder->data.memory.currentReadPos; /* Trying to seek too far backwards. */
+ }
- case 8:
- default:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[3] = MA_CHANNEL_BACK_RIGHT;
- pChannelMap[4] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[5] = MA_CHANNEL_LFE;
- pChannelMap[6] = MA_CHANNEL_SIDE_LEFT;
- pChannelMap[7] = MA_CHANNEL_SIDE_RIGHT;
- } break;
- }
+ pDecoder->data.memory.currentReadPos -= (size_t)-byteOffset;
+ }
+ } else {
+ if (origin == ma_seek_origin_end) {
+ if (byteOffset < 0) {
+ byteOffset = -byteOffset;
+ }
- /* Remainder. */
- if (channels > 8) {
- ma_uint32 iChannel;
- for (iChannel = 8; iChannel < channels; ++iChannel) {
- if (iChannel < MA_MAX_CHANNELS) {
- pChannelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-8));
+ if (byteOffset > (ma_int64)pDecoder->data.memory.dataSize) {
+ pDecoder->data.memory.currentReadPos = 0; /* Trying to seek too far back. */
+ } else {
+ pDecoder->data.memory.currentReadPos = pDecoder->data.memory.dataSize - (size_t)byteOffset;
+ }
+ } else {
+ if ((size_t)byteOffset <= pDecoder->data.memory.dataSize) {
+ pDecoder->data.memory.currentReadPos = (size_t)byteOffset;
} else {
- pChannelMap[iChannel] = MA_CHANNEL_NONE;
+ pDecoder->data.memory.currentReadPos = pDecoder->data.memory.dataSize; /* Trying to seek too far forward. */
}
}
}
+
+ return MA_SUCCESS;
}
-static void ma_get_standard_channel_map_rfc3551(ma_uint32 channels, ma_channel* pChannelMap)
+static ma_result ma_decoder__on_tell_memory(ma_decoder* pDecoder, ma_int64* pCursor)
{
- switch (channels)
- {
- case 1:
- {
- pChannelMap[0] = MA_CHANNEL_MONO;
- } break;
-
- case 2:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- } break;
-
- case 3:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_FRONT_CENTER;
- } break;
+ MA_ASSERT(pDecoder != NULL);
+ MA_ASSERT(pCursor != NULL);
- case 4:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[2] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[3] = MA_CHANNEL_BACK_CENTER;
- } break;
+ *pCursor = (ma_int64)pDecoder->data.memory.currentReadPos;
- case 5:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[3] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[4] = MA_CHANNEL_BACK_RIGHT;
- } break;
+ return MA_SUCCESS;
+}
- case 6:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_SIDE_LEFT;
- pChannelMap[2] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[3] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[4] = MA_CHANNEL_SIDE_RIGHT;
- pChannelMap[5] = MA_CHANNEL_BACK_CENTER;
- } break;
+static ma_result ma_decoder__preinit_memory(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+{
+ ma_result result = ma_decoder__preinit(ma_decoder__on_read_memory, ma_decoder__on_seek_memory, ma_decoder__on_tell_memory, NULL, pConfig, pDecoder);
+ if (result != MA_SUCCESS) {
+ return result;
}
- /* Remainder. */
- if (channels > 8) {
- ma_uint32 iChannel;
- for (iChannel = 6; iChannel < channels; ++iChannel) {
- if (iChannel < MA_MAX_CHANNELS) {
- pChannelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-6));
- } else {
- pChannelMap[iChannel] = MA_CHANNEL_NONE;
- }
- }
+ if (pData == NULL || dataSize == 0) {
+ return MA_INVALID_ARGS;
}
+
+ pDecoder->data.memory.pData = (const ma_uint8*)pData;
+ pDecoder->data.memory.dataSize = dataSize;
+ pDecoder->data.memory.currentReadPos = 0;
+
+ (void)pConfig;
+ return MA_SUCCESS;
}
-static void ma_get_standard_channel_map_flac(ma_uint32 channels, ma_channel* pChannelMap)
+MA_API ma_result ma_decoder_init_memory(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
- switch (channels)
- {
- case 1:
- {
- pChannelMap[0] = MA_CHANNEL_MONO;
- } break;
+ ma_decoder_config config;
+ ma_result result;
- case 2:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- } break;
+ config = ma_decoder_config_init_copy(pConfig); /* Make sure the config is not NULL. */
- case 3:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_FRONT_CENTER;
- } break;
+ result = ma_decoder__preinit_memory(pData, dataSize, &config, pDecoder);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- case 4:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[3] = MA_CHANNEL_BACK_RIGHT;
- } break;
+ return ma_decoder_init__internal(ma_decoder__on_read_memory, ma_decoder__on_seek_memory, NULL, &config, pDecoder);
+}
- case 5:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[3] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[4] = MA_CHANNEL_BACK_RIGHT;
- } break;
- case 6:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[3] = MA_CHANNEL_LFE;
- pChannelMap[4] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[5] = MA_CHANNEL_BACK_RIGHT;
- } break;
+#if defined(MA_HAS_WAV) || \
+ defined(MA_HAS_MP3) || \
+ defined(MA_HAS_FLAC) || \
+ defined(MA_HAS_VORBIS) || \
+ defined(MA_HAS_OPUS)
+#define MA_HAS_PATH_API
+#endif
- case 7:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[3] = MA_CHANNEL_LFE;
- pChannelMap[4] = MA_CHANNEL_BACK_CENTER;
- pChannelMap[5] = MA_CHANNEL_SIDE_LEFT;
- pChannelMap[6] = MA_CHANNEL_SIDE_RIGHT;
- } break;
+#if defined(MA_HAS_PATH_API)
+static const char* ma_path_file_name(const char* path)
+{
+ const char* fileName;
- case 8:
- default:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[3] = MA_CHANNEL_LFE;
- pChannelMap[4] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[5] = MA_CHANNEL_BACK_RIGHT;
- pChannelMap[6] = MA_CHANNEL_SIDE_LEFT;
- pChannelMap[7] = MA_CHANNEL_SIDE_RIGHT;
- } break;
+ if (path == NULL) {
+ return NULL;
}
- /* Remainder. */
- if (channels > 8) {
- ma_uint32 iChannel;
- for (iChannel = 8; iChannel < channels; ++iChannel) {
- if (iChannel < MA_MAX_CHANNELS) {
- pChannelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-8));
- } else {
- pChannelMap[iChannel] = MA_CHANNEL_NONE;
- }
+ fileName = path;
+
+ /* We just loop through the path until we find the last slash. */
+ while (path[0] != '\0') {
+ if (path[0] == '/' || path[0] == '\\') {
+ fileName = path;
}
- }
-}
-static void ma_get_standard_channel_map_vorbis(ma_uint32 channels, ma_channel* pChannelMap)
-{
- /* In Vorbis' type 0 channel mapping, the first two channels are not always the standard left/right - it will have the center speaker where the right usually goes. Why?! */
- switch (channels)
- {
- case 1:
- {
- pChannelMap[0] = MA_CHANNEL_MONO;
- } break;
+ path += 1;
+ }
- case 2:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- } break;
+ /* At this point the file name is sitting on a slash, so just move forward. */
+ while (fileName[0] != '\0' && (fileName[0] == '/' || fileName[0] == '\\')) {
+ fileName += 1;
+ }
- case 3:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[2] = MA_CHANNEL_FRONT_RIGHT;
- } break;
+ return fileName;
+}
- case 4:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[3] = MA_CHANNEL_BACK_RIGHT;
- } break;
+static const wchar_t* ma_path_file_name_w(const wchar_t* path)
+{
+ const wchar_t* fileName;
- case 5:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[2] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[3] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[4] = MA_CHANNEL_BACK_RIGHT;
- } break;
+ if (path == NULL) {
+ return NULL;
+ }
- case 6:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[2] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[3] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[4] = MA_CHANNEL_BACK_RIGHT;
- pChannelMap[5] = MA_CHANNEL_LFE;
- } break;
+ fileName = path;
- case 7:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[2] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[3] = MA_CHANNEL_SIDE_LEFT;
- pChannelMap[4] = MA_CHANNEL_SIDE_RIGHT;
- pChannelMap[5] = MA_CHANNEL_BACK_CENTER;
- pChannelMap[6] = MA_CHANNEL_LFE;
- } break;
+ /* We just loop through the path until we find the last slash. */
+ while (path[0] != '\0') {
+ if (path[0] == '/' || path[0] == '\\') {
+ fileName = path;
+ }
- case 8:
- default:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[2] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[3] = MA_CHANNEL_SIDE_LEFT;
- pChannelMap[4] = MA_CHANNEL_SIDE_RIGHT;
- pChannelMap[5] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[6] = MA_CHANNEL_BACK_RIGHT;
- pChannelMap[7] = MA_CHANNEL_LFE;
- } break;
+ path += 1;
}
- /* Remainder. */
- if (channels > 8) {
- ma_uint32 iChannel;
- for (iChannel = 8; iChannel < channels; ++iChannel) {
- if (iChannel < MA_MAX_CHANNELS) {
- pChannelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-8));
- } else {
- pChannelMap[iChannel] = MA_CHANNEL_NONE;
- }
- }
+ /* At this point the file name is sitting on a slash, so just move forward. */
+ while (fileName[0] != '\0' && (fileName[0] == '/' || fileName[0] == '\\')) {
+ fileName += 1;
}
+
+ return fileName;
}
-static void ma_get_standard_channel_map_sound4(ma_uint32 channels, ma_channel* pChannelMap)
+
+static const char* ma_path_extension(const char* path)
{
- switch (channels)
- {
- case 1:
- {
- pChannelMap[0] = MA_CHANNEL_MONO;
- } break;
+ const char* extension;
+ const char* lastOccurance;
- case 2:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- } break;
+ if (path == NULL) {
+ path = "";
+ }
- case 3:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_BACK_CENTER;
- } break;
+ extension = ma_path_file_name(path);
+ lastOccurance = NULL;
- case 4:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[3] = MA_CHANNEL_BACK_RIGHT;
- } break;
+ /* Just find the last '.' and return. */
+ while (extension[0] != '\0') {
+ if (extension[0] == '.') {
+ extension += 1;
+ lastOccurance = extension;
+ }
- case 5:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[3] = MA_CHANNEL_BACK_RIGHT;
- pChannelMap[4] = MA_CHANNEL_FRONT_CENTER;
- } break;
+ extension += 1;
+ }
- case 6:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[3] = MA_CHANNEL_BACK_RIGHT;
- pChannelMap[4] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[5] = MA_CHANNEL_LFE;
- } break;
+ return (lastOccurance != NULL) ? lastOccurance : extension;
+}
- case 7:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[3] = MA_CHANNEL_BACK_RIGHT;
- pChannelMap[4] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[5] = MA_CHANNEL_BACK_CENTER;
- pChannelMap[6] = MA_CHANNEL_LFE;
- } break;
+static const wchar_t* ma_path_extension_w(const wchar_t* path)
+{
+ const wchar_t* extension;
+ const wchar_t* lastOccurance;
- case 8:
- default:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[3] = MA_CHANNEL_BACK_RIGHT;
- pChannelMap[4] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[5] = MA_CHANNEL_LFE;
- pChannelMap[6] = MA_CHANNEL_SIDE_LEFT;
- pChannelMap[7] = MA_CHANNEL_SIDE_RIGHT;
- } break;
+ if (path == NULL) {
+ path = L"";
}
- /* Remainder. */
- if (channels > 8) {
- ma_uint32 iChannel;
- for (iChannel = 8; iChannel < MA_MAX_CHANNELS; ++iChannel) {
- if (iChannel < MA_MAX_CHANNELS) {
- pChannelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-8));
- } else {
- pChannelMap[iChannel] = MA_CHANNEL_NONE;
- }
+ extension = ma_path_file_name_w(path);
+ lastOccurance = NULL;
+
+ /* Just find the last '.' and return. */
+ while (extension[0] != '\0') {
+ if (extension[0] == '.') {
+ extension += 1;
+ lastOccurance = extension;
}
+
+ extension += 1;
}
+
+ return (lastOccurance != NULL) ? lastOccurance : extension;
}
-static void ma_get_standard_channel_map_sndio(ma_uint32 channels, ma_channel* pChannelMap)
+
+static ma_bool32 ma_path_extension_equal(const char* path, const char* extension)
{
- switch (channels)
- {
- case 1:
- {
- pChannelMap[0] = MA_CHANNEL_MONO;
- } break;
+ const char* ext1;
+ const char* ext2;
- case 2:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- } break;
+ if (path == NULL || extension == NULL) {
+ return MA_FALSE;
+ }
- case 3:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_FRONT_CENTER;
- } break;
+ ext1 = extension;
+ ext2 = ma_path_extension(path);
- case 4:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[3] = MA_CHANNEL_BACK_RIGHT;
- } break;
+#if defined(_MSC_VER) || defined(__DMC__)
+ return _stricmp(ext1, ext2) == 0;
+#else
+ return strcasecmp(ext1, ext2) == 0;
+#endif
+}
- case 5:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[3] = MA_CHANNEL_BACK_RIGHT;
- pChannelMap[4] = MA_CHANNEL_FRONT_CENTER;
- } break;
+static ma_bool32 ma_path_extension_equal_w(const wchar_t* path, const wchar_t* extension)
+{
+ const wchar_t* ext1;
+ const wchar_t* ext2;
- case 6:
- default:
- {
- pChannelMap[0] = MA_CHANNEL_FRONT_LEFT;
- pChannelMap[1] = MA_CHANNEL_FRONT_RIGHT;
- pChannelMap[2] = MA_CHANNEL_BACK_LEFT;
- pChannelMap[3] = MA_CHANNEL_BACK_RIGHT;
- pChannelMap[4] = MA_CHANNEL_FRONT_CENTER;
- pChannelMap[5] = MA_CHANNEL_LFE;
- } break;
+ if (path == NULL || extension == NULL) {
+ return MA_FALSE;
}
- /* Remainder. */
- if (channels > 6) {
- ma_uint32 iChannel;
- for (iChannel = 6; iChannel < channels && iChannel < MA_MAX_CHANNELS; ++iChannel) {
- if (iChannel < MA_MAX_CHANNELS) {
- pChannelMap[iChannel] = (ma_channel)(MA_CHANNEL_AUX_0 + (iChannel-6));
- } else {
- pChannelMap[iChannel] = MA_CHANNEL_NONE;
- }
- }
- }
-}
+ ext1 = extension;
+ ext2 = ma_path_extension_w(path);
-MA_API void ma_get_standard_channel_map(ma_standard_channel_map standardChannelMap, ma_uint32 channels, ma_channel* pChannelMap)
-{
- switch (standardChannelMap)
+#if defined(_MSC_VER) || defined(__WATCOMC__) || defined(__DMC__)
+ return _wcsicmp(ext1, ext2) == 0;
+#else
+ /*
+ I'm not aware of a wide character version of strcasecmp(). I'm therefore converting the extensions to multibyte strings and comparing those. This
+ isn't the most efficient way to do it, but it should work OK.
+ */
{
- case ma_standard_channel_map_alsa:
- {
- ma_get_standard_channel_map_alsa(channels, pChannelMap);
- } break;
+ char ext1MB[4096];
+ char ext2MB[4096];
+ const wchar_t* pext1 = ext1;
+ const wchar_t* pext2 = ext2;
+ mbstate_t mbs1;
+ mbstate_t mbs2;
- case ma_standard_channel_map_rfc3551:
- {
- ma_get_standard_channel_map_rfc3551(channels, pChannelMap);
- } break;
+ MA_ZERO_OBJECT(&mbs1);
+ MA_ZERO_OBJECT(&mbs2);
- case ma_standard_channel_map_flac:
- {
- ma_get_standard_channel_map_flac(channels, pChannelMap);
- } break;
+ if (wcsrtombs(ext1MB, &pext1, sizeof(ext1MB), &mbs1) == (size_t)-1) {
+ return MA_FALSE;
+ }
+ if (wcsrtombs(ext2MB, &pext2, sizeof(ext2MB), &mbs2) == (size_t)-1) {
+ return MA_FALSE;
+ }
- case ma_standard_channel_map_vorbis:
- {
- ma_get_standard_channel_map_vorbis(channels, pChannelMap);
- } break;
+ return strcasecmp(ext1MB, ext2MB) == 0;
+ }
+#endif
+}
+#endif /* MA_HAS_PATH_API */
- case ma_standard_channel_map_sound4:
- {
- ma_get_standard_channel_map_sound4(channels, pChannelMap);
- } break;
- case ma_standard_channel_map_sndio:
- {
- ma_get_standard_channel_map_sndio(channels, pChannelMap);
- } break;
- case ma_standard_channel_map_microsoft: /* Also default. */
- /*case ma_standard_channel_map_default;*/
- default:
- {
- ma_get_standard_channel_map_microsoft(channels, pChannelMap);
- } break;
- }
+static ma_result ma_decoder__on_read_vfs(ma_decoder* pDecoder, void* pBufferOut, size_t bytesToRead, size_t* pBytesRead)
+{
+ MA_ASSERT(pDecoder != NULL);
+ MA_ASSERT(pBufferOut != NULL);
+
+ return ma_vfs_or_default_read(pDecoder->data.vfs.pVFS, pDecoder->data.vfs.file, pBufferOut, bytesToRead, pBytesRead);
}
-MA_API void ma_channel_map_copy(ma_channel* pOut, const ma_channel* pIn, ma_uint32 channels)
+static ma_result ma_decoder__on_seek_vfs(ma_decoder* pDecoder, ma_int64 offset, ma_seek_origin origin)
{
- if (pOut != NULL && pIn != NULL && channels > 0) {
- MA_COPY_MEMORY(pOut, pIn, sizeof(*pOut) * channels);
- }
+ MA_ASSERT(pDecoder != NULL);
+
+ return ma_vfs_or_default_seek(pDecoder->data.vfs.pVFS, pDecoder->data.vfs.file, offset, origin);
}
-MA_API void ma_channel_map_copy_or_default(ma_channel* pOut, const ma_channel* pIn, ma_uint32 channels)
+static ma_result ma_decoder__on_tell_vfs(ma_decoder* pDecoder, ma_int64* pCursor)
{
- if (pOut == NULL || channels == 0) {
- return;
- }
+ MA_ASSERT(pDecoder != NULL);
- if (pIn != NULL) {
- ma_channel_map_copy(pOut, pIn, channels);
- } else {
- ma_get_standard_channel_map(ma_standard_channel_map_default, channels, pOut);
- }
+ return ma_vfs_or_default_tell(pDecoder->data.vfs.pVFS, pDecoder->data.vfs.file, pCursor);
}
-MA_API ma_bool32 ma_channel_map_valid(ma_uint32 channels, const ma_channel* pChannelMap)
+static ma_result ma_decoder__preinit_vfs(ma_vfs* pVFS, const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
- if (pChannelMap == NULL) {
- return MA_FALSE;
+ ma_result result;
+ ma_vfs_file file;
+
+ result = ma_decoder__preinit(ma_decoder__on_read_vfs, ma_decoder__on_seek_vfs, ma_decoder__on_tell_vfs, NULL, pConfig, pDecoder);
+ if (result != MA_SUCCESS) {
+ return result;
}
- /* A channel count of 0 is invalid. */
- if (channels == 0) {
- return MA_FALSE;
+ if (pFilePath == NULL || pFilePath[0] == '\0') {
+ return MA_INVALID_ARGS;
}
- /* It does not make sense to have a mono channel when there is more than 1 channel. */
- if (channels > 1) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < channels; ++iChannel) {
- if (pChannelMap[iChannel] == MA_CHANNEL_MONO) {
- return MA_FALSE;
- }
- }
+ result = ma_vfs_or_default_open(pVFS, pFilePath, MA_OPEN_MODE_READ, &file);
+ if (result != MA_SUCCESS) {
+ return result;
}
- return MA_TRUE;
+ pDecoder->data.vfs.pVFS = pVFS;
+ pDecoder->data.vfs.file = file;
+
+ return MA_SUCCESS;
}
-MA_API ma_bool32 ma_channel_map_equal(ma_uint32 channels, const ma_channel* pChannelMapA, const ma_channel* pChannelMapB)
+MA_API ma_result ma_decoder_init_vfs(ma_vfs* pVFS, const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
- ma_uint32 iChannel;
+ ma_result result;
+ ma_decoder_config config;
- if (pChannelMapA == pChannelMapB) {
- return MA_TRUE;
+ config = ma_decoder_config_init_copy(pConfig);
+ result = ma_decoder__preinit_vfs(pVFS, pFilePath, &config, pDecoder);
+ if (result != MA_SUCCESS) {
+ return result;
}
- for (iChannel = 0; iChannel < channels; ++iChannel) {
- if (ma_channel_map_get_channel(pChannelMapA, channels, iChannel) != ma_channel_map_get_channel(pChannelMapB, channels, iChannel)) {
- return MA_FALSE;
+ result = MA_NO_BACKEND;
+
+ if (config.encodingFormat != ma_encoding_format_unknown) {
+ #ifdef MA_HAS_WAV
+ if (config.encodingFormat == ma_encoding_format_wav) {
+ result = ma_decoder_init_wav__internal(&config, pDecoder);
+ }
+ #endif
+ #ifdef MA_HAS_FLAC
+ if (config.encodingFormat == ma_encoding_format_flac) {
+ result = ma_decoder_init_flac__internal(&config, pDecoder);
+ }
+ #endif
+ #ifdef MA_HAS_MP3
+ if (config.encodingFormat == ma_encoding_format_mp3) {
+ result = ma_decoder_init_mp3__internal(&config, pDecoder);
+ }
+ #endif
+ #ifdef MA_HAS_VORBIS
+ if (config.encodingFormat == ma_encoding_format_vorbis) {
+ result = ma_decoder_init_vorbis__internal(&config, pDecoder);
+ }
+ #endif
+
+ /* Make sure we seek back to the start if we didn't initialize a decoder successfully so the next attempts have a fresh start. */
+ if (result != MA_SUCCESS) {
+ ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
}
}
- return MA_TRUE;
-}
+ if (result != MA_SUCCESS) {
+ /* Getting here means we weren't able to initialize a decoder of a specific encoding format. */
-MA_API ma_bool32 ma_channel_map_blank(ma_uint32 channels, const ma_channel* pChannelMap)
-{
- ma_uint32 iChannel;
+ /*
+ We use trial and error to open a decoder. We prioritize custom decoders so that if they
+ implement the same encoding format they take priority over the built-in decoders.
+ */
+ if (result != MA_SUCCESS) {
+ result = ma_decoder_init_custom__internal(&config, pDecoder);
+ if (result != MA_SUCCESS) {
+ ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
+ }
+ }
- /* A null channel map is equivalent to the default channel map. */
- if (pChannelMap == NULL) {
- return MA_FALSE;
- }
+ /*
+ If we get to this point and we still haven't found a decoder, and the caller has requested a
+ specific encoding format, there's no hope for it. Abort.
+ */
+ if (config.encodingFormat != ma_encoding_format_unknown) {
+ return MA_NO_BACKEND;
+ }
- for (iChannel = 0; iChannel < channels; ++iChannel) {
- if (pChannelMap[iChannel] != MA_CHANNEL_NONE) {
- return MA_FALSE;
+ #ifdef MA_HAS_WAV
+ if (result != MA_SUCCESS && ma_path_extension_equal(pFilePath, "wav")) {
+ result = ma_decoder_init_wav__internal(&config, pDecoder);
+ if (result != MA_SUCCESS) {
+ ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
+ }
+ }
+ #endif
+ #ifdef MA_HAS_FLAC
+ if (result != MA_SUCCESS && ma_path_extension_equal(pFilePath, "flac")) {
+ result = ma_decoder_init_flac__internal(&config, pDecoder);
+ if (result != MA_SUCCESS) {
+ ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
+ }
+ }
+ #endif
+ #ifdef MA_HAS_MP3
+ if (result != MA_SUCCESS && ma_path_extension_equal(pFilePath, "mp3")) {
+ result = ma_decoder_init_mp3__internal(&config, pDecoder);
+ if (result != MA_SUCCESS) {
+ ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
+ }
}
+ #endif
}
- return MA_TRUE;
-}
-
-MA_API ma_bool32 ma_channel_map_contains_channel_position(ma_uint32 channels, const ma_channel* pChannelMap, ma_channel channelPosition)
-{
- ma_uint32 iChannel;
+ /* If we still haven't got a result just use trial and error. Otherwise we can finish up. */
+ if (result != MA_SUCCESS) {
+ result = ma_decoder_init__internal(ma_decoder__on_read_vfs, ma_decoder__on_seek_vfs, NULL, &config, pDecoder);
+ } else {
+ result = ma_decoder__postinit(&config, pDecoder);
+ }
- for (iChannel = 0; iChannel < channels; ++iChannel) {
- if (ma_channel_map_get_channel(pChannelMap, channels, iChannel) == channelPosition) {
- return MA_TRUE;
+ if (result != MA_SUCCESS) {
+ if (pDecoder->data.vfs.file != NULL) { /* <-- Will be reset to NULL if ma_decoder_uninit() is called in one of the steps above which allows us to avoid a double close of the file. */
+ ma_vfs_or_default_close(pVFS, pDecoder->data.vfs.file);
}
+
+ return result;
}
- return MA_FALSE;
+ return MA_SUCCESS;
}
+static ma_result ma_decoder__preinit_vfs_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+{
+ ma_result result;
+ ma_vfs_file file;
-/**************************************************************************************************************************************************************
+ result = ma_decoder__preinit(ma_decoder__on_read_vfs, ma_decoder__on_seek_vfs, ma_decoder__on_tell_vfs, NULL, pConfig, pDecoder);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
-Conversion Helpers
+ if (pFilePath == NULL || pFilePath[0] == '\0') {
+ return MA_INVALID_ARGS;
+ }
-**************************************************************************************************************************************************************/
-MA_API ma_uint64 ma_convert_frames(void* pOut, ma_uint64 frameCountOut, ma_format formatOut, ma_uint32 channelsOut, ma_uint32 sampleRateOut, const void* pIn, ma_uint64 frameCountIn, ma_format formatIn, ma_uint32 channelsIn, ma_uint32 sampleRateIn)
-{
- ma_data_converter_config config;
+ result = ma_vfs_or_default_open_w(pVFS, pFilePath, MA_OPEN_MODE_READ, &file);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- config = ma_data_converter_config_init(formatIn, formatOut, channelsIn, channelsOut, sampleRateIn, sampleRateOut);
- ma_get_standard_channel_map(ma_standard_channel_map_default, channelsOut, config.channelMapOut);
- ma_get_standard_channel_map(ma_standard_channel_map_default, channelsIn, config.channelMapIn);
- config.resampling.linear.lpfOrder = ma_min(MA_DEFAULT_RESAMPLER_LPF_ORDER, MA_MAX_FILTER_ORDER);
+ pDecoder->data.vfs.pVFS = pVFS;
+ pDecoder->data.vfs.file = file;
- return ma_convert_frames_ex(pOut, frameCountOut, pIn, frameCountIn, &config);
+ return MA_SUCCESS;
}
-MA_API ma_uint64 ma_convert_frames_ex(void* pOut, ma_uint64 frameCountOut, const void* pIn, ma_uint64 frameCountIn, const ma_data_converter_config* pConfig)
+MA_API ma_result ma_decoder_init_vfs_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
ma_result result;
- ma_data_converter converter;
-
- if (frameCountIn == 0 || pConfig == NULL) {
- return 0;
- }
+ ma_decoder_config config;
- result = ma_data_converter_init(pConfig, &converter);
+ config = ma_decoder_config_init_copy(pConfig);
+ result = ma_decoder__preinit_vfs_w(pVFS, pFilePath, &config, pDecoder);
if (result != MA_SUCCESS) {
- return 0; /* Failed to initialize the data converter. */
+ return result;
}
- if (pOut == NULL) {
- frameCountOut = ma_data_converter_get_expected_output_frame_count(&converter, frameCountIn);
- } else {
- result = ma_data_converter_process_pcm_frames(&converter, pIn, &frameCountIn, pOut, &frameCountOut);
+ result = MA_NO_BACKEND;
+
+ if (config.encodingFormat != ma_encoding_format_unknown) {
+ #ifdef MA_HAS_WAV
+ if (config.encodingFormat == ma_encoding_format_wav) {
+ result = ma_decoder_init_wav__internal(&config, pDecoder);
+ }
+ #endif
+ #ifdef MA_HAS_FLAC
+ if (config.encodingFormat == ma_encoding_format_flac) {
+ result = ma_decoder_init_flac__internal(&config, pDecoder);
+ }
+ #endif
+ #ifdef MA_HAS_MP3
+ if (config.encodingFormat == ma_encoding_format_mp3) {
+ result = ma_decoder_init_mp3__internal(&config, pDecoder);
+ }
+ #endif
+ #ifdef MA_HAS_VORBIS
+ if (config.encodingFormat == ma_encoding_format_vorbis) {
+ result = ma_decoder_init_vorbis__internal(&config, pDecoder);
+ }
+ #endif
+
+ /* Make sure we seek back to the start if we didn't initialize a decoder successfully so the next attempts have a fresh start. */
if (result != MA_SUCCESS) {
- frameCountOut = 0;
+ ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
}
}
- ma_data_converter_uninit(&converter);
- return frameCountOut;
-}
+ if (result != MA_SUCCESS) {
+ /* Getting here means we weren't able to initialize a decoder of a specific encoding format. */
+ /*
+ We use trial and error to open a decoder. We prioritize custom decoders so that if they
+ implement the same encoding format they take priority over the built-in decoders.
+ */
+ if (result != MA_SUCCESS) {
+ result = ma_decoder_init_custom__internal(&config, pDecoder);
+ if (result != MA_SUCCESS) {
+ ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
+ }
+ }
-/**************************************************************************************************************************************************************
+ /*
+ If we get to this point and we still haven't found a decoder, and the caller has requested a
+ specific encoding format, there's no hope for it. Abort.
+ */
+ if (config.encodingFormat != ma_encoding_format_unknown) {
+ return MA_NO_BACKEND;
+ }
-Ring Buffer
+ #ifdef MA_HAS_WAV
+ if (result != MA_SUCCESS && ma_path_extension_equal_w(pFilePath, L"wav")) {
+ result = ma_decoder_init_wav__internal(&config, pDecoder);
+ if (result != MA_SUCCESS) {
+ ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
+ }
+ }
+ #endif
+ #ifdef MA_HAS_FLAC
+ if (result != MA_SUCCESS && ma_path_extension_equal_w(pFilePath, L"flac")) {
+ result = ma_decoder_init_flac__internal(&config, pDecoder);
+ if (result != MA_SUCCESS) {
+ ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
+ }
+ }
+ #endif
+ #ifdef MA_HAS_MP3
+ if (result != MA_SUCCESS && ma_path_extension_equal_w(pFilePath, L"mp3")) {
+ result = ma_decoder_init_mp3__internal(&config, pDecoder);
+ if (result != MA_SUCCESS) {
+ ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
+ }
+ }
+ #endif
+ }
-**************************************************************************************************************************************************************/
-static MA_INLINE ma_uint32 ma_rb__extract_offset_in_bytes(ma_uint32 encodedOffset)
-{
- return encodedOffset & 0x7FFFFFFF;
-}
+ /* If we still haven't got a result just use trial and error. Otherwise we can finish up. */
+ if (result != MA_SUCCESS) {
+ result = ma_decoder_init__internal(ma_decoder__on_read_vfs, ma_decoder__on_seek_vfs, NULL, &config, pDecoder);
+ } else {
+ result = ma_decoder__postinit(&config, pDecoder);
+ }
-static MA_INLINE ma_uint32 ma_rb__extract_offset_loop_flag(ma_uint32 encodedOffset)
-{
- return encodedOffset & 0x80000000;
-}
+ if (result != MA_SUCCESS) {
+ ma_vfs_or_default_close(pVFS, pDecoder->data.vfs.file);
+ return result;
+ }
-static MA_INLINE void* ma_rb__get_read_ptr(ma_rb* pRB)
-{
- MA_ASSERT(pRB != NULL);
- return ma_offset_ptr(pRB->pBuffer, ma_rb__extract_offset_in_bytes(c89atomic_load_32(&pRB->encodedReadOffset)));
+ return MA_SUCCESS;
}
-static MA_INLINE void* ma_rb__get_write_ptr(ma_rb* pRB)
+MA_API ma_result ma_decoder_init_file(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
- MA_ASSERT(pRB != NULL);
- return ma_offset_ptr(pRB->pBuffer, ma_rb__extract_offset_in_bytes(c89atomic_load_32(&pRB->encodedWriteOffset)));
+ return ma_decoder_init_vfs(NULL, pFilePath, pConfig, pDecoder);
}
-static MA_INLINE ma_uint32 ma_rb__construct_offset(ma_uint32 offsetInBytes, ma_uint32 offsetLoopFlag)
+MA_API ma_result ma_decoder_init_file_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
- return offsetLoopFlag | offsetInBytes;
+ return ma_decoder_init_vfs_w(NULL, pFilePath, pConfig, pDecoder);
}
-static MA_INLINE void ma_rb__deconstruct_offset(ma_uint32 encodedOffset, ma_uint32* pOffsetInBytes, ma_uint32* pOffsetLoopFlag)
+MA_API ma_result ma_decoder_uninit(ma_decoder* pDecoder)
{
- MA_ASSERT(pOffsetInBytes != NULL);
- MA_ASSERT(pOffsetLoopFlag != NULL);
+ if (pDecoder == NULL) {
+ return MA_INVALID_ARGS;
+ }
- *pOffsetInBytes = ma_rb__extract_offset_in_bytes(encodedOffset);
- *pOffsetLoopFlag = ma_rb__extract_offset_loop_flag(encodedOffset);
-}
+ if (pDecoder->pBackend != NULL) {
+ if (pDecoder->pBackendVTable != NULL && pDecoder->pBackendVTable->onUninit != NULL) {
+ pDecoder->pBackendVTable->onUninit(pDecoder->pBackendUserData, pDecoder->pBackend, &pDecoder->allocationCallbacks);
+ }
+ }
+
+ if (pDecoder->onRead == ma_decoder__on_read_vfs) {
+ ma_vfs_or_default_close(pDecoder->data.vfs.pVFS, pDecoder->data.vfs.file);
+ pDecoder->data.vfs.file = NULL;
+ }
+ ma_data_converter_uninit(&pDecoder->converter, &pDecoder->allocationCallbacks);
+ ma_data_source_uninit(&pDecoder->ds);
-MA_API ma_result ma_rb_init_ex(size_t subbufferSizeInBytes, size_t subbufferCount, size_t subbufferStrideInBytes, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_rb* pRB)
+ if (pDecoder->pInputCache != NULL) {
+ ma_free(pDecoder->pInputCache, &pDecoder->allocationCallbacks);
+ }
+
+ return MA_SUCCESS;
+}
+
+MA_API ma_result ma_decoder_read_pcm_frames(ma_decoder* pDecoder, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
- ma_result result;
- const ma_uint32 maxSubBufferSize = 0x7FFFFFFF - (MA_SIMD_ALIGNMENT-1);
+ ma_result result = MA_SUCCESS;
+ ma_uint64 totalFramesReadOut;
+ void* pRunningFramesOut;
- if (pRB == NULL) {
+ if (pFramesRead != NULL) {
+ *pFramesRead = 0; /* Safety. */
+ }
+
+ if (frameCount == 0) {
return MA_INVALID_ARGS;
}
- if (subbufferSizeInBytes == 0 || subbufferCount == 0) {
+ if (pDecoder == NULL) {
return MA_INVALID_ARGS;
}
- if (subbufferSizeInBytes > maxSubBufferSize) {
- return MA_INVALID_ARGS; /* Maximum buffer size is ~2GB. The most significant bit is a flag for use internally. */
+ if (pDecoder->pBackend == NULL) {
+ return MA_INVALID_OPERATION;
}
+ /* Fast path. */
+ if (pDecoder->converter.isPassthrough) {
+ result = ma_data_source_read_pcm_frames(pDecoder->pBackend, pFramesOut, frameCount, &totalFramesReadOut);
+ } else {
+ /*
+ Getting here means we need to do data conversion. If we're seeking forward and are _not_ doing resampling we can run this in a fast path. If we're doing resampling we
+ need to run through each sample because we need to ensure it's internal cache is updated.
+ */
+ if (pFramesOut == NULL && pDecoder->converter.hasResampler == MA_FALSE) {
+ result = ma_data_source_read_pcm_frames(pDecoder->pBackend, NULL, frameCount, &totalFramesReadOut);
+ } else {
+ /* Slow path. Need to run everything through the data converter. */
+ ma_format internalFormat;
+ ma_uint32 internalChannels;
+
+ totalFramesReadOut = 0;
+ pRunningFramesOut = pFramesOut;
- MA_ZERO_OBJECT(pRB);
+ result = ma_data_source_get_data_format(pDecoder->pBackend, &internalFormat, &internalChannels, NULL, NULL, 0);
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to retrieve the internal format and channel count. */
+ }
- result = ma_allocation_callbacks_init_copy(&pRB->allocationCallbacks, pAllocationCallbacks);
- if (result != MA_SUCCESS) {
- return result;
- }
+ /*
+ We run a different path depending on whether or not we are using a heap-allocated
+ intermediary buffer or not. If the data converter does not support the calculation of
+ the required number of input frames, we'll use the heap-allocated path. Otherwise we'll
+ use the stack-allocated path.
+ */
+ if (pDecoder->pInputCache != NULL) {
+ /* We don't have a way of determining the required number of input frames, so need to persistently store input data in a cache. */
+ while (totalFramesReadOut < frameCount) {
+ ma_uint64 framesToReadThisIterationIn;
+ ma_uint64 framesToReadThisIterationOut;
+
+ /* If there's any data available in the cache, that needs to get processed first. */
+ if (pDecoder->inputCacheRemaining > 0) {
+ framesToReadThisIterationOut = (frameCount - totalFramesReadOut);
+ framesToReadThisIterationIn = framesToReadThisIterationOut;
+ if (framesToReadThisIterationIn > pDecoder->inputCacheRemaining) {
+ framesToReadThisIterationIn = pDecoder->inputCacheRemaining;
+ }
- pRB->subbufferSizeInBytes = (ma_uint32)subbufferSizeInBytes;
- pRB->subbufferCount = (ma_uint32)subbufferCount;
+ result = ma_data_converter_process_pcm_frames(&pDecoder->converter, ma_offset_pcm_frames_ptr(pDecoder->pInputCache, pDecoder->inputCacheConsumed, internalFormat, internalChannels), &framesToReadThisIterationIn, pRunningFramesOut, &framesToReadThisIterationOut);
+ if (result != MA_SUCCESS) {
+ break;
+ }
- if (pOptionalPreallocatedBuffer != NULL) {
- pRB->subbufferStrideInBytes = (ma_uint32)subbufferStrideInBytes;
- pRB->pBuffer = pOptionalPreallocatedBuffer;
- } else {
- size_t bufferSizeInBytes;
+ pDecoder->inputCacheConsumed += framesToReadThisIterationIn;
+ pDecoder->inputCacheRemaining -= framesToReadThisIterationIn;
- /*
- Here is where we allocate our own buffer. We always want to align this to MA_SIMD_ALIGNMENT for future SIMD optimization opportunity. To do this
- we need to make sure the stride is a multiple of MA_SIMD_ALIGNMENT.
- */
- pRB->subbufferStrideInBytes = (pRB->subbufferSizeInBytes + (MA_SIMD_ALIGNMENT-1)) & ~MA_SIMD_ALIGNMENT;
+ totalFramesReadOut += framesToReadThisIterationOut;
- bufferSizeInBytes = (size_t)pRB->subbufferCount*pRB->subbufferStrideInBytes;
- pRB->pBuffer = ma_aligned_malloc(bufferSizeInBytes, MA_SIMD_ALIGNMENT, &pRB->allocationCallbacks);
- if (pRB->pBuffer == NULL) {
- return MA_OUT_OF_MEMORY;
- }
+ if (pRunningFramesOut != NULL) {
+ pRunningFramesOut = ma_offset_ptr(pRunningFramesOut, framesToReadThisIterationOut * ma_get_bytes_per_frame(pDecoder->outputFormat, pDecoder->outputChannels));
+ }
- MA_ZERO_MEMORY(pRB->pBuffer, bufferSizeInBytes);
- pRB->ownsBuffer = MA_TRUE;
- }
+ if (framesToReadThisIterationIn == 0 && framesToReadThisIterationOut == 0) {
+ break; /* We're done. */
+ }
+ }
- return MA_SUCCESS;
-}
+ /* Getting here means there's no data in the cache and we need to fill it up from the data source. */
+ if (pDecoder->inputCacheRemaining == 0) {
+ pDecoder->inputCacheConsumed = 0;
-MA_API ma_result ma_rb_init(size_t bufferSizeInBytes, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_rb* pRB)
-{
- return ma_rb_init_ex(bufferSizeInBytes, 1, 0, pOptionalPreallocatedBuffer, pAllocationCallbacks, pRB);
-}
+ result = ma_data_source_read_pcm_frames(pDecoder->pBackend, pDecoder->pInputCache, pDecoder->inputCacheCap, &pDecoder->inputCacheRemaining);
+ if (result != MA_SUCCESS) {
+ break;
+ }
+ }
+ }
+ } else {
+ /* We have a way of determining the required number of input frames so just use the stack. */
+ while (totalFramesReadOut < frameCount) {
+ ma_uint8 pIntermediaryBuffer[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In internal format. */
+ ma_uint64 intermediaryBufferCap = sizeof(pIntermediaryBuffer) / ma_get_bytes_per_frame(internalFormat, internalChannels);
+ ma_uint64 framesToReadThisIterationIn;
+ ma_uint64 framesReadThisIterationIn;
+ ma_uint64 framesToReadThisIterationOut;
+ ma_uint64 framesReadThisIterationOut;
+ ma_uint64 requiredInputFrameCount;
+
+ framesToReadThisIterationOut = (frameCount - totalFramesReadOut);
+ framesToReadThisIterationIn = framesToReadThisIterationOut;
+ if (framesToReadThisIterationIn > intermediaryBufferCap) {
+ framesToReadThisIterationIn = intermediaryBufferCap;
+ }
-MA_API void ma_rb_uninit(ma_rb* pRB)
-{
- if (pRB == NULL) {
- return;
+ ma_data_converter_get_required_input_frame_count(&pDecoder->converter, framesToReadThisIterationOut, &requiredInputFrameCount);
+ if (framesToReadThisIterationIn > requiredInputFrameCount) {
+ framesToReadThisIterationIn = requiredInputFrameCount;
+ }
+
+ if (requiredInputFrameCount > 0) {
+ result = ma_data_source_read_pcm_frames(pDecoder->pBackend, pIntermediaryBuffer, framesToReadThisIterationIn, &framesReadThisIterationIn);
+ } else {
+ framesReadThisIterationIn = 0;
+ }
+
+ /*
+ At this point we have our decoded data in input format and now we need to convert to output format. Note that even if we didn't read any
+ input frames, we still want to try processing frames because there may some output frames generated from cached input data.
+ */
+ framesReadThisIterationOut = framesToReadThisIterationOut;
+ result = ma_data_converter_process_pcm_frames(&pDecoder->converter, pIntermediaryBuffer, &framesReadThisIterationIn, pRunningFramesOut, &framesReadThisIterationOut);
+ if (result != MA_SUCCESS) {
+ break;
+ }
+
+ totalFramesReadOut += framesReadThisIterationOut;
+
+ if (pRunningFramesOut != NULL) {
+ pRunningFramesOut = ma_offset_ptr(pRunningFramesOut, framesReadThisIterationOut * ma_get_bytes_per_frame(pDecoder->outputFormat, pDecoder->outputChannels));
+ }
+
+ if (framesReadThisIterationIn == 0 && framesReadThisIterationOut == 0) {
+ break; /* We're done. */
+ }
+ }
+ }
+ }
}
- if (pRB->ownsBuffer) {
- ma_aligned_free(pRB->pBuffer, &pRB->allocationCallbacks);
+ pDecoder->readPointerInPCMFrames += totalFramesReadOut;
+
+ if (pFramesRead != NULL) {
+ *pFramesRead = totalFramesReadOut;
}
-}
-MA_API void ma_rb_reset(ma_rb* pRB)
-{
- if (pRB == NULL) {
- return;
+ if (result == MA_SUCCESS && totalFramesReadOut == 0) {
+ result = MA_AT_END;
}
- c89atomic_exchange_32(&pRB->encodedReadOffset, 0);
- c89atomic_exchange_32(&pRB->encodedWriteOffset, 0);
+ return result;
}
-MA_API ma_result ma_rb_acquire_read(ma_rb* pRB, size_t* pSizeInBytes, void** ppBufferOut)
+MA_API ma_result ma_decoder_seek_to_pcm_frame(ma_decoder* pDecoder, ma_uint64 frameIndex)
{
- ma_uint32 writeOffset;
- ma_uint32 writeOffsetInBytes;
- ma_uint32 writeOffsetLoopFlag;
- ma_uint32 readOffset;
- ma_uint32 readOffsetInBytes;
- ma_uint32 readOffsetLoopFlag;
- size_t bytesAvailable;
- size_t bytesRequested;
-
- if (pRB == NULL || pSizeInBytes == NULL || ppBufferOut == NULL) {
+ if (pDecoder == NULL) {
return MA_INVALID_ARGS;
}
- /* The returned buffer should never move ahead of the write pointer. */
- writeOffset = c89atomic_load_32(&pRB->encodedWriteOffset);
- ma_rb__deconstruct_offset(writeOffset, &writeOffsetInBytes, &writeOffsetLoopFlag);
+ if (pDecoder->pBackend != NULL) {
+ ma_result result;
+ ma_uint64 internalFrameIndex;
+ ma_uint32 internalSampleRate;
- readOffset = c89atomic_load_32(&pRB->encodedReadOffset);
- ma_rb__deconstruct_offset(readOffset, &readOffsetInBytes, &readOffsetLoopFlag);
+ result = ma_data_source_get_data_format(pDecoder->pBackend, NULL, NULL, &internalSampleRate, NULL, 0);
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to retrieve the internal sample rate. */
+ }
- /*
- The number of bytes available depends on whether or not the read and write pointers are on the same loop iteration. If so, we
- can only read up to the write pointer. If not, we can only read up to the end of the buffer.
- */
- if (readOffsetLoopFlag == writeOffsetLoopFlag) {
- bytesAvailable = writeOffsetInBytes - readOffsetInBytes;
- } else {
- bytesAvailable = pRB->subbufferSizeInBytes - readOffsetInBytes;
- }
+ if (internalSampleRate == pDecoder->outputSampleRate) {
+ internalFrameIndex = frameIndex;
+ } else {
+ internalFrameIndex = ma_calculate_frame_count_after_resampling(internalSampleRate, pDecoder->outputSampleRate, frameIndex);
+ }
- bytesRequested = *pSizeInBytes;
- if (bytesRequested > bytesAvailable) {
- bytesRequested = bytesAvailable;
- }
+ result = ma_data_source_seek_to_pcm_frame(pDecoder->pBackend, internalFrameIndex);
+ if (result == MA_SUCCESS) {
+ pDecoder->readPointerInPCMFrames = frameIndex;
+ }
- *pSizeInBytes = bytesRequested;
- (*ppBufferOut) = ma_rb__get_read_ptr(pRB);
+ return result;
+ }
- return MA_SUCCESS;
+ /* Should never get here, but if we do it means onSeekToPCMFrame was not set by the backend. */
+ return MA_INVALID_ARGS;
}
-MA_API ma_result ma_rb_commit_read(ma_rb* pRB, size_t sizeInBytes, void* pBufferOut)
+MA_API ma_result ma_decoder_get_data_format(ma_decoder* pDecoder, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
- ma_uint32 readOffset;
- ma_uint32 readOffsetInBytes;
- ma_uint32 readOffsetLoopFlag;
- ma_uint32 newReadOffsetInBytes;
- ma_uint32 newReadOffsetLoopFlag;
-
- if (pRB == NULL) {
+ if (pDecoder == NULL) {
return MA_INVALID_ARGS;
}
- /* Validate the buffer. */
- if (pBufferOut != ma_rb__get_read_ptr(pRB)) {
- return MA_INVALID_ARGS;
+ if (pFormat != NULL) {
+ *pFormat = pDecoder->outputFormat;
}
- readOffset = c89atomic_load_32(&pRB->encodedReadOffset);
- ma_rb__deconstruct_offset(readOffset, &readOffsetInBytes, &readOffsetLoopFlag);
-
- /* Check that sizeInBytes is correct. It should never go beyond the end of the buffer. */
- newReadOffsetInBytes = (ma_uint32)(readOffsetInBytes + sizeInBytes);
- if (newReadOffsetInBytes > pRB->subbufferSizeInBytes) {
- return MA_INVALID_ARGS; /* <-- sizeInBytes will cause the read offset to overflow. */
+ if (pChannels != NULL) {
+ *pChannels = pDecoder->outputChannels;
}
- /* Move the read pointer back to the start if necessary. */
- newReadOffsetLoopFlag = readOffsetLoopFlag;
- if (newReadOffsetInBytes == pRB->subbufferSizeInBytes) {
- newReadOffsetInBytes = 0;
- newReadOffsetLoopFlag ^= 0x80000000;
+ if (pSampleRate != NULL) {
+ *pSampleRate = pDecoder->outputSampleRate;
}
- c89atomic_exchange_32(&pRB->encodedReadOffset, ma_rb__construct_offset(newReadOffsetLoopFlag, newReadOffsetInBytes));
-
- if (ma_rb_pointer_distance(pRB) == 0) {
- return MA_AT_END;
- } else {
- return MA_SUCCESS;
+ if (pChannelMap != NULL) {
+ ma_data_converter_get_output_channel_map(&pDecoder->converter, pChannelMap, channelMapCap);
}
+
+ return MA_SUCCESS;
}
-MA_API ma_result ma_rb_acquire_write(ma_rb* pRB, size_t* pSizeInBytes, void** ppBufferOut)
+MA_API ma_result ma_decoder_get_cursor_in_pcm_frames(ma_decoder* pDecoder, ma_uint64* pCursor)
{
- ma_uint32 readOffset;
- ma_uint32 readOffsetInBytes;
- ma_uint32 readOffsetLoopFlag;
- ma_uint32 writeOffset;
- ma_uint32 writeOffsetInBytes;
- ma_uint32 writeOffsetLoopFlag;
- size_t bytesAvailable;
- size_t bytesRequested;
-
- if (pRB == NULL || pSizeInBytes == NULL || ppBufferOut == NULL) {
+ if (pCursor == NULL) {
return MA_INVALID_ARGS;
}
- /* The returned buffer should never overtake the read buffer. */
- readOffset = c89atomic_load_32(&pRB->encodedReadOffset);
- ma_rb__deconstruct_offset(readOffset, &readOffsetInBytes, &readOffsetLoopFlag);
-
- writeOffset = c89atomic_load_32(&pRB->encodedWriteOffset);
- ma_rb__deconstruct_offset(writeOffset, &writeOffsetInBytes, &writeOffsetLoopFlag);
-
- /*
- In the case of writing, if the write pointer and the read pointer are on the same loop iteration we can only
- write up to the end of the buffer. Otherwise we can only write up to the read pointer. The write pointer should
- never overtake the read pointer.
- */
- if (writeOffsetLoopFlag == readOffsetLoopFlag) {
- bytesAvailable = pRB->subbufferSizeInBytes - writeOffsetInBytes;
- } else {
- bytesAvailable = readOffsetInBytes - writeOffsetInBytes;
- }
+ *pCursor = 0;
- bytesRequested = *pSizeInBytes;
- if (bytesRequested > bytesAvailable) {
- bytesRequested = bytesAvailable;
+ if (pDecoder == NULL) {
+ return MA_INVALID_ARGS;
}
- *pSizeInBytes = bytesRequested;
- *ppBufferOut = ma_rb__get_write_ptr(pRB);
-
- /* Clear the buffer if desired. */
- if (pRB->clearOnWriteAcquire) {
- MA_ZERO_MEMORY(*ppBufferOut, *pSizeInBytes);
- }
+ *pCursor = pDecoder->readPointerInPCMFrames;
return MA_SUCCESS;
}
-MA_API ma_result ma_rb_commit_write(ma_rb* pRB, size_t sizeInBytes, void* pBufferOut)
+MA_API ma_result ma_decoder_get_length_in_pcm_frames(ma_decoder* pDecoder, ma_uint64* pLength)
{
- ma_uint32 writeOffset;
- ma_uint32 writeOffsetInBytes;
- ma_uint32 writeOffsetLoopFlag;
- ma_uint32 newWriteOffsetInBytes;
- ma_uint32 newWriteOffsetLoopFlag;
-
- if (pRB == NULL) {
+ if (pLength == NULL) {
return MA_INVALID_ARGS;
}
- /* Validate the buffer. */
- if (pBufferOut != ma_rb__get_write_ptr(pRB)) {
+ *pLength = 0;
+
+ if (pDecoder == NULL) {
return MA_INVALID_ARGS;
}
- writeOffset = c89atomic_load_32(&pRB->encodedWriteOffset);
- ma_rb__deconstruct_offset(writeOffset, &writeOffsetInBytes, &writeOffsetLoopFlag);
+ if (pDecoder->pBackend != NULL) {
+ ma_result result;
+ ma_uint64 internalLengthInPCMFrames;
+ ma_uint32 internalSampleRate;
- /* Check that sizeInBytes is correct. It should never go beyond the end of the buffer. */
- newWriteOffsetInBytes = (ma_uint32)(writeOffsetInBytes + sizeInBytes);
- if (newWriteOffsetInBytes > pRB->subbufferSizeInBytes) {
- return MA_INVALID_ARGS; /* <-- sizeInBytes will cause the read offset to overflow. */
- }
+ result = ma_data_source_get_length_in_pcm_frames(pDecoder->pBackend, &internalLengthInPCMFrames);
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to retrieve the internal length. */
+ }
- /* Move the read pointer back to the start if necessary. */
- newWriteOffsetLoopFlag = writeOffsetLoopFlag;
- if (newWriteOffsetInBytes == pRB->subbufferSizeInBytes) {
- newWriteOffsetInBytes = 0;
- newWriteOffsetLoopFlag ^= 0x80000000;
- }
+ result = ma_data_source_get_data_format(pDecoder->pBackend, NULL, NULL, &internalSampleRate, NULL, 0);
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to retrieve the internal sample rate. */
+ }
- c89atomic_exchange_32(&pRB->encodedWriteOffset, ma_rb__construct_offset(newWriteOffsetLoopFlag, newWriteOffsetInBytes));
+ if (internalSampleRate == pDecoder->outputSampleRate) {
+ *pLength = internalLengthInPCMFrames;
+ } else {
+ *pLength = ma_calculate_frame_count_after_resampling(pDecoder->outputSampleRate, internalSampleRate, internalLengthInPCMFrames);
+ }
- if (ma_rb_pointer_distance(pRB) == 0) {
- return MA_AT_END;
- } else {
return MA_SUCCESS;
+ } else {
+ return MA_NO_BACKEND;
}
}
-MA_API ma_result ma_rb_seek_read(ma_rb* pRB, size_t offsetInBytes)
+MA_API ma_result ma_decoder_get_available_frames(ma_decoder* pDecoder, ma_uint64* pAvailableFrames)
{
- ma_uint32 readOffset;
- ma_uint32 readOffsetInBytes;
- ma_uint32 readOffsetLoopFlag;
- ma_uint32 writeOffset;
- ma_uint32 writeOffsetInBytes;
- ma_uint32 writeOffsetLoopFlag;
- ma_uint32 newReadOffsetInBytes;
- ma_uint32 newReadOffsetLoopFlag;
+ ma_result result;
+ ma_uint64 totalFrameCount;
- if (pRB == NULL || offsetInBytes > pRB->subbufferSizeInBytes) {
+ if (pAvailableFrames == NULL) {
return MA_INVALID_ARGS;
}
- readOffset = c89atomic_load_32(&pRB->encodedReadOffset);
- ma_rb__deconstruct_offset(readOffset, &readOffsetInBytes, &readOffsetLoopFlag);
+ *pAvailableFrames = 0;
- writeOffset = c89atomic_load_32(&pRB->encodedWriteOffset);
- ma_rb__deconstruct_offset(writeOffset, &writeOffsetInBytes, &writeOffsetLoopFlag);
+ if (pDecoder == NULL) {
+ return MA_INVALID_ARGS;
+ }
- newReadOffsetLoopFlag = readOffsetLoopFlag;
+ result = ma_decoder_get_length_in_pcm_frames(pDecoder, &totalFrameCount);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- /* We cannot go past the write buffer. */
- if (readOffsetLoopFlag == writeOffsetLoopFlag) {
- if ((readOffsetInBytes + offsetInBytes) > writeOffsetInBytes) {
- newReadOffsetInBytes = writeOffsetInBytes;
- } else {
- newReadOffsetInBytes = (ma_uint32)(readOffsetInBytes + offsetInBytes);
- }
+ if (totalFrameCount <= pDecoder->readPointerInPCMFrames) {
+ *pAvailableFrames = 0;
} else {
- /* May end up looping. */
- if ((readOffsetInBytes + offsetInBytes) >= pRB->subbufferSizeInBytes) {
- newReadOffsetInBytes = (ma_uint32)(readOffsetInBytes + offsetInBytes) - pRB->subbufferSizeInBytes;
- newReadOffsetLoopFlag ^= 0x80000000; /* <-- Looped. */
- } else {
- newReadOffsetInBytes = (ma_uint32)(readOffsetInBytes + offsetInBytes);
- }
+ *pAvailableFrames = totalFrameCount - pDecoder->readPointerInPCMFrames;
}
- c89atomic_exchange_32(&pRB->encodedReadOffset, ma_rb__construct_offset(newReadOffsetInBytes, newReadOffsetLoopFlag));
return MA_SUCCESS;
}
-MA_API ma_result ma_rb_seek_write(ma_rb* pRB, size_t offsetInBytes)
+
+static ma_result ma_decoder__full_decode_and_uninit(ma_decoder* pDecoder, ma_decoder_config* pConfigOut, ma_uint64* pFrameCountOut, void** ppPCMFramesOut)
{
- ma_uint32 readOffset;
- ma_uint32 readOffsetInBytes;
- ma_uint32 readOffsetLoopFlag;
- ma_uint32 writeOffset;
- ma_uint32 writeOffsetInBytes;
- ma_uint32 writeOffsetLoopFlag;
- ma_uint32 newWriteOffsetInBytes;
- ma_uint32 newWriteOffsetLoopFlag;
+ ma_result result;
+ ma_uint64 totalFrameCount;
+ ma_uint64 bpf;
+ ma_uint64 dataCapInFrames;
+ void* pPCMFramesOut;
+
+ MA_ASSERT(pDecoder != NULL);
- if (pRB == NULL) {
- return MA_INVALID_ARGS;
- }
+ totalFrameCount = 0;
+ bpf = ma_get_bytes_per_frame(pDecoder->outputFormat, pDecoder->outputChannels);
- readOffset = c89atomic_load_32(&pRB->encodedReadOffset);
- ma_rb__deconstruct_offset(readOffset, &readOffsetInBytes, &readOffsetLoopFlag);
+ /* The frame count is unknown until we try reading. Thus, we just run in a loop. */
+ dataCapInFrames = 0;
+ pPCMFramesOut = NULL;
+ for (;;) {
+ ma_uint64 frameCountToTryReading;
+ ma_uint64 framesJustRead;
- writeOffset = c89atomic_load_32(&pRB->encodedWriteOffset);
- ma_rb__deconstruct_offset(writeOffset, &writeOffsetInBytes, &writeOffsetLoopFlag);
+ /* Make room if there's not enough. */
+ if (totalFrameCount == dataCapInFrames) {
+ void* pNewPCMFramesOut;
+ ma_uint64 newDataCapInFrames = dataCapInFrames*2;
+ if (newDataCapInFrames == 0) {
+ newDataCapInFrames = 4096;
+ }
- newWriteOffsetLoopFlag = writeOffsetLoopFlag;
+ if ((newDataCapInFrames * bpf) > MA_SIZE_MAX) {
+ ma_free(pPCMFramesOut, &pDecoder->allocationCallbacks);
+ return MA_TOO_BIG;
+ }
- /* We cannot go past the write buffer. */
- if (readOffsetLoopFlag == writeOffsetLoopFlag) {
- /* May end up looping. */
- if ((writeOffsetInBytes + offsetInBytes) >= pRB->subbufferSizeInBytes) {
- newWriteOffsetInBytes = (ma_uint32)(writeOffsetInBytes + offsetInBytes) - pRB->subbufferSizeInBytes;
- newWriteOffsetLoopFlag ^= 0x80000000; /* <-- Looped. */
- } else {
- newWriteOffsetInBytes = (ma_uint32)(writeOffsetInBytes + offsetInBytes);
- }
- } else {
- if ((writeOffsetInBytes + offsetInBytes) > readOffsetInBytes) {
- newWriteOffsetInBytes = readOffsetInBytes;
- } else {
- newWriteOffsetInBytes = (ma_uint32)(writeOffsetInBytes + offsetInBytes);
+ pNewPCMFramesOut = (void*)ma_realloc(pPCMFramesOut, (size_t)(newDataCapInFrames * bpf), &pDecoder->allocationCallbacks);
+ if (pNewPCMFramesOut == NULL) {
+ ma_free(pPCMFramesOut, &pDecoder->allocationCallbacks);
+ return MA_OUT_OF_MEMORY;
+ }
+
+ dataCapInFrames = newDataCapInFrames;
+ pPCMFramesOut = pNewPCMFramesOut;
}
- }
- c89atomic_exchange_32(&pRB->encodedWriteOffset, ma_rb__construct_offset(newWriteOffsetInBytes, newWriteOffsetLoopFlag));
- return MA_SUCCESS;
-}
+ frameCountToTryReading = dataCapInFrames - totalFrameCount;
+ MA_ASSERT(frameCountToTryReading > 0);
-MA_API ma_int32 ma_rb_pointer_distance(ma_rb* pRB)
-{
- ma_uint32 readOffset;
- ma_uint32 readOffsetInBytes;
- ma_uint32 readOffsetLoopFlag;
- ma_uint32 writeOffset;
- ma_uint32 writeOffsetInBytes;
- ma_uint32 writeOffsetLoopFlag;
+ result = ma_decoder_read_pcm_frames(pDecoder, (ma_uint8*)pPCMFramesOut + (totalFrameCount * bpf), frameCountToTryReading, &framesJustRead);
+ totalFrameCount += framesJustRead;
- if (pRB == NULL) {
- return 0;
+ if (result != MA_SUCCESS) {
+ break;
+ }
+
+ if (framesJustRead < frameCountToTryReading) {
+ break;
+ }
}
- readOffset = c89atomic_load_32(&pRB->encodedReadOffset);
- ma_rb__deconstruct_offset(readOffset, &readOffsetInBytes, &readOffsetLoopFlag);
- writeOffset = c89atomic_load_32(&pRB->encodedWriteOffset);
- ma_rb__deconstruct_offset(writeOffset, &writeOffsetInBytes, &writeOffsetLoopFlag);
+ if (pConfigOut != NULL) {
+ pConfigOut->format = pDecoder->outputFormat;
+ pConfigOut->channels = pDecoder->outputChannels;
+ pConfigOut->sampleRate = pDecoder->outputSampleRate;
+ }
- if (readOffsetLoopFlag == writeOffsetLoopFlag) {
- return writeOffsetInBytes - readOffsetInBytes;
+ if (ppPCMFramesOut != NULL) {
+ *ppPCMFramesOut = pPCMFramesOut;
} else {
- return writeOffsetInBytes + (pRB->subbufferSizeInBytes - readOffsetInBytes);
+ ma_free(pPCMFramesOut, &pDecoder->allocationCallbacks);
+ }
+
+ if (pFrameCountOut != NULL) {
+ *pFrameCountOut = totalFrameCount;
}
+
+ ma_decoder_uninit(pDecoder);
+ return MA_SUCCESS;
}
-MA_API ma_uint32 ma_rb_available_read(ma_rb* pRB)
+MA_API ma_result ma_decode_from_vfs(ma_vfs* pVFS, const char* pFilePath, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut)
{
- ma_int32 dist;
+ ma_result result;
+ ma_decoder_config config;
+ ma_decoder decoder;
- if (pRB == NULL) {
- return 0;
+ if (pFrameCountOut != NULL) {
+ *pFrameCountOut = 0;
}
-
- dist = ma_rb_pointer_distance(pRB);
- if (dist < 0) {
- return 0;
+ if (ppPCMFramesOut != NULL) {
+ *ppPCMFramesOut = NULL;
}
- return dist;
-}
+ config = ma_decoder_config_init_copy(pConfig);
-MA_API ma_uint32 ma_rb_available_write(ma_rb* pRB)
-{
- if (pRB == NULL) {
- return 0;
+ result = ma_decoder_init_vfs(pVFS, pFilePath, &config, &decoder);
+ if (result != MA_SUCCESS) {
+ return result;
}
- return (ma_uint32)(ma_rb_get_subbuffer_size(pRB) - ma_rb_pointer_distance(pRB));
+ result = ma_decoder__full_decode_and_uninit(&decoder, pConfig, pFrameCountOut, ppPCMFramesOut);
+
+ return result;
}
-MA_API size_t ma_rb_get_subbuffer_size(ma_rb* pRB)
+MA_API ma_result ma_decode_file(const char* pFilePath, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut)
{
- if (pRB == NULL) {
- return 0;
- }
-
- return pRB->subbufferSizeInBytes;
+ return ma_decode_from_vfs(NULL, pFilePath, pConfig, pFrameCountOut, ppPCMFramesOut);
}
-MA_API size_t ma_rb_get_subbuffer_stride(ma_rb* pRB)
+MA_API ma_result ma_decode_memory(const void* pData, size_t dataSize, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut)
{
- if (pRB == NULL) {
- return 0;
- }
+ ma_decoder_config config;
+ ma_decoder decoder;
+ ma_result result;
- if (pRB->subbufferStrideInBytes == 0) {
- return (size_t)pRB->subbufferSizeInBytes;
+ if (pFrameCountOut != NULL) {
+ *pFrameCountOut = 0;
+ }
+ if (ppPCMFramesOut != NULL) {
+ *ppPCMFramesOut = NULL;
}
- return (size_t)pRB->subbufferStrideInBytes;
-}
-
-MA_API size_t ma_rb_get_subbuffer_offset(ma_rb* pRB, size_t subbufferIndex)
-{
- if (pRB == NULL) {
- return 0;
+ if (pData == NULL || dataSize == 0) {
+ return MA_INVALID_ARGS;
}
- return subbufferIndex * ma_rb_get_subbuffer_stride(pRB);
-}
+ config = ma_decoder_config_init_copy(pConfig);
-MA_API void* ma_rb_get_subbuffer_ptr(ma_rb* pRB, size_t subbufferIndex, void* pBuffer)
-{
- if (pRB == NULL) {
- return NULL;
+ result = ma_decoder_init_memory(pData, dataSize, &config, &decoder);
+ if (result != MA_SUCCESS) {
+ return result;
}
- return ma_offset_ptr(pBuffer, ma_rb_get_subbuffer_offset(pRB, subbufferIndex));
+ return ma_decoder__full_decode_and_uninit(&decoder, pConfig, pFrameCountOut, ppPCMFramesOut);
}
+#endif /* MA_NO_DECODING */
+#ifndef MA_NO_ENCODING
-static MA_INLINE ma_uint32 ma_pcm_rb_get_bpf(ma_pcm_rb* pRB)
+#if defined(MA_HAS_WAV)
+static size_t ma_encoder__internal_on_write_wav(void* pUserData, const void* pData, size_t bytesToWrite)
{
- MA_ASSERT(pRB != NULL);
+ ma_encoder* pEncoder = (ma_encoder*)pUserData;
+ size_t bytesWritten = 0;
- return ma_get_bytes_per_frame(pRB->format, pRB->channels);
+ MA_ASSERT(pEncoder != NULL);
+
+ pEncoder->onWrite(pEncoder, pData, bytesToWrite, &bytesWritten);
+ return bytesWritten;
}
-MA_API ma_result ma_pcm_rb_init_ex(ma_format format, ma_uint32 channels, ma_uint32 subbufferSizeInFrames, ma_uint32 subbufferCount, ma_uint32 subbufferStrideInFrames, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_pcm_rb* pRB)
+static drwav_bool32 ma_encoder__internal_on_seek_wav(void* pUserData, int offset, drwav_seek_origin origin)
{
- ma_uint32 bpf;
+ ma_encoder* pEncoder = (ma_encoder*)pUserData;
ma_result result;
- if (pRB == NULL) {
- return MA_INVALID_ARGS;
+ MA_ASSERT(pEncoder != NULL);
+
+ result = pEncoder->onSeek(pEncoder, offset, (origin == drwav_seek_origin_start) ? ma_seek_origin_start : ma_seek_origin_current);
+ if (result != MA_SUCCESS) {
+ return DRWAV_FALSE;
+ } else {
+ return DRWAV_TRUE;
}
+}
- MA_ZERO_OBJECT(pRB);
+static ma_result ma_encoder__on_init_wav(ma_encoder* pEncoder)
+{
+ drwav_data_format wavFormat;
+ drwav_allocation_callbacks allocationCallbacks;
+ drwav* pWav;
- bpf = ma_get_bytes_per_frame(format, channels);
- if (bpf == 0) {
- return MA_INVALID_ARGS;
+ MA_ASSERT(pEncoder != NULL);
+
+ pWav = (drwav*)ma_malloc(sizeof(*pWav), &pEncoder->config.allocationCallbacks);
+ if (pWav == NULL) {
+ return MA_OUT_OF_MEMORY;
}
- result = ma_rb_init_ex(subbufferSizeInFrames*bpf, subbufferCount, subbufferStrideInFrames*bpf, pOptionalPreallocatedBuffer, pAllocationCallbacks, &pRB->rb);
- if (result != MA_SUCCESS) {
- return result;
+ wavFormat.container = drwav_container_riff;
+ wavFormat.channels = pEncoder->config.channels;
+ wavFormat.sampleRate = pEncoder->config.sampleRate;
+ wavFormat.bitsPerSample = ma_get_bytes_per_sample(pEncoder->config.format) * 8;
+ if (pEncoder->config.format == ma_format_f32) {
+ wavFormat.format = DR_WAVE_FORMAT_IEEE_FLOAT;
+ } else {
+ wavFormat.format = DR_WAVE_FORMAT_PCM;
}
- pRB->format = format;
- pRB->channels = channels;
+ allocationCallbacks.pUserData = pEncoder->config.allocationCallbacks.pUserData;
+ allocationCallbacks.onMalloc = pEncoder->config.allocationCallbacks.onMalloc;
+ allocationCallbacks.onRealloc = pEncoder->config.allocationCallbacks.onRealloc;
+ allocationCallbacks.onFree = pEncoder->config.allocationCallbacks.onFree;
- return MA_SUCCESS;
-}
+ if (!drwav_init_write(pWav, &wavFormat, ma_encoder__internal_on_write_wav, ma_encoder__internal_on_seek_wav, pEncoder, &allocationCallbacks)) {
+ return MA_ERROR;
+ }
-MA_API ma_result ma_pcm_rb_init(ma_format format, ma_uint32 channels, ma_uint32 bufferSizeInFrames, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_pcm_rb* pRB)
-{
- return ma_pcm_rb_init_ex(format, channels, bufferSizeInFrames, 1, 0, pOptionalPreallocatedBuffer, pAllocationCallbacks, pRB);
+ pEncoder->pInternalEncoder = pWav;
+
+ return MA_SUCCESS;
}
-MA_API void ma_pcm_rb_uninit(ma_pcm_rb* pRB)
+static void ma_encoder__on_uninit_wav(ma_encoder* pEncoder)
{
- if (pRB == NULL) {
- return;
- }
+ drwav* pWav;
- ma_rb_uninit(&pRB->rb);
-}
+ MA_ASSERT(pEncoder != NULL);
-MA_API void ma_pcm_rb_reset(ma_pcm_rb* pRB)
-{
- if (pRB == NULL) {
- return;
- }
+ pWav = (drwav*)pEncoder->pInternalEncoder;
+ MA_ASSERT(pWav != NULL);
- ma_rb_reset(&pRB->rb);
+ drwav_uninit(pWav);
+ ma_free(pWav, &pEncoder->config.allocationCallbacks);
}
-MA_API ma_result ma_pcm_rb_acquire_read(ma_pcm_rb* pRB, ma_uint32* pSizeInFrames, void** ppBufferOut)
+static ma_result ma_encoder__on_write_pcm_frames_wav(ma_encoder* pEncoder, const void* pFramesIn, ma_uint64 frameCount, ma_uint64* pFramesWritten)
{
- size_t sizeInBytes;
- ma_result result;
+ drwav* pWav;
+ ma_uint64 framesWritten;
- if (pRB == NULL || pSizeInFrames == NULL) {
- return MA_INVALID_ARGS;
- }
+ MA_ASSERT(pEncoder != NULL);
- sizeInBytes = *pSizeInFrames * ma_pcm_rb_get_bpf(pRB);
+ pWav = (drwav*)pEncoder->pInternalEncoder;
+ MA_ASSERT(pWav != NULL);
- result = ma_rb_acquire_read(&pRB->rb, &sizeInBytes, ppBufferOut);
- if (result != MA_SUCCESS) {
- return result;
+ framesWritten = drwav_write_pcm_frames(pWav, frameCount, pFramesIn);
+
+ if (pFramesWritten != NULL) {
+ *pFramesWritten = framesWritten;
}
- *pSizeInFrames = (ma_uint32)(sizeInBytes / (size_t)ma_pcm_rb_get_bpf(pRB));
return MA_SUCCESS;
}
+#endif
-MA_API ma_result ma_pcm_rb_commit_read(ma_pcm_rb* pRB, ma_uint32 sizeInFrames, void* pBufferOut)
+MA_API ma_encoder_config ma_encoder_config_init(ma_encoding_format encodingFormat, ma_format format, ma_uint32 channels, ma_uint32 sampleRate)
{
- if (pRB == NULL) {
- return MA_INVALID_ARGS;
- }
+ ma_encoder_config config;
+
+ MA_ZERO_OBJECT(&config);
+ config.encodingFormat = encodingFormat;
+ config.format = format;
+ config.channels = channels;
+ config.sampleRate = sampleRate;
- return ma_rb_commit_read(&pRB->rb, sizeInFrames * ma_pcm_rb_get_bpf(pRB), pBufferOut);
+ return config;
}
-MA_API ma_result ma_pcm_rb_acquire_write(ma_pcm_rb* pRB, ma_uint32* pSizeInFrames, void** ppBufferOut)
+MA_API ma_result ma_encoder_preinit(const ma_encoder_config* pConfig, ma_encoder* pEncoder)
{
- size_t sizeInBytes;
ma_result result;
- if (pRB == NULL) {
+ if (pEncoder == NULL) {
return MA_INVALID_ARGS;
}
- sizeInBytes = *pSizeInFrames * ma_pcm_rb_get_bpf(pRB);
+ MA_ZERO_OBJECT(pEncoder);
- result = ma_rb_acquire_write(&pRB->rb, &sizeInBytes, ppBufferOut);
- if (result != MA_SUCCESS) {
- return result;
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- *pSizeInFrames = (ma_uint32)(sizeInBytes / ma_pcm_rb_get_bpf(pRB));
- return MA_SUCCESS;
-}
-
-MA_API ma_result ma_pcm_rb_commit_write(ma_pcm_rb* pRB, ma_uint32 sizeInFrames, void* pBufferOut)
-{
- if (pRB == NULL) {
+ if (pConfig->format == ma_format_unknown || pConfig->channels == 0 || pConfig->sampleRate == 0) {
return MA_INVALID_ARGS;
}
- return ma_rb_commit_write(&pRB->rb, sizeInFrames * ma_pcm_rb_get_bpf(pRB), pBufferOut);
-}
+ pEncoder->config = *pConfig;
-MA_API ma_result ma_pcm_rb_seek_read(ma_pcm_rb* pRB, ma_uint32 offsetInFrames)
-{
- if (pRB == NULL) {
- return MA_INVALID_ARGS;
+ result = ma_allocation_callbacks_init_copy(&pEncoder->config.allocationCallbacks, &pConfig->allocationCallbacks);
+ if (result != MA_SUCCESS) {
+ return result;
}
- return ma_rb_seek_read(&pRB->rb, offsetInFrames * ma_pcm_rb_get_bpf(pRB));
+ return MA_SUCCESS;
}
-MA_API ma_result ma_pcm_rb_seek_write(ma_pcm_rb* pRB, ma_uint32 offsetInFrames)
+MA_API ma_result ma_encoder_init__internal(ma_encoder_write_proc onWrite, ma_encoder_seek_proc onSeek, void* pUserData, ma_encoder* pEncoder)
{
- if (pRB == NULL) {
+ ma_result result = MA_SUCCESS;
+
+ /* This assumes ma_encoder_preinit() has been called prior. */
+ MA_ASSERT(pEncoder != NULL);
+
+ if (onWrite == NULL || onSeek == NULL) {
return MA_INVALID_ARGS;
}
- return ma_rb_seek_write(&pRB->rb, offsetInFrames * ma_pcm_rb_get_bpf(pRB));
-}
+ pEncoder->onWrite = onWrite;
+ pEncoder->onSeek = onSeek;
+ pEncoder->pUserData = pUserData;
-MA_API ma_int32 ma_pcm_rb_pointer_distance(ma_pcm_rb* pRB)
-{
- if (pRB == NULL) {
- return 0;
- }
+ switch (pEncoder->config.encodingFormat)
+ {
+ case ma_encoding_format_wav:
+ {
+ #if defined(MA_HAS_WAV)
+ pEncoder->onInit = ma_encoder__on_init_wav;
+ pEncoder->onUninit = ma_encoder__on_uninit_wav;
+ pEncoder->onWritePCMFrames = ma_encoder__on_write_pcm_frames_wav;
+ #else
+ result = MA_NO_BACKEND;
+ #endif
+ } break;
- return ma_rb_pointer_distance(&pRB->rb) / ma_pcm_rb_get_bpf(pRB);
-}
+ default:
+ {
+ result = MA_INVALID_ARGS;
+ } break;
+ }
-MA_API ma_uint32 ma_pcm_rb_available_read(ma_pcm_rb* pRB)
-{
- if (pRB == NULL) {
- return 0;
+ /* Getting here means we should have our backend callbacks set up. */
+ if (result == MA_SUCCESS) {
+ result = pEncoder->onInit(pEncoder);
}
- return ma_rb_available_read(&pRB->rb) / ma_pcm_rb_get_bpf(pRB);
+ return result;
}
-MA_API ma_uint32 ma_pcm_rb_available_write(ma_pcm_rb* pRB)
+static ma_result ma_encoder__on_write_vfs(ma_encoder* pEncoder, const void* pBufferIn, size_t bytesToWrite, size_t* pBytesWritten)
{
- if (pRB == NULL) {
- return 0;
- }
-
- return ma_rb_available_write(&pRB->rb) / ma_pcm_rb_get_bpf(pRB);
+ return ma_vfs_or_default_write(pEncoder->data.vfs.pVFS, pEncoder->data.vfs.file, pBufferIn, bytesToWrite, pBytesWritten);
}
-MA_API ma_uint32 ma_pcm_rb_get_subbuffer_size(ma_pcm_rb* pRB)
+static ma_result ma_encoder__on_seek_vfs(ma_encoder* pEncoder, ma_int64 offset, ma_seek_origin origin)
{
- if (pRB == NULL) {
- return 0;
- }
-
- return (ma_uint32)(ma_rb_get_subbuffer_size(&pRB->rb) / ma_pcm_rb_get_bpf(pRB));
+ return ma_vfs_or_default_seek(pEncoder->data.vfs.pVFS, pEncoder->data.vfs.file, offset, origin);
}
-MA_API ma_uint32 ma_pcm_rb_get_subbuffer_stride(ma_pcm_rb* pRB)
+MA_API ma_result ma_encoder_init_vfs(ma_vfs* pVFS, const char* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder)
{
- if (pRB == NULL) {
- return 0;
- }
+ ma_result result;
+ ma_vfs_file file;
- return (ma_uint32)(ma_rb_get_subbuffer_stride(&pRB->rb) / ma_pcm_rb_get_bpf(pRB));
-}
+ result = ma_encoder_preinit(pConfig, pEncoder);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
-MA_API ma_uint32 ma_pcm_rb_get_subbuffer_offset(ma_pcm_rb* pRB, ma_uint32 subbufferIndex)
-{
- if (pRB == NULL) {
- return 0;
+ /* Now open the file. If this fails we don't need to uninitialize the encoder. */
+ result = ma_vfs_or_default_open(pVFS, pFilePath, MA_OPEN_MODE_WRITE, &file);
+ if (result != MA_SUCCESS) {
+ return result;
}
- return (ma_uint32)(ma_rb_get_subbuffer_offset(&pRB->rb, subbufferIndex) / ma_pcm_rb_get_bpf(pRB));
-}
+ pEncoder->data.vfs.pVFS = pVFS;
+ pEncoder->data.vfs.file = file;
-MA_API void* ma_pcm_rb_get_subbuffer_ptr(ma_pcm_rb* pRB, ma_uint32 subbufferIndex, void* pBuffer)
-{
- if (pRB == NULL) {
- return NULL;
+ result = ma_encoder_init__internal(ma_encoder__on_write_vfs, ma_encoder__on_seek_vfs, NULL, pEncoder);
+ if (result != MA_SUCCESS) {
+ ma_vfs_or_default_close(pVFS, file);
+ return result;
}
- return ma_rb_get_subbuffer_ptr(&pRB->rb, subbufferIndex, pBuffer);
+ return MA_SUCCESS;
}
-
-
-MA_API ma_result ma_duplex_rb_init(ma_format captureFormat, ma_uint32 captureChannels, ma_uint32 sampleRate, ma_uint32 captureInternalSampleRate, ma_uint32 captureInternalPeriodSizeInFrames, const ma_allocation_callbacks* pAllocationCallbacks, ma_duplex_rb* pRB)
+MA_API ma_result ma_encoder_init_vfs_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder)
{
ma_result result;
- ma_uint32 sizeInFrames;
+ ma_vfs_file file;
- sizeInFrames = (ma_uint32)ma_calculate_frame_count_after_resampling(sampleRate, captureInternalSampleRate, captureInternalPeriodSizeInFrames * 5);
- if (sizeInFrames == 0) {
- return MA_INVALID_ARGS;
+ result = ma_encoder_preinit(pConfig, pEncoder);
+ if (result != MA_SUCCESS) {
+ return result;
}
- result = ma_pcm_rb_init(captureFormat, captureChannels, sizeInFrames, NULL, pAllocationCallbacks, &pRB->rb);
+ /* Now open the file. If this fails we don't need to uninitialize the encoder. */
+ result = ma_vfs_or_default_open_w(pVFS, pFilePath, MA_OPEN_MODE_WRITE, &file);
if (result != MA_SUCCESS) {
return result;
}
- /* Seek forward a bit so we have a bit of a buffer in case of desyncs. */
- ma_pcm_rb_seek_write((ma_pcm_rb*)pRB, captureInternalPeriodSizeInFrames * 2);
+ pEncoder->data.vfs.pVFS = pVFS;
+ pEncoder->data.vfs.file = file;
+
+ result = ma_encoder_init__internal(ma_encoder__on_write_vfs, ma_encoder__on_seek_vfs, NULL, pEncoder);
+ if (result != MA_SUCCESS) {
+ ma_vfs_or_default_close(pVFS, file);
+ return result;
+ }
return MA_SUCCESS;
}
-MA_API ma_result ma_duplex_rb_uninit(ma_duplex_rb* pRB)
+MA_API ma_result ma_encoder_init_file(const char* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder)
{
- ma_pcm_rb_uninit((ma_pcm_rb*)pRB);
- return MA_SUCCESS;
+ return ma_encoder_init_vfs(NULL, pFilePath, pConfig, pEncoder);
+}
+
+MA_API ma_result ma_encoder_init_file_w(const wchar_t* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder)
+{
+ return ma_encoder_init_vfs_w(NULL, pFilePath, pConfig, pEncoder);
}
+MA_API ma_result ma_encoder_init(ma_encoder_write_proc onWrite, ma_encoder_seek_proc onSeek, void* pUserData, const ma_encoder_config* pConfig, ma_encoder* pEncoder)
+{
+ ma_result result;
+ result = ma_encoder_preinit(pConfig, pEncoder);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
-/**************************************************************************************************************************************************************
+ return ma_encoder_init__internal(onWrite, onSeek, pUserData, pEncoder);
+}
-Miscellaneous Helpers
-**************************************************************************************************************************************************************/
-MA_API const char* ma_result_description(ma_result result)
+MA_API void ma_encoder_uninit(ma_encoder* pEncoder)
{
- switch (result)
- {
- case MA_SUCCESS: return "No error";
- case MA_ERROR: return "Unknown error";
- case MA_INVALID_ARGS: return "Invalid argument";
- case MA_INVALID_OPERATION: return "Invalid operation";
- case MA_OUT_OF_MEMORY: return "Out of memory";
- case MA_OUT_OF_RANGE: return "Out of range";
- case MA_ACCESS_DENIED: return "Permission denied";
- case MA_DOES_NOT_EXIST: return "Resource does not exist";
- case MA_ALREADY_EXISTS: return "Resource already exists";
- case MA_TOO_MANY_OPEN_FILES: return "Too many open files";
- case MA_INVALID_FILE: return "Invalid file";
- case MA_TOO_BIG: return "Too large";
- case MA_PATH_TOO_LONG: return "Path too long";
- case MA_NAME_TOO_LONG: return "Name too long";
- case MA_NOT_DIRECTORY: return "Not a directory";
- case MA_IS_DIRECTORY: return "Is a directory";
- case MA_DIRECTORY_NOT_EMPTY: return "Directory not empty";
- case MA_AT_END: return "At end";
- case MA_NO_SPACE: return "No space available";
- case MA_BUSY: return "Device or resource busy";
- case MA_IO_ERROR: return "Input/output error";
- case MA_INTERRUPT: return "Interrupted";
- case MA_UNAVAILABLE: return "Resource unavailable";
- case MA_ALREADY_IN_USE: return "Resource already in use";
- case MA_BAD_ADDRESS: return "Bad address";
- case MA_BAD_SEEK: return "Illegal seek";
- case MA_BAD_PIPE: return "Broken pipe";
- case MA_DEADLOCK: return "Deadlock";
- case MA_TOO_MANY_LINKS: return "Too many links";
- case MA_NOT_IMPLEMENTED: return "Not implemented";
- case MA_NO_MESSAGE: return "No message of desired type";
- case MA_BAD_MESSAGE: return "Invalid message";
- case MA_NO_DATA_AVAILABLE: return "No data available";
- case MA_INVALID_DATA: return "Invalid data";
- case MA_TIMEOUT: return "Timeout";
- case MA_NO_NETWORK: return "Network unavailable";
- case MA_NOT_UNIQUE: return "Not unique";
- case MA_NOT_SOCKET: return "Socket operation on non-socket";
- case MA_NO_ADDRESS: return "Destination address required";
- case MA_BAD_PROTOCOL: return "Protocol wrong type for socket";
- case MA_PROTOCOL_UNAVAILABLE: return "Protocol not available";
- case MA_PROTOCOL_NOT_SUPPORTED: return "Protocol not supported";
- case MA_PROTOCOL_FAMILY_NOT_SUPPORTED: return "Protocol family not supported";
- case MA_ADDRESS_FAMILY_NOT_SUPPORTED: return "Address family not supported";
- case MA_SOCKET_NOT_SUPPORTED: return "Socket type not supported";
- case MA_CONNECTION_RESET: return "Connection reset";
- case MA_ALREADY_CONNECTED: return "Already connected";
- case MA_NOT_CONNECTED: return "Not connected";
- case MA_CONNECTION_REFUSED: return "Connection refused";
- case MA_NO_HOST: return "No host";
- case MA_IN_PROGRESS: return "Operation in progress";
- case MA_CANCELLED: return "Operation cancelled";
- case MA_MEMORY_ALREADY_MAPPED: return "Memory already mapped";
+ if (pEncoder == NULL) {
+ return;
+ }
- case MA_FORMAT_NOT_SUPPORTED: return "Format not supported";
- case MA_DEVICE_TYPE_NOT_SUPPORTED: return "Device type not supported";
- case MA_SHARE_MODE_NOT_SUPPORTED: return "Share mode not supported";
- case MA_NO_BACKEND: return "No backend";
- case MA_NO_DEVICE: return "No device";
- case MA_API_NOT_FOUND: return "API not found";
- case MA_INVALID_DEVICE_CONFIG: return "Invalid device config";
+ if (pEncoder->onUninit) {
+ pEncoder->onUninit(pEncoder);
+ }
- case MA_DEVICE_NOT_INITIALIZED: return "Device not initialized";
- case MA_DEVICE_NOT_STARTED: return "Device not started";
+ /* If we have a file handle, close it. */
+ if (pEncoder->onWrite == ma_encoder__on_write_vfs) {
+ ma_vfs_or_default_close(pEncoder->data.vfs.pVFS, pEncoder->data.vfs.file);
+ pEncoder->data.vfs.file = NULL;
+ }
+}
- case MA_FAILED_TO_INIT_BACKEND: return "Failed to initialize backend";
- case MA_FAILED_TO_OPEN_BACKEND_DEVICE: return "Failed to open backend device";
- case MA_FAILED_TO_START_BACKEND_DEVICE: return "Failed to start backend device";
- case MA_FAILED_TO_STOP_BACKEND_DEVICE: return "Failed to stop backend device";
- default: return "Unknown error";
+MA_API ma_result ma_encoder_write_pcm_frames(ma_encoder* pEncoder, const void* pFramesIn, ma_uint64 frameCount, ma_uint64* pFramesWritten)
+{
+ if (pFramesWritten != NULL) {
+ *pFramesWritten = 0;
+ }
+
+ if (pEncoder == NULL || pFramesIn == NULL) {
+ return MA_INVALID_ARGS;
}
+
+ return pEncoder->onWritePCMFrames(pEncoder, pFramesIn, frameCount, pFramesWritten);
}
+#endif /* MA_NO_ENCODING */
-MA_API void* ma_malloc(size_t sz, const ma_allocation_callbacks* pAllocationCallbacks)
+
+
+/**************************************************************************************************************************************************************
+
+Generation
+
+**************************************************************************************************************************************************************/
+#ifndef MA_NO_GENERATION
+MA_API ma_waveform_config ma_waveform_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_waveform_type type, double amplitude, double frequency)
{
- if (pAllocationCallbacks != NULL) {
- return ma__malloc_from_callbacks(sz, pAllocationCallbacks);
- } else {
- return ma__malloc_default(sz, NULL);
- }
+ ma_waveform_config config;
+
+ MA_ZERO_OBJECT(&config);
+ config.format = format;
+ config.channels = channels;
+ config.sampleRate = sampleRate;
+ config.type = type;
+ config.amplitude = amplitude;
+ config.frequency = frequency;
+
+ return config;
}
-MA_API void* ma_realloc(void* p, size_t sz, const ma_allocation_callbacks* pAllocationCallbacks)
+static ma_result ma_waveform__data_source_on_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
- if (pAllocationCallbacks != NULL) {
- if (pAllocationCallbacks->onRealloc != NULL) {
- return pAllocationCallbacks->onRealloc(p, sz, pAllocationCallbacks->pUserData);
- } else {
- return NULL; /* This requires a native implementation of realloc(). */
- }
- } else {
- return ma__realloc_default(p, sz, NULL);
- }
+ return ma_waveform_read_pcm_frames((ma_waveform*)pDataSource, pFramesOut, frameCount, pFramesRead);
}
-MA_API void ma_free(void* p, const ma_allocation_callbacks* pAllocationCallbacks)
+static ma_result ma_waveform__data_source_on_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
- if (pAllocationCallbacks != NULL) {
- ma__free_from_callbacks(p, pAllocationCallbacks);
- } else {
- ma__free_default(p, NULL);
- }
+ return ma_waveform_seek_to_pcm_frame((ma_waveform*)pDataSource, frameIndex);
}
-MA_API void* ma_aligned_malloc(size_t sz, size_t alignment, const ma_allocation_callbacks* pAllocationCallbacks)
+static ma_result ma_waveform__data_source_on_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
- size_t extraBytes;
- void* pUnaligned;
- void* pAligned;
+ ma_waveform* pWaveform = (ma_waveform*)pDataSource;
- if (alignment == 0) {
- return 0;
- }
+ *pFormat = pWaveform->config.format;
+ *pChannels = pWaveform->config.channels;
+ *pSampleRate = pWaveform->config.sampleRate;
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, pWaveform->config.channels);
- extraBytes = alignment-1 + sizeof(void*);
+ return MA_SUCCESS;
+}
- pUnaligned = ma_malloc(sz + extraBytes, pAllocationCallbacks);
- if (pUnaligned == NULL) {
- return NULL;
- }
+static ma_result ma_waveform__data_source_on_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
+{
+ ma_waveform* pWaveform = (ma_waveform*)pDataSource;
- pAligned = (void*)(((ma_uintptr)pUnaligned + extraBytes) & ~((ma_uintptr)(alignment-1)));
- ((void**)pAligned)[-1] = pUnaligned;
+ *pCursor = (ma_uint64)(pWaveform->time / pWaveform->advance);
- return pAligned;
+ return MA_SUCCESS;
}
-MA_API void ma_aligned_free(void* p, const ma_allocation_callbacks* pAllocationCallbacks)
+static double ma_waveform__calculate_advance(ma_uint32 sampleRate, double frequency)
{
- ma_free(((void**)p)[-1], pAllocationCallbacks);
+ return (1.0 / (sampleRate / frequency));
}
-MA_API const char* ma_get_format_name(ma_format format)
+static void ma_waveform__update_advance(ma_waveform* pWaveform)
{
- switch (format)
- {
- case ma_format_unknown: return "Unknown";
- case ma_format_u8: return "8-bit Unsigned Integer";
- case ma_format_s16: return "16-bit Signed Integer";
- case ma_format_s24: return "24-bit Signed Integer (Tightly Packed)";
- case ma_format_s32: return "32-bit Signed Integer";
- case ma_format_f32: return "32-bit IEEE Floating Point";
- default: return "Invalid";
- }
+ pWaveform->advance = ma_waveform__calculate_advance(pWaveform->config.sampleRate, pWaveform->config.frequency);
}
-MA_API void ma_blend_f32(float* pOut, float* pInA, float* pInB, float factor, ma_uint32 channels)
+static ma_data_source_vtable g_ma_waveform_data_source_vtable =
{
- ma_uint32 i;
- for (i = 0; i < channels; ++i) {
- pOut[i] = ma_mix_f32(pInA[i], pInB[i], factor);
- }
-}
-
+ ma_waveform__data_source_on_read,
+ ma_waveform__data_source_on_seek,
+ ma_waveform__data_source_on_get_data_format,
+ ma_waveform__data_source_on_get_cursor,
+ NULL, /* onGetLength. There's no notion of a length in waveforms. */
+ NULL, /* onSetLooping */
+ 0
+};
-MA_API ma_uint32 ma_get_bytes_per_sample(ma_format format)
+MA_API ma_result ma_waveform_init(const ma_waveform_config* pConfig, ma_waveform* pWaveform)
{
- ma_uint32 sizes[] = {
- 0, /* unknown */
- 1, /* u8 */
- 2, /* s16 */
- 3, /* s24 */
- 4, /* s32 */
- 4, /* f32 */
- };
- return sizes[format];
-}
+ ma_result result;
+ ma_data_source_config dataSourceConfig;
+ if (pWaveform == NULL) {
+ return MA_INVALID_ARGS;
+ }
+ MA_ZERO_OBJECT(pWaveform);
-MA_API ma_data_source_config ma_data_source_config_init(void)
-{
- ma_data_source_config config;
+ dataSourceConfig = ma_data_source_config_init();
+ dataSourceConfig.vtable = &g_ma_waveform_data_source_vtable;
- MA_ZERO_OBJECT(&config);
+ result = ma_data_source_init(&dataSourceConfig, &pWaveform->ds);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- return config;
-}
+ pWaveform->config = *pConfig;
+ pWaveform->advance = ma_waveform__calculate_advance(pWaveform->config.sampleRate, pWaveform->config.frequency);
+ pWaveform->time = 0;
+ return MA_SUCCESS;
+}
-MA_API ma_result ma_data_source_init(const ma_data_source_config* pConfig, ma_data_source* pDataSource)
+MA_API void ma_waveform_uninit(ma_waveform* pWaveform)
{
- ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
+ if (pWaveform == NULL) {
+ return;
+ }
- if (pDataSource == NULL) {
+ ma_data_source_uninit(&pWaveform->ds);
+}
+
+MA_API ma_result ma_waveform_set_amplitude(ma_waveform* pWaveform, double amplitude)
+{
+ if (pWaveform == NULL) {
return MA_INVALID_ARGS;
}
- MA_ZERO_OBJECT(pDataSourceBase);
+ pWaveform->config.amplitude = amplitude;
+ return MA_SUCCESS;
+}
- if (pConfig == NULL) {
+MA_API ma_result ma_waveform_set_frequency(ma_waveform* pWaveform, double frequency)
+{
+ if (pWaveform == NULL) {
return MA_INVALID_ARGS;
}
- pDataSourceBase->vtable = pConfig->vtable;
- pDataSourceBase->rangeBegInFrames = 0;
- pDataSourceBase->rangeEndInFrames = ~((ma_uint64)0);
- pDataSourceBase->loopBegInFrames = 0;
- pDataSourceBase->loopEndInFrames = ~((ma_uint64)0);
- pDataSourceBase->pCurrent = pDataSource; /* Always read from ourself by default. */
- pDataSourceBase->pNext = NULL;
- pDataSourceBase->onGetNext = NULL;
+ pWaveform->config.frequency = frequency;
+ ma_waveform__update_advance(pWaveform);
- /* Compatibility: Need to make a copy of the callbacks. This will be removed in version 0.11. */
- if (pConfig->vtable != NULL) {
- pDataSourceBase->cb = *pConfig->vtable;
+ return MA_SUCCESS;
+}
+
+MA_API ma_result ma_waveform_set_type(ma_waveform* pWaveform, ma_waveform_type type)
+{
+ if (pWaveform == NULL) {
+ return MA_INVALID_ARGS;
}
+ pWaveform->config.type = type;
return MA_SUCCESS;
}
-MA_API void ma_data_source_uninit(ma_data_source* pDataSource)
+MA_API ma_result ma_waveform_set_sample_rate(ma_waveform* pWaveform, ma_uint32 sampleRate)
{
- if (pDataSource == NULL) {
- return;
+ if (pWaveform == NULL) {
+ return MA_INVALID_ARGS;
}
- /*
- This is placeholder in case we need this later. Data sources need to call this in their
- uninitialization routine to ensure things work later on if something is added here.
- */
+ pWaveform->config.sampleRate = sampleRate;
+ ma_waveform__update_advance(pWaveform);
+
+ return MA_SUCCESS;
}
-#if defined(MA_EXPERIMENTAL__DATA_LOOPING_AND_CHAINING)
-static ma_result ma_data_source_resolve_current(ma_data_source* pDataSource, ma_data_source** ppCurrentDataSource)
+static float ma_waveform_sine_f32(double time, double amplitude)
{
- ma_data_source_base* pCurrentDataSource = (ma_data_source_base*)pDataSource;
+ return (float)(ma_sind(MA_TAU_D * time) * amplitude);
+}
- MA_ASSERT(pDataSource != NULL);
- MA_ASSERT(ppCurrentDataSource != NULL);
+static ma_int16 ma_waveform_sine_s16(double time, double amplitude)
+{
+ return ma_pcm_sample_f32_to_s16(ma_waveform_sine_f32(time, amplitude));
+}
- if (pCurrentDataSource->pCurrent == NULL) {
- /*
- The current data source is NULL. If we're using this in the context of a chain we need to return NULL
- here so that we don't end up looping. Otherwise we just return the data source itself.
- */
- if (pCurrentDataSource->pNext != NULL || pCurrentDataSource->onGetNext != NULL) {
- pCurrentDataSource = NULL;
- } else {
- pCurrentDataSource = (ma_data_source_base*)pDataSource; /* Not being used in a chain. Make sure we just always read from the data source itself at all times. */
- }
+static float ma_waveform_square_f32(double time, double amplitude)
+{
+ double f = time - (ma_int64)time;
+ double r;
+
+ if (f < 0.5) {
+ r = amplitude;
} else {
- pCurrentDataSource = (ma_data_source_base*)pCurrentDataSource->pCurrent;
+ r = -amplitude;
}
- *ppCurrentDataSource = pCurrentDataSource;
+ return (float)r;
+}
- return MA_SUCCESS;
+static ma_int16 ma_waveform_square_s16(double time, double amplitude)
+{
+ return ma_pcm_sample_f32_to_s16(ma_waveform_square_f32(time, amplitude));
}
-static ma_result ma_data_source_read_pcm_frames_within_range(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead, ma_bool32 loop)
+static float ma_waveform_triangle_f32(double time, double amplitude)
{
- ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
-
- if (pDataSourceBase == NULL) {
- return MA_AT_END;
- }
+ double f = time - (ma_int64)time;
+ double r;
- if (pDataSourceBase->rangeEndInFrames == ~((ma_uint64)0) && (pDataSourceBase->loopEndInFrames == ~((ma_uint64)0) || loop == MA_FALSE)) {
- /* No range is set - just read like normal. The data source itself will tell us when the end is reached. */
- return pDataSourceBase->cb.onRead(pDataSourceBase, pFramesOut, frameCount, pFramesRead);
- } else {
- /* Need to clamp to within the range. */
- ma_result result;
- ma_uint64 cursor;
- ma_uint64 framesRead = 0;
- ma_uint64 rangeEnd;
+ r = 2 * ma_abs(2 * (f - 0.5)) - 1;
- result = ma_data_source_get_cursor_in_pcm_frames(pDataSourceBase, &cursor);
- if (result != MA_SUCCESS) {
- /* Failed to retrieve the cursor. Cannot read within a range or loop points. Just read like normal - this may happen for things like noise data sources where it doesn't really matter. */
- return pDataSourceBase->cb.onRead(pDataSourceBase, pFramesOut, frameCount, pFramesRead);
- }
+ return (float)(r * amplitude);
+}
+
+static ma_int16 ma_waveform_triangle_s16(double time, double amplitude)
+{
+ return ma_pcm_sample_f32_to_s16(ma_waveform_triangle_f32(time, amplitude));
+}
- /* We have the cursor. We need to make sure we don't read beyond our range. */
- rangeEnd = pDataSourceBase->rangeEndInFrames;
+static float ma_waveform_sawtooth_f32(double time, double amplitude)
+{
+ double f = time - (ma_int64)time;
+ double r;
+
+ r = 2 * (f - 0.5);
+
+ return (float)(r * amplitude);
+}
+
+static ma_int16 ma_waveform_sawtooth_s16(double time, double amplitude)
+{
+ return ma_pcm_sample_f32_to_s16(ma_waveform_sawtooth_f32(time, amplitude));
+}
+
+static void ma_waveform_read_pcm_frames__sine(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount)
+{
+ ma_uint64 iFrame;
+ ma_uint64 iChannel;
+ ma_uint32 bps = ma_get_bytes_per_sample(pWaveform->config.format);
+ ma_uint32 bpf = bps * pWaveform->config.channels;
+
+ MA_ASSERT(pWaveform != NULL);
+ MA_ASSERT(pFramesOut != NULL);
+
+ if (pWaveform->config.format == ma_format_f32) {
+ float* pFramesOutF32 = (float*)pFramesOut;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ float s = ma_waveform_sine_f32(pWaveform->time, pWaveform->config.amplitude);
+ pWaveform->time += pWaveform->advance;
+
+ for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
+ pFramesOutF32[iFrame*pWaveform->config.channels + iChannel] = s;
+ }
+ }
+ } else if (pWaveform->config.format == ma_format_s16) {
+ ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_int16 s = ma_waveform_sine_s16(pWaveform->time, pWaveform->config.amplitude);
+ pWaveform->time += pWaveform->advance;
- /* If looping, make sure we're within range. */
- if (loop) {
- if (pDataSourceBase->loopEndInFrames != ~((ma_uint64)0)) {
- rangeEnd = ma_min(rangeEnd, pDataSourceBase->rangeBegInFrames + pDataSourceBase->loopEndInFrames);
+ for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
+ pFramesOutS16[iFrame*pWaveform->config.channels + iChannel] = s;
}
}
+ } else {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ float s = ma_waveform_sine_f32(pWaveform->time, pWaveform->config.amplitude);
+ pWaveform->time += pWaveform->advance;
- if (frameCount > (rangeEnd - cursor) && rangeEnd != ~((ma_uint64)0)) {
- frameCount = (rangeEnd - cursor);
+ for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
+ ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pWaveform->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
+ }
}
+ }
+}
+
+static void ma_waveform_read_pcm_frames__square(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount)
+{
+ ma_uint64 iFrame;
+ ma_uint64 iChannel;
+ ma_uint32 bps = ma_get_bytes_per_sample(pWaveform->config.format);
+ ma_uint32 bpf = bps * pWaveform->config.channels;
- result = pDataSourceBase->cb.onRead(pDataSourceBase, pFramesOut, frameCount, &framesRead);
+ MA_ASSERT(pWaveform != NULL);
+ MA_ASSERT(pFramesOut != NULL);
- if (pFramesRead != NULL) {
- *pFramesRead = framesRead;
+ if (pWaveform->config.format == ma_format_f32) {
+ float* pFramesOutF32 = (float*)pFramesOut;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ float s = ma_waveform_square_f32(pWaveform->time, pWaveform->config.amplitude);
+ pWaveform->time += pWaveform->advance;
+
+ for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
+ pFramesOutF32[iFrame*pWaveform->config.channels + iChannel] = s;
+ }
}
+ } else if (pWaveform->config.format == ma_format_s16) {
+ ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_int16 s = ma_waveform_square_s16(pWaveform->time, pWaveform->config.amplitude);
+ pWaveform->time += pWaveform->advance;
- /* We need to make sure MA_AT_END is returned if we hit the end of the range. */
- if (result != MA_AT_END && framesRead == 0) {
- result = MA_AT_END;
+ for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
+ pFramesOutS16[iFrame*pWaveform->config.channels + iChannel] = s;
+ }
}
+ } else {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ float s = ma_waveform_square_f32(pWaveform->time, pWaveform->config.amplitude);
+ pWaveform->time += pWaveform->advance;
- return result;
+ for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
+ ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pWaveform->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
+ }
+ }
}
}
-#endif
-MA_API ma_result ma_data_source_read_pcm_frames(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead, ma_bool32 loop)
+static void ma_waveform_read_pcm_frames__triangle(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount)
{
-#if defined(MA_EXPERIMENTAL__DATA_LOOPING_AND_CHAINING)
- ma_result result = MA_SUCCESS;
- ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
- ma_data_source_base* pCurrentDataSource;
- void* pRunningFramesOut = pFramesOut;
- ma_uint64 totalFramesProcessed = 0;
- ma_format format;
- ma_uint32 channels;
- ma_uint32 emptyLoopCounter = 0; /* Keeps track of how many times 0 frames have been read. For infinite loop detection of sounds with no audio data. */
+ ma_uint64 iFrame;
+ ma_uint64 iChannel;
+ ma_uint32 bps = ma_get_bytes_per_sample(pWaveform->config.format);
+ ma_uint32 bpf = bps * pWaveform->config.channels;
- if (pFramesRead != NULL) {
- *pFramesRead = 0;
- }
+ MA_ASSERT(pWaveform != NULL);
+ MA_ASSERT(pFramesOut != NULL);
- if (pDataSourceBase == NULL) {
- return MA_INVALID_ARGS;
- }
+ if (pWaveform->config.format == ma_format_f32) {
+ float* pFramesOutF32 = (float*)pFramesOut;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ float s = ma_waveform_triangle_f32(pWaveform->time, pWaveform->config.amplitude);
+ pWaveform->time += pWaveform->advance;
- /*
- We need to know the data format so we can advance the output buffer as we read frames. If this
- fails, chaining will not work and we'll just read as much as we can from the current source.
- */
- if (ma_data_source_get_data_format(pDataSource, &format, &channels, NULL) != MA_SUCCESS) {
- result = ma_data_source_resolve_current(pDataSource, (ma_data_source**)&pCurrentDataSource);
- if (result != MA_SUCCESS) {
- return result;
+ for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
+ pFramesOutF32[iFrame*pWaveform->config.channels + iChannel] = s;
+ }
+ }
+ } else if (pWaveform->config.format == ma_format_s16) {
+ ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_int16 s = ma_waveform_triangle_s16(pWaveform->time, pWaveform->config.amplitude);
+ pWaveform->time += pWaveform->advance;
+
+ for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
+ pFramesOutS16[iFrame*pWaveform->config.channels + iChannel] = s;
+ }
}
+ } else {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ float s = ma_waveform_triangle_f32(pWaveform->time, pWaveform->config.amplitude);
+ pWaveform->time += pWaveform->advance;
- return ma_data_source_read_pcm_frames_within_range(pCurrentDataSource, pFramesOut, frameCount, pFramesRead, loop);
+ for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
+ ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pWaveform->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
+ }
+ }
}
+}
- /*
- Looping is a bit of a special case. When the `loop` argument is true, chaining will not work and
- only the current data source will be read from.
- */
+static void ma_waveform_read_pcm_frames__sawtooth(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount)
+{
+ ma_uint64 iFrame;
+ ma_uint64 iChannel;
+ ma_uint32 bps = ma_get_bytes_per_sample(pWaveform->config.format);
+ ma_uint32 bpf = bps * pWaveform->config.channels;
- /* Keep reading until we've read as many frames as possible. */
- while (totalFramesProcessed < frameCount) {
- ma_uint64 framesProcessed;
- ma_uint64 framesRemaining = frameCount - totalFramesProcessed;
+ MA_ASSERT(pWaveform != NULL);
+ MA_ASSERT(pFramesOut != NULL);
- /* We need to resolve the data source that we'll actually be reading from. */
- result = ma_data_source_resolve_current(pDataSource, (ma_data_source**)&pCurrentDataSource);
- if (result != MA_SUCCESS) {
- break;
- }
+ if (pWaveform->config.format == ma_format_f32) {
+ float* pFramesOutF32 = (float*)pFramesOut;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ float s = ma_waveform_sawtooth_f32(pWaveform->time, pWaveform->config.amplitude);
+ pWaveform->time += pWaveform->advance;
- if (pCurrentDataSource == NULL) {
- break;
+ for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
+ pFramesOutF32[iFrame*pWaveform->config.channels + iChannel] = s;
+ }
}
+ } else if (pWaveform->config.format == ma_format_s16) {
+ ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_int16 s = ma_waveform_sawtooth_s16(pWaveform->time, pWaveform->config.amplitude);
+ pWaveform->time += pWaveform->advance;
- result = ma_data_source_read_pcm_frames_within_range(pCurrentDataSource, pRunningFramesOut, framesRemaining, &framesProcessed, loop);
- totalFramesProcessed += framesProcessed;
+ for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
+ pFramesOutS16[iFrame*pWaveform->config.channels + iChannel] = s;
+ }
+ }
+ } else {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ float s = ma_waveform_sawtooth_f32(pWaveform->time, pWaveform->config.amplitude);
+ pWaveform->time += pWaveform->advance;
- /*
- If we encounted an error from the read callback, make sure it's propagated to the caller. The caller may need to know whether or not MA_BUSY is returned which is
- not necessarily considered an error.
- */
- if (result != MA_SUCCESS && result != MA_AT_END) {
- break;
+ for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
+ ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pWaveform->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
+ }
}
+ }
+}
- /*
- We can determine if we've reached the end by checking the return value of the onRead()
- callback. To loop back to the start, all we need to do is seek back to the first frame.
- */
- if (result == MA_AT_END) {
- /*
- We reached the end. If we're looping, we just loop back to the start of the current
- data source. If we're not looping we need to check if we have another in the chain, and
- if so, switch to it.
- */
- if (loop) {
- if (framesProcessed == 0) {
- emptyLoopCounter += 1;
- if (emptyLoopCounter > 1) {
- break; /* Infinite loop detected. Get out. */
- }
- } else {
- emptyLoopCounter = 0;
- }
+MA_API ma_result ma_waveform_read_pcm_frames(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+{
+ if (pFramesRead != NULL) {
+ *pFramesRead = 0;
+ }
- if (ma_data_source_seek_to_pcm_frame(pCurrentDataSource, pCurrentDataSource->loopBegInFrames) != MA_SUCCESS) {
- break; /* Failed to loop. Abort. */
- }
+ if (frameCount == 0) {
+ return MA_INVALID_ARGS;
+ }
- /* Don't return MA_AT_END for looping sounds. */
- result = MA_SUCCESS;
- } else {
- if (pCurrentDataSource->pNext != NULL) {
- pDataSourceBase->pCurrent = pCurrentDataSource->pNext;
- } else if (pCurrentDataSource->onGetNext != NULL) {
- pDataSourceBase->pCurrent = pCurrentDataSource->onGetNext(pCurrentDataSource);
- if (pDataSourceBase->pCurrent == NULL) {
- break; /* Our callback did not return a next data source. We're done. */
- }
- } else {
- /* Reached the end of the chain. We're done. */
- break;
- }
+ if (pWaveform == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (pFramesOut != NULL) {
+ switch (pWaveform->config.type)
+ {
+ case ma_waveform_type_sine:
+ {
+ ma_waveform_read_pcm_frames__sine(pWaveform, pFramesOut, frameCount);
+ } break;
+
+ case ma_waveform_type_square:
+ {
+ ma_waveform_read_pcm_frames__square(pWaveform, pFramesOut, frameCount);
+ } break;
- /* The next data source needs to be rewound to ensure data is read in looping scenarios. */
- ma_data_source_seek_to_pcm_frame(pDataSourceBase->pCurrent, 0);
+ case ma_waveform_type_triangle:
+ {
+ ma_waveform_read_pcm_frames__triangle(pWaveform, pFramesOut, frameCount);
+ } break;
- /*
- We need to make sure we clear the MA_AT_END result so we don't accidentally return
- it in the event that we coincidentally ended reading at the exact transition point
- of two data sources in a chain.
- */
- result = MA_SUCCESS;
- }
- }
+ case ma_waveform_type_sawtooth:
+ {
+ ma_waveform_read_pcm_frames__sawtooth(pWaveform, pFramesOut, frameCount);
+ } break;
- if (pRunningFramesOut != NULL) {
- pRunningFramesOut = ma_offset_ptr(pRunningFramesOut, framesProcessed * ma_get_bytes_per_frame(format, channels));
+ default: return MA_INVALID_OPERATION; /* Unknown waveform type. */
}
+ } else {
+ pWaveform->time += pWaveform->advance * (ma_int64)frameCount; /* Cast to int64 required for VC6. Won't affect anything in practice. */
}
if (pFramesRead != NULL) {
- *pFramesRead = totalFramesProcessed;
+ *pFramesRead = frameCount;
}
- return result;
-#else
- ma_data_source_callbacks* pCallbacks = (ma_data_source_callbacks*)pDataSource;
-
- /* Safety. */
- if (pFramesRead != NULL) {
- *pFramesRead = 0;
- }
+ return MA_SUCCESS;
+}
- if (pCallbacks == NULL) {
+MA_API ma_result ma_waveform_seek_to_pcm_frame(ma_waveform* pWaveform, ma_uint64 frameIndex)
+{
+ if (pWaveform == NULL) {
return MA_INVALID_ARGS;
}
- if (pCallbacks->onRead == NULL) {
- return MA_NOT_IMPLEMENTED;
- }
+ pWaveform->time = pWaveform->advance * (ma_int64)frameIndex; /* Casting for VC6. Won't be an issue in practice. */
- /* A very small optimization for the non looping case. */
- if (loop == MA_FALSE) {
- return pCallbacks->onRead(pDataSource, pFramesOut, frameCount, pFramesRead);
- } else {
- ma_format format;
- ma_uint32 channels;
- ma_uint32 sampleRate;
- if (ma_data_source_get_data_format(pDataSource, &format, &channels, &sampleRate) != MA_SUCCESS) {
- return pCallbacks->onRead(pDataSource, pFramesOut, frameCount, pFramesRead); /* We don't have a way to retrieve the data format which means we don't know how to offset the output buffer. Just read as much as we can. */
- } else {
- ma_result result = MA_SUCCESS;
- ma_uint64 totalFramesProcessed;
- void* pRunningFramesOut = pFramesOut;
+ return MA_SUCCESS;
+}
- totalFramesProcessed = 0;
- while (totalFramesProcessed < frameCount) {
- ma_uint64 framesProcessed;
- ma_uint64 framesRemaining = frameCount - totalFramesProcessed;
- result = pCallbacks->onRead(pDataSource, pRunningFramesOut, framesRemaining, &framesProcessed);
- totalFramesProcessed += framesProcessed;
+MA_API ma_noise_config ma_noise_config_init(ma_format format, ma_uint32 channels, ma_noise_type type, ma_int32 seed, double amplitude)
+{
+ ma_noise_config config;
+ MA_ZERO_OBJECT(&config);
- /*
- If we encounted an error from the read callback, make sure it's propagated to the caller. The caller may need to know whether or not MA_BUSY is returned which is
- not necessarily considered an error.
- */
- if (result != MA_SUCCESS && result != MA_AT_END) {
- break;
- }
+ config.format = format;
+ config.channels = channels;
+ config.type = type;
+ config.seed = seed;
+ config.amplitude = amplitude;
- /*
- We can determine if we've reached the end by checking the return value of the onRead() callback. If it's less than what we requested it means
- we've reached the end. To loop back to the start, all we need to do is seek back to the first frame.
- */
- if (framesProcessed < framesRemaining || result == MA_AT_END) {
- if (ma_data_source_seek_to_pcm_frame(pDataSource, 0) != MA_SUCCESS) {
- break;
- }
- }
+ if (config.seed == 0) {
+ config.seed = MA_DEFAULT_LCG_SEED;
+ }
- if (pRunningFramesOut != NULL) {
- pRunningFramesOut = ma_offset_ptr(pRunningFramesOut, framesProcessed * ma_get_bytes_per_frame(format, channels));
- }
- }
+ return config;
+}
- if (pFramesRead != NULL) {
- *pFramesRead = totalFramesProcessed;
- }
- return result;
- }
- }
-#endif
+static ma_result ma_noise__data_source_on_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+{
+ return ma_noise_read_pcm_frames((ma_noise*)pDataSource, pFramesOut, frameCount, pFramesRead);
}
-MA_API ma_result ma_data_source_seek_pcm_frames(ma_data_source* pDataSource, ma_uint64 frameCount, ma_uint64* pFramesSeeked, ma_bool32 loop)
+static ma_result ma_noise__data_source_on_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
- return ma_data_source_read_pcm_frames(pDataSource, NULL, frameCount, pFramesSeeked, loop);
+ /* No-op. Just pretend to be successful. */
+ (void)pDataSource;
+ (void)frameIndex;
+ return MA_SUCCESS;
}
-MA_API ma_result ma_data_source_seek_to_pcm_frame(ma_data_source* pDataSource, ma_uint64 frameIndex)
+static ma_result ma_noise__data_source_on_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
-#if defined(MA_EXPERIMENTAL__DATA_LOOPING_AND_CHAINING)
- ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
+ ma_noise* pNoise = (ma_noise*)pDataSource;
- if (pDataSourceBase == NULL) {
- return MA_SUCCESS;
- }
+ *pFormat = pNoise->config.format;
+ *pChannels = pNoise->config.channels;
+ *pSampleRate = 0; /* There is no notion of sample rate with noise generation. */
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, pNoise->config.channels);
- if (pDataSourceBase->cb.onSeek == NULL) {
- return MA_NOT_IMPLEMENTED;
- }
+ return MA_SUCCESS;
+}
- if (frameIndex > pDataSourceBase->rangeEndInFrames) {
- return MA_INVALID_OPERATION; /* Trying to seek to far forward. */
- }
+static ma_data_source_vtable g_ma_noise_data_source_vtable =
+{
+ ma_noise__data_source_on_read,
+ ma_noise__data_source_on_seek, /* No-op for noise. */
+ ma_noise__data_source_on_get_data_format,
+ NULL, /* onGetCursor. No notion of a cursor for noise. */
+ NULL, /* onGetLength. No notion of a length for noise. */
+ NULL, /* onSetLooping */
+ 0
+};
- return pDataSourceBase->cb.onSeek(pDataSource, pDataSourceBase->rangeBegInFrames + frameIndex);
-#else
- ma_data_source_callbacks* pCallbacks = (ma_data_source_callbacks*)pDataSource;
- if (pCallbacks == NULL) {
+
+#ifndef MA_PINK_NOISE_BIN_SIZE
+#define MA_PINK_NOISE_BIN_SIZE 16
+#endif
+
+typedef struct
+{
+ size_t sizeInBytes;
+ struct
+ {
+ size_t binOffset;
+ size_t accumulationOffset;
+ size_t counterOffset;
+ } pink;
+ struct
+ {
+ size_t accumulationOffset;
+ } brownian;
+} ma_noise_heap_layout;
+
+static ma_result ma_noise_get_heap_layout(const ma_noise_config* pConfig, ma_noise_heap_layout* pHeapLayout)
+{
+ MA_ASSERT(pHeapLayout != NULL);
+
+ MA_ZERO_OBJECT(pHeapLayout);
+
+ if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
- if (pCallbacks->onSeek == NULL) {
- return MA_NOT_IMPLEMENTED;
+ if (pConfig->channels == 0) {
+ return MA_INVALID_ARGS;
}
- return pCallbacks->onSeek(pDataSource, frameIndex);
-#endif
-}
+ pHeapLayout->sizeInBytes = 0;
-MA_API ma_result ma_data_source_map(ma_data_source* pDataSource, void** ppFramesOut, ma_uint64* pFrameCount)
-{
- ma_data_source_callbacks* pCallbacks = (ma_data_source_callbacks*)pDataSource;
- if (pCallbacks == NULL) {
- return MA_INVALID_ARGS;
+ /* Pink. */
+ if (pConfig->type == ma_noise_type_pink) {
+ /* bin */
+ pHeapLayout->pink.binOffset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += sizeof(double*) * pConfig->channels;
+ pHeapLayout->sizeInBytes += sizeof(double ) * pConfig->channels * MA_PINK_NOISE_BIN_SIZE;
+
+ /* accumulation */
+ pHeapLayout->pink.accumulationOffset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += sizeof(double) * pConfig->channels;
+
+ /* counter */
+ pHeapLayout->pink.counterOffset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += sizeof(ma_uint32) * pConfig->channels;
}
- if (pCallbacks->onMap == NULL) {
- return MA_NOT_IMPLEMENTED;
+ /* Brownian. */
+ if (pConfig->type == ma_noise_type_brownian) {
+ /* accumulation */
+ pHeapLayout->brownian.accumulationOffset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += sizeof(double) * pConfig->channels;
}
- return pCallbacks->onMap(pDataSource, ppFramesOut, pFrameCount);
+ /* Make sure allocation size is aligned. */
+ pHeapLayout->sizeInBytes = ma_align_64(pHeapLayout->sizeInBytes);
+
+ return MA_SUCCESS;
}
-MA_API ma_result ma_data_source_unmap(ma_data_source* pDataSource, ma_uint64 frameCount)
+MA_API ma_result ma_noise_get_heap_size(const ma_noise_config* pConfig, size_t* pHeapSizeInBytes)
{
- ma_data_source_callbacks* pCallbacks = (ma_data_source_callbacks*)pDataSource;
- if (pCallbacks == NULL) {
+ ma_result result;
+ ma_noise_heap_layout heapLayout;
+
+ if (pHeapSizeInBytes == NULL) {
return MA_INVALID_ARGS;
}
- if (pCallbacks->onUnmap == NULL) {
- return MA_NOT_IMPLEMENTED;
+ *pHeapSizeInBytes = 0;
+
+ result = ma_noise_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
}
- return pCallbacks->onUnmap(pDataSource, frameCount);
+ *pHeapSizeInBytes = heapLayout.sizeInBytes;
+
+ return MA_SUCCESS;
}
-MA_API ma_result ma_data_source_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate)
+MA_API ma_result ma_noise_init_preallocated(const ma_noise_config* pConfig, void* pHeap, ma_noise* pNoise)
{
ma_result result;
- ma_format format;
- ma_uint32 channels;
- ma_uint32 sampleRate;
- ma_data_source_callbacks* pCallbacks = (ma_data_source_callbacks*)pDataSource;
+ ma_noise_heap_layout heapLayout;
+ ma_data_source_config dataSourceConfig;
+ ma_uint32 iChannel;
- if (pFormat != NULL) {
- *pFormat = ma_format_unknown;
+ if (pNoise == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pChannels != NULL) {
- *pChannels = 0;
- }
+ MA_ZERO_OBJECT(pNoise);
- if (pSampleRate != NULL) {
- *pSampleRate = 0;
+ result = ma_noise_get_heap_layout(pConfig, &heapLayout);
+ if (result != MA_SUCCESS) {
+ return result;
}
- if (pCallbacks == NULL) {
- return MA_INVALID_ARGS;
- }
+ pNoise->_pHeap = pHeap;
+ MA_ZERO_MEMORY(pNoise->_pHeap, heapLayout.sizeInBytes);
- if (pCallbacks->onGetDataFormat == NULL) {
- return MA_NOT_IMPLEMENTED;
- }
+ dataSourceConfig = ma_data_source_config_init();
+ dataSourceConfig.vtable = &g_ma_noise_data_source_vtable;
- result = pCallbacks->onGetDataFormat(pDataSource, &format, &channels, &sampleRate);
+ result = ma_data_source_init(&dataSourceConfig, &pNoise->ds);
if (result != MA_SUCCESS) {
return result;
}
- if (pFormat != NULL) {
- *pFormat = format;
- }
- if (pChannels != NULL) {
- *pChannels = channels;
+ pNoise->config = *pConfig;
+ ma_lcg_seed(&pNoise->lcg, pConfig->seed);
+
+ if (pNoise->config.type == ma_noise_type_pink) {
+ pNoise->state.pink.bin = (double** )ma_offset_ptr(pHeap, heapLayout.pink.binOffset);
+ pNoise->state.pink.accumulation = (double* )ma_offset_ptr(pHeap, heapLayout.pink.accumulationOffset);
+ pNoise->state.pink.counter = (ma_uint32*)ma_offset_ptr(pHeap, heapLayout.pink.counterOffset);
+
+ for (iChannel = 0; iChannel < pConfig->channels; iChannel += 1) {
+ pNoise->state.pink.bin[iChannel] = (double*)ma_offset_ptr(pHeap, heapLayout.pink.binOffset + (sizeof(double*) * pConfig->channels) + (sizeof(double) * MA_PINK_NOISE_BIN_SIZE * iChannel));
+ pNoise->state.pink.accumulation[iChannel] = 0;
+ pNoise->state.pink.counter[iChannel] = 1;
+ }
}
- if (pSampleRate != NULL) {
- *pSampleRate = sampleRate;
+
+ if (pNoise->config.type == ma_noise_type_brownian) {
+ pNoise->state.brownian.accumulation = (double*)ma_offset_ptr(pHeap, heapLayout.brownian.accumulationOffset);
+
+ for (iChannel = 0; iChannel < pConfig->channels; iChannel += 1) {
+ pNoise->state.brownian.accumulation[iChannel] = 0;
+ }
}
return MA_SUCCESS;
}
-MA_API ma_result ma_data_source_get_cursor_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pCursor)
+MA_API ma_result ma_noise_init(const ma_noise_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_noise* pNoise)
{
-#if defined(MA_EXPERIMENTAL__DATA_LOOPING_AND_CHAINING)
- ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
ma_result result;
- ma_uint64 cursor;
+ size_t heapSizeInBytes;
+ void* pHeap;
- if (pCursor == NULL) {
- return MA_INVALID_ARGS;
+ result = ma_noise_get_heap_size(pConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
}
- *pCursor = 0;
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+ } else {
+ pHeap = NULL;
+ }
- if (pDataSourceBase == NULL) {
- return MA_SUCCESS;
+ result = ma_noise_init_preallocated(pConfig, pHeap, pNoise);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
}
- if (pDataSourceBase->cb.onGetCursor == NULL) {
- return MA_NOT_IMPLEMENTED;
+ pNoise->_ownsHeap = MA_TRUE;
+ return MA_SUCCESS;
+}
+
+MA_API void ma_noise_uninit(ma_noise* pNoise, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ if (pNoise == NULL) {
+ return;
}
- result = pDataSourceBase->cb.onGetCursor(pDataSourceBase, &cursor);
- if (result != MA_SUCCESS) {
- return result;
+ ma_data_source_uninit(&pNoise->ds);
+
+ if (pNoise->_ownsHeap) {
+ ma_free(pNoise->_pHeap, pAllocationCallbacks);
}
+}
- /* The cursor needs to be made relative to the start of the range. */
- if (cursor < pDataSourceBase->rangeBegInFrames) { /* Safety check so we don't return some huge number. */
- *pCursor = 0;
- } else {
- *pCursor = cursor - pDataSourceBase->rangeBegInFrames;
+MA_API ma_result ma_noise_set_amplitude(ma_noise* pNoise, double amplitude)
+{
+ if (pNoise == NULL) {
+ return MA_INVALID_ARGS;
}
-
+
+ pNoise->config.amplitude = amplitude;
return MA_SUCCESS;
-#else
- ma_data_source_callbacks* pCallbacks = (ma_data_source_callbacks*)pDataSource;
+}
- if (pCursor == NULL) {
+MA_API ma_result ma_noise_set_seed(ma_noise* pNoise, ma_int32 seed)
+{
+ if (pNoise == NULL) {
return MA_INVALID_ARGS;
}
- *pCursor = 0;
+ pNoise->lcg.state = seed;
+ return MA_SUCCESS;
+}
- if (pCallbacks == NULL) {
+
+MA_API ma_result ma_noise_set_type(ma_noise* pNoise, ma_noise_type type)
+{
+ if (pNoise == NULL) {
return MA_INVALID_ARGS;
}
- if (pCallbacks->onGetCursor == NULL) {
- return MA_NOT_IMPLEMENTED;
- }
+ pNoise->config.type = type;
+ return MA_SUCCESS;
+}
- return pCallbacks->onGetCursor(pDataSource, pCursor);
-#endif
+static MA_INLINE float ma_noise_f32_white(ma_noise* pNoise)
+{
+ return (float)(ma_lcg_rand_f64(&pNoise->lcg) * pNoise->config.amplitude);
}
-MA_API ma_result ma_data_source_get_length_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pLength)
+static MA_INLINE ma_int16 ma_noise_s16_white(ma_noise* pNoise)
{
-#if defined(MA_EXPERIMENTAL__DATA_LOOPING_AND_CHAINING)
- ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
+ return ma_pcm_sample_f32_to_s16(ma_noise_f32_white(pNoise));
+}
- if (pLength == NULL) {
- return MA_INVALID_ARGS;
- }
+static MA_INLINE ma_uint64 ma_noise_read_pcm_frames__white(ma_noise* pNoise, void* pFramesOut, ma_uint64 frameCount)
+{
+ ma_uint64 iFrame;
+ ma_uint32 iChannel;
+ const ma_uint32 channels = pNoise->config.channels;
+ MA_ASSUME(channels > 0);
- *pLength = 0;
+ if (pNoise->config.format == ma_format_f32) {
+ float* pFramesOutF32 = (float*)pFramesOut;
+ if (pNoise->config.duplicateChannels) {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ float s = ma_noise_f32_white(pNoise);
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ pFramesOutF32[iFrame*channels + iChannel] = s;
+ }
+ }
+ } else {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ pFramesOutF32[iFrame*channels + iChannel] = ma_noise_f32_white(pNoise);
+ }
+ }
+ }
+ } else if (pNoise->config.format == ma_format_s16) {
+ ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
+ if (pNoise->config.duplicateChannels) {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_int16 s = ma_noise_s16_white(pNoise);
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ pFramesOutS16[iFrame*channels + iChannel] = s;
+ }
+ }
+ } else {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ pFramesOutS16[iFrame*channels + iChannel] = ma_noise_s16_white(pNoise);
+ }
+ }
+ }
+ } else {
+ const ma_uint32 bps = ma_get_bytes_per_sample(pNoise->config.format);
+ const ma_uint32 bpf = bps * channels;
- if (pDataSourceBase == NULL) {
- return MA_INVALID_ARGS;
+ if (pNoise->config.duplicateChannels) {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ float s = ma_noise_f32_white(pNoise);
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pNoise->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
+ }
+ }
+ } else {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ float s = ma_noise_f32_white(pNoise);
+ ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pNoise->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
+ }
+ }
+ }
}
- /*
- If we have a range defined we'll use that to determine the length. This is one of rare times
- where we'll actually trust the caller. If they've set the range, I think it's mostly safe to
- assume they've set it based on some higher level knowledge of the structure of the sound bank.
- */
- if (pDataSourceBase->rangeEndInFrames != ~((ma_uint64)0)) {
- *pLength = pDataSourceBase->rangeEndInFrames - pDataSourceBase->rangeBegInFrames;
- return MA_SUCCESS;
+ return frameCount;
+}
+
+
+static MA_INLINE unsigned int ma_tzcnt32(unsigned int x)
+{
+ unsigned int n;
+
+ /* Special case for odd numbers since they should happen about half the time. */
+ if (x & 0x1) {
+ return 0;
}
- /*
- Getting here means a range is not defined so we'll need to get the data source itself to tell
- us the length.
- */
- if (pDataSourceBase->cb.onGetLength == NULL) {
- return MA_NOT_IMPLEMENTED;
+ if (x == 0) {
+ return sizeof(x) << 3;
}
- return pDataSourceBase->cb.onGetLength(pDataSource, pLength);
-#else
- ma_data_source_callbacks* pCallbacks = (ma_data_source_callbacks*)pDataSource;
+ n = 1;
+ if ((x & 0x0000FFFF) == 0) { x >>= 16; n += 16; }
+ if ((x & 0x000000FF) == 0) { x >>= 8; n += 8; }
+ if ((x & 0x0000000F) == 0) { x >>= 4; n += 4; }
+ if ((x & 0x00000003) == 0) { x >>= 2; n += 2; }
+ n -= x & 0x00000001;
- if (pLength == NULL) {
- return MA_INVALID_ARGS;
- }
+ return n;
+}
- *pLength = 0;
+/*
+Pink noise generation based on Tonic (public domain) with modifications. https://github.com/TonicAudio/Tonic/blob/master/src/Tonic/Noise.h
- if (pCallbacks == NULL) {
- return MA_INVALID_ARGS;
+This is basically _the_ reference for pink noise from what I've found: http://www.firstpr.com.au/dsp/pink-noise/
+*/
+static MA_INLINE float ma_noise_f32_pink(ma_noise* pNoise, ma_uint32 iChannel)
+{
+ double result;
+ double binPrev;
+ double binNext;
+ unsigned int ibin;
+
+ ibin = ma_tzcnt32(pNoise->state.pink.counter[iChannel]) & (MA_PINK_NOISE_BIN_SIZE - 1);
+
+ binPrev = pNoise->state.pink.bin[iChannel][ibin];
+ binNext = ma_lcg_rand_f64(&pNoise->lcg);
+ pNoise->state.pink.bin[iChannel][ibin] = binNext;
+
+ pNoise->state.pink.accumulation[iChannel] += (binNext - binPrev);
+ pNoise->state.pink.counter[iChannel] += 1;
+
+ result = (ma_lcg_rand_f64(&pNoise->lcg) + pNoise->state.pink.accumulation[iChannel]);
+ result /= 10;
+
+ return (float)(result * pNoise->config.amplitude);
+}
+
+static MA_INLINE ma_int16 ma_noise_s16_pink(ma_noise* pNoise, ma_uint32 iChannel)
+{
+ return ma_pcm_sample_f32_to_s16(ma_noise_f32_pink(pNoise, iChannel));
+}
+
+static MA_INLINE ma_uint64 ma_noise_read_pcm_frames__pink(ma_noise* pNoise, void* pFramesOut, ma_uint64 frameCount)
+{
+ ma_uint64 iFrame;
+ ma_uint32 iChannel;
+ const ma_uint32 channels = pNoise->config.channels;
+ MA_ASSUME(channels > 0);
+
+ if (pNoise->config.format == ma_format_f32) {
+ float* pFramesOutF32 = (float*)pFramesOut;
+ if (pNoise->config.duplicateChannels) {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ float s = ma_noise_f32_pink(pNoise, 0);
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ pFramesOutF32[iFrame*channels + iChannel] = s;
+ }
+ }
+ } else {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ pFramesOutF32[iFrame*channels + iChannel] = ma_noise_f32_pink(pNoise, iChannel);
+ }
+ }
+ }
+ } else if (pNoise->config.format == ma_format_s16) {
+ ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
+ if (pNoise->config.duplicateChannels) {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_int16 s = ma_noise_s16_pink(pNoise, 0);
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ pFramesOutS16[iFrame*channels + iChannel] = s;
+ }
+ }
+ } else {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ pFramesOutS16[iFrame*channels + iChannel] = ma_noise_s16_pink(pNoise, iChannel);
+ }
+ }
+ }
+ } else {
+ const ma_uint32 bps = ma_get_bytes_per_sample(pNoise->config.format);
+ const ma_uint32 bpf = bps * channels;
+
+ if (pNoise->config.duplicateChannels) {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ float s = ma_noise_f32_pink(pNoise, 0);
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pNoise->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
+ }
+ }
+ } else {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ float s = ma_noise_f32_pink(pNoise, iChannel);
+ ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pNoise->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
+ }
+ }
+ }
}
- if (pCallbacks->onGetLength == NULL) {
- return MA_NOT_IMPLEMENTED;
+ return frameCount;
+}
+
+
+static MA_INLINE float ma_noise_f32_brownian(ma_noise* pNoise, ma_uint32 iChannel)
+{
+ double result;
+
+ result = (ma_lcg_rand_f64(&pNoise->lcg) + pNoise->state.brownian.accumulation[iChannel]);
+ result /= 1.005; /* Don't escape the -1..1 range on average. */
+
+ pNoise->state.brownian.accumulation[iChannel] = result;
+ result /= 20;
+
+ return (float)(result * pNoise->config.amplitude);
+}
+
+static MA_INLINE ma_int16 ma_noise_s16_brownian(ma_noise* pNoise, ma_uint32 iChannel)
+{
+ return ma_pcm_sample_f32_to_s16(ma_noise_f32_brownian(pNoise, iChannel));
+}
+
+static MA_INLINE ma_uint64 ma_noise_read_pcm_frames__brownian(ma_noise* pNoise, void* pFramesOut, ma_uint64 frameCount)
+{
+ ma_uint64 iFrame;
+ ma_uint32 iChannel;
+ const ma_uint32 channels = pNoise->config.channels;
+ MA_ASSUME(channels > 0);
+
+ if (pNoise->config.format == ma_format_f32) {
+ float* pFramesOutF32 = (float*)pFramesOut;
+ if (pNoise->config.duplicateChannels) {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ float s = ma_noise_f32_brownian(pNoise, 0);
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ pFramesOutF32[iFrame*channels + iChannel] = s;
+ }
+ }
+ } else {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ pFramesOutF32[iFrame*channels + iChannel] = ma_noise_f32_brownian(pNoise, iChannel);
+ }
+ }
+ }
+ } else if (pNoise->config.format == ma_format_s16) {
+ ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
+ if (pNoise->config.duplicateChannels) {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ ma_int16 s = ma_noise_s16_brownian(pNoise, 0);
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ pFramesOutS16[iFrame*channels + iChannel] = s;
+ }
+ }
+ } else {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ pFramesOutS16[iFrame*channels + iChannel] = ma_noise_s16_brownian(pNoise, iChannel);
+ }
+ }
+ }
+ } else {
+ const ma_uint32 bps = ma_get_bytes_per_sample(pNoise->config.format);
+ const ma_uint32 bpf = bps * channels;
+
+ if (pNoise->config.duplicateChannels) {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ float s = ma_noise_f32_brownian(pNoise, 0);
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pNoise->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
+ }
+ }
+ } else {
+ for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
+ for (iChannel = 0; iChannel < channels; iChannel += 1) {
+ float s = ma_noise_f32_brownian(pNoise, iChannel);
+ ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pNoise->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
+ }
+ }
+ }
}
- return pCallbacks->onGetLength(pDataSource, pLength);
-#endif
+ return frameCount;
}
-
-#if defined(MA_EXPERIMENTAL__DATA_LOOPING_AND_CHAINING)
-MA_API ma_result ma_data_source_set_range_in_pcm_frames(ma_data_source* pDataSource, ma_uint64 rangeBegInFrames, ma_uint64 rangeEndInFrames)
+MA_API ma_result ma_noise_read_pcm_frames(ma_noise* pNoise, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
- ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
- ma_result result;
- ma_uint64 cursor;
- ma_uint64 loopBegAbsolute;
- ma_uint64 loopEndAbsolute;
-
- if (pDataSource == NULL) {
- return MA_INVALID_ARGS;
- }
+ ma_uint64 framesRead = 0;
- if (rangeEndInFrames < rangeBegInFrames) {
- return MA_INVALID_ARGS; /* The end of the range must come after the beginning. */
+ if (pFramesRead != NULL) {
+ *pFramesRead = 0;
}
- /*
- The loop points need to be updated. We'll be storing the loop points relative to the range. We'll update
- these so that they maintain their absolute positioning. The loop points will then be clamped to the range.
- */
- loopBegAbsolute = pDataSourceBase->loopBegInFrames + pDataSourceBase->rangeBegInFrames;
- loopEndAbsolute = pDataSourceBase->loopEndInFrames + ((pDataSourceBase->loopEndInFrames != ~((ma_uint64)0)) ? pDataSourceBase->rangeBegInFrames : 0);
-
- pDataSourceBase->rangeBegInFrames = rangeBegInFrames;
- pDataSourceBase->rangeEndInFrames = rangeEndInFrames;
-
- /* Make the loop points relative again, and make sure they're clamped to within the range. */
- if (loopBegAbsolute > pDataSourceBase->rangeBegInFrames) {
- pDataSourceBase->loopBegInFrames = loopBegAbsolute - pDataSourceBase->rangeBegInFrames;
- } else {
- pDataSourceBase->loopBegInFrames = 0;
+ if (frameCount == 0) {
+ return MA_INVALID_ARGS;
}
- if (pDataSourceBase->loopBegInFrames > pDataSourceBase->rangeEndInFrames) {
- pDataSourceBase->loopBegInFrames = pDataSourceBase->rangeEndInFrames;
+ if (pNoise == NULL) {
+ return MA_INVALID_ARGS;
}
- /* Only need to update the loop end point if it's not -1. */
- if (loopEndAbsolute != ~((ma_uint64)0)) {
- if (loopEndAbsolute > pDataSourceBase->rangeBegInFrames) {
- pDataSourceBase->loopEndInFrames = loopEndAbsolute - pDataSourceBase->rangeBegInFrames;
- } else {
- pDataSourceBase->loopEndInFrames = 0;
- }
-
- if (pDataSourceBase->loopEndInFrames > pDataSourceBase->rangeEndInFrames && pDataSourceBase->loopEndInFrames) {
- pDataSourceBase->loopEndInFrames = pDataSourceBase->rangeEndInFrames;
+ /* The output buffer is allowed to be NULL. Since we aren't tracking cursors or anything we can just do nothing and pretend to be successful. */
+ if (pFramesOut == NULL) {
+ framesRead = frameCount;
+ } else {
+ switch (pNoise->config.type) {
+ case ma_noise_type_white: framesRead = ma_noise_read_pcm_frames__white (pNoise, pFramesOut, frameCount); break;
+ case ma_noise_type_pink: framesRead = ma_noise_read_pcm_frames__pink (pNoise, pFramesOut, frameCount); break;
+ case ma_noise_type_brownian: framesRead = ma_noise_read_pcm_frames__brownian(pNoise, pFramesOut, frameCount); break;
+ default: return MA_INVALID_OPERATION; /* Unknown noise type. */
}
}
-
- /* If the new range is past the current cursor position we need to seek to it. */
- result = ma_data_source_get_cursor_in_pcm_frames(pDataSource, &cursor);
- if (result == MA_SUCCESS) {
- /* Seek to within range. Note that our seek positions here are relative to the new range. */
- if (cursor < rangeBegInFrames) {
- ma_data_source_seek_to_pcm_frame(pDataSource, 0);
- } else if (cursor > rangeEndInFrames) {
- ma_data_source_seek_to_pcm_frame(pDataSource, rangeEndInFrames - rangeBegInFrames);
- }
- } else {
- /* We failed to get the cursor position. Probably means the data source has no notion of a cursor such a noise data source. Just pretend the seeking worked. */
+ if (pFramesRead != NULL) {
+ *pFramesRead = framesRead;
}
return MA_SUCCESS;
}
+#endif /* MA_NO_GENERATION */
-MA_API void ma_data_source_get_range_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pRangeBegInFrames, ma_uint64* pRangeEndInFrames)
-{
- ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
- if (pDataSource == NULL) {
- return;
- }
- if (pRangeBegInFrames != NULL) {
- *pRangeBegInFrames = pDataSourceBase->rangeBegInFrames;
- }
+#ifndef MA_NO_RESOURCE_MANAGER
+#ifndef MA_RESOURCE_MANAGER_PAGE_SIZE_IN_MILLISECONDS
+#define MA_RESOURCE_MANAGER_PAGE_SIZE_IN_MILLISECONDS 1000
+#endif
- if (pRangeEndInFrames != NULL) {
- *pRangeEndInFrames = pDataSourceBase->rangeEndInFrames;
- }
-}
+#ifndef MA_JOB_TYPE_RESOURCE_MANAGER_QUEUE_CAPACITY
+#define MA_JOB_TYPE_RESOURCE_MANAGER_QUEUE_CAPACITY 1024
+#endif
-MA_API ma_result ma_data_source_set_loop_point_in_pcm_frames(ma_data_source* pDataSource, ma_uint64 loopBegInFrames, ma_uint64 loopEndInFrames)
+MA_API ma_resource_manager_pipeline_notifications ma_resource_manager_pipeline_notifications_init(void)
{
- ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
+ ma_resource_manager_pipeline_notifications notifications;
- if (pDataSource == NULL) {
- return MA_INVALID_ARGS;
- }
+ MA_ZERO_OBJECT(¬ifications);
- if (loopEndInFrames < loopBegInFrames) {
- return MA_INVALID_ARGS; /* The end of the loop point must come after the beginning. */
- }
+ return notifications;
+}
- if (loopEndInFrames > pDataSourceBase->rangeEndInFrames && loopEndInFrames != ~((ma_uint64)0)) {
- return MA_INVALID_ARGS; /* The end of the loop point must not go beyond the range. */
+static void ma_resource_manager_pipeline_notifications_signal_all_notifications(const ma_resource_manager_pipeline_notifications* pPipelineNotifications)
+{
+ if (pPipelineNotifications == NULL) {
+ return;
}
- pDataSourceBase->loopBegInFrames = loopBegInFrames;
- pDataSourceBase->loopEndInFrames = loopEndInFrames;
+ if (pPipelineNotifications->init.pNotification) { ma_async_notification_signal(pPipelineNotifications->init.pNotification); }
+ if (pPipelineNotifications->done.pNotification) { ma_async_notification_signal(pPipelineNotifications->done.pNotification); }
+}
- /* The end cannot exceed the range. */
- if (pDataSourceBase->loopEndInFrames > (pDataSourceBase->rangeEndInFrames - pDataSourceBase->rangeBegInFrames) && pDataSourceBase->loopEndInFrames != ~((ma_uint64)0)) {
- pDataSourceBase->loopEndInFrames = (pDataSourceBase->rangeEndInFrames - pDataSourceBase->rangeBegInFrames);
+static void ma_resource_manager_pipeline_notifications_acquire_all_fences(const ma_resource_manager_pipeline_notifications* pPipelineNotifications)
+{
+ if (pPipelineNotifications == NULL) {
+ return;
}
- return MA_SUCCESS;
+ if (pPipelineNotifications->init.pFence != NULL) { ma_fence_acquire(pPipelineNotifications->init.pFence); }
+ if (pPipelineNotifications->done.pFence != NULL) { ma_fence_acquire(pPipelineNotifications->done.pFence); }
}
-MA_API void ma_data_source_get_loop_point_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pLoopBegInFrames, ma_uint64* pLoopEndInFrames)
+static void ma_resource_manager_pipeline_notifications_release_all_fences(const ma_resource_manager_pipeline_notifications* pPipelineNotifications)
{
- ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
-
- if (pDataSource == NULL) {
+ if (pPipelineNotifications == NULL) {
return;
}
- if (pLoopBegInFrames != NULL) {
- *pLoopBegInFrames = pDataSourceBase->loopBegInFrames;
- }
-
- if (pLoopEndInFrames != NULL) {
- *pLoopEndInFrames = pDataSourceBase->loopEndInFrames;
- }
+ if (pPipelineNotifications->init.pFence != NULL) { ma_fence_release(pPipelineNotifications->init.pFence); }
+ if (pPipelineNotifications->done.pFence != NULL) { ma_fence_release(pPipelineNotifications->done.pFence); }
}
-MA_API ma_result ma_data_source_set_current(ma_data_source* pDataSource, ma_data_source* pCurrentDataSource)
-{
- ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
- if (pDataSource == NULL) {
- return MA_INVALID_ARGS;
- }
- pDataSourceBase->pCurrent = pCurrentDataSource;
+#ifndef MA_DEFAULT_HASH_SEED
+#define MA_DEFAULT_HASH_SEED 42
+#endif
- return MA_SUCCESS;
-}
+/* MurmurHash3. Based on code from https://github.com/PeterScott/murmur3/blob/master/murmur3.c (public domain). */
+#if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
+ #pragma GCC diagnostic push
+ #if __GNUC__ >= 7
+ #pragma GCC diagnostic ignored "-Wimplicit-fallthrough"
+ #endif
+#endif
-MA_API ma_data_source* ma_data_source_get_current(ma_data_source* pDataSource)
+static MA_INLINE ma_uint32 ma_rotl32(ma_uint32 x, ma_int8 r)
{
- ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
-
- if (pDataSource == NULL) {
- return NULL;
- }
-
- return pDataSourceBase->pCurrent;
+ return (x << r) | (x >> (32 - r));
}
-MA_API ma_result ma_data_source_set_next(ma_data_source* pDataSource, ma_data_source* pNextDataSource)
+static MA_INLINE ma_uint32 ma_hash_getblock(const ma_uint32* blocks, int i)
{
- ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
-
- if (pDataSource == NULL) {
- return MA_INVALID_ARGS;
+ if (ma_is_little_endian()) {
+ return blocks[i];
+ } else {
+ return ma_swap_endian_uint32(blocks[i]);
}
-
- pDataSourceBase->pNext = pNextDataSource;
-
- return MA_SUCCESS;
}
-MA_API ma_data_source* ma_data_source_get_next(ma_data_source* pDataSource)
+static MA_INLINE ma_uint32 ma_hash_fmix32(ma_uint32 h)
{
- ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
-
- if (pDataSource == NULL) {
- return NULL;
- }
+ h ^= h >> 16;
+ h *= 0x85ebca6b;
+ h ^= h >> 13;
+ h *= 0xc2b2ae35;
+ h ^= h >> 16;
- return pDataSourceBase->pNext;
+ return h;
}
-MA_API ma_result ma_data_source_set_next_callback(ma_data_source* pDataSource, ma_data_source_get_next_proc onGetNext)
+static ma_uint32 ma_hash_32(const void* key, int len, ma_uint32 seed)
{
- ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
-
- if (pDataSource == NULL) {
- return MA_INVALID_ARGS;
- }
+ const ma_uint8* data = (const ma_uint8*)key;
+ const ma_uint32* blocks;
+ const ma_uint8* tail;
+ const int nblocks = len / 4;
+ ma_uint32 h1 = seed;
+ ma_uint32 c1 = 0xcc9e2d51;
+ ma_uint32 c2 = 0x1b873593;
+ ma_uint32 k1;
+ int i;
- pDataSourceBase->onGetNext = onGetNext;
+ blocks = (const ma_uint32 *)(data + nblocks*4);
- return MA_SUCCESS;
-}
+ for(i = -nblocks; i; i++) {
+ k1 = ma_hash_getblock(blocks,i);
-MA_API ma_data_source_get_next_proc ma_data_source_get_next_callback(ma_data_source* pDataSource)
-{
- ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
+ k1 *= c1;
+ k1 = ma_rotl32(k1, 15);
+ k1 *= c2;
- if (pDataSource == NULL) {
- return NULL;
+ h1 ^= k1;
+ h1 = ma_rotl32(h1, 13);
+ h1 = h1*5 + 0xe6546b64;
}
- return pDataSourceBase->onGetNext;
-}
-#endif
+ tail = (const ma_uint8*)(data + nblocks*4);
-static ma_result ma_audio_buffer_ref__data_source_on_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
-{
- ma_audio_buffer_ref* pAudioBufferRef = (ma_audio_buffer_ref*)pDataSource;
- ma_uint64 framesRead = ma_audio_buffer_ref_read_pcm_frames(pAudioBufferRef, pFramesOut, frameCount, MA_FALSE);
+ k1 = 0;
+ switch(len & 3) {
+ case 3: k1 ^= tail[2] << 16;
+ case 2: k1 ^= tail[1] << 8;
+ case 1: k1 ^= tail[0];
+ k1 *= c1; k1 = ma_rotl32(k1, 15); k1 *= c2; h1 ^= k1;
+ };
- if (pFramesRead != NULL) {
- *pFramesRead = framesRead;
- }
- if (framesRead < frameCount || framesRead == 0) {
- return MA_AT_END;
- }
+ h1 ^= len;
+ h1 = ma_hash_fmix32(h1);
- return MA_SUCCESS;
+ return h1;
}
-static ma_result ma_audio_buffer_ref__data_source_on_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
-{
- return ma_audio_buffer_ref_seek_to_pcm_frame((ma_audio_buffer_ref*)pDataSource, frameIndex);
-}
+#if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
+ #pragma GCC diagnostic push
+#endif
+/* End MurmurHash3 */
-static ma_result ma_audio_buffer_ref__data_source_on_map(ma_data_source* pDataSource, void** ppFramesOut, ma_uint64* pFrameCount)
+static ma_uint32 ma_hash_string_32(const char* str)
{
- return ma_audio_buffer_ref_map((ma_audio_buffer_ref*)pDataSource, ppFramesOut, pFrameCount);
+ return ma_hash_32(str, (int)strlen(str), MA_DEFAULT_HASH_SEED);
}
-static ma_result ma_audio_buffer_ref__data_source_on_unmap(ma_data_source* pDataSource, ma_uint64 frameCount)
+static ma_uint32 ma_hash_string_w_32(const wchar_t* str)
{
- return ma_audio_buffer_ref_unmap((ma_audio_buffer_ref*)pDataSource, frameCount);
+ return ma_hash_32(str, (int)wcslen(str) * sizeof(*str), MA_DEFAULT_HASH_SEED);
}
-static ma_result ma_audio_buffer_ref__data_source_on_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate)
+
+
+
+/*
+Basic BST Functions
+*/
+static ma_result ma_resource_manager_data_buffer_node_search(ma_resource_manager* pResourceManager, ma_uint32 hashedName32, ma_resource_manager_data_buffer_node** ppDataBufferNode)
{
- ma_audio_buffer_ref* pAudioBufferRef = (ma_audio_buffer_ref*)pDataSource;
+ ma_resource_manager_data_buffer_node* pCurrentNode;
- *pFormat = pAudioBufferRef->format;
- *pChannels = pAudioBufferRef->channels;
- *pSampleRate = 0; /* There is no notion of a sample rate with audio buffers. */
+ MA_ASSERT(pResourceManager != NULL);
+ MA_ASSERT(ppDataBufferNode != NULL);
- return MA_SUCCESS;
+ pCurrentNode = pResourceManager->pRootDataBufferNode;
+ while (pCurrentNode != NULL) {
+ if (hashedName32 == pCurrentNode->hashedName32) {
+ break; /* Found. */
+ } else if (hashedName32 < pCurrentNode->hashedName32) {
+ pCurrentNode = pCurrentNode->pChildLo;
+ } else {
+ pCurrentNode = pCurrentNode->pChildHi;
+ }
+ }
+
+ *ppDataBufferNode = pCurrentNode;
+
+ if (pCurrentNode == NULL) {
+ return MA_DOES_NOT_EXIST;
+ } else {
+ return MA_SUCCESS;
+ }
}
-static ma_result ma_audio_buffer_ref__data_source_on_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
+static ma_result ma_resource_manager_data_buffer_node_insert_point(ma_resource_manager* pResourceManager, ma_uint32 hashedName32, ma_resource_manager_data_buffer_node** ppInsertPoint)
{
- ma_audio_buffer_ref* pAudioBufferRef = (ma_audio_buffer_ref*)pDataSource;
+ ma_result result = MA_SUCCESS;
+ ma_resource_manager_data_buffer_node* pCurrentNode;
- *pCursor = pAudioBufferRef->cursor;
+ MA_ASSERT(pResourceManager != NULL);
+ MA_ASSERT(ppInsertPoint != NULL);
- return MA_SUCCESS;
+ *ppInsertPoint = NULL;
+
+ if (pResourceManager->pRootDataBufferNode == NULL) {
+ return MA_SUCCESS; /* No items. */
+ }
+
+ /* We need to find the node that will become the parent of the new node. If a node is found that already has the same hashed name we need to return MA_ALREADY_EXISTS. */
+ pCurrentNode = pResourceManager->pRootDataBufferNode;
+ while (pCurrentNode != NULL) {
+ if (hashedName32 == pCurrentNode->hashedName32) {
+ result = MA_ALREADY_EXISTS;
+ break;
+ } else {
+ if (hashedName32 < pCurrentNode->hashedName32) {
+ if (pCurrentNode->pChildLo == NULL) {
+ result = MA_SUCCESS;
+ break;
+ } else {
+ pCurrentNode = pCurrentNode->pChildLo;
+ }
+ } else {
+ if (pCurrentNode->pChildHi == NULL) {
+ result = MA_SUCCESS;
+ break;
+ } else {
+ pCurrentNode = pCurrentNode->pChildHi;
+ }
+ }
+ }
+ }
+
+ *ppInsertPoint = pCurrentNode;
+ return result;
}
-static ma_result ma_audio_buffer_ref__data_source_on_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
+static ma_result ma_resource_manager_data_buffer_node_insert_at(ma_resource_manager* pResourceManager, ma_resource_manager_data_buffer_node* pDataBufferNode, ma_resource_manager_data_buffer_node* pInsertPoint)
{
- ma_audio_buffer_ref* pAudioBufferRef = (ma_audio_buffer_ref*)pDataSource;
+ MA_ASSERT(pResourceManager != NULL);
+ MA_ASSERT(pDataBufferNode != NULL);
- *pLength = pAudioBufferRef->sizeInFrames;
+ /* The key must have been set before calling this function. */
+ MA_ASSERT(pDataBufferNode->hashedName32 != 0);
+
+ if (pInsertPoint == NULL) {
+ /* It's the first node. */
+ pResourceManager->pRootDataBufferNode = pDataBufferNode;
+ } else {
+ /* It's not the first node. It needs to be inserted. */
+ if (pDataBufferNode->hashedName32 < pInsertPoint->hashedName32) {
+ MA_ASSERT(pInsertPoint->pChildLo == NULL);
+ pInsertPoint->pChildLo = pDataBufferNode;
+ } else {
+ MA_ASSERT(pInsertPoint->pChildHi == NULL);
+ pInsertPoint->pChildHi = pDataBufferNode;
+ }
+ }
+
+ pDataBufferNode->pParent = pInsertPoint;
return MA_SUCCESS;
}
-static ma_data_source_vtable g_ma_audio_buffer_ref_data_source_vtable =
-{
- ma_audio_buffer_ref__data_source_on_read,
- ma_audio_buffer_ref__data_source_on_seek,
- ma_audio_buffer_ref__data_source_on_map,
- ma_audio_buffer_ref__data_source_on_unmap,
- ma_audio_buffer_ref__data_source_on_get_data_format,
- ma_audio_buffer_ref__data_source_on_get_cursor,
- ma_audio_buffer_ref__data_source_on_get_length
-};
-
-MA_API ma_result ma_audio_buffer_ref_init(ma_format format, ma_uint32 channels, const void* pData, ma_uint64 sizeInFrames, ma_audio_buffer_ref* pAudioBufferRef)
+#if 0 /* Unused for now. */
+static ma_result ma_resource_manager_data_buffer_node_insert(ma_resource_manager* pResourceManager, ma_resource_manager_data_buffer_node* pDataBufferNode)
{
ma_result result;
- ma_data_source_config dataSourceConfig;
+ ma_resource_manager_data_buffer_node* pInsertPoint;
- if (pAudioBufferRef == NULL) {
+ MA_ASSERT(pResourceManager != NULL);
+ MA_ASSERT(pDataBufferNode != NULL);
+
+ result = ma_resource_manager_data_buffer_node_insert_point(pResourceManager, pDataBufferNode->hashedName32, &pInsertPoint);
+ if (result != MA_SUCCESS) {
return MA_INVALID_ARGS;
}
- MA_ZERO_OBJECT(pAudioBufferRef);
+ return ma_resource_manager_data_buffer_node_insert_at(pResourceManager, pDataBufferNode, pInsertPoint);
+}
+#endif
- dataSourceConfig = ma_data_source_config_init();
- dataSourceConfig.vtable = &g_ma_audio_buffer_ref_data_source_vtable;
+static MA_INLINE ma_resource_manager_data_buffer_node* ma_resource_manager_data_buffer_node_find_min(ma_resource_manager_data_buffer_node* pDataBufferNode)
+{
+ ma_resource_manager_data_buffer_node* pCurrentNode;
- result = ma_data_source_init(&dataSourceConfig, &pAudioBufferRef->ds);
- if (result != MA_SUCCESS) {
- return result;
- }
+ MA_ASSERT(pDataBufferNode != NULL);
- pAudioBufferRef->format = format;
- pAudioBufferRef->channels = channels;
- pAudioBufferRef->cursor = 0;
- pAudioBufferRef->sizeInFrames = sizeInFrames;
- pAudioBufferRef->pData = pData;
+ pCurrentNode = pDataBufferNode;
+ while (pCurrentNode->pChildLo != NULL) {
+ pCurrentNode = pCurrentNode->pChildLo;
+ }
- return MA_SUCCESS;
+ return pCurrentNode;
}
-MA_API void ma_audio_buffer_ref_uninit(ma_audio_buffer_ref* pAudioBufferRef)
+static MA_INLINE ma_resource_manager_data_buffer_node* ma_resource_manager_data_buffer_node_find_max(ma_resource_manager_data_buffer_node* pDataBufferNode)
{
- if (pAudioBufferRef == NULL) {
- return;
+ ma_resource_manager_data_buffer_node* pCurrentNode;
+
+ MA_ASSERT(pDataBufferNode != NULL);
+
+ pCurrentNode = pDataBufferNode;
+ while (pCurrentNode->pChildHi != NULL) {
+ pCurrentNode = pCurrentNode->pChildHi;
}
- ma_data_source_uninit(&pAudioBufferRef->ds);
+ return pCurrentNode;
}
-MA_API ma_result ma_audio_buffer_ref_set_data(ma_audio_buffer_ref* pAudioBufferRef, const void* pData, ma_uint64 sizeInFrames)
+static MA_INLINE ma_resource_manager_data_buffer_node* ma_resource_manager_data_buffer_node_find_inorder_successor(ma_resource_manager_data_buffer_node* pDataBufferNode)
{
- if (pAudioBufferRef == NULL) {
- return MA_INVALID_ARGS;
- }
-
- pAudioBufferRef->cursor = 0;
- pAudioBufferRef->sizeInFrames = sizeInFrames;
- pAudioBufferRef->pData = pData;
+ MA_ASSERT(pDataBufferNode != NULL);
+ MA_ASSERT(pDataBufferNode->pChildHi != NULL);
- return MA_SUCCESS;
+ return ma_resource_manager_data_buffer_node_find_min(pDataBufferNode->pChildHi);
}
-MA_API ma_uint64 ma_audio_buffer_ref_read_pcm_frames(ma_audio_buffer_ref* pAudioBufferRef, void* pFramesOut, ma_uint64 frameCount, ma_bool32 loop)
+static MA_INLINE ma_resource_manager_data_buffer_node* ma_resource_manager_data_buffer_node_find_inorder_predecessor(ma_resource_manager_data_buffer_node* pDataBufferNode)
{
- ma_uint64 totalFramesRead = 0;
+ MA_ASSERT(pDataBufferNode != NULL);
+ MA_ASSERT(pDataBufferNode->pChildLo != NULL);
- if (pAudioBufferRef == NULL) {
- return 0;
- }
+ return ma_resource_manager_data_buffer_node_find_max(pDataBufferNode->pChildLo);
+}
- if (frameCount == 0) {
- return 0;
- }
+static ma_result ma_resource_manager_data_buffer_node_remove(ma_resource_manager* pResourceManager, ma_resource_manager_data_buffer_node* pDataBufferNode)
+{
+ MA_ASSERT(pResourceManager != NULL);
+ MA_ASSERT(pDataBufferNode != NULL);
- while (totalFramesRead < frameCount) {
- ma_uint64 framesAvailable = pAudioBufferRef->sizeInFrames - pAudioBufferRef->cursor;
- ma_uint64 framesRemaining = frameCount - totalFramesRead;
- ma_uint64 framesToRead;
+ if (pDataBufferNode->pChildLo == NULL) {
+ if (pDataBufferNode->pChildHi == NULL) {
+ /* Simple case - deleting a buffer with no children. */
+ if (pDataBufferNode->pParent == NULL) {
+ MA_ASSERT(pResourceManager->pRootDataBufferNode == pDataBufferNode); /* There is only a single buffer in the tree which should be equal to the root node. */
+ pResourceManager->pRootDataBufferNode = NULL;
+ } else {
+ if (pDataBufferNode->pParent->pChildLo == pDataBufferNode) {
+ pDataBufferNode->pParent->pChildLo = NULL;
+ } else {
+ pDataBufferNode->pParent->pChildHi = NULL;
+ }
+ }
+ } else {
+ /* Node has one child - pChildHi != NULL. */
+ pDataBufferNode->pChildHi->pParent = pDataBufferNode->pParent;
- framesToRead = framesRemaining;
- if (framesToRead > framesAvailable) {
- framesToRead = framesAvailable;
+ if (pDataBufferNode->pParent == NULL) {
+ MA_ASSERT(pResourceManager->pRootDataBufferNode == pDataBufferNode);
+ pResourceManager->pRootDataBufferNode = pDataBufferNode->pChildHi;
+ } else {
+ if (pDataBufferNode->pParent->pChildLo == pDataBufferNode) {
+ pDataBufferNode->pParent->pChildLo = pDataBufferNode->pChildHi;
+ } else {
+ pDataBufferNode->pParent->pChildHi = pDataBufferNode->pChildHi;
+ }
+ }
}
+ } else {
+ if (pDataBufferNode->pChildHi == NULL) {
+ /* Node has one child - pChildLo != NULL. */
+ pDataBufferNode->pChildLo->pParent = pDataBufferNode->pParent;
- if (pFramesOut != NULL) {
- ma_copy_pcm_frames(pFramesOut, ma_offset_ptr(pAudioBufferRef->pData, pAudioBufferRef->cursor * ma_get_bytes_per_frame(pAudioBufferRef->format, pAudioBufferRef->channels)), framesToRead, pAudioBufferRef->format, pAudioBufferRef->channels);
- }
+ if (pDataBufferNode->pParent == NULL) {
+ MA_ASSERT(pResourceManager->pRootDataBufferNode == pDataBufferNode);
+ pResourceManager->pRootDataBufferNode = pDataBufferNode->pChildLo;
+ } else {
+ if (pDataBufferNode->pParent->pChildLo == pDataBufferNode) {
+ pDataBufferNode->pParent->pChildLo = pDataBufferNode->pChildLo;
+ } else {
+ pDataBufferNode->pParent->pChildHi = pDataBufferNode->pChildLo;
+ }
+ }
+ } else {
+ /* Complex case - deleting a node with two children. */
+ ma_resource_manager_data_buffer_node* pReplacementDataBufferNode;
- totalFramesRead += framesToRead;
+ /* For now we are just going to use the in-order successor as the replacement, but we may want to try to keep this balanced by switching between the two. */
+ pReplacementDataBufferNode = ma_resource_manager_data_buffer_node_find_inorder_successor(pDataBufferNode);
+ MA_ASSERT(pReplacementDataBufferNode != NULL);
- pAudioBufferRef->cursor += framesToRead;
- if (pAudioBufferRef->cursor == pAudioBufferRef->sizeInFrames) {
- if (loop) {
- pAudioBufferRef->cursor = 0;
+ /*
+ Now that we have our replacement node we can make the change. The simple way to do this would be to just exchange the values, and then remove the replacement
+ node, however we track specific nodes via pointers which means we can't just swap out the values. We need to instead just change the pointers around. The
+ replacement node should have at most 1 child. Therefore, we can detach it in terms of our simpler cases above. What we're essentially doing is detaching the
+ replacement node and reinserting it into the same position as the deleted node.
+ */
+ MA_ASSERT(pReplacementDataBufferNode->pParent != NULL); /* The replacement node should never be the root which means it should always have a parent. */
+ MA_ASSERT(pReplacementDataBufferNode->pChildLo == NULL); /* Because we used in-order successor. This would be pChildHi == NULL if we used in-order predecessor. */
+
+ if (pReplacementDataBufferNode->pChildHi == NULL) {
+ if (pReplacementDataBufferNode->pParent->pChildLo == pReplacementDataBufferNode) {
+ pReplacementDataBufferNode->pParent->pChildLo = NULL;
+ } else {
+ pReplacementDataBufferNode->pParent->pChildHi = NULL;
+ }
} else {
- break; /* We've reached the end and we're not looping. Done. */
+ pReplacementDataBufferNode->pChildHi->pParent = pReplacementDataBufferNode->pParent;
+ if (pReplacementDataBufferNode->pParent->pChildLo == pReplacementDataBufferNode) {
+ pReplacementDataBufferNode->pParent->pChildLo = pReplacementDataBufferNode->pChildHi;
+ } else {
+ pReplacementDataBufferNode->pParent->pChildHi = pReplacementDataBufferNode->pChildHi;
+ }
}
- }
- MA_ASSERT(pAudioBufferRef->cursor < pAudioBufferRef->sizeInFrames);
+
+ /* The replacement node has essentially been detached from the binary tree, so now we need to replace the old data buffer with it. The first thing to update is the parent */
+ if (pDataBufferNode->pParent != NULL) {
+ if (pDataBufferNode->pParent->pChildLo == pDataBufferNode) {
+ pDataBufferNode->pParent->pChildLo = pReplacementDataBufferNode;
+ } else {
+ pDataBufferNode->pParent->pChildHi = pReplacementDataBufferNode;
+ }
+ }
+
+ /* Now need to update the replacement node's pointers. */
+ pReplacementDataBufferNode->pParent = pDataBufferNode->pParent;
+ pReplacementDataBufferNode->pChildLo = pDataBufferNode->pChildLo;
+ pReplacementDataBufferNode->pChildHi = pDataBufferNode->pChildHi;
+
+ /* Now the children of the replacement node need to have their parent pointers updated. */
+ if (pReplacementDataBufferNode->pChildLo != NULL) {
+ pReplacementDataBufferNode->pChildLo->pParent = pReplacementDataBufferNode;
+ }
+ if (pReplacementDataBufferNode->pChildHi != NULL) {
+ pReplacementDataBufferNode->pChildHi->pParent = pReplacementDataBufferNode;
+ }
+
+ /* Now the root node needs to be updated. */
+ if (pResourceManager->pRootDataBufferNode == pDataBufferNode) {
+ pResourceManager->pRootDataBufferNode = pReplacementDataBufferNode;
+ }
+ }
}
- return totalFramesRead;
+ return MA_SUCCESS;
}
-MA_API ma_result ma_audio_buffer_ref_seek_to_pcm_frame(ma_audio_buffer_ref* pAudioBufferRef, ma_uint64 frameIndex)
+#if 0 /* Unused for now. */
+static ma_result ma_resource_manager_data_buffer_node_remove_by_key(ma_resource_manager* pResourceManager, ma_uint32 hashedName32)
{
- if (pAudioBufferRef == NULL) {
- return MA_INVALID_ARGS;
- }
+ ma_result result;
+ ma_resource_manager_data_buffer_node* pDataBufferNode;
- if (frameIndex > pAudioBufferRef->sizeInFrames) {
- return MA_INVALID_ARGS;
+ result = ma_resource_manager_data_buffer_search(pResourceManager, hashedName32, &pDataBufferNode);
+ if (result != MA_SUCCESS) {
+ return result; /* Could not find the data buffer. */
}
- pAudioBufferRef->cursor = (size_t)frameIndex;
+ return ma_resource_manager_data_buffer_remove(pResourceManager, pDataBufferNode);
+}
+#endif
- return MA_SUCCESS;
+static ma_resource_manager_data_supply_type ma_resource_manager_data_buffer_node_get_data_supply_type(ma_resource_manager_data_buffer_node* pDataBufferNode)
+{
+ return (ma_resource_manager_data_supply_type)c89atomic_load_i32(&pDataBufferNode->data.type);
}
-MA_API ma_result ma_audio_buffer_ref_map(ma_audio_buffer_ref* pAudioBufferRef, void** ppFramesOut, ma_uint64* pFrameCount)
+static void ma_resource_manager_data_buffer_node_set_data_supply_type(ma_resource_manager_data_buffer_node* pDataBufferNode, ma_resource_manager_data_supply_type supplyType)
{
- ma_uint64 framesAvailable;
- ma_uint64 frameCount = 0;
+ c89atomic_exchange_i32(&pDataBufferNode->data.type, supplyType);
+}
- if (ppFramesOut != NULL) {
- *ppFramesOut = NULL; /* Safety. */
- }
+static ma_result ma_resource_manager_data_buffer_node_increment_ref(ma_resource_manager* pResourceManager, ma_resource_manager_data_buffer_node* pDataBufferNode, ma_uint32* pNewRefCount)
+{
+ ma_uint32 refCount;
- if (pFrameCount != NULL) {
- frameCount = *pFrameCount;
- *pFrameCount = 0; /* Safety. */
- }
+ MA_ASSERT(pResourceManager != NULL);
+ MA_ASSERT(pDataBufferNode != NULL);
- if (pAudioBufferRef == NULL || ppFramesOut == NULL || pFrameCount == NULL) {
- return MA_INVALID_ARGS;
- }
+ (void)pResourceManager;
- framesAvailable = pAudioBufferRef->sizeInFrames - pAudioBufferRef->cursor;
- if (frameCount > framesAvailable) {
- frameCount = framesAvailable;
- }
+ refCount = c89atomic_fetch_add_32(&pDataBufferNode->refCount, 1) + 1;
- *ppFramesOut = ma_offset_ptr(pAudioBufferRef->pData, pAudioBufferRef->cursor * ma_get_bytes_per_frame(pAudioBufferRef->format, pAudioBufferRef->channels));
- *pFrameCount = frameCount;
+ if (pNewRefCount != NULL) {
+ *pNewRefCount = refCount;
+ }
return MA_SUCCESS;
}
-MA_API ma_result ma_audio_buffer_ref_unmap(ma_audio_buffer_ref* pAudioBufferRef, ma_uint64 frameCount)
+static ma_result ma_resource_manager_data_buffer_node_decrement_ref(ma_resource_manager* pResourceManager, ma_resource_manager_data_buffer_node* pDataBufferNode, ma_uint32* pNewRefCount)
{
- ma_uint64 framesAvailable;
-
- if (pAudioBufferRef == NULL) {
- return MA_INVALID_ARGS;
- }
+ ma_uint32 refCount;
- framesAvailable = pAudioBufferRef->sizeInFrames - pAudioBufferRef->cursor;
- if (frameCount > framesAvailable) {
- return MA_INVALID_ARGS; /* The frame count was too big. This should never happen in an unmapping. Need to make sure the caller is aware of this. */
- }
+ MA_ASSERT(pResourceManager != NULL);
+ MA_ASSERT(pDataBufferNode != NULL);
- pAudioBufferRef->cursor += frameCount;
+ (void)pResourceManager;
- if (pAudioBufferRef->cursor == pAudioBufferRef->sizeInFrames) {
- return MA_AT_END; /* Successful. Need to tell the caller that the end has been reached so that it can loop if desired. */
- } else {
- return MA_SUCCESS;
- }
-}
+ refCount = c89atomic_fetch_sub_32(&pDataBufferNode->refCount, 1) - 1;
-MA_API ma_bool32 ma_audio_buffer_ref_at_end(const ma_audio_buffer_ref* pAudioBufferRef)
-{
- if (pAudioBufferRef == NULL) {
- return MA_FALSE;
+ if (pNewRefCount != NULL) {
+ *pNewRefCount = refCount;
}
- return pAudioBufferRef->cursor == pAudioBufferRef->sizeInFrames;
+ return MA_SUCCESS;
}
-MA_API ma_result ma_audio_buffer_ref_get_cursor_in_pcm_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pCursor)
+static void ma_resource_manager_data_buffer_node_free(ma_resource_manager* pResourceManager, ma_resource_manager_data_buffer_node* pDataBufferNode)
{
- if (pCursor == NULL) {
- return MA_INVALID_ARGS;
- }
-
- *pCursor = 0;
+ MA_ASSERT(pResourceManager != NULL);
+ MA_ASSERT(pDataBufferNode != NULL);
- if (pAudioBufferRef == NULL) {
- return MA_INVALID_ARGS;
+ if (pDataBufferNode->isDataOwnedByResourceManager) {
+ if (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBufferNode) == ma_resource_manager_data_supply_type_encoded) {
+ ma_free((void*)pDataBufferNode->data.backend.encoded.pData, &pResourceManager->config.allocationCallbacks);
+ pDataBufferNode->data.backend.encoded.pData = NULL;
+ pDataBufferNode->data.backend.encoded.sizeInBytes = 0;
+ } else if (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBufferNode) == ma_resource_manager_data_supply_type_decoded) {
+ ma_free((void*)pDataBufferNode->data.backend.decoded.pData, &pResourceManager->config.allocationCallbacks);
+ pDataBufferNode->data.backend.decoded.pData = NULL;
+ pDataBufferNode->data.backend.decoded.totalFrameCount = 0;
+ } else if (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBufferNode) == ma_resource_manager_data_supply_type_decoded_paged) {
+ ma_paged_audio_buffer_data_uninit(&pDataBufferNode->data.backend.decodedPaged.data, &pResourceManager->config.allocationCallbacks);
+ } else {
+ /* Should never hit this if the node was successfully initialized. */
+ MA_ASSERT(pDataBufferNode->result != MA_SUCCESS);
+ }
}
- *pCursor = pAudioBufferRef->cursor;
-
- return MA_SUCCESS;
+ /* The data buffer itself needs to be freed. */
+ ma_free(pDataBufferNode, &pResourceManager->config.allocationCallbacks);
}
-MA_API ma_result ma_audio_buffer_ref_get_length_in_pcm_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pLength)
+static ma_result ma_resource_manager_data_buffer_node_result(const ma_resource_manager_data_buffer_node* pDataBufferNode)
{
- if (pLength == NULL) {
- return MA_INVALID_ARGS;
- }
+ MA_ASSERT(pDataBufferNode != NULL);
- *pLength = 0;
+ return (ma_result)c89atomic_load_i32((ma_result*)&pDataBufferNode->result); /* Need a naughty const-cast here. */
+}
- if (pAudioBufferRef == NULL) {
- return MA_INVALID_ARGS;
- }
- *pLength = pAudioBufferRef->sizeInFrames;
+static ma_bool32 ma_resource_manager_is_threading_enabled(const ma_resource_manager* pResourceManager)
+{
+ MA_ASSERT(pResourceManager != NULL);
- return MA_SUCCESS;
+ return (pResourceManager->config.flags & MA_RESOURCE_MANAGER_FLAG_NO_THREADING) == 0;
}
-MA_API ma_result ma_audio_buffer_ref_get_available_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pAvailableFrames)
+
+typedef struct
{
- if (pAvailableFrames == NULL) {
- return MA_INVALID_ARGS;
- }
+ union
+ {
+ ma_async_notification_event e;
+ ma_async_notification_poll p;
+ } backend; /* Must be the first member. */
+ ma_resource_manager* pResourceManager;
+} ma_resource_manager_inline_notification;
- *pAvailableFrames = 0;
+static ma_result ma_resource_manager_inline_notification_init(ma_resource_manager* pResourceManager, ma_resource_manager_inline_notification* pNotification)
+{
+ MA_ASSERT(pResourceManager != NULL);
+ MA_ASSERT(pNotification != NULL);
- if (pAudioBufferRef == NULL) {
- return MA_INVALID_ARGS;
- }
+ pNotification->pResourceManager = pResourceManager;
- if (pAudioBufferRef->sizeInFrames <= pAudioBufferRef->cursor) {
- *pAvailableFrames = 0;
+ if (ma_resource_manager_is_threading_enabled(pResourceManager)) {
+ return ma_async_notification_event_init(&pNotification->backend.e);
} else {
- *pAvailableFrames = pAudioBufferRef->sizeInFrames - pAudioBufferRef->cursor;
+ return ma_async_notification_poll_init(&pNotification->backend.p);
}
-
- return MA_SUCCESS;
}
+static void ma_resource_manager_inline_notification_uninit(ma_resource_manager_inline_notification* pNotification)
+{
+ MA_ASSERT(pNotification != NULL);
+ if (ma_resource_manager_is_threading_enabled(pNotification->pResourceManager)) {
+ ma_async_notification_event_uninit(&pNotification->backend.e);
+ } else {
+ /* No need to uninitialize a polling notification. */
+ }
+}
-
-MA_API ma_audio_buffer_config ma_audio_buffer_config_init(ma_format format, ma_uint32 channels, ma_uint64 sizeInFrames, const void* pData, const ma_allocation_callbacks* pAllocationCallbacks)
+static void ma_resource_manager_inline_notification_wait(ma_resource_manager_inline_notification* pNotification)
{
- ma_audio_buffer_config config;
-
- MA_ZERO_OBJECT(&config);
- config.format = format;
- config.channels = channels;
- config.sizeInFrames = sizeInFrames;
- config.pData = pData;
- ma_allocation_callbacks_init_copy(&config.allocationCallbacks, pAllocationCallbacks);
+ MA_ASSERT(pNotification != NULL);
- return config;
+ if (ma_resource_manager_is_threading_enabled(pNotification->pResourceManager)) {
+ ma_async_notification_event_wait(&pNotification->backend.e);
+ } else {
+ while (ma_async_notification_poll_is_signalled(&pNotification->backend.p) == MA_FALSE) {
+ ma_result result = ma_resource_manager_process_next_job(pNotification->pResourceManager);
+ if (result == MA_NO_DATA_AVAILABLE || result == MA_CANCELLED) {
+ break;
+ }
+ }
+ }
}
-static ma_result ma_audio_buffer_init_ex(const ma_audio_buffer_config* pConfig, ma_bool32 doCopy, ma_audio_buffer* pAudioBuffer)
+static void ma_resource_manager_inline_notification_wait_and_uninit(ma_resource_manager_inline_notification* pNotification)
{
- ma_result result;
+ ma_resource_manager_inline_notification_wait(pNotification);
+ ma_resource_manager_inline_notification_uninit(pNotification);
+}
- if (pAudioBuffer == NULL) {
- return MA_INVALID_ARGS;
- }
- MA_ZERO_MEMORY(pAudioBuffer, sizeof(*pAudioBuffer) - sizeof(pAudioBuffer->_pExtraData)); /* Safety. Don't overwrite the extra data. */
+static void ma_resource_manager_data_buffer_bst_lock(ma_resource_manager* pResourceManager)
+{
+ MA_ASSERT(pResourceManager != NULL);
- if (pConfig == NULL) {
- return MA_INVALID_ARGS;
+ if (ma_resource_manager_is_threading_enabled(pResourceManager)) {
+ #ifndef MA_NO_THREADING
+ {
+ ma_mutex_lock(&pResourceManager->dataBufferBSTLock);
+ }
+ #else
+ {
+ MA_ASSERT(MA_FALSE); /* Should never hit this. */
+ }
+ #endif
+ } else {
+ /* Threading not enabled. Do nothing. */
}
+}
- if (pConfig->sizeInFrames == 0) {
- return MA_INVALID_ARGS; /* Not allowing buffer sizes of 0 frames. */
- }
+static void ma_resource_manager_data_buffer_bst_unlock(ma_resource_manager* pResourceManager)
+{
+ MA_ASSERT(pResourceManager != NULL);
- result = ma_audio_buffer_ref_init(pConfig->format, pConfig->channels, NULL, 0, &pAudioBuffer->ref);
- if (result != MA_SUCCESS) {
- return result;
+ if (ma_resource_manager_is_threading_enabled(pResourceManager)) {
+ #ifndef MA_NO_THREADING
+ {
+ ma_mutex_unlock(&pResourceManager->dataBufferBSTLock);
+ }
+ #else
+ {
+ MA_ASSERT(MA_FALSE); /* Should never hit this. */
+ }
+ #endif
+ } else {
+ /* Threading not enabled. Do nothing. */
}
+}
- ma_allocation_callbacks_init_copy(&pAudioBuffer->allocationCallbacks, &pConfig->allocationCallbacks);
-
- if (doCopy) {
- ma_uint64 allocationSizeInBytes;
- void* pData;
+#ifndef MA_NO_THREADING
+static ma_thread_result MA_THREADCALL ma_resource_manager_job_thread(void* pUserData)
+{
+ ma_resource_manager* pResourceManager = (ma_resource_manager*)pUserData;
+ MA_ASSERT(pResourceManager != NULL);
- allocationSizeInBytes = pConfig->sizeInFrames * ma_get_bytes_per_frame(pConfig->format, pConfig->channels);
- if (allocationSizeInBytes > MA_SIZE_MAX) {
- return MA_OUT_OF_MEMORY; /* Too big. */
- }
+ for (;;) {
+ ma_result result;
+ ma_job job;
- pData = ma__malloc_from_callbacks((size_t)allocationSizeInBytes, &pAudioBuffer->allocationCallbacks); /* Safe cast to size_t. */
- if (pData == NULL) {
- return MA_OUT_OF_MEMORY;
+ result = ma_resource_manager_next_job(pResourceManager, &job);
+ if (result != MA_SUCCESS) {
+ break;
}
- if (pConfig->pData != NULL) {
- ma_copy_pcm_frames(pData, pConfig->pData, pConfig->sizeInFrames, pConfig->format, pConfig->channels);
- } else {
- ma_silence_pcm_frames(pData, pConfig->sizeInFrames, pConfig->format, pConfig->channels);
+ /* Terminate if we got a quit message. */
+ if (job.toc.breakup.code == MA_JOB_TYPE_QUIT) {
+ break;
}
- ma_audio_buffer_ref_set_data(&pAudioBuffer->ref, pData, pConfig->sizeInFrames);
- pAudioBuffer->ownsData = MA_TRUE;
- } else {
- ma_audio_buffer_ref_set_data(&pAudioBuffer->ref, pConfig->pData, pConfig->sizeInFrames);
- pAudioBuffer->ownsData = MA_FALSE;
+ ma_job_process(&job);
}
- return MA_SUCCESS;
+ return (ma_thread_result)0;
}
+#endif
-static void ma_audio_buffer_uninit_ex(ma_audio_buffer* pAudioBuffer, ma_bool32 doFree)
+MA_API ma_resource_manager_config ma_resource_manager_config_init(void)
{
- if (pAudioBuffer == NULL) {
- return;
- }
+ ma_resource_manager_config config;
- if (pAudioBuffer->ownsData && pAudioBuffer->ref.pData != &pAudioBuffer->_pExtraData[0]) {
- ma__free_from_callbacks((void*)pAudioBuffer->ref.pData, &pAudioBuffer->allocationCallbacks); /* Naugty const cast, but OK in this case since we've guarded it with the ownsData check. */
- }
-
- if (doFree) {
- ma__free_from_callbacks(pAudioBuffer, &pAudioBuffer->allocationCallbacks);
+ MA_ZERO_OBJECT(&config);
+ config.decodedFormat = ma_format_unknown;
+ config.decodedChannels = 0;
+ config.decodedSampleRate = 0;
+ config.jobThreadCount = 1; /* A single miniaudio-managed job thread by default. */
+ config.jobQueueCapacity = MA_JOB_TYPE_RESOURCE_MANAGER_QUEUE_CAPACITY;
+
+ /* Flags. */
+ config.flags = 0;
+ #ifdef MA_NO_THREADING
+ {
+ /* Threading is disabled at compile time so disable threading at runtime as well by default. */
+ config.flags |= MA_RESOURCE_MANAGER_FLAG_NO_THREADING;
+ config.jobThreadCount = 0;
}
+ #endif
- ma_audio_buffer_ref_uninit(&pAudioBuffer->ref);
-}
-
-MA_API ma_result ma_audio_buffer_init(const ma_audio_buffer_config* pConfig, ma_audio_buffer* pAudioBuffer)
-{
- return ma_audio_buffer_init_ex(pConfig, MA_FALSE, pAudioBuffer);
+ return config;
}
-MA_API ma_result ma_audio_buffer_init_copy(const ma_audio_buffer_config* pConfig, ma_audio_buffer* pAudioBuffer)
-{
- return ma_audio_buffer_init_ex(pConfig, MA_TRUE, pAudioBuffer);
-}
-MA_API ma_result ma_audio_buffer_alloc_and_init(const ma_audio_buffer_config* pConfig, ma_audio_buffer** ppAudioBuffer)
+MA_API ma_result ma_resource_manager_init(const ma_resource_manager_config* pConfig, ma_resource_manager* pResourceManager)
{
ma_result result;
- ma_audio_buffer* pAudioBuffer;
- ma_audio_buffer_config innerConfig; /* We'll be making some changes to the config, so need to make a copy. */
- ma_uint64 allocationSizeInBytes;
+ ma_job_queue_config jobQueueConfig;
- if (ppAudioBuffer == NULL) {
+ if (pResourceManager == NULL) {
return MA_INVALID_ARGS;
}
- *ppAudioBuffer = NULL; /* Safety. */
+ MA_ZERO_OBJECT(pResourceManager);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
- innerConfig = *pConfig;
- ma_allocation_callbacks_init_copy(&innerConfig.allocationCallbacks, &pConfig->allocationCallbacks);
+ #ifndef MA_NO_THREADING
+ {
+ if (pConfig->jobThreadCount > ma_countof(pResourceManager->jobThreads)) {
+ return MA_INVALID_ARGS; /* Requesting too many job threads. */
+ }
+ }
+ #endif
- allocationSizeInBytes = sizeof(*pAudioBuffer) - sizeof(pAudioBuffer->_pExtraData) + (pConfig->sizeInFrames * ma_get_bytes_per_frame(pConfig->format, pConfig->channels));
- if (allocationSizeInBytes > MA_SIZE_MAX) {
- return MA_OUT_OF_MEMORY; /* Too big. */
+ pResourceManager->config = *pConfig;
+ ma_allocation_callbacks_init_copy(&pResourceManager->config.allocationCallbacks, &pConfig->allocationCallbacks);
+
+ /* Get the log set up early so we can start using it as soon as possible. */
+ if (pResourceManager->config.pLog == NULL) {
+ result = ma_log_init(&pResourceManager->config.allocationCallbacks, &pResourceManager->log);
+ if (result == MA_SUCCESS) {
+ pResourceManager->config.pLog = &pResourceManager->log;
+ } else {
+ pResourceManager->config.pLog = NULL; /* Logging is unavailable. */
+ }
}
- pAudioBuffer = (ma_audio_buffer*)ma__malloc_from_callbacks((size_t)allocationSizeInBytes, &innerConfig.allocationCallbacks); /* Safe cast to size_t. */
- if (pAudioBuffer == NULL) {
- return MA_OUT_OF_MEMORY;
+ if (pResourceManager->config.pVFS == NULL) {
+ result = ma_default_vfs_init(&pResourceManager->defaultVFS, &pResourceManager->config.allocationCallbacks);
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to initialize the default file system. */
+ }
+
+ pResourceManager->config.pVFS = &pResourceManager->defaultVFS;
}
- if (pConfig->pData != NULL) {
- ma_copy_pcm_frames(&pAudioBuffer->_pExtraData[0], pConfig->pData, pConfig->sizeInFrames, pConfig->format, pConfig->channels);
- } else {
- ma_silence_pcm_frames(&pAudioBuffer->_pExtraData[0], pConfig->sizeInFrames, pConfig->format, pConfig->channels);
+ /* If threading has been disabled at compile time, enfore it at run time as well. */
+ #ifdef MA_NO_THREADING
+ {
+ pResourceManager->config.flags |= MA_RESOURCE_MANAGER_FLAG_NO_THREADING;
}
+ #endif
- innerConfig.pData = &pAudioBuffer->_pExtraData[0];
+ /* We need to force MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING if MA_RESOURCE_MANAGER_FLAG_NO_THREADING is set. */
+ if ((pResourceManager->config.flags & MA_RESOURCE_MANAGER_FLAG_NO_THREADING) != 0) {
+ pResourceManager->config.flags |= MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING;
- result = ma_audio_buffer_init_ex(&innerConfig, MA_FALSE, pAudioBuffer);
+ /* We cannot allow job threads when MA_RESOURCE_MANAGER_FLAG_NO_THREADING has been set. This is an invalid use case. */
+ if (pResourceManager->config.jobThreadCount > 0) {
+ return MA_INVALID_ARGS;
+ }
+ }
+
+ /* Job queue. */
+ jobQueueConfig.capacity = pResourceManager->config.jobQueueCapacity;
+ jobQueueConfig.flags = 0;
+ if ((pResourceManager->config.flags & MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING) != 0) {
+ if (pResourceManager->config.jobThreadCount > 0) {
+ return MA_INVALID_ARGS; /* Non-blocking mode is only valid for self-managed job threads. */
+ }
+
+ jobQueueConfig.flags |= MA_JOB_QUEUE_FLAG_NON_BLOCKING;
+ }
+
+ result = ma_job_queue_init(&jobQueueConfig, &pResourceManager->config.allocationCallbacks, &pResourceManager->jobQueue);
if (result != MA_SUCCESS) {
- ma__free_from_callbacks(pAudioBuffer, &innerConfig.allocationCallbacks);
return result;
}
- *ppAudioBuffer = pAudioBuffer;
- return MA_SUCCESS;
-}
+ /* Custom decoding backends. */
+ if (pConfig->ppCustomDecodingBackendVTables != NULL && pConfig->customDecodingBackendCount > 0) {
+ size_t sizeInBytes = sizeof(*pResourceManager->config.ppCustomDecodingBackendVTables) * pConfig->customDecodingBackendCount;
-MA_API void ma_audio_buffer_uninit(ma_audio_buffer* pAudioBuffer)
-{
- ma_audio_buffer_uninit_ex(pAudioBuffer, MA_FALSE);
-}
+ pResourceManager->config.ppCustomDecodingBackendVTables = (ma_decoding_backend_vtable**)ma_malloc(sizeInBytes, &pResourceManager->config.allocationCallbacks);
+ if (pResourceManager->config.ppCustomDecodingBackendVTables == NULL) {
+ ma_job_queue_uninit(&pResourceManager->jobQueue, &pResourceManager->config.allocationCallbacks);
+ return MA_OUT_OF_MEMORY;
+ }
-MA_API void ma_audio_buffer_uninit_and_free(ma_audio_buffer* pAudioBuffer)
-{
- ma_audio_buffer_uninit_ex(pAudioBuffer, MA_TRUE);
-}
+ MA_COPY_MEMORY(pResourceManager->config.ppCustomDecodingBackendVTables, pConfig->ppCustomDecodingBackendVTables, sizeInBytes);
-MA_API ma_uint64 ma_audio_buffer_read_pcm_frames(ma_audio_buffer* pAudioBuffer, void* pFramesOut, ma_uint64 frameCount, ma_bool32 loop)
-{
- if (pAudioBuffer == NULL) {
- return 0;
+ pResourceManager->config.customDecodingBackendCount = pConfig->customDecodingBackendCount;
+ pResourceManager->config.pCustomDecodingBackendUserData = pConfig->pCustomDecodingBackendUserData;
}
- return ma_audio_buffer_ref_read_pcm_frames(&pAudioBuffer->ref, pFramesOut, frameCount, loop);
-}
-MA_API ma_result ma_audio_buffer_seek_to_pcm_frame(ma_audio_buffer* pAudioBuffer, ma_uint64 frameIndex)
-{
- if (pAudioBuffer == NULL) {
- return MA_INVALID_ARGS;
- }
- return ma_audio_buffer_ref_seek_to_pcm_frame(&pAudioBuffer->ref, frameIndex);
-}
+ /* Here is where we initialize our threading stuff. We don't do this if we don't support threading. */
+ if (ma_resource_manager_is_threading_enabled(pResourceManager)) {
+ #ifndef MA_NO_THREADING
+ {
+ ma_uint32 iJobThread;
-MA_API ma_result ma_audio_buffer_map(ma_audio_buffer* pAudioBuffer, void** ppFramesOut, ma_uint64* pFrameCount)
-{
- if (ppFramesOut != NULL) {
- *ppFramesOut = NULL; /* Safety. */
- }
+ /* Data buffer lock. */
+ result = ma_mutex_init(&pResourceManager->dataBufferBSTLock);
+ if (result != MA_SUCCESS) {
+ ma_job_queue_uninit(&pResourceManager->jobQueue, &pResourceManager->config.allocationCallbacks);
+ return result;
+ }
- if (pAudioBuffer == NULL) {
- if (pFrameCount != NULL) {
- *pFrameCount = 0;
+ /* Create the job threads last to ensure the threads has access to valid data. */
+ for (iJobThread = 0; iJobThread < pResourceManager->config.jobThreadCount; iJobThread += 1) {
+ result = ma_thread_create(&pResourceManager->jobThreads[iJobThread], ma_thread_priority_normal, 0, ma_resource_manager_job_thread, pResourceManager, &pResourceManager->config.allocationCallbacks);
+ if (result != MA_SUCCESS) {
+ ma_mutex_uninit(&pResourceManager->dataBufferBSTLock);
+ ma_job_queue_uninit(&pResourceManager->jobQueue, &pResourceManager->config.allocationCallbacks);
+ return result;
+ }
+ }
}
-
- return MA_INVALID_ARGS;
+ #else
+ {
+ /* Threading is disabled at compile time. We should never get here because validation checks should have already been performed. */
+ MA_ASSERT(MA_FALSE);
+ }
+ #endif
}
- return ma_audio_buffer_ref_map(&pAudioBuffer->ref, ppFramesOut, pFrameCount);
+ return MA_SUCCESS;
}
-MA_API ma_result ma_audio_buffer_unmap(ma_audio_buffer* pAudioBuffer, ma_uint64 frameCount)
+
+static void ma_resource_manager_delete_all_data_buffer_nodes(ma_resource_manager* pResourceManager)
{
- if (pAudioBuffer == NULL) {
- return MA_INVALID_ARGS;
- }
+ MA_ASSERT(pResourceManager);
- return ma_audio_buffer_ref_unmap(&pAudioBuffer->ref, frameCount);
-}
+ /* If everything was done properly, there shouldn't be any active data buffers. */
+ while (pResourceManager->pRootDataBufferNode != NULL) {
+ ma_resource_manager_data_buffer_node* pDataBufferNode = pResourceManager->pRootDataBufferNode;
+ ma_resource_manager_data_buffer_node_remove(pResourceManager, pDataBufferNode);
-MA_API ma_bool32 ma_audio_buffer_at_end(const ma_audio_buffer* pAudioBuffer)
-{
- if (pAudioBuffer == NULL) {
- return MA_FALSE;
+ /* The data buffer has been removed from the BST, so now we need to free it's data. */
+ ma_resource_manager_data_buffer_node_free(pResourceManager, pDataBufferNode);
}
-
- return ma_audio_buffer_ref_at_end(&pAudioBuffer->ref);
}
-MA_API ma_result ma_audio_buffer_get_cursor_in_pcm_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pCursor)
+MA_API void ma_resource_manager_uninit(ma_resource_manager* pResourceManager)
{
- if (pAudioBuffer == NULL) {
- return MA_INVALID_ARGS;
+ if (pResourceManager == NULL) {
+ return;
}
- return ma_audio_buffer_ref_get_cursor_in_pcm_frames(&pAudioBuffer->ref, pCursor);
-}
+ /*
+ Job threads need to be killed first. To do this we need to post a quit message to the message queue and then wait for the thread. The quit message will never be removed from the
+ queue which means it will never not be returned after being encounted for the first time which means all threads will eventually receive it.
+ */
+ ma_resource_manager_post_job_quit(pResourceManager);
-MA_API ma_result ma_audio_buffer_get_length_in_pcm_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pLength)
-{
- if (pAudioBuffer == NULL) {
- return MA_INVALID_ARGS;
+ /* Wait for every job to finish before continuing to ensure nothing is sill trying to access any of our objects below. */
+ if (ma_resource_manager_is_threading_enabled(pResourceManager)) {
+ #ifndef MA_NO_THREADING
+ {
+ ma_uint32 iJobThread;
+
+ for (iJobThread = 0; iJobThread < pResourceManager->config.jobThreadCount; iJobThread += 1) {
+ ma_thread_wait(&pResourceManager->jobThreads[iJobThread]);
+ }
+ }
+ #else
+ {
+ MA_ASSERT(MA_FALSE); /* Should never hit this. */
+ }
+ #endif
}
- return ma_audio_buffer_ref_get_length_in_pcm_frames(&pAudioBuffer->ref, pLength);
-}
+ /* At this point the thread should have returned and no other thread should be accessing our data. We can now delete all data buffers. */
+ ma_resource_manager_delete_all_data_buffer_nodes(pResourceManager);
-MA_API ma_result ma_audio_buffer_get_available_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pAvailableFrames)
-{
- if (pAvailableFrames == NULL) {
- return MA_INVALID_ARGS;
+ /* The job queue is no longer needed. */
+ ma_job_queue_uninit(&pResourceManager->jobQueue, &pResourceManager->config.allocationCallbacks);
+
+ /* We're no longer doing anything with data buffers so the lock can now be uninitialized. */
+ if (ma_resource_manager_is_threading_enabled(pResourceManager)) {
+ #ifndef MA_NO_THREADING
+ {
+ ma_mutex_uninit(&pResourceManager->dataBufferBSTLock);
+ }
+ #else
+ {
+ MA_ASSERT(MA_FALSE); /* Should never hit this. */
+ }
+ #endif
}
- *pAvailableFrames = 0;
+ ma_free(pResourceManager->config.ppCustomDecodingBackendVTables, &pResourceManager->config.allocationCallbacks);
- if (pAudioBuffer == NULL) {
- return MA_INVALID_ARGS;
+ if (pResourceManager->config.pLog == &pResourceManager->log) {
+ ma_log_uninit(&pResourceManager->log);
}
-
- return ma_audio_buffer_ref_get_available_frames(&pAudioBuffer->ref, pAvailableFrames);
}
+MA_API ma_log* ma_resource_manager_get_log(ma_resource_manager* pResourceManager)
+{
+ if (pResourceManager == NULL) {
+ return NULL;
+ }
+ return pResourceManager->config.pLog;
+}
-/**************************************************************************************************************************************************************
-VFS
-**************************************************************************************************************************************************************/
-MA_API ma_result ma_vfs_open(ma_vfs* pVFS, const char* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
+MA_API ma_resource_manager_data_source_config ma_resource_manager_data_source_config_init(void)
{
- ma_vfs_callbacks* pCallbacks = (ma_vfs_callbacks*)pVFS;
+ ma_resource_manager_data_source_config config;
- if (pFile == NULL) {
- return MA_INVALID_ARGS;
- }
+ MA_ZERO_OBJECT(&config);
+ config.rangeEndInPCMFrames = ~((ma_uint64)0);
+ config.loopPointEndInPCMFrames = ~((ma_uint64)0);
- *pFile = NULL;
+ return config;
+}
- if (pVFS == NULL || pFilePath == NULL || openMode == 0) {
- return MA_INVALID_ARGS;
- }
- if (pCallbacks->onOpen == NULL) {
- return MA_NOT_IMPLEMENTED;
- }
+static ma_decoder_config ma_resource_manager__init_decoder_config(ma_resource_manager* pResourceManager)
+{
+ ma_decoder_config config;
- return pCallbacks->onOpen(pVFS, pFilePath, openMode, pFile);
+ config = ma_decoder_config_init(pResourceManager->config.decodedFormat, pResourceManager->config.decodedChannels, pResourceManager->config.decodedSampleRate);
+ config.allocationCallbacks = pResourceManager->config.allocationCallbacks;
+ config.ppCustomBackendVTables = pResourceManager->config.ppCustomDecodingBackendVTables;
+ config.customBackendCount = pResourceManager->config.customDecodingBackendCount;
+ config.pCustomBackendUserData = pResourceManager->config.pCustomDecodingBackendUserData;
+
+ return config;
}
-MA_API ma_result ma_vfs_open_w(ma_vfs* pVFS, const wchar_t* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
+static ma_result ma_resource_manager__init_decoder(ma_resource_manager* pResourceManager, const char* pFilePath, const wchar_t* pFilePathW, ma_decoder* pDecoder)
{
- ma_vfs_callbacks* pCallbacks = (ma_vfs_callbacks*)pVFS;
-
- if (pFile == NULL) {
- return MA_INVALID_ARGS;
- }
+ ma_result result;
+ ma_decoder_config config;
- *pFile = NULL;
+ MA_ASSERT(pResourceManager != NULL);
+ MA_ASSERT(pFilePath != NULL || pFilePathW != NULL);
+ MA_ASSERT(pDecoder != NULL);
- if (pVFS == NULL || pFilePath == NULL || openMode == 0) {
- return MA_INVALID_ARGS;
- }
+ config = ma_resource_manager__init_decoder_config(pResourceManager);
- if (pCallbacks->onOpenW == NULL) {
- return MA_NOT_IMPLEMENTED;
+ if (pFilePath != NULL) {
+ result = ma_decoder_init_vfs(pResourceManager->config.pVFS, pFilePath, &config, pDecoder);
+ if (result != MA_SUCCESS) {
+ ma_log_postf(ma_resource_manager_get_log(pResourceManager), MA_LOG_LEVEL_WARNING, "Failed to load file \"%s\". %s.\n", pFilePath, ma_result_description(result));
+ return result;
+ }
+ } else {
+ result = ma_decoder_init_vfs_w(pResourceManager->config.pVFS, pFilePathW, &config, pDecoder);
+ if (result != MA_SUCCESS) {
+ #if (defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L) || defined(_MSC_VER)
+ ma_log_postf(ma_resource_manager_get_log(pResourceManager), MA_LOG_LEVEL_WARNING, "Failed to load file \"%ls\". %s.\n", pFilePathW, ma_result_description(result));
+ #endif
+ return result;
+ }
}
- return pCallbacks->onOpenW(pVFS, pFilePath, openMode, pFile);
+ return MA_SUCCESS;
}
-MA_API ma_result ma_vfs_close(ma_vfs* pVFS, ma_vfs_file file)
+static ma_data_source* ma_resource_manager_data_buffer_get_connector(ma_resource_manager_data_buffer* pDataBuffer)
{
- ma_vfs_callbacks* pCallbacks = (ma_vfs_callbacks*)pVFS;
-
- if (pVFS == NULL || file == NULL) {
- return MA_INVALID_ARGS;
- }
-
- if (pCallbacks->onClose == NULL) {
- return MA_NOT_IMPLEMENTED;
- }
+ switch (pDataBuffer->pNode->data.type)
+ {
+ case ma_resource_manager_data_supply_type_encoded: return &pDataBuffer->connector.decoder;
+ case ma_resource_manager_data_supply_type_decoded: return &pDataBuffer->connector.buffer;
+ case ma_resource_manager_data_supply_type_decoded_paged: return &pDataBuffer->connector.pagedBuffer;
- return pCallbacks->onClose(pVFS, file);
+ case ma_resource_manager_data_supply_type_unknown:
+ default:
+ {
+ ma_log_postf(ma_resource_manager_get_log(pDataBuffer->pResourceManager), MA_LOG_LEVEL_ERROR, "Failed to retrieve data buffer connector. Unknown data supply type.\n");
+ return NULL;
+ };
+ };
}
-MA_API ma_result ma_vfs_read(ma_vfs* pVFS, ma_vfs_file file, void* pDst, size_t sizeInBytes, size_t* pBytesRead)
+static ma_result ma_resource_manager_data_buffer_init_connector(ma_resource_manager_data_buffer* pDataBuffer, ma_async_notification* pInitNotification, ma_fence* pInitFence)
{
- ma_vfs_callbacks* pCallbacks = (ma_vfs_callbacks*)pVFS;
-
- if (pBytesRead != NULL) {
- *pBytesRead = 0;
- }
+ ma_result result;
- if (pVFS == NULL || file == NULL || pDst == NULL) {
- return MA_INVALID_ARGS;
- }
+ MA_ASSERT(pDataBuffer != NULL);
+ MA_ASSERT(pDataBuffer->isConnectorInitialized == MA_FALSE);
- if (pCallbacks->onRead == NULL) {
- return MA_NOT_IMPLEMENTED;
+ /* The underlying data buffer must be initialized before we'll be able to know how to initialize the backend. */
+ result = ma_resource_manager_data_buffer_node_result(pDataBuffer->pNode);
+ if (result != MA_SUCCESS && result != MA_BUSY) {
+ return result; /* The data buffer is in an erroneous state. */
}
- return pCallbacks->onRead(pVFS, file, pDst, sizeInBytes, pBytesRead);
-}
-
-MA_API ma_result ma_vfs_write(ma_vfs* pVFS, ma_vfs_file file, const void* pSrc, size_t sizeInBytes, size_t* pBytesWritten)
-{
- ma_vfs_callbacks* pCallbacks = (ma_vfs_callbacks*)pVFS;
+ /*
+ We need to initialize either a ma_decoder or an ma_audio_buffer depending on whether or not the backing data is encoded or decoded. These act as the
+ "instance" to the data and are used to form the connection between underlying data buffer and the data source. If the data buffer is decoded, we can use
+ an ma_audio_buffer. This enables us to use memory mapping when mixing which saves us a bit of data movement overhead.
+ */
+ switch (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode))
+ {
+ case ma_resource_manager_data_supply_type_encoded: /* Connector is a decoder. */
+ {
+ ma_decoder_config config;
+ config = ma_resource_manager__init_decoder_config(pDataBuffer->pResourceManager);
+ result = ma_decoder_init_memory(pDataBuffer->pNode->data.backend.encoded.pData, pDataBuffer->pNode->data.backend.encoded.sizeInBytes, &config, &pDataBuffer->connector.decoder);
+ } break;
- if (pBytesWritten != NULL) {
- *pBytesWritten = 0;
- }
+ case ma_resource_manager_data_supply_type_decoded: /* Connector is an audio buffer. */
+ {
+ ma_audio_buffer_config config;
+ config = ma_audio_buffer_config_init(pDataBuffer->pNode->data.backend.decoded.format, pDataBuffer->pNode->data.backend.decoded.channels, pDataBuffer->pNode->data.backend.decoded.totalFrameCount, pDataBuffer->pNode->data.backend.decoded.pData, NULL);
+ result = ma_audio_buffer_init(&config, &pDataBuffer->connector.buffer);
+ } break;
- if (pVFS == NULL || file == NULL || pSrc == NULL) {
- return MA_INVALID_ARGS;
- }
+ case ma_resource_manager_data_supply_type_decoded_paged: /* Connector is a paged audio buffer. */
+ {
+ ma_paged_audio_buffer_config config;
+ config = ma_paged_audio_buffer_config_init(&pDataBuffer->pNode->data.backend.decodedPaged.data);
+ result = ma_paged_audio_buffer_init(&config, &pDataBuffer->connector.pagedBuffer);
+ } break;
- if (pCallbacks->onWrite == NULL) {
- return MA_NOT_IMPLEMENTED;
+ case ma_resource_manager_data_supply_type_unknown:
+ default:
+ {
+ /* Unknown data supply type. Should never happen. Need to post an error here. */
+ return MA_INVALID_ARGS;
+ };
}
- return pCallbacks->onWrite(pVFS, file, pSrc, sizeInBytes, pBytesWritten);
-}
+ /*
+ Initialization of the connector is when we can fire the init notification. This will give the application access to
+ the format/channels/rate of the data source.
+ */
+ if (result == MA_SUCCESS) {
+ /*
+ Make sure the looping state is set before returning in order to handle the case where the
+ loop state was set on the data buffer before the connector was initialized.
+ */
+ ma_data_source_set_looping(ma_resource_manager_data_buffer_get_connector(pDataBuffer), ma_resource_manager_data_buffer_is_looping(pDataBuffer));
-MA_API ma_result ma_vfs_seek(ma_vfs* pVFS, ma_vfs_file file, ma_int64 offset, ma_seek_origin origin)
-{
- ma_vfs_callbacks* pCallbacks = (ma_vfs_callbacks*)pVFS;
+ pDataBuffer->isConnectorInitialized = MA_TRUE;
- if (pVFS == NULL || file == NULL) {
- return MA_INVALID_ARGS;
- }
+ if (pInitNotification != NULL) {
+ ma_async_notification_signal(pInitNotification);
+ }
- if (pCallbacks->onSeek == NULL) {
- return MA_NOT_IMPLEMENTED;
+ if (pInitFence != NULL) {
+ ma_fence_release(pInitFence);
+ }
}
- return pCallbacks->onSeek(pVFS, file, offset, origin);
+ /* At this point the backend should be initialized. We do *not* want to set pDataSource->result here - that needs to be done at a higher level to ensure it's done as the last step. */
+ return result;
}
-MA_API ma_result ma_vfs_tell(ma_vfs* pVFS, ma_vfs_file file, ma_int64* pCursor)
+static ma_result ma_resource_manager_data_buffer_uninit_connector(ma_resource_manager* pResourceManager, ma_resource_manager_data_buffer* pDataBuffer)
{
- ma_vfs_callbacks* pCallbacks = (ma_vfs_callbacks*)pVFS;
+ MA_ASSERT(pResourceManager != NULL);
+ MA_ASSERT(pDataBuffer != NULL);
- if (pCursor == NULL) {
- return MA_INVALID_ARGS;
- }
+ switch (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode))
+ {
+ case ma_resource_manager_data_supply_type_encoded: /* Connector is a decoder. */
+ {
+ ma_decoder_uninit(&pDataBuffer->connector.decoder);
+ } break;
- *pCursor = 0;
+ case ma_resource_manager_data_supply_type_decoded: /* Connector is an audio buffer. */
+ {
+ ma_audio_buffer_uninit(&pDataBuffer->connector.buffer);
+ } break;
- if (pVFS == NULL || file == NULL) {
- return MA_INVALID_ARGS;
- }
+ case ma_resource_manager_data_supply_type_decoded_paged: /* Connector is a paged audio buffer. */
+ {
+ ma_paged_audio_buffer_uninit(&pDataBuffer->connector.pagedBuffer);
+ } break;
- if (pCallbacks->onTell == NULL) {
- return MA_NOT_IMPLEMENTED;
+ case ma_resource_manager_data_supply_type_unknown:
+ default:
+ {
+ /* Unknown data supply type. Should never happen. Need to post an error here. */
+ return MA_INVALID_ARGS;
+ };
}
- return pCallbacks->onTell(pVFS, file, pCursor);
+ return MA_SUCCESS;
}
-MA_API ma_result ma_vfs_info(ma_vfs* pVFS, ma_vfs_file file, ma_file_info* pInfo)
+static ma_uint32 ma_resource_manager_data_buffer_node_next_execution_order(ma_resource_manager_data_buffer_node* pDataBufferNode)
{
- ma_vfs_callbacks* pCallbacks = (ma_vfs_callbacks*)pVFS;
+ MA_ASSERT(pDataBufferNode != NULL);
+ return c89atomic_fetch_add_32(&pDataBufferNode->executionCounter, 1);
+}
- if (pInfo == NULL) {
- return MA_INVALID_ARGS;
- }
+static ma_result ma_resource_manager_data_buffer_node_init_supply_encoded(ma_resource_manager* pResourceManager, ma_resource_manager_data_buffer_node* pDataBufferNode, const char* pFilePath, const wchar_t* pFilePathW)
+{
+ ma_result result;
+ size_t dataSizeInBytes;
+ void* pData;
- MA_ZERO_OBJECT(pInfo);
+ MA_ASSERT(pResourceManager != NULL);
+ MA_ASSERT(pDataBufferNode != NULL);
+ MA_ASSERT(pFilePath != NULL || pFilePathW != NULL);
- if (pVFS == NULL || file == NULL) {
- return MA_INVALID_ARGS;
- }
+ result = ma_vfs_open_and_read_file_ex(pResourceManager->config.pVFS, pFilePath, pFilePathW, &pData, &dataSizeInBytes, &pResourceManager->config.allocationCallbacks);
+ if (result != MA_SUCCESS) {
+ if (pFilePath != NULL) {
+ ma_log_postf(ma_resource_manager_get_log(pResourceManager), MA_LOG_LEVEL_WARNING, "Failed to load file \"%s\". %s.\n", pFilePath, ma_result_description(result));
+ } else {
+ #if (defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L) || defined(_MSC_VER)
+ ma_log_postf(ma_resource_manager_get_log(pResourceManager), MA_LOG_LEVEL_WARNING, "Failed to load file \"%ls\". %s.\n", pFilePathW, ma_result_description(result));
+ #endif
+ }
- if (pCallbacks->onInfo == NULL) {
- return MA_NOT_IMPLEMENTED;
+ return result;
}
- return pCallbacks->onInfo(pVFS, file, pInfo);
-}
+ pDataBufferNode->data.backend.encoded.pData = pData;
+ pDataBufferNode->data.backend.encoded.sizeInBytes = dataSizeInBytes;
+ ma_resource_manager_data_buffer_node_set_data_supply_type(pDataBufferNode, ma_resource_manager_data_supply_type_encoded); /* <-- Must be set last. */
+ return MA_SUCCESS;
+}
-static ma_result ma_vfs_open_and_read_file_ex(ma_vfs* pVFS, const char* pFilePath, const wchar_t* pFilePathW, void** ppData, size_t* pSize, const ma_allocation_callbacks* pAllocationCallbacks, ma_uint32 allocationType)
+static ma_result ma_resource_manager_data_buffer_node_init_supply_decoded(ma_resource_manager* pResourceManager, ma_resource_manager_data_buffer_node* pDataBufferNode, const char* pFilePath, const wchar_t* pFilePathW, ma_decoder** ppDecoder)
{
- ma_result result;
- ma_vfs_file file;
- ma_file_info info;
- void* pData;
- size_t bytesRead;
+ ma_result result = MA_SUCCESS;
+ ma_decoder* pDecoder;
+ ma_uint64 totalFrameCount;
- (void)allocationType;
+ MA_ASSERT(pResourceManager != NULL);
+ MA_ASSERT(pDataBufferNode != NULL);
+ MA_ASSERT(ppDecoder != NULL);
+ MA_ASSERT(pFilePath != NULL || pFilePathW != NULL);
- if (ppData != NULL) {
- *ppData = NULL;
- }
- if (pSize != NULL) {
- *pSize = 0;
- }
+ *ppDecoder = NULL; /* For safety. */
- if (ppData == NULL) {
- return MA_INVALID_ARGS;
+ pDecoder = (ma_decoder*)ma_malloc(sizeof(*pDecoder), &pResourceManager->config.allocationCallbacks);
+ if (pDecoder == NULL) {
+ return MA_OUT_OF_MEMORY;
}
- if (pFilePath != NULL) {
- result = ma_vfs_open(pVFS, pFilePath, MA_OPEN_MODE_READ, &file);
- } else {
- result = ma_vfs_open_w(pVFS, pFilePathW, MA_OPEN_MODE_READ, &file);
- }
+ result = ma_resource_manager__init_decoder(pResourceManager, pFilePath, pFilePathW, pDecoder);
if (result != MA_SUCCESS) {
+ ma_free(pDecoder, &pResourceManager->config.allocationCallbacks);
return result;
}
- result = ma_vfs_info(pVFS, file, &info);
+ /*
+ At this point we have the decoder and we now need to initialize the data supply. This will
+ be either a decoded buffer, or a decoded paged buffer. A regular buffer is just one big heap
+ allocated buffer, whereas a paged buffer is a linked list of paged-sized buffers. The latter
+ is used when the length of a sound is unknown until a full decode has been performed.
+ */
+ result = ma_decoder_get_length_in_pcm_frames(pDecoder, &totalFrameCount);
if (result != MA_SUCCESS) {
- ma_vfs_close(pVFS, file);
return result;
}
- if (info.sizeInBytes > MA_SIZE_MAX) {
- ma_vfs_close(pVFS, file);
- return MA_TOO_BIG;
- }
+ if (totalFrameCount > 0) {
+ /* It's a known length. The data supply is a regular decoded buffer. */
+ ma_uint64 dataSizeInBytes;
+ void* pData;
- pData = ma__malloc_from_callbacks((size_t)info.sizeInBytes, pAllocationCallbacks); /* Safe cast. */
- if (pData == NULL) {
- ma_vfs_close(pVFS, file);
- return result;
- }
+ dataSizeInBytes = totalFrameCount * ma_get_bytes_per_frame(pDecoder->outputFormat, pDecoder->outputChannels);
+ if (dataSizeInBytes > MA_SIZE_MAX) {
+ ma_decoder_uninit(pDecoder);
+ ma_free(pDecoder, &pResourceManager->config.allocationCallbacks);
+ return MA_TOO_BIG;
+ }
- result = ma_vfs_read(pVFS, file, pData, (size_t)info.sizeInBytes, &bytesRead); /* Safe cast. */
- ma_vfs_close(pVFS, file);
+ pData = ma_malloc((size_t)dataSizeInBytes, &pResourceManager->config.allocationCallbacks);
+ if (pData == NULL) {
+ ma_decoder_uninit(pDecoder);
+ ma_free(pDecoder, &pResourceManager->config.allocationCallbacks);
+ return MA_OUT_OF_MEMORY;
+ }
- if (result != MA_SUCCESS) {
- ma__free_from_callbacks(pData, pAllocationCallbacks);
- return result;
- }
+ /* The buffer needs to be initialized to silence in case the caller reads from it. */
+ ma_silence_pcm_frames(pData, totalFrameCount, pDecoder->outputFormat, pDecoder->outputChannels);
- if (pSize != NULL) {
- *pSize = bytesRead;
+ /* Data has been allocated and the data supply can now be initialized. */
+ pDataBufferNode->data.backend.decoded.pData = pData;
+ pDataBufferNode->data.backend.decoded.totalFrameCount = totalFrameCount;
+ pDataBufferNode->data.backend.decoded.format = pDecoder->outputFormat;
+ pDataBufferNode->data.backend.decoded.channels = pDecoder->outputChannels;
+ pDataBufferNode->data.backend.decoded.sampleRate = pDecoder->outputSampleRate;
+ pDataBufferNode->data.backend.decoded.decodedFrameCount = 0;
+ ma_resource_manager_data_buffer_node_set_data_supply_type(pDataBufferNode, ma_resource_manager_data_supply_type_decoded); /* <-- Must be set last. */
+ } else {
+ /*
+ It's an unknown length. The data supply is a paged decoded buffer. Setting this up is
+ actually easier than the non-paged decoded buffer because we just need to initialize
+ a ma_paged_audio_buffer object.
+ */
+ result = ma_paged_audio_buffer_data_init(pDecoder->outputFormat, pDecoder->outputChannels, &pDataBufferNode->data.backend.decodedPaged.data);
+ if (result != MA_SUCCESS) {
+ ma_decoder_uninit(pDecoder);
+ ma_free(pDecoder, &pResourceManager->config.allocationCallbacks);
+ return result;
+ }
+
+ pDataBufferNode->data.backend.decodedPaged.sampleRate = pDecoder->outputSampleRate;
+ pDataBufferNode->data.backend.decodedPaged.decodedFrameCount = 0;
+ ma_resource_manager_data_buffer_node_set_data_supply_type(pDataBufferNode, ma_resource_manager_data_supply_type_decoded_paged); /* <-- Must be set last. */
}
- MA_ASSERT(ppData != NULL);
- *ppData = pData;
+ *ppDecoder = pDecoder;
return MA_SUCCESS;
}
-MA_API ma_result ma_vfs_open_and_read_file(ma_vfs* pVFS, const char* pFilePath, void** ppData, size_t* pSize, const ma_allocation_callbacks* pAllocationCallbacks)
+static ma_result ma_resource_manager_data_buffer_node_decode_next_page(ma_resource_manager* pResourceManager, ma_resource_manager_data_buffer_node* pDataBufferNode, ma_decoder* pDecoder)
{
- return ma_vfs_open_and_read_file_ex(pVFS, pFilePath, NULL, ppData, pSize, pAllocationCallbacks, 0 /*MA_ALLOCATION_TYPE_GENERAL*/);
-}
+ ma_result result = MA_SUCCESS;
+ ma_uint64 pageSizeInFrames;
+ ma_uint64 framesToTryReading;
+ ma_uint64 framesRead;
-MA_API ma_result ma_vfs_open_and_read_file_w(ma_vfs* pVFS, const wchar_t* pFilePath, void** ppData, size_t* pSize, const ma_allocation_callbacks* pAllocationCallbacks)
-{
- return ma_vfs_open_and_read_file_ex(pVFS, NULL, pFilePath, ppData, pSize, pAllocationCallbacks, 0 /*MA_ALLOCATION_TYPE_GENERAL*/);
-}
+ MA_ASSERT(pResourceManager != NULL);
+ MA_ASSERT(pDataBufferNode != NULL);
+ MA_ASSERT(pDecoder != NULL);
+ /* We need to know the size of a page in frames to know how many frames to decode. */
+ pageSizeInFrames = MA_RESOURCE_MANAGER_PAGE_SIZE_IN_MILLISECONDS * (pDecoder->outputSampleRate/1000);
+ framesToTryReading = pageSizeInFrames;
-#if defined(MA_WIN32) && defined(MA_WIN32_DESKTOP) && !defined(MA_NO_WIN32_FILEIO)
-static void ma_default_vfs__get_open_settings_win32(ma_uint32 openMode, DWORD* pDesiredAccess, DWORD* pShareMode, DWORD* pCreationDisposition)
-{
- *pDesiredAccess = 0;
- if ((openMode & MA_OPEN_MODE_READ) != 0) {
- *pDesiredAccess |= GENERIC_READ;
+ /*
+ Here is where we do the decoding of the next page. We'll run a slightly different path depending
+ on whether or not we're using a flat or paged buffer because the allocation of the page differs
+ between the two. For a flat buffer it's an offset to an already-allocated buffer. For a paged
+ buffer, we need to allocate a new page and attach it to the linked list.
+ */
+ switch (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBufferNode))
+ {
+ case ma_resource_manager_data_supply_type_decoded:
+ {
+ /* The destination buffer is an offset to the existing buffer. Don't read more than we originally retrieved when we first initialized the decoder. */
+ void* pDst;
+ ma_uint64 framesRemaining = pDataBufferNode->data.backend.decoded.totalFrameCount - pDataBufferNode->data.backend.decoded.decodedFrameCount;
+ if (framesToTryReading > framesRemaining) {
+ framesToTryReading = framesRemaining;
+ }
+
+ if (framesToTryReading > 0) {
+ pDst = ma_offset_ptr(
+ pDataBufferNode->data.backend.decoded.pData,
+ pDataBufferNode->data.backend.decoded.decodedFrameCount * ma_get_bytes_per_frame(pDataBufferNode->data.backend.decoded.format, pDataBufferNode->data.backend.decoded.channels)
+ );
+ MA_ASSERT(pDst != NULL);
+
+ result = ma_decoder_read_pcm_frames(pDecoder, pDst, framesToTryReading, &framesRead);
+ if (framesRead > 0) {
+ pDataBufferNode->data.backend.decoded.decodedFrameCount += framesRead;
+ }
+ } else {
+ framesRead = 0;
+ }
+ } break;
+
+ case ma_resource_manager_data_supply_type_decoded_paged:
+ {
+ /* The destination buffer is a freshly allocated page. */
+ ma_paged_audio_buffer_page* pPage;
+
+ result = ma_paged_audio_buffer_data_allocate_page(&pDataBufferNode->data.backend.decodedPaged.data, framesToTryReading, NULL, &pResourceManager->config.allocationCallbacks, &pPage);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ result = ma_decoder_read_pcm_frames(pDecoder, pPage->pAudioData, framesToTryReading, &framesRead);
+ if (framesRead > 0) {
+ pPage->sizeInFrames = framesRead;
+
+ result = ma_paged_audio_buffer_data_append_page(&pDataBufferNode->data.backend.decodedPaged.data, pPage);
+ if (result == MA_SUCCESS) {
+ pDataBufferNode->data.backend.decodedPaged.decodedFrameCount += framesRead;
+ } else {
+ /* Failed to append the page. Just abort and set the status to MA_AT_END. */
+ ma_paged_audio_buffer_data_free_page(&pDataBufferNode->data.backend.decodedPaged.data, pPage, &pResourceManager->config.allocationCallbacks);
+ result = MA_AT_END;
+ }
+ } else {
+ /* No frames were read. Free the page and just set the status to MA_AT_END. */
+ ma_paged_audio_buffer_data_free_page(&pDataBufferNode->data.backend.decodedPaged.data, pPage, &pResourceManager->config.allocationCallbacks);
+ result = MA_AT_END;
+ }
+ } break;
+
+ case ma_resource_manager_data_supply_type_encoded:
+ case ma_resource_manager_data_supply_type_unknown:
+ default:
+ {
+ /* Unexpected data supply type. */
+ ma_log_postf(ma_resource_manager_get_log(pResourceManager), MA_LOG_LEVEL_ERROR, "Unexpected data supply type (%d) when decoding page.", ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBufferNode));
+ return MA_ERROR;
+ };
}
- if ((openMode & MA_OPEN_MODE_WRITE) != 0) {
- *pDesiredAccess |= GENERIC_WRITE;
+
+ if (result == MA_SUCCESS && framesRead == 0) {
+ result = MA_AT_END;
}
- *pShareMode = 0;
- if ((openMode & MA_OPEN_MODE_READ) != 0) {
- *pShareMode |= FILE_SHARE_READ;
+ return result;
+}
+
+static ma_result ma_resource_manager_data_buffer_node_acquire_critical_section(ma_resource_manager* pResourceManager, const char* pFilePath, const wchar_t* pFilePathW, ma_uint32 hashedName32, ma_uint32 flags, const ma_resource_manager_data_supply* pExistingData, ma_fence* pInitFence, ma_fence* pDoneFence, ma_resource_manager_inline_notification* pInitNotification, ma_resource_manager_data_buffer_node** ppDataBufferNode)
+{
+ ma_result result = MA_SUCCESS;
+ ma_resource_manager_data_buffer_node* pDataBufferNode = NULL;
+ ma_resource_manager_data_buffer_node* pInsertPoint;
+
+ if (ppDataBufferNode != NULL) {
+ *ppDataBufferNode = NULL;
}
- if ((openMode & MA_OPEN_MODE_WRITE) != 0) {
- *pCreationDisposition = CREATE_ALWAYS; /* Opening in write mode. Truncate. */
+ result = ma_resource_manager_data_buffer_node_insert_point(pResourceManager, hashedName32, &pInsertPoint);
+ if (result == MA_ALREADY_EXISTS) {
+ /* The node already exists. We just need to increment the reference count. */
+ pDataBufferNode = pInsertPoint;
+
+ result = ma_resource_manager_data_buffer_node_increment_ref(pResourceManager, pDataBufferNode, NULL);
+ if (result != MA_SUCCESS) {
+ return result; /* Should never happen. Failed to increment the reference count. */
+ }
+
+ result = MA_ALREADY_EXISTS;
+ goto done;
} else {
- *pCreationDisposition = OPEN_EXISTING; /* Opening in read mode. File must exist. */
- }
-}
+ /*
+ The node does not already exist. We need to post a LOAD_DATA_BUFFER_NODE job here. This
+ needs to be done inside the critical section to ensure an uninitialization of the node
+ does not occur before initialization on another thread.
+ */
+ pDataBufferNode = (ma_resource_manager_data_buffer_node*)ma_malloc(sizeof(*pDataBufferNode), &pResourceManager->config.allocationCallbacks);
+ if (pDataBufferNode == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+
+ MA_ZERO_OBJECT(pDataBufferNode);
+ pDataBufferNode->hashedName32 = hashedName32;
+ pDataBufferNode->refCount = 1; /* Always set to 1 by default (this is our first reference). */
+
+ if (pExistingData == NULL) {
+ pDataBufferNode->data.type = ma_resource_manager_data_supply_type_unknown; /* <-- We won't know this until we start decoding. */
+ pDataBufferNode->result = MA_BUSY; /* Must be set to MA_BUSY before we leave the critical section, so might as well do it now. */
+ pDataBufferNode->isDataOwnedByResourceManager = MA_TRUE;
+ } else {
+ pDataBufferNode->data = *pExistingData;
+ pDataBufferNode->result = MA_SUCCESS; /* Not loading asynchronously, so just set the status */
+ pDataBufferNode->isDataOwnedByResourceManager = MA_FALSE;
+ }
+
+ result = ma_resource_manager_data_buffer_node_insert_at(pResourceManager, pDataBufferNode, pInsertPoint);
+ if (result != MA_SUCCESS) {
+ ma_free(pDataBufferNode, &pResourceManager->config.allocationCallbacks);
+ return result; /* Should never happen. Failed to insert the data buffer into the BST. */
+ }
+
+ /*
+ Here is where we'll post the job, but only if we're loading asynchronously. If we're
+ loading synchronously we'll defer loading to a later stage, outside of the critical
+ section.
+ */
+ if (pDataBufferNode->isDataOwnedByResourceManager && (flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC) != 0) {
+ /* Loading asynchronously. Post the job. */
+ ma_job job;
+ char* pFilePathCopy = NULL;
+ wchar_t* pFilePathWCopy = NULL;
+
+ /* We need a copy of the file path. We should probably make this more efficient, but for now we'll do a transient memory allocation. */
+ if (pFilePath != NULL) {
+ pFilePathCopy = ma_copy_string(pFilePath, &pResourceManager->config.allocationCallbacks);
+ } else {
+ pFilePathWCopy = ma_copy_string_w(pFilePathW, &pResourceManager->config.allocationCallbacks);
+ }
+
+ if (pFilePathCopy == NULL && pFilePathWCopy == NULL) {
+ ma_resource_manager_data_buffer_node_remove(pResourceManager, pDataBufferNode);
+ ma_free(pDataBufferNode, &pResourceManager->config.allocationCallbacks);
+ return MA_OUT_OF_MEMORY;
+ }
+
+ if ((flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT) != 0) {
+ ma_resource_manager_inline_notification_init(pResourceManager, pInitNotification);
+ }
-static ma_result ma_default_vfs_open__win32(ma_vfs* pVFS, const char* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
-{
- HANDLE hFile;
- DWORD dwDesiredAccess;
- DWORD dwShareMode;
- DWORD dwCreationDisposition;
+ /* Acquire init and done fences before posting the job. These will be unacquired by the job thread. */
+ if (pInitFence != NULL) { ma_fence_acquire(pInitFence); }
+ if (pDoneFence != NULL) { ma_fence_acquire(pDoneFence); }
- (void)pVFS;
+ /* We now have everything we need to post the job to the job thread. */
+ job = ma_job_init(MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_BUFFER_NODE);
+ job.order = ma_resource_manager_data_buffer_node_next_execution_order(pDataBufferNode);
+ job.data.resourceManager.loadDataBufferNode.pResourceManager = pResourceManager;
+ job.data.resourceManager.loadDataBufferNode.pDataBufferNode = pDataBufferNode;
+ job.data.resourceManager.loadDataBufferNode.pFilePath = pFilePathCopy;
+ job.data.resourceManager.loadDataBufferNode.pFilePathW = pFilePathWCopy;
+ job.data.resourceManager.loadDataBufferNode.decode = (flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE ) != 0;
+ job.data.resourceManager.loadDataBufferNode.pInitNotification = ((flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT) != 0) ? pInitNotification : NULL;
+ job.data.resourceManager.loadDataBufferNode.pDoneNotification = NULL;
+ job.data.resourceManager.loadDataBufferNode.pInitFence = pInitFence;
+ job.data.resourceManager.loadDataBufferNode.pDoneFence = pDoneFence;
- ma_default_vfs__get_open_settings_win32(openMode, &dwDesiredAccess, &dwShareMode, &dwCreationDisposition);
+ result = ma_resource_manager_post_job(pResourceManager, &job);
+ if (result != MA_SUCCESS) {
+ /* Failed to post job. Probably ran out of memory. */
+ ma_log_postf(ma_resource_manager_get_log(pResourceManager), MA_LOG_LEVEL_ERROR, "Failed to post MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_BUFFER_NODE job. %s.\n", ma_result_description(result));
- hFile = CreateFileA(pFilePath, dwDesiredAccess, dwShareMode, NULL, dwCreationDisposition, FILE_ATTRIBUTE_NORMAL, NULL);
- if (hFile == INVALID_HANDLE_VALUE) {
- return ma_result_from_GetLastError(GetLastError());
- }
+ /*
+ Fences were acquired before posting the job, but since the job was not able to
+ be posted, we need to make sure we release them so nothing gets stuck waiting.
+ */
+ if (pInitFence != NULL) { ma_fence_release(pInitFence); }
+ if (pDoneFence != NULL) { ma_fence_release(pDoneFence); }
- *pFile = hFile;
- return MA_SUCCESS;
-}
+ if ((flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT) != 0) {
+ ma_resource_manager_inline_notification_init(pResourceManager, pInitNotification);
+ }
-static ma_result ma_default_vfs_open_w__win32(ma_vfs* pVFS, const wchar_t* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
-{
- HANDLE hFile;
- DWORD dwDesiredAccess;
- DWORD dwShareMode;
- DWORD dwCreationDisposition;
+ ma_free(pFilePathCopy, &pResourceManager->config.allocationCallbacks);
+ ma_free(pFilePathWCopy, &pResourceManager->config.allocationCallbacks);
- (void)pVFS;
+ ma_resource_manager_data_buffer_node_remove(pResourceManager, pDataBufferNode);
+ ma_free(pDataBufferNode, &pResourceManager->config.allocationCallbacks);
- ma_default_vfs__get_open_settings_win32(openMode, &dwDesiredAccess, &dwShareMode, &dwCreationDisposition);
+ return result;
+ }
+ }
+ }
- hFile = CreateFileW(pFilePath, dwDesiredAccess, dwShareMode, NULL, dwCreationDisposition, FILE_ATTRIBUTE_NORMAL, NULL);
- if (hFile == INVALID_HANDLE_VALUE) {
- return ma_result_from_GetLastError(GetLastError());
+done:
+ if (ppDataBufferNode != NULL) {
+ *ppDataBufferNode = pDataBufferNode;
}
- *pFile = hFile;
- return MA_SUCCESS;
+ return result;
}
-static ma_result ma_default_vfs_close__win32(ma_vfs* pVFS, ma_vfs_file file)
+static ma_result ma_resource_manager_data_buffer_node_acquire(ma_resource_manager* pResourceManager, const char* pFilePath, const wchar_t* pFilePathW, ma_uint32 hashedName32, ma_uint32 flags, const ma_resource_manager_data_supply* pExistingData, ma_fence* pInitFence, ma_fence* pDoneFence, ma_resource_manager_data_buffer_node** ppDataBufferNode)
{
- (void)pVFS;
+ ma_result result = MA_SUCCESS;
+ ma_bool32 nodeAlreadyExists = MA_FALSE;
+ ma_resource_manager_data_buffer_node* pDataBufferNode = NULL;
+ ma_resource_manager_inline_notification initNotification; /* Used when the WAIT_INIT flag is set. */
- if (CloseHandle((HANDLE)file) == 0) {
- return ma_result_from_GetLastError(GetLastError());
+ if (ppDataBufferNode != NULL) {
+ *ppDataBufferNode = NULL; /* Safety. */
}
- return MA_SUCCESS;
-}
+ if (pResourceManager == NULL || (pFilePath == NULL && pFilePathW == NULL && hashedName32 == 0)) {
+ return MA_INVALID_ARGS;
+ }
+ /* If we're specifying existing data, it must be valid. */
+ if (pExistingData != NULL && pExistingData->type == ma_resource_manager_data_supply_type_unknown) {
+ return MA_INVALID_ARGS;
+ }
-static ma_result ma_default_vfs_read__win32(ma_vfs* pVFS, ma_vfs_file file, void* pDst, size_t sizeInBytes, size_t* pBytesRead)
-{
- ma_result result = MA_SUCCESS;
- size_t totalBytesRead;
+ /* If we don't support threading, remove the ASYNC flag to make the rest of this a bit simpler. */
+ if (ma_resource_manager_is_threading_enabled(pResourceManager) == MA_FALSE) {
+ flags &= ~MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC;
+ }
- (void)pVFS;
+ if (hashedName32 == 0) {
+ if (pFilePath != NULL) {
+ hashedName32 = ma_hash_string_32(pFilePath);
+ } else {
+ hashedName32 = ma_hash_string_w_32(pFilePathW);
+ }
+ }
- totalBytesRead = 0;
- while (totalBytesRead < sizeInBytes) {
- size_t bytesRemaining;
- DWORD bytesToRead;
- DWORD bytesRead;
- BOOL readResult;
+ /*
+ Here is where we either increment the node's reference count or allocate a new one and add it
+ to the BST. When allocating a new node, we need to make sure the LOAD_DATA_BUFFER_NODE job is
+ posted inside the critical section just in case the caller immediately uninitializes the node
+ as this will ensure the FREE_DATA_BUFFER_NODE job is given an execution order such that the
+ node is not uninitialized before initialization.
+ */
+ ma_resource_manager_data_buffer_bst_lock(pResourceManager);
+ {
+ result = ma_resource_manager_data_buffer_node_acquire_critical_section(pResourceManager, pFilePath, pFilePathW, hashedName32, flags, pExistingData, pInitFence, pDoneFence, &initNotification, &pDataBufferNode);
+ }
+ ma_resource_manager_data_buffer_bst_unlock(pResourceManager);
- bytesRemaining = sizeInBytes - totalBytesRead;
- if (bytesRemaining >= 0xFFFFFFFF) {
- bytesToRead = 0xFFFFFFFF;
- } else {
- bytesToRead = (DWORD)bytesRemaining;
+ if (result == MA_ALREADY_EXISTS) {
+ nodeAlreadyExists = MA_TRUE;
+ result = MA_SUCCESS;
+ } else {
+ if (result != MA_SUCCESS) {
+ return result;
}
+ }
- readResult = ReadFile((HANDLE)file, ma_offset_ptr(pDst, totalBytesRead), bytesToRead, &bytesRead, NULL);
- if (readResult == 1 && bytesRead == 0) {
- result = MA_AT_END;
- break; /* EOF */
+ /*
+ If we're loading synchronously, we'll need to load everything now. When loading asynchronously,
+ a job will have been posted inside the BST critical section so that an uninitialization can be
+ allocated an appropriate execution order thereby preventing it from being uninitialized before
+ the node is initialized by the decoding thread(s).
+ */
+ if (nodeAlreadyExists == MA_FALSE) { /* Don't need to try loading anything if the node already exists. */
+ if (pFilePath == NULL && pFilePathW == NULL) {
+ /*
+ If this path is hit, it means a buffer is being copied (i.e. initialized from only the
+ hashed name), but that node has been freed in the meantime, probably from some other
+ thread. This is an invalid operation.
+ */
+ ma_log_postf(ma_resource_manager_get_log(pResourceManager), MA_LOG_LEVEL_WARNING, "Cloning data buffer node failed because the source node was released. The source node must remain valid until the cloning has completed.\n");
+ result = MA_INVALID_OPERATION;
+ goto done;
}
- totalBytesRead += bytesRead;
+ if (pDataBufferNode->isDataOwnedByResourceManager) {
+ if ((flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC) == 0) {
+ /* Loading synchronously. Load the sound in it's entirety here. */
+ if ((flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE) == 0) {
+ /* No decoding. This is the simple case - just store the file contents in memory. */
+ result = ma_resource_manager_data_buffer_node_init_supply_encoded(pResourceManager, pDataBufferNode, pFilePath, pFilePathW);
+ if (result != MA_SUCCESS) {
+ goto done;
+ }
+ } else {
+ /* Decoding. We do this the same way as we do when loading asynchronously. */
+ ma_decoder* pDecoder;
+ result = ma_resource_manager_data_buffer_node_init_supply_decoded(pResourceManager, pDataBufferNode, pFilePath, pFilePathW, &pDecoder);
+ if (result != MA_SUCCESS) {
+ goto done;
+ }
- if (bytesRead < bytesToRead) {
- break; /* EOF */
+ /* We have the decoder, now decode page by page just like we do when loading asynchronously. */
+ for (;;) {
+ /* Decode next page. */
+ result = ma_resource_manager_data_buffer_node_decode_next_page(pResourceManager, pDataBufferNode, pDecoder);
+ if (result != MA_SUCCESS) {
+ break; /* Will return MA_AT_END when the last page has been decoded. */
+ }
+ }
+
+ /* Reaching the end needs to be considered successful. */
+ if (result == MA_AT_END) {
+ result = MA_SUCCESS;
+ }
+
+ /*
+ At this point the data buffer is either fully decoded or some error occurred. Either
+ way, the decoder is no longer necessary.
+ */
+ ma_decoder_uninit(pDecoder);
+ ma_free(pDecoder, &pResourceManager->config.allocationCallbacks);
+ }
+
+ /* Getting here means we were successful. Make sure the status of the node is updated accordingly. */
+ c89atomic_exchange_i32(&pDataBufferNode->result, result);
+ } else {
+ /* Loading asynchronously. We may need to wait for initialization. */
+ if ((flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT) != 0) {
+ ma_resource_manager_inline_notification_wait(&initNotification);
+ }
+ }
+ } else {
+ /* The data is not managed by the resource manager so there's nothing else to do. */
+ MA_ASSERT(pExistingData != NULL);
}
+ }
- if (readResult == 0) {
- result = ma_result_from_GetLastError(GetLastError());
- break;
+done:
+ /* If we failed to initialize the data buffer we need to free it. */
+ if (result != MA_SUCCESS) {
+ if (nodeAlreadyExists == MA_FALSE) {
+ ma_resource_manager_data_buffer_node_remove(pResourceManager, pDataBufferNode);
+ ma_free(pDataBufferNode, &pResourceManager->config.allocationCallbacks);
}
}
- if (pBytesRead != NULL) {
- *pBytesRead = totalBytesRead;
+ /*
+ The init notification needs to be uninitialized. This will be used if the node does not already
+ exist, and we've specified ASYNC | WAIT_INIT.
+ */
+ if (nodeAlreadyExists == MA_FALSE && pDataBufferNode->isDataOwnedByResourceManager && (flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC) != 0) {
+ if ((flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT) != 0) {
+ ma_resource_manager_inline_notification_uninit(&initNotification);
+ }
+ }
+
+ if (ppDataBufferNode != NULL) {
+ *ppDataBufferNode = pDataBufferNode;
}
return result;
}
-static ma_result ma_default_vfs_write__win32(ma_vfs* pVFS, ma_vfs_file file, const void* pSrc, size_t sizeInBytes, size_t* pBytesWritten)
+static ma_result ma_resource_manager_data_buffer_node_unacquire(ma_resource_manager* pResourceManager, ma_resource_manager_data_buffer_node* pDataBufferNode, const char* pName, const wchar_t* pNameW)
{
ma_result result = MA_SUCCESS;
- size_t totalBytesWritten;
+ ma_uint32 refCount = 0xFFFFFFFF; /* The new reference count of the node after decrementing. Initialize to non-0 to be safe we don't fall into the freeing path. */
+ ma_uint32 hashedName32 = 0;
- (void)pVFS;
+ if (pResourceManager == NULL) {
+ return MA_INVALID_ARGS;
+ }
- totalBytesWritten = 0;
- while (totalBytesWritten < sizeInBytes) {
- size_t bytesRemaining;
- DWORD bytesToWrite;
- DWORD bytesWritten;
- BOOL writeResult;
+ if (pDataBufferNode == NULL) {
+ if (pName == NULL && pNameW == NULL) {
+ return MA_INVALID_ARGS;
+ }
- bytesRemaining = sizeInBytes - totalBytesWritten;
- if (bytesRemaining >= 0xFFFFFFFF) {
- bytesToWrite = 0xFFFFFFFF;
+ if (pName != NULL) {
+ hashedName32 = ma_hash_string_32(pName);
} else {
- bytesToWrite = (DWORD)bytesRemaining;
+ hashedName32 = ma_hash_string_w_32(pNameW);
}
+ }
- writeResult = WriteFile((HANDLE)file, ma_offset_ptr(pSrc, totalBytesWritten), bytesToWrite, &bytesWritten, NULL);
- totalBytesWritten += bytesWritten;
+ /*
+ The first thing to do is decrement the reference counter of the node. Then, if the reference
+ count is zero, we need to free the node. If the node is still in the process of loading, we'll
+ need to post a job to the job queue to free the node. Otherwise we'll just do it here.
+ */
+ ma_resource_manager_data_buffer_bst_lock(pResourceManager);
+ {
+ /* Might need to find the node. Must be done inside the critical section. */
+ if (pDataBufferNode == NULL) {
+ result = ma_resource_manager_data_buffer_node_search(pResourceManager, hashedName32, &pDataBufferNode);
+ if (result != MA_SUCCESS) {
+ goto stage2; /* Couldn't find the node. */
+ }
+ }
- if (writeResult == 0) {
- result = ma_result_from_GetLastError(GetLastError());
- break;
+ result = ma_resource_manager_data_buffer_node_decrement_ref(pResourceManager, pDataBufferNode, &refCount);
+ if (result != MA_SUCCESS) {
+ goto stage2; /* Should never happen. */
}
- }
- if (pBytesWritten != NULL) {
- *pBytesWritten = totalBytesWritten;
+ if (refCount == 0) {
+ result = ma_resource_manager_data_buffer_node_remove(pResourceManager, pDataBufferNode);
+ if (result != MA_SUCCESS) {
+ goto stage2; /* An error occurred when trying to remove the data buffer. This should never happen. */
+ }
+ }
}
+ ma_resource_manager_data_buffer_bst_unlock(pResourceManager);
- return result;
-}
+stage2:
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ /*
+ Here is where we need to free the node. We don't want to do this inside the critical section
+ above because we want to keep that as small as possible for multi-threaded efficiency.
+ */
+ if (refCount == 0) {
+ if (ma_resource_manager_data_buffer_node_result(pDataBufferNode) == MA_BUSY) {
+ /* The sound is still loading. We need to delay the freeing of the node to a safe time. */
+ ma_job job;
-static ma_result ma_default_vfs_seek__win32(ma_vfs* pVFS, ma_vfs_file file, ma_int64 offset, ma_seek_origin origin)
-{
- LARGE_INTEGER liDistanceToMove;
- DWORD dwMoveMethod;
- BOOL result;
+ /* We need to mark the node as unavailable for the sake of the resource manager worker threads. */
+ c89atomic_exchange_i32(&pDataBufferNode->result, MA_UNAVAILABLE);
- (void)pVFS;
+ job = ma_job_init(MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_BUFFER_NODE);
+ job.order = ma_resource_manager_data_buffer_node_next_execution_order(pDataBufferNode);
+ job.data.resourceManager.freeDataBufferNode.pResourceManager = pResourceManager;
+ job.data.resourceManager.freeDataBufferNode.pDataBufferNode = pDataBufferNode;
- liDistanceToMove.QuadPart = offset;
+ result = ma_resource_manager_post_job(pResourceManager, &job);
+ if (result != MA_SUCCESS) {
+ ma_log_postf(ma_resource_manager_get_log(pResourceManager), MA_LOG_LEVEL_ERROR, "Failed to post MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_BUFFER_NODE job. %s.\n", ma_result_description(result));
+ return result;
+ }
- /* */ if (origin == ma_seek_origin_current) {
- dwMoveMethod = FILE_CURRENT;
- } else if (origin == ma_seek_origin_end) {
- dwMoveMethod = FILE_END;
- } else {
- dwMoveMethod = FILE_BEGIN;
+ /* If we don't support threading, process the job queue here. */
+ if (ma_resource_manager_is_threading_enabled(pResourceManager) == MA_FALSE) {
+ while (ma_resource_manager_data_buffer_node_result(pDataBufferNode) == MA_BUSY) {
+ result = ma_resource_manager_process_next_job(pResourceManager);
+ if (result == MA_NO_DATA_AVAILABLE || result == MA_CANCELLED) {
+ result = MA_SUCCESS;
+ break;
+ }
+ }
+ } else {
+ /* Threading is enabled. The job queue will deal with the rest of the cleanup from here. */
+ }
+ } else {
+ /* The sound isn't loading so we can just free the node here. */
+ ma_resource_manager_data_buffer_node_free(pResourceManager, pDataBufferNode);
+ }
}
-#if (defined(_MSC_VER) && _MSC_VER <= 1200) || defined(__DMC__)
- /* No SetFilePointerEx() so restrict to 31 bits. */
- if (origin > 0x7FFFFFFF) {
- return MA_OUT_OF_RANGE;
- }
+ return result;
+}
- result = SetFilePointer((HANDLE)file, (LONG)liDistanceToMove.QuadPart, NULL, dwMoveMethod);
-#else
- result = SetFilePointerEx((HANDLE)file, liDistanceToMove, NULL, dwMoveMethod);
-#endif
- if (result == 0) {
- return ma_result_from_GetLastError(GetLastError());
- }
- return MA_SUCCESS;
-}
-static ma_result ma_default_vfs_tell__win32(ma_vfs* pVFS, ma_vfs_file file, ma_int64* pCursor)
+static ma_uint32 ma_resource_manager_data_buffer_next_execution_order(ma_resource_manager_data_buffer* pDataBuffer)
{
- LARGE_INTEGER liZero;
- LARGE_INTEGER liTell;
- BOOL result;
-#if (defined(_MSC_VER) && _MSC_VER <= 1200) || defined(__DMC__)
- LONG tell;
-#endif
+ MA_ASSERT(pDataBuffer != NULL);
+ return c89atomic_fetch_add_32(&pDataBuffer->executionCounter, 1);
+}
- (void)pVFS;
+static ma_result ma_resource_manager_data_buffer_cb__read_pcm_frames(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+{
+ return ma_resource_manager_data_buffer_read_pcm_frames((ma_resource_manager_data_buffer*)pDataSource, pFramesOut, frameCount, pFramesRead);
+}
- liZero.QuadPart = 0;
+static ma_result ma_resource_manager_data_buffer_cb__seek_to_pcm_frame(ma_data_source* pDataSource, ma_uint64 frameIndex)
+{
+ return ma_resource_manager_data_buffer_seek_to_pcm_frame((ma_resource_manager_data_buffer*)pDataSource, frameIndex);
+}
-#if (defined(_MSC_VER) && _MSC_VER <= 1200) || defined(__DMC__)
- result = SetFilePointer((HANDLE)file, (LONG)liZero.QuadPart, &tell, FILE_CURRENT);
- liTell.QuadPart = tell;
-#else
- result = SetFilePointerEx((HANDLE)file, liZero, &liTell, FILE_CURRENT);
-#endif
- if (result == 0) {
- return ma_result_from_GetLastError(GetLastError());
- }
+static ma_result ma_resource_manager_data_buffer_cb__get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
+{
+ return ma_resource_manager_data_buffer_get_data_format((ma_resource_manager_data_buffer*)pDataSource, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
+}
- if (pCursor != NULL) {
- *pCursor = liTell.QuadPart;
- }
+static ma_result ma_resource_manager_data_buffer_cb__get_cursor_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pCursor)
+{
+ return ma_resource_manager_data_buffer_get_cursor_in_pcm_frames((ma_resource_manager_data_buffer*)pDataSource, pCursor);
+}
- return MA_SUCCESS;
+static ma_result ma_resource_manager_data_buffer_cb__get_length_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pLength)
+{
+ return ma_resource_manager_data_buffer_get_length_in_pcm_frames((ma_resource_manager_data_buffer*)pDataSource, pLength);
}
-static ma_result ma_default_vfs_info__win32(ma_vfs* pVFS, ma_vfs_file file, ma_file_info* pInfo)
+static ma_result ma_resource_manager_data_buffer_cb__set_looping(ma_data_source* pDataSource, ma_bool32 isLooping)
{
- BY_HANDLE_FILE_INFORMATION fi;
- BOOL result;
+ return ma_resource_manager_data_buffer_set_looping((ma_resource_manager_data_buffer*)pDataSource, isLooping);
+}
- (void)pVFS;
+static ma_data_source_vtable g_ma_resource_manager_data_buffer_vtable =
+{
+ ma_resource_manager_data_buffer_cb__read_pcm_frames,
+ ma_resource_manager_data_buffer_cb__seek_to_pcm_frame,
+ ma_resource_manager_data_buffer_cb__get_data_format,
+ ma_resource_manager_data_buffer_cb__get_cursor_in_pcm_frames,
+ ma_resource_manager_data_buffer_cb__get_length_in_pcm_frames,
+ ma_resource_manager_data_buffer_cb__set_looping,
+ 0
+};
- result = GetFileInformationByHandle((HANDLE)file, &fi);
- if (result == 0) {
- return ma_result_from_GetLastError(GetLastError());
- }
+static ma_result ma_resource_manager_data_buffer_init_ex_internal(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source_config* pConfig, ma_uint32 hashedName32, ma_resource_manager_data_buffer* pDataBuffer)
+{
+ ma_result result = MA_SUCCESS;
+ ma_resource_manager_data_buffer_node* pDataBufferNode;
+ ma_data_source_config dataSourceConfig;
+ ma_bool32 async;
+ ma_uint32 flags;
+ ma_resource_manager_pipeline_notifications notifications;
- pInfo->sizeInBytes = ((ma_uint64)fi.nFileSizeHigh << 32) | ((ma_uint64)fi.nFileSizeLow);
+ if (pDataBuffer == NULL) {
+ if (pConfig != NULL && pConfig->pNotifications != NULL) {
+ ma_resource_manager_pipeline_notifications_signal_all_notifications(pConfig->pNotifications);
+ }
- return MA_SUCCESS;
-}
-#else
-static ma_result ma_default_vfs_open__stdio(ma_vfs* pVFS, const char* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
-{
- ma_result result;
- FILE* pFileStd;
- const char* pOpenModeStr;
+ return MA_INVALID_ARGS;
+ }
- MA_ASSERT(pFilePath != NULL);
- MA_ASSERT(openMode != 0);
- MA_ASSERT(pFile != NULL);
+ MA_ZERO_OBJECT(pDataBuffer);
- (void)pVFS;
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
- if ((openMode & MA_OPEN_MODE_READ) != 0) {
- if ((openMode & MA_OPEN_MODE_WRITE) != 0) {
- pOpenModeStr = "r+";
- } else {
- pOpenModeStr = "rb";
- }
+ if (pConfig->pNotifications != NULL) {
+ notifications = *pConfig->pNotifications; /* From here on out we should be referencing `notifications` instead of `pNotifications`. Set this to NULL to catch errors at testing time. */
} else {
- pOpenModeStr = "wb";
+ MA_ZERO_OBJECT(¬ifications);
}
- result = ma_fopen(&pFileStd, pFilePath, pOpenModeStr);
- if (result != MA_SUCCESS) {
- return result;
+ /* For safety, always remove the ASYNC flag if threading is disabled on the resource manager. */
+ flags = pConfig->flags;
+ if (ma_resource_manager_is_threading_enabled(pResourceManager) == MA_FALSE) {
+ flags &= ~MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC;
}
- *pFile = pFileStd;
+ async = (flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC) != 0;
- return MA_SUCCESS;
-}
+ /*
+ Fences need to be acquired before doing anything. These must be aquired and released outside of
+ the node to ensure there's no holes where ma_fence_wait() could prematurely return before the
+ data buffer has completed initialization.
-static ma_result ma_default_vfs_open_w__stdio(ma_vfs* pVFS, const wchar_t* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
-{
- ma_result result;
- FILE* pFileStd;
- const wchar_t* pOpenModeStr;
+ When loading asynchronously, the node acquisition routine below will acquire the fences on this
+ thread and then release them on the async thread when the operation is complete.
- MA_ASSERT(pFilePath != NULL);
- MA_ASSERT(openMode != 0);
- MA_ASSERT(pFile != NULL);
+ These fences are always released at the "done" tag at the end of this function. They'll be
+ acquired a second if loading asynchronously. This double acquisition system is just done to
+ simplify code maintanence.
+ */
+ ma_resource_manager_pipeline_notifications_acquire_all_fences(¬ifications);
+ {
+ /* We first need to acquire a node. If ASYNC is not set, this will not return until the entire sound has been loaded. */
+ result = ma_resource_manager_data_buffer_node_acquire(pResourceManager, pConfig->pFilePath, pConfig->pFilePathW, hashedName32, flags, NULL, notifications.init.pFence, notifications.done.pFence, &pDataBufferNode);
+ if (result != MA_SUCCESS) {
+ ma_resource_manager_pipeline_notifications_signal_all_notifications(¬ifications);
+ goto done;
+ }
- (void)pVFS;
+ dataSourceConfig = ma_data_source_config_init();
+ dataSourceConfig.vtable = &g_ma_resource_manager_data_buffer_vtable;
- if ((openMode & MA_OPEN_MODE_READ) != 0) {
- if ((openMode & MA_OPEN_MODE_WRITE) != 0) {
- pOpenModeStr = L"r+";
- } else {
- pOpenModeStr = L"rb";
+ result = ma_data_source_init(&dataSourceConfig, &pDataBuffer->ds);
+ if (result != MA_SUCCESS) {
+ ma_resource_manager_data_buffer_node_unacquire(pResourceManager, pDataBufferNode, NULL, NULL);
+ ma_resource_manager_pipeline_notifications_signal_all_notifications(¬ifications);
+ goto done;
}
- } else {
- pOpenModeStr = L"wb";
- }
- result = ma_wfopen(&pFileStd, pFilePath, pOpenModeStr, (pVFS != NULL) ? &((ma_default_vfs*)pVFS)->allocationCallbacks : NULL);
- if (result != MA_SUCCESS) {
- return result;
- }
+ pDataBuffer->pResourceManager = pResourceManager;
+ pDataBuffer->pNode = pDataBufferNode;
+ pDataBuffer->flags = flags;
+ pDataBuffer->result = MA_BUSY; /* Always default to MA_BUSY for safety. It'll be overwritten when loading completes or an error occurs. */
- *pFile = pFileStd;
+ /* If we're loading asynchronously we need to post a job to the job queue to initialize the connector. */
+ if (async == MA_FALSE || ma_resource_manager_data_buffer_node_result(pDataBufferNode) == MA_SUCCESS) {
+ /* Loading synchronously or the data has already been fully loaded. We can just initialize the connector from here without a job. */
+ result = ma_resource_manager_data_buffer_init_connector(pDataBuffer, NULL, NULL);
+ c89atomic_exchange_i32(&pDataBuffer->result, result);
- return MA_SUCCESS;
-}
+ ma_resource_manager_pipeline_notifications_signal_all_notifications(¬ifications);
+ goto done;
+ } else {
+ /* The node's data supply isn't initialized yet. The caller has requested that we load asynchronously so we need to post a job to do this. */
+ ma_job job;
+ ma_resource_manager_inline_notification initNotification; /* Used when the WAIT_INIT flag is set. */
-static ma_result ma_default_vfs_close__stdio(ma_vfs* pVFS, ma_vfs_file file)
-{
- MA_ASSERT(file != NULL);
+ if ((flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT) != 0) {
+ ma_resource_manager_inline_notification_init(pResourceManager, &initNotification);
+ }
- (void)pVFS;
+ /*
+ The status of the data buffer needs to be set to MA_BUSY before posting the job so that the
+ worker thread is aware of it's busy state. If the LOAD_DATA_BUFFER job sees a status other
+ than MA_BUSY, it'll assume an error and fall through to an early exit.
+ */
+ c89atomic_exchange_i32(&pDataBuffer->result, MA_BUSY);
- fclose((FILE*)file);
+ /* Acquire fences a second time. These will be released by the async thread. */
+ ma_resource_manager_pipeline_notifications_acquire_all_fences(¬ifications);
- return MA_SUCCESS;
-}
+ job = ma_job_init(MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_BUFFER);
+ job.order = ma_resource_manager_data_buffer_next_execution_order(pDataBuffer);
+ job.data.resourceManager.loadDataBuffer.pDataBuffer = pDataBuffer;
+ job.data.resourceManager.loadDataBuffer.pInitNotification = ((flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT) != 0) ? &initNotification : notifications.init.pNotification;
+ job.data.resourceManager.loadDataBuffer.pDoneNotification = notifications.done.pNotification;
+ job.data.resourceManager.loadDataBuffer.pInitFence = notifications.init.pFence;
+ job.data.resourceManager.loadDataBuffer.pDoneFence = notifications.done.pFence;
-static ma_result ma_default_vfs_read__stdio(ma_vfs* pVFS, ma_vfs_file file, void* pDst, size_t sizeInBytes, size_t* pBytesRead)
-{
- size_t result;
+ result = ma_resource_manager_post_job(pResourceManager, &job);
+ if (result != MA_SUCCESS) {
+ /* We failed to post the job. Most likely there isn't enough room in the queue's buffer. */
+ ma_log_postf(ma_resource_manager_get_log(pResourceManager), MA_LOG_LEVEL_ERROR, "Failed to post MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_BUFFER job. %s.\n", ma_result_description(result));
+ c89atomic_exchange_i32(&pDataBuffer->result, result);
- MA_ASSERT(file != NULL);
- MA_ASSERT(pDst != NULL);
+ /* Release the fences after the result has been set on the data buffer. */
+ ma_resource_manager_pipeline_notifications_release_all_fences(¬ifications);
+ } else {
+ if ((flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT) != 0) {
+ ma_resource_manager_inline_notification_wait(&initNotification);
- (void)pVFS;
+ if (notifications.init.pNotification != NULL) {
+ ma_async_notification_signal(notifications.init.pNotification);
+ }
- result = fread(pDst, 1, sizeInBytes, (FILE*)file);
+ /* NOTE: Do not release the init fence here. It will have been done by the job. */
- if (pBytesRead != NULL) {
- *pBytesRead = result;
- }
+ /* Make sure we return an error if initialization failed on the async thread. */
+ result = ma_resource_manager_data_buffer_result(pDataBuffer);
+ if (result == MA_BUSY) {
+ result = MA_SUCCESS;
+ }
+ }
+ }
- if (result != sizeInBytes) {
- if (result == 0 && feof((FILE*)file)) {
- return MA_AT_END;
- } else {
- return ma_result_from_errno(ferror((FILE*)file));
+ if ((flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT) != 0) {
+ ma_resource_manager_inline_notification_uninit(&initNotification);
+ }
+ }
+
+ if (result != MA_SUCCESS) {
+ ma_resource_manager_data_buffer_node_unacquire(pResourceManager, pDataBufferNode, NULL, NULL);
+ goto done;
+ }
+ }
+done:
+ if (result == MA_SUCCESS) {
+ if (pConfig->initialSeekPointInPCMFrames > 0) {
+ ma_resource_manager_data_buffer_seek_to_pcm_frame(pDataBuffer, pConfig->initialSeekPointInPCMFrames);
}
}
- return MA_SUCCESS;
+ ma_resource_manager_pipeline_notifications_release_all_fences(¬ifications);
+
+ return result;
}
-static ma_result ma_default_vfs_write__stdio(ma_vfs* pVFS, ma_vfs_file file, const void* pSrc, size_t sizeInBytes, size_t* pBytesWritten)
+MA_API ma_result ma_resource_manager_data_buffer_init_ex(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source_config* pConfig, ma_resource_manager_data_buffer* pDataBuffer)
{
- size_t result;
+ return ma_resource_manager_data_buffer_init_ex_internal(pResourceManager, pConfig, 0, pDataBuffer);
+}
- MA_ASSERT(file != NULL);
- MA_ASSERT(pSrc != NULL);
+MA_API ma_result ma_resource_manager_data_buffer_init(ma_resource_manager* pResourceManager, const char* pFilePath, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_buffer* pDataBuffer)
+{
+ ma_resource_manager_data_source_config config;
- (void)pVFS;
+ config = ma_resource_manager_data_source_config_init();
+ config.pFilePath = pFilePath;
+ config.flags = flags;
+ config.pNotifications = pNotifications;
- result = fwrite(pSrc, 1, sizeInBytes, (FILE*)file);
+ return ma_resource_manager_data_buffer_init_ex(pResourceManager, &config, pDataBuffer);
+}
- if (pBytesWritten != NULL) {
- *pBytesWritten = result;
- }
+MA_API ma_result ma_resource_manager_data_buffer_init_w(ma_resource_manager* pResourceManager, const wchar_t* pFilePath, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_buffer* pDataBuffer)
+{
+ ma_resource_manager_data_source_config config;
- if (result != sizeInBytes) {
- return ma_result_from_errno(ferror((FILE*)file));
- }
+ config = ma_resource_manager_data_source_config_init();
+ config.pFilePathW = pFilePath;
+ config.flags = flags;
+ config.pNotifications = pNotifications;
- return MA_SUCCESS;
+ return ma_resource_manager_data_buffer_init_ex(pResourceManager, &config, pDataBuffer);
}
-static ma_result ma_default_vfs_seek__stdio(ma_vfs* pVFS, ma_vfs_file file, ma_int64 offset, ma_seek_origin origin)
+MA_API ma_result ma_resource_manager_data_buffer_init_copy(ma_resource_manager* pResourceManager, const ma_resource_manager_data_buffer* pExistingDataBuffer, ma_resource_manager_data_buffer* pDataBuffer)
{
- int result;
- int whence;
-
- MA_ASSERT(file != NULL);
-
- (void)pVFS;
+ ma_resource_manager_data_source_config config;
- if (origin == ma_seek_origin_start) {
- whence = SEEK_SET;
- } else if (origin == ma_seek_origin_end) {
- whence = SEEK_END;
- } else {
- whence = SEEK_CUR;
+ if (pExistingDataBuffer == NULL) {
+ return MA_INVALID_ARGS;
}
-#if defined(_WIN32)
- #if defined(_MSC_VER) && _MSC_VER > 1200
- result = _fseeki64((FILE*)file, offset, whence);
- #else
- /* No _fseeki64() so restrict to 31 bits. */
- if (origin > 0x7FFFFFFF) {
- return MA_OUT_OF_RANGE;
- }
+ MA_ASSERT(pExistingDataBuffer->pNode != NULL); /* <-- If you've triggered this, you've passed in an invalid existing data buffer. */
- result = fseek((FILE*)file, (int)offset, whence);
- #endif
-#else
- result = fseek((FILE*)file, (long int)offset, whence);
-#endif
- if (result != 0) {
- return MA_ERROR;
- }
+ config = ma_resource_manager_data_source_config_init();
+ config.flags = pExistingDataBuffer->flags;
- return MA_SUCCESS;
+ return ma_resource_manager_data_buffer_init_ex_internal(pResourceManager, &config, pExistingDataBuffer->pNode->hashedName32, pDataBuffer);
}
-static ma_result ma_default_vfs_tell__stdio(ma_vfs* pVFS, ma_vfs_file file, ma_int64* pCursor)
+static ma_result ma_resource_manager_data_buffer_uninit_internal(ma_resource_manager_data_buffer* pDataBuffer)
{
- ma_int64 result;
-
- MA_ASSERT(file != NULL);
- MA_ASSERT(pCursor != NULL);
+ MA_ASSERT(pDataBuffer != NULL);
- (void)pVFS;
+ /* The connector should be uninitialized first. */
+ ma_resource_manager_data_buffer_uninit_connector(pDataBuffer->pResourceManager, pDataBuffer);
-#if defined(_WIN32)
- #if defined(_MSC_VER) && _MSC_VER > 1200
- result = _ftelli64((FILE*)file);
- #else
- result = ftell((FILE*)file);
- #endif
-#else
- result = ftell((FILE*)file);
-#endif
+ /* With the connector uninitialized we can unacquire the node. */
+ ma_resource_manager_data_buffer_node_unacquire(pDataBuffer->pResourceManager, pDataBuffer->pNode, NULL, NULL);
- *pCursor = result;
+ /* The base data source needs to be uninitialized as well. */
+ ma_data_source_uninit(&pDataBuffer->ds);
return MA_SUCCESS;
}
-#if !defined(_MSC_VER) && !((defined(_POSIX_C_SOURCE) && _POSIX_C_SOURCE >= 1) || defined(_XOPEN_SOURCE) || defined(_POSIX_SOURCE)) && !defined(MA_BSD)
-int fileno(FILE *stream);
-#endif
-
-static ma_result ma_default_vfs_info__stdio(ma_vfs* pVFS, ma_vfs_file file, ma_file_info* pInfo)
+MA_API ma_result ma_resource_manager_data_buffer_uninit(ma_resource_manager_data_buffer* pDataBuffer)
{
- int fd;
- struct stat info;
+ ma_result result;
- MA_ASSERT(file != NULL);
- MA_ASSERT(pInfo != NULL);
+ if (pDataBuffer == NULL) {
+ return MA_INVALID_ARGS;
+ }
- (void)pVFS;
+ if (ma_resource_manager_data_buffer_result(pDataBuffer) == MA_SUCCESS) {
+ /* The data buffer can be deleted synchronously. */
+ return ma_resource_manager_data_buffer_uninit_internal(pDataBuffer);
+ } else {
+ /*
+ The data buffer needs to be deleted asynchronously because it's still loading. With the status set to MA_UNAVAILABLE, no more pages will
+ be loaded and the uninitialization should happen fairly quickly. Since the caller owns the data buffer, we need to wait for this event
+ to get processed before returning.
+ */
+ ma_resource_manager_inline_notification notification;
+ ma_job job;
-#if defined(_MSC_VER)
- fd = _fileno((FILE*)file);
-#else
- fd = fileno((FILE*)file);
-#endif
+ /*
+ We need to mark the node as unavailable so we don't try reading from it anymore, but also to
+ let the loading thread know that it needs to abort it's loading procedure.
+ */
+ c89atomic_exchange_i32(&pDataBuffer->result, MA_UNAVAILABLE);
- if (fstat(fd, &info) != 0) {
- return ma_result_from_errno(errno);
- }
+ result = ma_resource_manager_inline_notification_init(pDataBuffer->pResourceManager, ¬ification);
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to create the notification. This should rarely, if ever, happen. */
+ }
- pInfo->sizeInBytes = info.st_size;
+ job = ma_job_init(MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_BUFFER);
+ job.order = ma_resource_manager_data_buffer_next_execution_order(pDataBuffer);
+ job.data.resourceManager.freeDataBuffer.pDataBuffer = pDataBuffer;
+ job.data.resourceManager.freeDataBuffer.pDoneNotification = ¬ification;
+ job.data.resourceManager.freeDataBuffer.pDoneFence = NULL;
- return MA_SUCCESS;
-}
-#endif
+ result = ma_resource_manager_post_job(pDataBuffer->pResourceManager, &job);
+ if (result != MA_SUCCESS) {
+ ma_resource_manager_inline_notification_uninit(¬ification);
+ return result;
+ }
+
+ ma_resource_manager_inline_notification_wait_and_uninit(¬ification);
+ }
+ return result;
+}
-static ma_result ma_default_vfs_open(ma_vfs* pVFS, const char* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
+MA_API ma_result ma_resource_manager_data_buffer_read_pcm_frames(ma_resource_manager_data_buffer* pDataBuffer, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
- if (pFile == NULL) {
- return MA_INVALID_ARGS;
- }
+ ma_result result = MA_SUCCESS;
+ ma_uint64 framesRead = 0;
+ ma_bool32 isDecodedBufferBusy = MA_FALSE;
- *pFile = NULL;
+ /* Safety. */
+ if (pFramesRead != NULL) {
+ *pFramesRead = 0;
+ }
- if (pFilePath == NULL || openMode == 0) {
+ if (frameCount == 0) {
return MA_INVALID_ARGS;
}
-#if defined(MA_WIN32) && defined(MA_WIN32_DESKTOP) && !defined(MA_NO_WIN32_FILEIO)
- return ma_default_vfs_open__win32(pVFS, pFilePath, openMode, pFile);
-#else
- return ma_default_vfs_open__stdio(pVFS, pFilePath, openMode, pFile);
-#endif
-}
+ /*
+ We cannot be using the data buffer after it's been uninitialized. If you trigger this assert it means you're trying to read from the data buffer after
+ it's been uninitialized or is in the process of uninitializing.
+ */
+ MA_ASSERT(ma_resource_manager_data_buffer_node_result(pDataBuffer->pNode) != MA_UNAVAILABLE);
-static ma_result ma_default_vfs_open_w(ma_vfs* pVFS, const wchar_t* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
-{
- if (pFile == NULL) {
- return MA_INVALID_ARGS;
+ /* If the node is not initialized we need to abort with a busy code. */
+ if (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode) == ma_resource_manager_data_supply_type_unknown) {
+ return MA_BUSY; /* Still loading. */
}
- *pFile = NULL;
+ if (pDataBuffer->seekToCursorOnNextRead) {
+ pDataBuffer->seekToCursorOnNextRead = MA_FALSE;
- if (pFilePath == NULL || openMode == 0) {
- return MA_INVALID_ARGS;
+ result = ma_data_source_seek_to_pcm_frame(ma_resource_manager_data_buffer_get_connector(pDataBuffer), pDataBuffer->seekTargetInPCMFrames);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
}
-#if defined(MA_WIN32) && defined(MA_WIN32_DESKTOP) && !defined(MA_NO_WIN32_FILEIO)
- return ma_default_vfs_open_w__win32(pVFS, pFilePath, openMode, pFile);
-#else
- return ma_default_vfs_open_w__stdio(pVFS, pFilePath, openMode, pFile);
-#endif
-}
+ /*
+ For decoded buffers (not paged) we need to check beforehand how many frames we have available. We cannot
+ exceed this amount. We'll read as much as we can, and then return MA_BUSY.
+ */
+ if (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode) == ma_resource_manager_data_supply_type_decoded) {
+ ma_uint64 availableFrames;
-static ma_result ma_default_vfs_close(ma_vfs* pVFS, ma_vfs_file file)
-{
- if (file == NULL) {
- return MA_INVALID_ARGS;
+ isDecodedBufferBusy = (ma_resource_manager_data_buffer_node_result(pDataBuffer->pNode) == MA_BUSY);
+
+ if (ma_resource_manager_data_buffer_get_available_frames(pDataBuffer, &availableFrames) == MA_SUCCESS) {
+ /* Don't try reading more than the available frame count. */
+ if (frameCount > availableFrames) {
+ frameCount = availableFrames;
+
+ /*
+ If there's no frames available we want to set the status to MA_AT_END. The logic below
+ will check if the node is busy, and if so, change it to MA_BUSY. The reason we do this
+ is because we don't want to call `ma_data_source_read_pcm_frames()` if the frame count
+ is 0 because that'll result in a situation where it's possible MA_AT_END won't get
+ returned.
+ */
+ if (frameCount == 0) {
+ result = MA_AT_END;
+ }
+ } else {
+ isDecodedBufferBusy = MA_FALSE; /* We have enough frames available in the buffer to avoid a MA_BUSY status. */
+ }
+ }
}
-#if defined(MA_WIN32) && defined(MA_WIN32_DESKTOP) && !defined(MA_NO_WIN32_FILEIO)
- return ma_default_vfs_close__win32(pVFS, file);
-#else
- return ma_default_vfs_close__stdio(pVFS, file);
-#endif
-}
-
-static ma_result ma_default_vfs_read(ma_vfs* pVFS, ma_vfs_file file, void* pDst, size_t sizeInBytes, size_t* pBytesRead)
-{
- if (pBytesRead != NULL) {
- *pBytesRead = 0;
+ /* Don't attempt to read anything if we've got no frames available. */
+ if (frameCount > 0) {
+ result = ma_data_source_read_pcm_frames(ma_resource_manager_data_buffer_get_connector(pDataBuffer), pFramesOut, frameCount, &framesRead);
}
- if (file == NULL || pDst == NULL) {
- return MA_INVALID_ARGS;
+ /*
+ If we returned MA_AT_END, but the node is still loading, we don't want to return that code or else the caller will interpret the sound
+ as at the end and terminate decoding.
+ */
+ if (result == MA_AT_END) {
+ if (ma_resource_manager_data_buffer_node_result(pDataBuffer->pNode) == MA_BUSY) {
+ result = MA_BUSY;
+ }
}
-#if defined(MA_WIN32) && defined(MA_WIN32_DESKTOP) && !defined(MA_NO_WIN32_FILEIO)
- return ma_default_vfs_read__win32(pVFS, file, pDst, sizeInBytes, pBytesRead);
-#else
- return ma_default_vfs_read__stdio(pVFS, file, pDst, sizeInBytes, pBytesRead);
-#endif
-}
+ if (isDecodedBufferBusy) {
+ result = MA_BUSY;
+ }
-static ma_result ma_default_vfs_write(ma_vfs* pVFS, ma_vfs_file file, const void* pSrc, size_t sizeInBytes, size_t* pBytesWritten)
-{
- if (pBytesWritten != NULL) {
- *pBytesWritten = 0;
+ if (pFramesRead != NULL) {
+ *pFramesRead = framesRead;
}
- if (file == NULL || pSrc == NULL) {
- return MA_INVALID_ARGS;
+ if (result == MA_SUCCESS && framesRead == 0) {
+ result = MA_AT_END;
}
-#if defined(MA_WIN32) && defined(MA_WIN32_DESKTOP) && !defined(MA_NO_WIN32_FILEIO)
- return ma_default_vfs_write__win32(pVFS, file, pSrc, sizeInBytes, pBytesWritten);
-#else
- return ma_default_vfs_write__stdio(pVFS, file, pSrc, sizeInBytes, pBytesWritten);
-#endif
+ return result;
}
-static ma_result ma_default_vfs_seek(ma_vfs* pVFS, ma_vfs_file file, ma_int64 offset, ma_seek_origin origin)
+MA_API ma_result ma_resource_manager_data_buffer_seek_to_pcm_frame(ma_resource_manager_data_buffer* pDataBuffer, ma_uint64 frameIndex)
{
- if (file == NULL) {
- return MA_INVALID_ARGS;
- }
+ ma_result result;
-#if defined(MA_WIN32) && defined(MA_WIN32_DESKTOP) && !defined(MA_NO_WIN32_FILEIO)
- return ma_default_vfs_seek__win32(pVFS, file, offset, origin);
-#else
- return ma_default_vfs_seek__stdio(pVFS, file, offset, origin);
-#endif
-}
+ /* We cannot be using the data source after it's been uninitialized. */
+ MA_ASSERT(ma_resource_manager_data_buffer_node_result(pDataBuffer->pNode) != MA_UNAVAILABLE);
-static ma_result ma_default_vfs_tell(ma_vfs* pVFS, ma_vfs_file file, ma_int64* pCursor)
-{
- if (pCursor == NULL) {
- return MA_INVALID_ARGS;
+ /* If we haven't yet got a connector we need to abort. */
+ if (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode) == ma_resource_manager_data_supply_type_unknown) {
+ pDataBuffer->seekTargetInPCMFrames = frameIndex;
+ pDataBuffer->seekToCursorOnNextRead = MA_TRUE;
+ return MA_BUSY; /* Still loading. */
}
- *pCursor = 0;
-
- if (file == NULL) {
- return MA_INVALID_ARGS;
+ result = ma_data_source_seek_to_pcm_frame(ma_resource_manager_data_buffer_get_connector(pDataBuffer), frameIndex);
+ if (result != MA_SUCCESS) {
+ return result;
}
-#if defined(MA_WIN32) && defined(MA_WIN32_DESKTOP) && !defined(MA_NO_WIN32_FILEIO)
- return ma_default_vfs_tell__win32(pVFS, file, pCursor);
-#else
- return ma_default_vfs_tell__stdio(pVFS, file, pCursor);
-#endif
+ pDataBuffer->seekTargetInPCMFrames = ~(ma_uint64)0; /* <-- For identification purposes. */
+ pDataBuffer->seekToCursorOnNextRead = MA_FALSE;
+
+ return MA_SUCCESS;
}
-static ma_result ma_default_vfs_info(ma_vfs* pVFS, ma_vfs_file file, ma_file_info* pInfo)
+MA_API ma_result ma_resource_manager_data_buffer_get_data_format(ma_resource_manager_data_buffer* pDataBuffer, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
- if (pInfo == NULL) {
- return MA_INVALID_ARGS;
- }
+ /* We cannot be using the data source after it's been uninitialized. */
+ MA_ASSERT(ma_resource_manager_data_buffer_node_result(pDataBuffer->pNode) != MA_UNAVAILABLE);
- MA_ZERO_OBJECT(pInfo);
+ switch (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode))
+ {
+ case ma_resource_manager_data_supply_type_encoded:
+ {
+ return ma_data_source_get_data_format(&pDataBuffer->connector.decoder, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
+ };
- if (file == NULL) {
- return MA_INVALID_ARGS;
- }
+ case ma_resource_manager_data_supply_type_decoded:
+ {
+ *pFormat = pDataBuffer->pNode->data.backend.decoded.format;
+ *pChannels = pDataBuffer->pNode->data.backend.decoded.channels;
+ *pSampleRate = pDataBuffer->pNode->data.backend.decoded.sampleRate;
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, pDataBuffer->pNode->data.backend.decoded.channels);
+ return MA_SUCCESS;
+ };
-#if defined(MA_WIN32) && defined(MA_WIN32_DESKTOP) && !defined(MA_NO_WIN32_FILEIO)
- return ma_default_vfs_info__win32(pVFS, file, pInfo);
-#else
- return ma_default_vfs_info__stdio(pVFS, file, pInfo);
-#endif
-}
+ case ma_resource_manager_data_supply_type_decoded_paged:
+ {
+ *pFormat = pDataBuffer->pNode->data.backend.decodedPaged.data.format;
+ *pChannels = pDataBuffer->pNode->data.backend.decodedPaged.data.channels;
+ *pSampleRate = pDataBuffer->pNode->data.backend.decodedPaged.sampleRate;
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, pDataBuffer->pNode->data.backend.decoded.channels);
+ return MA_SUCCESS;
+ };
+ case ma_resource_manager_data_supply_type_unknown:
+ {
+ return MA_BUSY; /* Still loading. */
+ };
-MA_API ma_result ma_default_vfs_init(ma_default_vfs* pVFS, const ma_allocation_callbacks* pAllocationCallbacks)
+ default:
+ {
+ /* Unknown supply type. Should never hit this. */
+ return MA_INVALID_ARGS;
+ }
+ }
+}
+
+MA_API ma_result ma_resource_manager_data_buffer_get_cursor_in_pcm_frames(ma_resource_manager_data_buffer* pDataBuffer, ma_uint64* pCursor)
{
- if (pVFS == NULL) {
+ /* We cannot be using the data source after it's been uninitialized. */
+ MA_ASSERT(ma_resource_manager_data_buffer_node_result(pDataBuffer->pNode) != MA_UNAVAILABLE);
+
+ if (pDataBuffer == NULL || pCursor == NULL) {
return MA_INVALID_ARGS;
}
- pVFS->cb.onOpen = ma_default_vfs_open;
- pVFS->cb.onOpenW = ma_default_vfs_open_w;
- pVFS->cb.onClose = ma_default_vfs_close;
- pVFS->cb.onRead = ma_default_vfs_read;
- pVFS->cb.onWrite = ma_default_vfs_write;
- pVFS->cb.onSeek = ma_default_vfs_seek;
- pVFS->cb.onTell = ma_default_vfs_tell;
- pVFS->cb.onInfo = ma_default_vfs_info;
- ma_allocation_callbacks_init_copy(&pVFS->allocationCallbacks, pAllocationCallbacks);
+ *pCursor = 0;
- return MA_SUCCESS;
-}
+ switch (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode))
+ {
+ case ma_resource_manager_data_supply_type_encoded:
+ {
+ return ma_decoder_get_cursor_in_pcm_frames(&pDataBuffer->connector.decoder, pCursor);
+ };
+ case ma_resource_manager_data_supply_type_decoded:
+ {
+ return ma_audio_buffer_get_cursor_in_pcm_frames(&pDataBuffer->connector.buffer, pCursor);
+ };
-MA_API ma_result ma_vfs_or_default_open(ma_vfs* pVFS, const char* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
-{
- if (pVFS != NULL) {
- return ma_vfs_open(pVFS, pFilePath, openMode, pFile);
- } else {
- return ma_default_vfs_open(pVFS, pFilePath, openMode, pFile);
- }
-}
+ case ma_resource_manager_data_supply_type_decoded_paged:
+ {
+ return ma_paged_audio_buffer_get_cursor_in_pcm_frames(&pDataBuffer->connector.pagedBuffer, pCursor);
+ };
-MA_API ma_result ma_vfs_or_default_open_w(ma_vfs* pVFS, const wchar_t* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
-{
- if (pVFS != NULL) {
- return ma_vfs_open_w(pVFS, pFilePath, openMode, pFile);
- } else {
- return ma_default_vfs_open_w(pVFS, pFilePath, openMode, pFile);
+ case ma_resource_manager_data_supply_type_unknown:
+ {
+ return MA_BUSY;
+ };
+
+ default:
+ {
+ return MA_INVALID_ARGS;
+ }
}
}
-MA_API ma_result ma_vfs_or_default_close(ma_vfs* pVFS, ma_vfs_file file)
+MA_API ma_result ma_resource_manager_data_buffer_get_length_in_pcm_frames(ma_resource_manager_data_buffer* pDataBuffer, ma_uint64* pLength)
{
- if (pVFS != NULL) {
- return ma_vfs_close(pVFS, file);
- } else {
- return ma_default_vfs_close(pVFS, file);
+ /* We cannot be using the data source after it's been uninitialized. */
+ MA_ASSERT(ma_resource_manager_data_buffer_node_result(pDataBuffer->pNode) != MA_UNAVAILABLE);
+
+ if (pDataBuffer == NULL || pLength == NULL) {
+ return MA_INVALID_ARGS;
}
-}
-MA_API ma_result ma_vfs_or_default_read(ma_vfs* pVFS, ma_vfs_file file, void* pDst, size_t sizeInBytes, size_t* pBytesRead)
-{
- if (pVFS != NULL) {
- return ma_vfs_read(pVFS, file, pDst, sizeInBytes, pBytesRead);
- } else {
- return ma_default_vfs_read(pVFS, file, pDst, sizeInBytes, pBytesRead);
+ if (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode) == ma_resource_manager_data_supply_type_unknown) {
+ return MA_BUSY; /* Still loading. */
}
+
+ return ma_data_source_get_length_in_pcm_frames(ma_resource_manager_data_buffer_get_connector(pDataBuffer), pLength);
}
-MA_API ma_result ma_vfs_or_default_write(ma_vfs* pVFS, ma_vfs_file file, const void* pSrc, size_t sizeInBytes, size_t* pBytesWritten)
+MA_API ma_result ma_resource_manager_data_buffer_result(const ma_resource_manager_data_buffer* pDataBuffer)
{
- if (pVFS != NULL) {
- return ma_vfs_write(pVFS, file, pSrc, sizeInBytes, pBytesWritten);
- } else {
- return ma_default_vfs_write(pVFS, file, pSrc, sizeInBytes, pBytesWritten);
+ if (pDataBuffer == NULL) {
+ return MA_INVALID_ARGS;
}
+
+ return (ma_result)c89atomic_load_i32((ma_result*)&pDataBuffer->result); /* Need a naughty const-cast here. */
}
-MA_API ma_result ma_vfs_or_default_seek(ma_vfs* pVFS, ma_vfs_file file, ma_int64 offset, ma_seek_origin origin)
+MA_API ma_result ma_resource_manager_data_buffer_set_looping(ma_resource_manager_data_buffer* pDataBuffer, ma_bool32 isLooping)
{
- if (pVFS != NULL) {
- return ma_vfs_seek(pVFS, file, offset, origin);
- } else {
- return ma_default_vfs_seek(pVFS, file, offset, origin);
+ if (pDataBuffer == NULL) {
+ return MA_INVALID_ARGS;
}
+
+ c89atomic_exchange_32(&pDataBuffer->isLooping, isLooping);
+
+ /* The looping state needs to be set on the connector as well or else looping won't work when we read audio data. */
+ ma_data_source_set_looping(ma_resource_manager_data_buffer_get_connector(pDataBuffer), isLooping);
+
+ return MA_SUCCESS;
}
-MA_API ma_result ma_vfs_or_default_tell(ma_vfs* pVFS, ma_vfs_file file, ma_int64* pCursor)
+MA_API ma_bool32 ma_resource_manager_data_buffer_is_looping(const ma_resource_manager_data_buffer* pDataBuffer)
{
- if (pVFS != NULL) {
- return ma_vfs_tell(pVFS, file, pCursor);
- } else {
- return ma_default_vfs_tell(pVFS, file, pCursor);
+ if (pDataBuffer == NULL) {
+ return MA_FALSE;
}
+
+ return c89atomic_load_32((ma_bool32*)&pDataBuffer->isLooping);
}
-MA_API ma_result ma_vfs_or_default_info(ma_vfs* pVFS, ma_vfs_file file, ma_file_info* pInfo)
+MA_API ma_result ma_resource_manager_data_buffer_get_available_frames(ma_resource_manager_data_buffer* pDataBuffer, ma_uint64* pAvailableFrames)
{
- if (pVFS != NULL) {
- return ma_vfs_info(pVFS, file, pInfo);
- } else {
- return ma_default_vfs_info(pVFS, file, pInfo);
+ if (pAvailableFrames == NULL) {
+ return MA_INVALID_ARGS;
}
-}
-
+ *pAvailableFrames = 0;
-/**************************************************************************************************************************************************************
+ if (pDataBuffer == NULL) {
+ return MA_INVALID_ARGS;
+ }
-Decoding and Encoding Headers. These are auto-generated from a tool.
+ if (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode) == ma_resource_manager_data_supply_type_unknown) {
+ if (ma_resource_manager_data_buffer_node_result(pDataBuffer->pNode) == MA_BUSY) {
+ return MA_BUSY;
+ } else {
+ return MA_INVALID_OPERATION; /* No connector. */
+ }
+ }
-**************************************************************************************************************************************************************/
-#if !defined(MA_NO_WAV) && (!defined(MA_NO_DECODING) || !defined(MA_NO_ENCODING))
-/* dr_wav_h begin */
-#ifndef dr_wav_h
-#define dr_wav_h
-#ifdef __cplusplus
-extern "C" {
-#endif
-#define DRWAV_STRINGIFY(x) #x
-#define DRWAV_XSTRINGIFY(x) DRWAV_STRINGIFY(x)
-#define DRWAV_VERSION_MAJOR 0
-#define DRWAV_VERSION_MINOR 13
-#define DRWAV_VERSION_REVISION 1
-#define DRWAV_VERSION_STRING DRWAV_XSTRINGIFY(DRWAV_VERSION_MAJOR) "." DRWAV_XSTRINGIFY(DRWAV_VERSION_MINOR) "." DRWAV_XSTRINGIFY(DRWAV_VERSION_REVISION)
-#include
-typedef signed char drwav_int8;
-typedef unsigned char drwav_uint8;
-typedef signed short drwav_int16;
-typedef unsigned short drwav_uint16;
-typedef signed int drwav_int32;
-typedef unsigned int drwav_uint32;
-#if defined(_MSC_VER) && !defined(__clang__)
- typedef signed __int64 drwav_int64;
- typedef unsigned __int64 drwav_uint64;
-#else
- #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
- #pragma GCC diagnostic push
- #pragma GCC diagnostic ignored "-Wlong-long"
- #if defined(__clang__)
- #pragma GCC diagnostic ignored "-Wc++11-long-long"
- #endif
- #endif
- typedef signed long long drwav_int64;
- typedef unsigned long long drwav_uint64;
- #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
- #pragma GCC diagnostic pop
- #endif
-#endif
-#if defined(__LP64__) || defined(_WIN64) || (defined(__x86_64__) && !defined(__ILP32__)) || defined(_M_X64) || defined(__ia64) || defined (_M_IA64) || defined(__aarch64__) || defined(_M_ARM64) || defined(__powerpc64__)
- typedef drwav_uint64 drwav_uintptr;
-#else
- typedef drwav_uint32 drwav_uintptr;
-#endif
-typedef drwav_uint8 drwav_bool8;
-typedef drwav_uint32 drwav_bool32;
-#define DRWAV_TRUE 1
-#define DRWAV_FALSE 0
-#if !defined(DRWAV_API)
- #if defined(DRWAV_DLL)
- #if defined(_WIN32)
- #define DRWAV_DLL_IMPORT __declspec(dllimport)
- #define DRWAV_DLL_EXPORT __declspec(dllexport)
- #define DRWAV_DLL_PRIVATE static
- #else
- #if defined(__GNUC__) && __GNUC__ >= 4
- #define DRWAV_DLL_IMPORT __attribute__((visibility("default")))
- #define DRWAV_DLL_EXPORT __attribute__((visibility("default")))
- #define DRWAV_DLL_PRIVATE __attribute__((visibility("hidden")))
- #else
- #define DRWAV_DLL_IMPORT
- #define DRWAV_DLL_EXPORT
- #define DRWAV_DLL_PRIVATE static
- #endif
- #endif
- #if defined(DR_WAV_IMPLEMENTATION) || defined(DRWAV_IMPLEMENTATION)
- #define DRWAV_API DRWAV_DLL_EXPORT
- #else
- #define DRWAV_API DRWAV_DLL_IMPORT
- #endif
- #define DRWAV_PRIVATE DRWAV_DLL_PRIVATE
- #else
- #define DRWAV_API extern
- #define DRWAV_PRIVATE static
- #endif
-#endif
-typedef drwav_int32 drwav_result;
-#define DRWAV_SUCCESS 0
-#define DRWAV_ERROR -1
-#define DRWAV_INVALID_ARGS -2
-#define DRWAV_INVALID_OPERATION -3
-#define DRWAV_OUT_OF_MEMORY -4
-#define DRWAV_OUT_OF_RANGE -5
-#define DRWAV_ACCESS_DENIED -6
-#define DRWAV_DOES_NOT_EXIST -7
-#define DRWAV_ALREADY_EXISTS -8
-#define DRWAV_TOO_MANY_OPEN_FILES -9
-#define DRWAV_INVALID_FILE -10
-#define DRWAV_TOO_BIG -11
-#define DRWAV_PATH_TOO_LONG -12
-#define DRWAV_NAME_TOO_LONG -13
-#define DRWAV_NOT_DIRECTORY -14
-#define DRWAV_IS_DIRECTORY -15
-#define DRWAV_DIRECTORY_NOT_EMPTY -16
-#define DRWAV_END_OF_FILE -17
-#define DRWAV_NO_SPACE -18
-#define DRWAV_BUSY -19
-#define DRWAV_IO_ERROR -20
-#define DRWAV_INTERRUPT -21
-#define DRWAV_UNAVAILABLE -22
-#define DRWAV_ALREADY_IN_USE -23
-#define DRWAV_BAD_ADDRESS -24
-#define DRWAV_BAD_SEEK -25
-#define DRWAV_BAD_PIPE -26
-#define DRWAV_DEADLOCK -27
-#define DRWAV_TOO_MANY_LINKS -28
-#define DRWAV_NOT_IMPLEMENTED -29
-#define DRWAV_NO_MESSAGE -30
-#define DRWAV_BAD_MESSAGE -31
-#define DRWAV_NO_DATA_AVAILABLE -32
-#define DRWAV_INVALID_DATA -33
-#define DRWAV_TIMEOUT -34
-#define DRWAV_NO_NETWORK -35
-#define DRWAV_NOT_UNIQUE -36
-#define DRWAV_NOT_SOCKET -37
-#define DRWAV_NO_ADDRESS -38
-#define DRWAV_BAD_PROTOCOL -39
-#define DRWAV_PROTOCOL_UNAVAILABLE -40
-#define DRWAV_PROTOCOL_NOT_SUPPORTED -41
-#define DRWAV_PROTOCOL_FAMILY_NOT_SUPPORTED -42
-#define DRWAV_ADDRESS_FAMILY_NOT_SUPPORTED -43
-#define DRWAV_SOCKET_NOT_SUPPORTED -44
-#define DRWAV_CONNECTION_RESET -45
-#define DRWAV_ALREADY_CONNECTED -46
-#define DRWAV_NOT_CONNECTED -47
-#define DRWAV_CONNECTION_REFUSED -48
-#define DRWAV_NO_HOST -49
-#define DRWAV_IN_PROGRESS -50
-#define DRWAV_CANCELLED -51
-#define DRWAV_MEMORY_ALREADY_MAPPED -52
-#define DRWAV_AT_END -53
-#define DR_WAVE_FORMAT_PCM 0x1
-#define DR_WAVE_FORMAT_ADPCM 0x2
-#define DR_WAVE_FORMAT_IEEE_FLOAT 0x3
-#define DR_WAVE_FORMAT_ALAW 0x6
-#define DR_WAVE_FORMAT_MULAW 0x7
-#define DR_WAVE_FORMAT_DVI_ADPCM 0x11
-#define DR_WAVE_FORMAT_EXTENSIBLE 0xFFFE
-#define DRWAV_SEQUENTIAL 0x00000001
-DRWAV_API void drwav_version(drwav_uint32* pMajor, drwav_uint32* pMinor, drwav_uint32* pRevision);
-DRWAV_API const char* drwav_version_string(void);
-typedef enum
-{
- drwav_seek_origin_start,
- drwav_seek_origin_current
-} drwav_seek_origin;
-typedef enum
-{
- drwav_container_riff,
- drwav_container_w64,
- drwav_container_rf64
-} drwav_container;
-typedef struct
-{
- union
+ switch (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode))
{
- drwav_uint8 fourcc[4];
- drwav_uint8 guid[16];
- } id;
- drwav_uint64 sizeInBytes;
- unsigned int paddingSize;
-} drwav_chunk_header;
-typedef struct
-{
- drwav_uint16 formatTag;
- drwav_uint16 channels;
- drwav_uint32 sampleRate;
- drwav_uint32 avgBytesPerSec;
- drwav_uint16 blockAlign;
- drwav_uint16 bitsPerSample;
- drwav_uint16 extendedSize;
- drwav_uint16 validBitsPerSample;
- drwav_uint32 channelMask;
- drwav_uint8 subFormat[16];
-} drwav_fmt;
-DRWAV_API drwav_uint16 drwav_fmt_get_format(const drwav_fmt* pFMT);
-typedef size_t (* drwav_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead);
-typedef size_t (* drwav_write_proc)(void* pUserData, const void* pData, size_t bytesToWrite);
-typedef drwav_bool32 (* drwav_seek_proc)(void* pUserData, int offset, drwav_seek_origin origin);
-typedef drwav_uint64 (* drwav_chunk_proc)(void* pChunkUserData, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pReadSeekUserData, const drwav_chunk_header* pChunkHeader, drwav_container container, const drwav_fmt* pFMT);
-typedef struct
-{
- void* pUserData;
- void* (* onMalloc)(size_t sz, void* pUserData);
- void* (* onRealloc)(void* p, size_t sz, void* pUserData);
- void (* onFree)(void* p, void* pUserData);
-} drwav_allocation_callbacks;
-typedef struct
-{
- const drwav_uint8* data;
- size_t dataSize;
- size_t currentReadPos;
-} drwav__memory_stream;
-typedef struct
-{
- void** ppData;
- size_t* pDataSize;
- size_t dataSize;
- size_t dataCapacity;
- size_t currentWritePos;
-} drwav__memory_stream_write;
-typedef struct
-{
- drwav_container container;
- drwav_uint32 format;
- drwav_uint32 channels;
- drwav_uint32 sampleRate;
- drwav_uint32 bitsPerSample;
-} drwav_data_format;
-typedef enum
-{
- drwav_metadata_type_none = 0,
- drwav_metadata_type_unknown = 1 << 0,
- drwav_metadata_type_smpl = 1 << 1,
- drwav_metadata_type_inst = 1 << 2,
- drwav_metadata_type_cue = 1 << 3,
- drwav_metadata_type_acid = 1 << 4,
- drwav_metadata_type_bext = 1 << 5,
- drwav_metadata_type_list_label = 1 << 6,
- drwav_metadata_type_list_note = 1 << 7,
- drwav_metadata_type_list_labelled_cue_region = 1 << 8,
- drwav_metadata_type_list_info_software = 1 << 9,
- drwav_metadata_type_list_info_copyright = 1 << 10,
- drwav_metadata_type_list_info_title = 1 << 11,
- drwav_metadata_type_list_info_artist = 1 << 12,
- drwav_metadata_type_list_info_comment = 1 << 13,
- drwav_metadata_type_list_info_date = 1 << 14,
- drwav_metadata_type_list_info_genre = 1 << 15,
- drwav_metadata_type_list_info_album = 1 << 16,
- drwav_metadata_type_list_info_tracknumber = 1 << 17,
- drwav_metadata_type_list_all_info_strings = drwav_metadata_type_list_info_software
- | drwav_metadata_type_list_info_copyright
- | drwav_metadata_type_list_info_title
- | drwav_metadata_type_list_info_artist
- | drwav_metadata_type_list_info_comment
- | drwav_metadata_type_list_info_date
- | drwav_metadata_type_list_info_genre
- | drwav_metadata_type_list_info_album
- | drwav_metadata_type_list_info_tracknumber,
- drwav_metadata_type_list_all_adtl = drwav_metadata_type_list_label
- | drwav_metadata_type_list_note
- | drwav_metadata_type_list_labelled_cue_region,
- drwav_metadata_type_all = -2,
- drwav_metadata_type_all_including_unknown = -1
-} drwav_metadata_type;
-typedef enum
-{
- drwav_smpl_loop_type_forward = 0,
- drwav_smpl_loop_type_pingpong = 1,
- drwav_smpl_loop_type_backward = 2
-} drwav_smpl_loop_type;
-typedef struct
-{
- drwav_uint32 cuePointId;
- drwav_uint32 type;
- drwav_uint32 firstSampleByteOffset;
- drwav_uint32 lastSampleByteOffset;
- drwav_uint32 sampleFraction;
- drwav_uint32 playCount;
-} drwav_smpl_loop;
-typedef struct
-{
- drwav_uint32 manufacturerId;
- drwav_uint32 productId;
- drwav_uint32 samplePeriodNanoseconds;
- drwav_uint32 midiUnityNote;
- drwav_uint32 midiPitchFraction;
- drwav_uint32 smpteFormat;
- drwav_uint32 smpteOffset;
- drwav_uint32 sampleLoopCount;
- drwav_uint32 samplerSpecificDataSizeInBytes;
- drwav_smpl_loop* pLoops;
- drwav_uint8* pSamplerSpecificData;
-} drwav_smpl;
-typedef struct
-{
- drwav_int8 midiUnityNote;
- drwav_int8 fineTuneCents;
- drwav_int8 gainDecibels;
- drwav_int8 lowNote;
- drwav_int8 highNote;
- drwav_int8 lowVelocity;
- drwav_int8 highVelocity;
-} drwav_inst;
-typedef struct
-{
- drwav_uint32 id;
- drwav_uint32 playOrderPosition;
- drwav_uint8 dataChunkId[4];
- drwav_uint32 chunkStart;
- drwav_uint32 blockStart;
- drwav_uint32 sampleByteOffset;
-} drwav_cue_point;
-typedef struct
-{
- drwav_uint32 cuePointCount;
- drwav_cue_point *pCuePoints;
-} drwav_cue;
-typedef enum
-{
- drwav_acid_flag_one_shot = 1,
- drwav_acid_flag_root_note_set = 2,
- drwav_acid_flag_stretch = 4,
- drwav_acid_flag_disk_based = 8,
- drwav_acid_flag_acidizer = 16
-} drwav_acid_flag;
-typedef struct
-{
- drwav_uint32 flags;
- drwav_uint16 midiUnityNote;
- drwav_uint16 reserved1;
- float reserved2;
- drwav_uint32 numBeats;
- drwav_uint16 meterDenominator;
- drwav_uint16 meterNumerator;
- float tempo;
-} drwav_acid;
-typedef struct
+ case ma_resource_manager_data_supply_type_encoded:
+ {
+ return ma_decoder_get_available_frames(&pDataBuffer->connector.decoder, pAvailableFrames);
+ };
+
+ case ma_resource_manager_data_supply_type_decoded:
+ {
+ return ma_audio_buffer_get_available_frames(&pDataBuffer->connector.buffer, pAvailableFrames);
+ };
+
+ case ma_resource_manager_data_supply_type_decoded_paged:
+ {
+ ma_uint64 cursor;
+ ma_paged_audio_buffer_get_cursor_in_pcm_frames(&pDataBuffer->connector.pagedBuffer, &cursor);
+
+ if (pDataBuffer->pNode->data.backend.decodedPaged.decodedFrameCount > cursor) {
+ *pAvailableFrames = pDataBuffer->pNode->data.backend.decodedPaged.decodedFrameCount - cursor;
+ } else {
+ *pAvailableFrames = 0;
+ }
+
+ return MA_SUCCESS;
+ };
+
+ case ma_resource_manager_data_supply_type_unknown:
+ default:
+ {
+ /* Unknown supply type. Should never hit this. */
+ return MA_INVALID_ARGS;
+ }
+ }
+}
+
+MA_API ma_result ma_resource_manager_register_file(ma_resource_manager* pResourceManager, const char* pFilePath, ma_uint32 flags)
{
- drwav_uint32 cuePointId;
- drwav_uint32 stringLength;
- char* pString;
-} drwav_list_label_or_note;
-typedef struct
+ return ma_resource_manager_data_buffer_node_acquire(pResourceManager, pFilePath, NULL, 0, flags, NULL, NULL, NULL, NULL);
+}
+
+MA_API ma_result ma_resource_manager_register_file_w(ma_resource_manager* pResourceManager, const wchar_t* pFilePath, ma_uint32 flags)
{
- char* pDescription;
- char* pOriginatorName;
- char* pOriginatorReference;
- char pOriginationDate[10];
- char pOriginationTime[8];
- drwav_uint64 timeReference;
- drwav_uint16 version;
- char* pCodingHistory;
- drwav_uint32 codingHistorySize;
- drwav_uint8* pUMID;
- drwav_uint16 loudnessValue;
- drwav_uint16 loudnessRange;
- drwav_uint16 maxTruePeakLevel;
- drwav_uint16 maxMomentaryLoudness;
- drwav_uint16 maxShortTermLoudness;
-} drwav_bext;
-typedef struct
+ return ma_resource_manager_data_buffer_node_acquire(pResourceManager, NULL, pFilePath, 0, flags, NULL, NULL, NULL, NULL);
+}
+
+
+static ma_result ma_resource_manager_register_data(ma_resource_manager* pResourceManager, const char* pName, const wchar_t* pNameW, ma_resource_manager_data_supply* pExistingData)
{
- drwav_uint32 stringLength;
- char* pString;
-} drwav_list_info_text;
-typedef struct
+ return ma_resource_manager_data_buffer_node_acquire(pResourceManager, pName, pNameW, 0, 0, pExistingData, NULL, NULL, NULL);
+}
+
+static ma_result ma_resource_manager_register_decoded_data_internal(ma_resource_manager* pResourceManager, const char* pName, const wchar_t* pNameW, const void* pData, ma_uint64 frameCount, ma_format format, ma_uint32 channels, ma_uint32 sampleRate)
{
- drwav_uint32 cuePointId;
- drwav_uint32 sampleLength;
- drwav_uint8 purposeId[4];
- drwav_uint16 country;
- drwav_uint16 language;
- drwav_uint16 dialect;
- drwav_uint16 codePage;
- drwav_uint32 stringLength;
- char* pString;
-} drwav_list_labelled_cue_region;
-typedef enum
+ ma_resource_manager_data_supply data;
+ data.type = ma_resource_manager_data_supply_type_decoded;
+ data.backend.decoded.pData = pData;
+ data.backend.decoded.totalFrameCount = frameCount;
+ data.backend.decoded.format = format;
+ data.backend.decoded.channels = channels;
+ data.backend.decoded.sampleRate = sampleRate;
+
+ return ma_resource_manager_register_data(pResourceManager, pName, pNameW, &data);
+}
+
+MA_API ma_result ma_resource_manager_register_decoded_data(ma_resource_manager* pResourceManager, const char* pName, const void* pData, ma_uint64 frameCount, ma_format format, ma_uint32 channels, ma_uint32 sampleRate)
{
- drwav_metadata_location_invalid,
- drwav_metadata_location_top_level,
- drwav_metadata_location_inside_info_list,
- drwav_metadata_location_inside_adtl_list
-} drwav_metadata_location;
-typedef struct
+ return ma_resource_manager_register_decoded_data_internal(pResourceManager, pName, NULL, pData, frameCount, format, channels, sampleRate);
+}
+
+MA_API ma_result ma_resource_manager_register_decoded_data_w(ma_resource_manager* pResourceManager, const wchar_t* pName, const void* pData, ma_uint64 frameCount, ma_format format, ma_uint32 channels, ma_uint32 sampleRate)
{
- drwav_uint8 id[4];
- drwav_metadata_location chunkLocation;
- drwav_uint32 dataSizeInBytes;
- drwav_uint8* pData;
-} drwav_unknown_metadata;
-typedef struct
+ return ma_resource_manager_register_decoded_data_internal(pResourceManager, NULL, pName, pData, frameCount, format, channels, sampleRate);
+}
+
+
+static ma_result ma_resource_manager_register_encoded_data_internal(ma_resource_manager* pResourceManager, const char* pName, const wchar_t* pNameW, const void* pData, size_t sizeInBytes)
{
- drwav_metadata_type type;
- union
- {
- drwav_cue cue;
- drwav_smpl smpl;
- drwav_acid acid;
- drwav_inst inst;
- drwav_bext bext;
- drwav_list_label_or_note labelOrNote;
- drwav_list_labelled_cue_region labelledCueRegion;
- drwav_list_info_text infoText;
- drwav_unknown_metadata unknown;
- } data;
-} drwav_metadata;
-typedef struct
+ ma_resource_manager_data_supply data;
+ data.type = ma_resource_manager_data_supply_type_encoded;
+ data.backend.encoded.pData = pData;
+ data.backend.encoded.sizeInBytes = sizeInBytes;
+
+ return ma_resource_manager_register_data(pResourceManager, pName, pNameW, &data);
+}
+
+MA_API ma_result ma_resource_manager_register_encoded_data(ma_resource_manager* pResourceManager, const char* pName, const void* pData, size_t sizeInBytes)
{
- drwav_read_proc onRead;
- drwav_write_proc onWrite;
- drwav_seek_proc onSeek;
- void* pUserData;
- drwav_allocation_callbacks allocationCallbacks;
- drwav_container container;
- drwav_fmt fmt;
- drwav_uint32 sampleRate;
- drwav_uint16 channels;
- drwav_uint16 bitsPerSample;
- drwav_uint16 translatedFormatTag;
- drwav_uint64 totalPCMFrameCount;
- drwav_uint64 dataChunkDataSize;
- drwav_uint64 dataChunkDataPos;
- drwav_uint64 bytesRemaining;
- drwav_uint64 readCursorInPCMFrames;
- drwav_uint64 dataChunkDataSizeTargetWrite;
- drwav_bool32 isSequentialWrite;
- drwav_metadata_type allowedMetadataTypes;
- drwav_metadata* pMetadata;
- drwav_uint32 metadataCount;
- drwav__memory_stream memoryStream;
- drwav__memory_stream_write memoryStreamWrite;
- struct
- {
- drwav_uint32 bytesRemainingInBlock;
- drwav_uint16 predictor[2];
- drwav_int32 delta[2];
- drwav_int32 cachedFrames[4];
- drwav_uint32 cachedFrameCount;
- drwav_int32 prevFrames[2][2];
- } msadpcm;
- struct
- {
- drwav_uint32 bytesRemainingInBlock;
- drwav_int32 predictor[2];
- drwav_int32 stepIndex[2];
- drwav_int32 cachedFrames[16];
- drwav_uint32 cachedFrameCount;
- } ima;
-} drwav;
-DRWAV_API drwav_bool32 drwav_init(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_ex(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_chunk_proc onChunk, void* pReadSeekUserData, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_with_metadata(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_write(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_write_sequential(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_write_sequential_pcm_frames(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_write_with_metadata(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks, drwav_metadata* pMetadata, drwav_uint32 metadataCount);
-DRWAV_API drwav_uint64 drwav_target_write_size_bytes(const drwav_data_format* pFormat, drwav_uint64 totalFrameCount, drwav_metadata* pMetadata, drwav_uint32 metadataCount);
-DRWAV_API drwav_metadata* drwav_take_ownership_of_metadata(drwav* pWav);
-DRWAV_API drwav_result drwav_uninit(drwav* pWav);
-DRWAV_API size_t drwav_read_raw(drwav* pWav, size_t bytesToRead, void* pBufferOut);
-DRWAV_API drwav_uint64 drwav_read_pcm_frames(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut);
-DRWAV_API drwav_uint64 drwav_read_pcm_frames_le(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut);
-DRWAV_API drwav_uint64 drwav_read_pcm_frames_be(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut);
-DRWAV_API drwav_bool32 drwav_seek_to_pcm_frame(drwav* pWav, drwav_uint64 targetFrameIndex);
-DRWAV_API drwav_result drwav_get_cursor_in_pcm_frames(drwav* pWav, drwav_uint64* pCursor);
-DRWAV_API drwav_result drwav_get_length_in_pcm_frames(drwav* pWav, drwav_uint64* pLength);
-DRWAV_API size_t drwav_write_raw(drwav* pWav, size_t bytesToWrite, const void* pData);
-DRWAV_API drwav_uint64 drwav_write_pcm_frames(drwav* pWav, drwav_uint64 framesToWrite, const void* pData);
-DRWAV_API drwav_uint64 drwav_write_pcm_frames_le(drwav* pWav, drwav_uint64 framesToWrite, const void* pData);
-DRWAV_API drwav_uint64 drwav_write_pcm_frames_be(drwav* pWav, drwav_uint64 framesToWrite, const void* pData);
-#ifndef DR_WAV_NO_CONVERSION_API
-DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut);
-DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16le(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut);
-DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16be(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut);
-DRWAV_API void drwav_u8_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount);
-DRWAV_API void drwav_s24_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount);
-DRWAV_API void drwav_s32_to_s16(drwav_int16* pOut, const drwav_int32* pIn, size_t sampleCount);
-DRWAV_API void drwav_f32_to_s16(drwav_int16* pOut, const float* pIn, size_t sampleCount);
-DRWAV_API void drwav_f64_to_s16(drwav_int16* pOut, const double* pIn, size_t sampleCount);
-DRWAV_API void drwav_alaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount);
-DRWAV_API void drwav_mulaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount);
-DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut);
-DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32le(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut);
-DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32be(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut);
-DRWAV_API void drwav_u8_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount);
-DRWAV_API void drwav_s16_to_f32(float* pOut, const drwav_int16* pIn, size_t sampleCount);
-DRWAV_API void drwav_s24_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount);
-DRWAV_API void drwav_s32_to_f32(float* pOut, const drwav_int32* pIn, size_t sampleCount);
-DRWAV_API void drwav_f64_to_f32(float* pOut, const double* pIn, size_t sampleCount);
-DRWAV_API void drwav_alaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount);
-DRWAV_API void drwav_mulaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount);
-DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut);
-DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32le(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut);
-DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32be(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut);
-DRWAV_API void drwav_u8_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount);
-DRWAV_API void drwav_s16_to_s32(drwav_int32* pOut, const drwav_int16* pIn, size_t sampleCount);
-DRWAV_API void drwav_s24_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount);
-DRWAV_API void drwav_f32_to_s32(drwav_int32* pOut, const float* pIn, size_t sampleCount);
-DRWAV_API void drwav_f64_to_s32(drwav_int32* pOut, const double* pIn, size_t sampleCount);
-DRWAV_API void drwav_alaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount);
-DRWAV_API void drwav_mulaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount);
-#endif
-#ifndef DR_WAV_NO_STDIO
-DRWAV_API drwav_bool32 drwav_init_file(drwav* pWav, const char* filename, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_file_ex(drwav* pWav, const char* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_file_w(drwav* pWav, const wchar_t* filename, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_file_ex_w(drwav* pWav, const wchar_t* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_file_with_metadata(drwav* pWav, const char* filename, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_file_with_metadata_w(drwav* pWav, const wchar_t* filename, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_file_write(drwav* pWav, const char* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_file_write_sequential(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_file_write_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_file_write_sequential_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks);
-#endif
-DRWAV_API drwav_bool32 drwav_init_memory(drwav* pWav, const void* data, size_t dataSize, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_memory_ex(drwav* pWav, const void* data, size_t dataSize, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_memory_with_metadata(drwav* pWav, const void* data, size_t dataSize, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_memory_write(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_memory_write_sequential(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_bool32 drwav_init_memory_write_sequential_pcm_frames(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks);
-#ifndef DR_WAV_NO_CONVERSION_API
-DRWAV_API drwav_int16* drwav_open_and_read_pcm_frames_s16(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API float* drwav_open_and_read_pcm_frames_f32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_int32* drwav_open_and_read_pcm_frames_s32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
-#ifndef DR_WAV_NO_STDIO
-DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
-#endif
-DRWAV_API drwav_int16* drwav_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API float* drwav_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_int32* drwav_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
-#endif
-DRWAV_API void drwav_free(void* p, const drwav_allocation_callbacks* pAllocationCallbacks);
-DRWAV_API drwav_uint16 drwav_bytes_to_u16(const drwav_uint8* data);
-DRWAV_API drwav_int16 drwav_bytes_to_s16(const drwav_uint8* data);
-DRWAV_API drwav_uint32 drwav_bytes_to_u32(const drwav_uint8* data);
-DRWAV_API drwav_int32 drwav_bytes_to_s32(const drwav_uint8* data);
-DRWAV_API drwav_uint64 drwav_bytes_to_u64(const drwav_uint8* data);
-DRWAV_API drwav_int64 drwav_bytes_to_s64(const drwav_uint8* data);
-DRWAV_API float drwav_bytes_to_f32(const drwav_uint8* data);
-DRWAV_API drwav_bool32 drwav_guid_equal(const drwav_uint8 a[16], const drwav_uint8 b[16]);
-DRWAV_API drwav_bool32 drwav_fourcc_equal(const drwav_uint8* a, const char* b);
-#ifdef __cplusplus
+ return ma_resource_manager_register_encoded_data_internal(pResourceManager, pName, NULL, pData, sizeInBytes);
}
-#endif
-#endif
-/* dr_wav_h end */
-#endif /* MA_NO_WAV */
-#if !defined(MA_NO_FLAC) && !defined(MA_NO_DECODING)
-/* dr_flac_h begin */
-#ifndef dr_flac_h
-#define dr_flac_h
-#ifdef __cplusplus
-extern "C" {
-#endif
-#define DRFLAC_STRINGIFY(x) #x
-#define DRFLAC_XSTRINGIFY(x) DRFLAC_STRINGIFY(x)
-#define DRFLAC_VERSION_MAJOR 0
-#define DRFLAC_VERSION_MINOR 12
-#define DRFLAC_VERSION_REVISION 31
-#define DRFLAC_VERSION_STRING DRFLAC_XSTRINGIFY(DRFLAC_VERSION_MAJOR) "." DRFLAC_XSTRINGIFY(DRFLAC_VERSION_MINOR) "." DRFLAC_XSTRINGIFY(DRFLAC_VERSION_REVISION)
-#include
-typedef signed char drflac_int8;
-typedef unsigned char drflac_uint8;
-typedef signed short drflac_int16;
-typedef unsigned short drflac_uint16;
-typedef signed int drflac_int32;
-typedef unsigned int drflac_uint32;
-#if defined(_MSC_VER)
- typedef signed __int64 drflac_int64;
- typedef unsigned __int64 drflac_uint64;
-#else
- #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
- #pragma GCC diagnostic push
- #pragma GCC diagnostic ignored "-Wlong-long"
- #if defined(__clang__)
- #pragma GCC diagnostic ignored "-Wc++11-long-long"
- #endif
- #endif
- typedef signed long long drflac_int64;
- typedef unsigned long long drflac_uint64;
- #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
- #pragma GCC diagnostic pop
- #endif
-#endif
-#if defined(__LP64__) || defined(_WIN64) || (defined(__x86_64__) && !defined(__ILP32__)) || defined(_M_X64) || defined(__ia64) || defined (_M_IA64) || defined(__aarch64__) || defined(_M_ARM64) || defined(__powerpc64__)
- typedef drflac_uint64 drflac_uintptr;
-#else
- typedef drflac_uint32 drflac_uintptr;
-#endif
-typedef drflac_uint8 drflac_bool8;
-typedef drflac_uint32 drflac_bool32;
-#define DRFLAC_TRUE 1
-#define DRFLAC_FALSE 0
-#if !defined(DRFLAC_API)
- #if defined(DRFLAC_DLL)
- #if defined(_WIN32)
- #define DRFLAC_DLL_IMPORT __declspec(dllimport)
- #define DRFLAC_DLL_EXPORT __declspec(dllexport)
- #define DRFLAC_DLL_PRIVATE static
- #else
- #if defined(__GNUC__) && __GNUC__ >= 4
- #define DRFLAC_DLL_IMPORT __attribute__((visibility("default")))
- #define DRFLAC_DLL_EXPORT __attribute__((visibility("default")))
- #define DRFLAC_DLL_PRIVATE __attribute__((visibility("hidden")))
- #else
- #define DRFLAC_DLL_IMPORT
- #define DRFLAC_DLL_EXPORT
- #define DRFLAC_DLL_PRIVATE static
- #endif
- #endif
- #if defined(DR_FLAC_IMPLEMENTATION) || defined(DRFLAC_IMPLEMENTATION)
- #define DRFLAC_API DRFLAC_DLL_EXPORT
- #else
- #define DRFLAC_API DRFLAC_DLL_IMPORT
- #endif
- #define DRFLAC_PRIVATE DRFLAC_DLL_PRIVATE
- #else
- #define DRFLAC_API extern
- #define DRFLAC_PRIVATE static
- #endif
-#endif
-#if defined(_MSC_VER) && _MSC_VER >= 1700
- #define DRFLAC_DEPRECATED __declspec(deprecated)
-#elif (defined(__GNUC__) && __GNUC__ >= 4)
- #define DRFLAC_DEPRECATED __attribute__((deprecated))
-#elif defined(__has_feature)
- #if __has_feature(attribute_deprecated)
- #define DRFLAC_DEPRECATED __attribute__((deprecated))
- #else
- #define DRFLAC_DEPRECATED
- #endif
-#else
- #define DRFLAC_DEPRECATED
-#endif
-DRFLAC_API void drflac_version(drflac_uint32* pMajor, drflac_uint32* pMinor, drflac_uint32* pRevision);
-DRFLAC_API const char* drflac_version_string(void);
-#ifndef DR_FLAC_BUFFER_SIZE
-#define DR_FLAC_BUFFER_SIZE 4096
-#endif
-#if defined(_WIN64) || defined(_LP64) || defined(__LP64__)
-#define DRFLAC_64BIT
-#endif
-#ifdef DRFLAC_64BIT
-typedef drflac_uint64 drflac_cache_t;
-#else
-typedef drflac_uint32 drflac_cache_t;
-#endif
-#define DRFLAC_METADATA_BLOCK_TYPE_STREAMINFO 0
-#define DRFLAC_METADATA_BLOCK_TYPE_PADDING 1
-#define DRFLAC_METADATA_BLOCK_TYPE_APPLICATION 2
-#define DRFLAC_METADATA_BLOCK_TYPE_SEEKTABLE 3
-#define DRFLAC_METADATA_BLOCK_TYPE_VORBIS_COMMENT 4
-#define DRFLAC_METADATA_BLOCK_TYPE_CUESHEET 5
-#define DRFLAC_METADATA_BLOCK_TYPE_PICTURE 6
-#define DRFLAC_METADATA_BLOCK_TYPE_INVALID 127
-#define DRFLAC_PICTURE_TYPE_OTHER 0
-#define DRFLAC_PICTURE_TYPE_FILE_ICON 1
-#define DRFLAC_PICTURE_TYPE_OTHER_FILE_ICON 2
-#define DRFLAC_PICTURE_TYPE_COVER_FRONT 3
-#define DRFLAC_PICTURE_TYPE_COVER_BACK 4
-#define DRFLAC_PICTURE_TYPE_LEAFLET_PAGE 5
-#define DRFLAC_PICTURE_TYPE_MEDIA 6
-#define DRFLAC_PICTURE_TYPE_LEAD_ARTIST 7
-#define DRFLAC_PICTURE_TYPE_ARTIST 8
-#define DRFLAC_PICTURE_TYPE_CONDUCTOR 9
-#define DRFLAC_PICTURE_TYPE_BAND 10
-#define DRFLAC_PICTURE_TYPE_COMPOSER 11
-#define DRFLAC_PICTURE_TYPE_LYRICIST 12
-#define DRFLAC_PICTURE_TYPE_RECORDING_LOCATION 13
-#define DRFLAC_PICTURE_TYPE_DURING_RECORDING 14
-#define DRFLAC_PICTURE_TYPE_DURING_PERFORMANCE 15
-#define DRFLAC_PICTURE_TYPE_SCREEN_CAPTURE 16
-#define DRFLAC_PICTURE_TYPE_BRIGHT_COLORED_FISH 17
-#define DRFLAC_PICTURE_TYPE_ILLUSTRATION 18
-#define DRFLAC_PICTURE_TYPE_BAND_LOGOTYPE 19
-#define DRFLAC_PICTURE_TYPE_PUBLISHER_LOGOTYPE 20
-typedef enum
+MA_API ma_result ma_resource_manager_register_encoded_data_w(ma_resource_manager* pResourceManager, const wchar_t* pName, const void* pData, size_t sizeInBytes)
{
- drflac_container_native,
- drflac_container_ogg,
- drflac_container_unknown
-} drflac_container;
-typedef enum
+ return ma_resource_manager_register_encoded_data_internal(pResourceManager, NULL, pName, pData, sizeInBytes);
+}
+
+
+MA_API ma_result ma_resource_manager_unregister_file(ma_resource_manager* pResourceManager, const char* pFilePath)
{
- drflac_seek_origin_start,
- drflac_seek_origin_current
-} drflac_seek_origin;
-#pragma pack(2)
-typedef struct
+ return ma_resource_manager_unregister_data(pResourceManager, pFilePath);
+}
+
+MA_API ma_result ma_resource_manager_unregister_file_w(ma_resource_manager* pResourceManager, const wchar_t* pFilePath)
{
- drflac_uint64 firstPCMFrame;
- drflac_uint64 flacFrameOffset;
- drflac_uint16 pcmFrameCount;
-} drflac_seekpoint;
-#pragma pack()
-typedef struct
+ return ma_resource_manager_unregister_data_w(pResourceManager, pFilePath);
+}
+
+MA_API ma_result ma_resource_manager_unregister_data(ma_resource_manager* pResourceManager, const char* pName)
{
- drflac_uint16 minBlockSizeInPCMFrames;
- drflac_uint16 maxBlockSizeInPCMFrames;
- drflac_uint32 minFrameSizeInPCMFrames;
- drflac_uint32 maxFrameSizeInPCMFrames;
- drflac_uint32 sampleRate;
- drflac_uint8 channels;
- drflac_uint8 bitsPerSample;
- drflac_uint64 totalPCMFrameCount;
- drflac_uint8 md5[16];
-} drflac_streaminfo;
-typedef struct
+ return ma_resource_manager_data_buffer_node_unacquire(pResourceManager, NULL, pName, NULL);
+}
+
+MA_API ma_result ma_resource_manager_unregister_data_w(ma_resource_manager* pResourceManager, const wchar_t* pName)
{
- drflac_uint32 type;
- const void* pRawData;
- drflac_uint32 rawDataSize;
- union
- {
- drflac_streaminfo streaminfo;
- struct
- {
- int unused;
- } padding;
- struct
- {
- drflac_uint32 id;
- const void* pData;
- drflac_uint32 dataSize;
- } application;
- struct
- {
- drflac_uint32 seekpointCount;
- const drflac_seekpoint* pSeekpoints;
- } seektable;
- struct
- {
- drflac_uint32 vendorLength;
- const char* vendor;
- drflac_uint32 commentCount;
- const void* pComments;
- } vorbis_comment;
- struct
- {
- char catalog[128];
- drflac_uint64 leadInSampleCount;
- drflac_bool32 isCD;
- drflac_uint8 trackCount;
- const void* pTrackData;
- } cuesheet;
- struct
- {
- drflac_uint32 type;
- drflac_uint32 mimeLength;
- const char* mime;
- drflac_uint32 descriptionLength;
- const char* description;
- drflac_uint32 width;
- drflac_uint32 height;
- drflac_uint32 colorDepth;
- drflac_uint32 indexColorCount;
- drflac_uint32 pictureDataSize;
- const drflac_uint8* pPictureData;
- } picture;
- } data;
-} drflac_metadata;
-typedef size_t (* drflac_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead);
-typedef drflac_bool32 (* drflac_seek_proc)(void* pUserData, int offset, drflac_seek_origin origin);
-typedef void (* drflac_meta_proc)(void* pUserData, drflac_metadata* pMetadata);
-typedef struct
+ return ma_resource_manager_data_buffer_node_unacquire(pResourceManager, NULL, NULL, pName);
+}
+
+
+static ma_uint32 ma_resource_manager_data_stream_next_execution_order(ma_resource_manager_data_stream* pDataStream)
{
- void* pUserData;
- void* (* onMalloc)(size_t sz, void* pUserData);
- void* (* onRealloc)(void* p, size_t sz, void* pUserData);
- void (* onFree)(void* p, void* pUserData);
-} drflac_allocation_callbacks;
-typedef struct
+ MA_ASSERT(pDataStream != NULL);
+ return c89atomic_fetch_add_32(&pDataStream->executionCounter, 1);
+}
+
+static ma_bool32 ma_resource_manager_data_stream_is_decoder_at_end(const ma_resource_manager_data_stream* pDataStream)
{
- const drflac_uint8* data;
- size_t dataSize;
- size_t currentReadPos;
-} drflac__memory_stream;
-typedef struct
+ MA_ASSERT(pDataStream != NULL);
+ return c89atomic_load_32((ma_bool32*)&pDataStream->isDecoderAtEnd);
+}
+
+static ma_uint32 ma_resource_manager_data_stream_seek_counter(const ma_resource_manager_data_stream* pDataStream)
{
- drflac_read_proc onRead;
- drflac_seek_proc onSeek;
- void* pUserData;
- size_t unalignedByteCount;
- drflac_cache_t unalignedCache;
- drflac_uint32 nextL2Line;
- drflac_uint32 consumedBits;
- drflac_cache_t cacheL2[DR_FLAC_BUFFER_SIZE/sizeof(drflac_cache_t)];
- drflac_cache_t cache;
- drflac_uint16 crc16;
- drflac_cache_t crc16Cache;
- drflac_uint32 crc16CacheIgnoredBytes;
-} drflac_bs;
-typedef struct
+ MA_ASSERT(pDataStream != NULL);
+ return c89atomic_load_32((ma_uint32*)&pDataStream->seekCounter);
+}
+
+
+static ma_result ma_resource_manager_data_stream_cb__read_pcm_frames(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
- drflac_uint8 subframeType;
- drflac_uint8 wastedBitsPerSample;
- drflac_uint8 lpcOrder;
- drflac_int32* pSamplesS32;
-} drflac_subframe;
-typedef struct
+ return ma_resource_manager_data_stream_read_pcm_frames((ma_resource_manager_data_stream*)pDataSource, pFramesOut, frameCount, pFramesRead);
+}
+
+static ma_result ma_resource_manager_data_stream_cb__seek_to_pcm_frame(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
- drflac_uint64 pcmFrameNumber;
- drflac_uint32 flacFrameNumber;
- drflac_uint32 sampleRate;
- drflac_uint16 blockSizeInPCMFrames;
- drflac_uint8 channelAssignment;
- drflac_uint8 bitsPerSample;
- drflac_uint8 crc8;
-} drflac_frame_header;
-typedef struct
+ return ma_resource_manager_data_stream_seek_to_pcm_frame((ma_resource_manager_data_stream*)pDataSource, frameIndex);
+}
+
+static ma_result ma_resource_manager_data_stream_cb__get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
- drflac_frame_header header;
- drflac_uint32 pcmFramesRemaining;
- drflac_subframe subframes[8];
-} drflac_frame;
-typedef struct
+ return ma_resource_manager_data_stream_get_data_format((ma_resource_manager_data_stream*)pDataSource, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
+}
+
+static ma_result ma_resource_manager_data_stream_cb__get_cursor_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pCursor)
{
- drflac_meta_proc onMeta;
- void* pUserDataMD;
- drflac_allocation_callbacks allocationCallbacks;
- drflac_uint32 sampleRate;
- drflac_uint8 channels;
- drflac_uint8 bitsPerSample;
- drflac_uint16 maxBlockSizeInPCMFrames;
- drflac_uint64 totalPCMFrameCount;
- drflac_container container;
- drflac_uint32 seekpointCount;
- drflac_frame currentFLACFrame;
- drflac_uint64 currentPCMFrame;
- drflac_uint64 firstFLACFramePosInBytes;
- drflac__memory_stream memoryStream;
- drflac_int32* pDecodedSamples;
- drflac_seekpoint* pSeekpoints;
- void* _oggbs;
- drflac_bool32 _noSeekTableSeek : 1;
- drflac_bool32 _noBinarySearchSeek : 1;
- drflac_bool32 _noBruteForceSeek : 1;
- drflac_bs bs;
- drflac_uint8 pExtraData[1];
-} drflac;
-DRFLAC_API drflac* drflac_open(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
-DRFLAC_API drflac* drflac_open_relaxed(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_container container, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
-DRFLAC_API drflac* drflac_open_with_metadata(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
-DRFLAC_API drflac* drflac_open_with_metadata_relaxed(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, drflac_container container, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
-DRFLAC_API void drflac_close(drflac* pFlac);
-DRFLAC_API drflac_uint64 drflac_read_pcm_frames_s32(drflac* pFlac, drflac_uint64 framesToRead, drflac_int32* pBufferOut);
-DRFLAC_API drflac_uint64 drflac_read_pcm_frames_s16(drflac* pFlac, drflac_uint64 framesToRead, drflac_int16* pBufferOut);
-DRFLAC_API drflac_uint64 drflac_read_pcm_frames_f32(drflac* pFlac, drflac_uint64 framesToRead, float* pBufferOut);
-DRFLAC_API drflac_bool32 drflac_seek_to_pcm_frame(drflac* pFlac, drflac_uint64 pcmFrameIndex);
-#ifndef DR_FLAC_NO_STDIO
-DRFLAC_API drflac* drflac_open_file(const char* pFileName, const drflac_allocation_callbacks* pAllocationCallbacks);
-DRFLAC_API drflac* drflac_open_file_w(const wchar_t* pFileName, const drflac_allocation_callbacks* pAllocationCallbacks);
-DRFLAC_API drflac* drflac_open_file_with_metadata(const char* pFileName, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
-DRFLAC_API drflac* drflac_open_file_with_metadata_w(const wchar_t* pFileName, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
-#endif
-DRFLAC_API drflac* drflac_open_memory(const void* pData, size_t dataSize, const drflac_allocation_callbacks* pAllocationCallbacks);
-DRFLAC_API drflac* drflac_open_memory_with_metadata(const void* pData, size_t dataSize, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
-DRFLAC_API drflac_int32* drflac_open_and_read_pcm_frames_s32(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
-DRFLAC_API drflac_int16* drflac_open_and_read_pcm_frames_s16(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
-DRFLAC_API float* drflac_open_and_read_pcm_frames_f32(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
-#ifndef DR_FLAC_NO_STDIO
-DRFLAC_API drflac_int32* drflac_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
-DRFLAC_API drflac_int16* drflac_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
-DRFLAC_API float* drflac_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
-#endif
-DRFLAC_API drflac_int32* drflac_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
-DRFLAC_API drflac_int16* drflac_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
-DRFLAC_API float* drflac_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
-DRFLAC_API void drflac_free(void* p, const drflac_allocation_callbacks* pAllocationCallbacks);
-typedef struct
+ return ma_resource_manager_data_stream_get_cursor_in_pcm_frames((ma_resource_manager_data_stream*)pDataSource, pCursor);
+}
+
+static ma_result ma_resource_manager_data_stream_cb__get_length_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pLength)
{
- drflac_uint32 countRemaining;
- const char* pRunningData;
-} drflac_vorbis_comment_iterator;
-DRFLAC_API void drflac_init_vorbis_comment_iterator(drflac_vorbis_comment_iterator* pIter, drflac_uint32 commentCount, const void* pComments);
-DRFLAC_API const char* drflac_next_vorbis_comment(drflac_vorbis_comment_iterator* pIter, drflac_uint32* pCommentLengthOut);
-typedef struct
+ return ma_resource_manager_data_stream_get_length_in_pcm_frames((ma_resource_manager_data_stream*)pDataSource, pLength);
+}
+
+static ma_result ma_resource_manager_data_stream_cb__set_looping(ma_data_source* pDataSource, ma_bool32 isLooping)
{
- drflac_uint32 countRemaining;
- const char* pRunningData;
-} drflac_cuesheet_track_iterator;
-#pragma pack(4)
-typedef struct
+ return ma_resource_manager_data_stream_set_looping((ma_resource_manager_data_stream*)pDataSource, isLooping);
+}
+
+static ma_data_source_vtable g_ma_resource_manager_data_stream_vtable =
{
- drflac_uint64 offset;
- drflac_uint8 index;
- drflac_uint8 reserved[3];
-} drflac_cuesheet_track_index;
-#pragma pack()
-typedef struct
+ ma_resource_manager_data_stream_cb__read_pcm_frames,
+ ma_resource_manager_data_stream_cb__seek_to_pcm_frame,
+ ma_resource_manager_data_stream_cb__get_data_format,
+ ma_resource_manager_data_stream_cb__get_cursor_in_pcm_frames,
+ ma_resource_manager_data_stream_cb__get_length_in_pcm_frames,
+ ma_resource_manager_data_stream_cb__set_looping,
+ MA_DATA_SOURCE_SELF_MANAGED_RANGE_AND_LOOP_POINT
+};
+
+static void ma_resource_manager_data_stream_set_absolute_cursor(ma_resource_manager_data_stream* pDataStream, ma_uint64 absoluteCursor)
{
- drflac_uint64 offset;
- drflac_uint8 trackNumber;
- char ISRC[12];
- drflac_bool8 isAudio;
- drflac_bool8 preEmphasis;
- drflac_uint8 indexCount;
- const drflac_cuesheet_track_index* pIndexPoints;
-} drflac_cuesheet_track;
-DRFLAC_API void drflac_init_cuesheet_track_iterator(drflac_cuesheet_track_iterator* pIter, drflac_uint32 trackCount, const void* pTrackData);
-DRFLAC_API drflac_bool32 drflac_next_cuesheet_track(drflac_cuesheet_track_iterator* pIter, drflac_cuesheet_track* pCuesheetTrack);
-#ifdef __cplusplus
+ /* Loop if possible. */
+ if (absoluteCursor > pDataStream->totalLengthInPCMFrames && pDataStream->totalLengthInPCMFrames > 0) {
+ absoluteCursor = absoluteCursor % pDataStream->totalLengthInPCMFrames;
+ }
+
+ c89atomic_exchange_64(&pDataStream->absoluteCursor, absoluteCursor);
}
-#endif
-#endif
-/* dr_flac_h end */
-#endif /* MA_NO_FLAC */
-#if !defined(MA_NO_MP3) && !defined(MA_NO_DECODING)
-/* dr_mp3_h begin */
-#ifndef dr_mp3_h
-#define dr_mp3_h
-#ifdef __cplusplus
-extern "C" {
-#endif
-#define DRMP3_STRINGIFY(x) #x
-#define DRMP3_XSTRINGIFY(x) DRMP3_STRINGIFY(x)
-#define DRMP3_VERSION_MAJOR 0
-#define DRMP3_VERSION_MINOR 6
-#define DRMP3_VERSION_REVISION 31
-#define DRMP3_VERSION_STRING DRMP3_XSTRINGIFY(DRMP3_VERSION_MAJOR) "." DRMP3_XSTRINGIFY(DRMP3_VERSION_MINOR) "." DRMP3_XSTRINGIFY(DRMP3_VERSION_REVISION)
-#include
-typedef signed char drmp3_int8;
-typedef unsigned char drmp3_uint8;
-typedef signed short drmp3_int16;
-typedef unsigned short drmp3_uint16;
-typedef signed int drmp3_int32;
-typedef unsigned int drmp3_uint32;
-#if defined(_MSC_VER)
- typedef signed __int64 drmp3_int64;
- typedef unsigned __int64 drmp3_uint64;
-#else
- #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
- #pragma GCC diagnostic push
- #pragma GCC diagnostic ignored "-Wlong-long"
- #if defined(__clang__)
- #pragma GCC diagnostic ignored "-Wc++11-long-long"
- #endif
- #endif
- typedef signed long long drmp3_int64;
- typedef unsigned long long drmp3_uint64;
- #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
- #pragma GCC diagnostic pop
- #endif
-#endif
-#if defined(__LP64__) || defined(_WIN64) || (defined(__x86_64__) && !defined(__ILP32__)) || defined(_M_X64) || defined(__ia64) || defined (_M_IA64) || defined(__aarch64__) || defined(_M_ARM64) || defined(__powerpc64__)
- typedef drmp3_uint64 drmp3_uintptr;
-#else
- typedef drmp3_uint32 drmp3_uintptr;
-#endif
-typedef drmp3_uint8 drmp3_bool8;
-typedef drmp3_uint32 drmp3_bool32;
-#define DRMP3_TRUE 1
-#define DRMP3_FALSE 0
-#if !defined(DRMP3_API)
- #if defined(DRMP3_DLL)
- #if defined(_WIN32)
- #define DRMP3_DLL_IMPORT __declspec(dllimport)
- #define DRMP3_DLL_EXPORT __declspec(dllexport)
- #define DRMP3_DLL_PRIVATE static
- #else
- #if defined(__GNUC__) && __GNUC__ >= 4
- #define DRMP3_DLL_IMPORT __attribute__((visibility("default")))
- #define DRMP3_DLL_EXPORT __attribute__((visibility("default")))
- #define DRMP3_DLL_PRIVATE __attribute__((visibility("hidden")))
- #else
- #define DRMP3_DLL_IMPORT
- #define DRMP3_DLL_EXPORT
- #define DRMP3_DLL_PRIVATE static
- #endif
- #endif
- #if defined(DR_MP3_IMPLEMENTATION) || defined(DRMP3_IMPLEMENTATION)
- #define DRMP3_API DRMP3_DLL_EXPORT
- #else
- #define DRMP3_API DRMP3_DLL_IMPORT
- #endif
- #define DRMP3_PRIVATE DRMP3_DLL_PRIVATE
- #else
- #define DRMP3_API extern
- #define DRMP3_PRIVATE static
- #endif
-#endif
-typedef drmp3_int32 drmp3_result;
-#define DRMP3_SUCCESS 0
-#define DRMP3_ERROR -1
-#define DRMP3_INVALID_ARGS -2
-#define DRMP3_INVALID_OPERATION -3
-#define DRMP3_OUT_OF_MEMORY -4
-#define DRMP3_OUT_OF_RANGE -5
-#define DRMP3_ACCESS_DENIED -6
-#define DRMP3_DOES_NOT_EXIST -7
-#define DRMP3_ALREADY_EXISTS -8
-#define DRMP3_TOO_MANY_OPEN_FILES -9
-#define DRMP3_INVALID_FILE -10
-#define DRMP3_TOO_BIG -11
-#define DRMP3_PATH_TOO_LONG -12
-#define DRMP3_NAME_TOO_LONG -13
-#define DRMP3_NOT_DIRECTORY -14
-#define DRMP3_IS_DIRECTORY -15
-#define DRMP3_DIRECTORY_NOT_EMPTY -16
-#define DRMP3_END_OF_FILE -17
-#define DRMP3_NO_SPACE -18
-#define DRMP3_BUSY -19
-#define DRMP3_IO_ERROR -20
-#define DRMP3_INTERRUPT -21
-#define DRMP3_UNAVAILABLE -22
-#define DRMP3_ALREADY_IN_USE -23
-#define DRMP3_BAD_ADDRESS -24
-#define DRMP3_BAD_SEEK -25
-#define DRMP3_BAD_PIPE -26
-#define DRMP3_DEADLOCK -27
-#define DRMP3_TOO_MANY_LINKS -28
-#define DRMP3_NOT_IMPLEMENTED -29
-#define DRMP3_NO_MESSAGE -30
-#define DRMP3_BAD_MESSAGE -31
-#define DRMP3_NO_DATA_AVAILABLE -32
-#define DRMP3_INVALID_DATA -33
-#define DRMP3_TIMEOUT -34
-#define DRMP3_NO_NETWORK -35
-#define DRMP3_NOT_UNIQUE -36
-#define DRMP3_NOT_SOCKET -37
-#define DRMP3_NO_ADDRESS -38
-#define DRMP3_BAD_PROTOCOL -39
-#define DRMP3_PROTOCOL_UNAVAILABLE -40
-#define DRMP3_PROTOCOL_NOT_SUPPORTED -41
-#define DRMP3_PROTOCOL_FAMILY_NOT_SUPPORTED -42
-#define DRMP3_ADDRESS_FAMILY_NOT_SUPPORTED -43
-#define DRMP3_SOCKET_NOT_SUPPORTED -44
-#define DRMP3_CONNECTION_RESET -45
-#define DRMP3_ALREADY_CONNECTED -46
-#define DRMP3_NOT_CONNECTED -47
-#define DRMP3_CONNECTION_REFUSED -48
-#define DRMP3_NO_HOST -49
-#define DRMP3_IN_PROGRESS -50
-#define DRMP3_CANCELLED -51
-#define DRMP3_MEMORY_ALREADY_MAPPED -52
-#define DRMP3_AT_END -53
-#define DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME 1152
-#define DRMP3_MAX_SAMPLES_PER_FRAME (DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME*2)
-#ifdef _MSC_VER
- #define DRMP3_INLINE __forceinline
-#elif defined(__GNUC__)
- #if defined(__STRICT_ANSI__)
- #define DRMP3_INLINE __inline__ __attribute__((always_inline))
- #else
- #define DRMP3_INLINE inline __attribute__((always_inline))
- #endif
-#elif defined(__WATCOMC__)
- #define DRMP3_INLINE __inline
-#else
- #define DRMP3_INLINE
-#endif
-DRMP3_API void drmp3_version(drmp3_uint32* pMajor, drmp3_uint32* pMinor, drmp3_uint32* pRevision);
-DRMP3_API const char* drmp3_version_string(void);
-typedef struct
+MA_API ma_result ma_resource_manager_data_stream_init_ex(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source_config* pConfig, ma_resource_manager_data_stream* pDataStream)
{
- int frame_bytes, channels, hz, layer, bitrate_kbps;
-} drmp3dec_frame_info;
-typedef struct
+ ma_result result;
+ ma_data_source_config dataSourceConfig;
+ char* pFilePathCopy = NULL;
+ wchar_t* pFilePathWCopy = NULL;
+ ma_job job;
+ ma_bool32 waitBeforeReturning = MA_FALSE;
+ ma_resource_manager_inline_notification waitNotification;
+ ma_resource_manager_pipeline_notifications notifications;
+
+ if (pDataStream == NULL) {
+ if (pConfig != NULL && pConfig->pNotifications != NULL) {
+ ma_resource_manager_pipeline_notifications_signal_all_notifications(pConfig->pNotifications);
+ }
+
+ return MA_INVALID_ARGS;
+ }
+
+ MA_ZERO_OBJECT(pDataStream);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (pConfig->pNotifications != NULL) {
+ notifications = *pConfig->pNotifications; /* From here on out, `notifications` should be used instead of `pNotifications`. Setting this to NULL to catch any errors at testing time. */
+ } else {
+ MA_ZERO_OBJECT(¬ifications);
+ }
+
+ dataSourceConfig = ma_data_source_config_init();
+ dataSourceConfig.vtable = &g_ma_resource_manager_data_stream_vtable;
+
+ result = ma_data_source_init(&dataSourceConfig, &pDataStream->ds);
+ if (result != MA_SUCCESS) {
+ ma_resource_manager_pipeline_notifications_signal_all_notifications(¬ifications);
+ return result;
+ }
+
+ pDataStream->pResourceManager = pResourceManager;
+ pDataStream->flags = pConfig->flags;
+ pDataStream->result = MA_BUSY;
+
+ ma_data_source_set_range_in_pcm_frames(pDataStream, pConfig->rangeBegInPCMFrames, pConfig->rangeEndInPCMFrames);
+ ma_data_source_set_loop_point_in_pcm_frames(pDataStream, pConfig->loopPointBegInPCMFrames, pConfig->loopPointEndInPCMFrames);
+ ma_data_source_set_looping(pDataStream, pConfig->isLooping);
+
+ if (pResourceManager == NULL || (pConfig->pFilePath == NULL && pConfig->pFilePathW == NULL)) {
+ ma_resource_manager_pipeline_notifications_signal_all_notifications(¬ifications);
+ return MA_INVALID_ARGS;
+ }
+
+ /* We want all access to the VFS and the internal decoder to happen on the job thread just to keep things easier to manage for the VFS. */
+
+ /* We need a copy of the file path. We should probably make this more efficient, but for now we'll do a transient memory allocation. */
+ if (pConfig->pFilePath != NULL) {
+ pFilePathCopy = ma_copy_string(pConfig->pFilePath, &pResourceManager->config.allocationCallbacks);
+ } else {
+ pFilePathWCopy = ma_copy_string_w(pConfig->pFilePathW, &pResourceManager->config.allocationCallbacks);
+ }
+
+ if (pFilePathCopy == NULL && pFilePathWCopy == NULL) {
+ ma_resource_manager_pipeline_notifications_signal_all_notifications(¬ifications);
+ return MA_OUT_OF_MEMORY;
+ }
+
+ /*
+ We need to check for the presence of MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC. If it's not set, we need to wait before returning. Otherwise we
+ can return immediately. Likewise, we'll also check for MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT and do the same.
+ */
+ if ((pConfig->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC) == 0 || (pConfig->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT) != 0) {
+ waitBeforeReturning = MA_TRUE;
+ ma_resource_manager_inline_notification_init(pResourceManager, &waitNotification);
+ }
+
+ ma_resource_manager_pipeline_notifications_acquire_all_fences(¬ifications);
+
+ /* Set the absolute cursor to our initial seek position so retrieval of the cursor returns a good value. */
+ ma_resource_manager_data_stream_set_absolute_cursor(pDataStream, pConfig->initialSeekPointInPCMFrames);
+
+ /* We now have everything we need to post the job. This is the last thing we need to do from here. The rest will be done by the job thread. */
+ job = ma_job_init(MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_STREAM);
+ job.order = ma_resource_manager_data_stream_next_execution_order(pDataStream);
+ job.data.resourceManager.loadDataStream.pDataStream = pDataStream;
+ job.data.resourceManager.loadDataStream.pFilePath = pFilePathCopy;
+ job.data.resourceManager.loadDataStream.pFilePathW = pFilePathWCopy;
+ job.data.resourceManager.loadDataStream.initialSeekPoint = pConfig->initialSeekPointInPCMFrames;
+ job.data.resourceManager.loadDataStream.pInitNotification = (waitBeforeReturning == MA_TRUE) ? &waitNotification : notifications.init.pNotification;
+ job.data.resourceManager.loadDataStream.pInitFence = notifications.init.pFence;
+ result = ma_resource_manager_post_job(pResourceManager, &job);
+ if (result != MA_SUCCESS) {
+ ma_resource_manager_pipeline_notifications_signal_all_notifications(¬ifications);
+ ma_resource_manager_pipeline_notifications_release_all_fences(¬ifications);
+
+ if (waitBeforeReturning) {
+ ma_resource_manager_inline_notification_uninit(&waitNotification);
+ }
+
+ ma_free(pFilePathCopy, &pResourceManager->config.allocationCallbacks);
+ ma_free(pFilePathWCopy, &pResourceManager->config.allocationCallbacks);
+ return result;
+ }
+
+ /* Wait if needed. */
+ if (waitBeforeReturning) {
+ ma_resource_manager_inline_notification_wait_and_uninit(&waitNotification);
+
+ if (notifications.init.pNotification != NULL) {
+ ma_async_notification_signal(notifications.init.pNotification);
+ }
+
+ /* NOTE: Do not release pInitFence here. That will be done by the job. */
+ }
+
+ return MA_SUCCESS;
+}
+
+MA_API ma_result ma_resource_manager_data_stream_init(ma_resource_manager* pResourceManager, const char* pFilePath, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_stream* pDataStream)
{
- float mdct_overlap[2][9*32], qmf_state[15*2*32];
- int reserv, free_format_bytes;
- drmp3_uint8 header[4], reserv_buf[511];
-} drmp3dec;
-DRMP3_API void drmp3dec_init(drmp3dec *dec);
-DRMP3_API int drmp3dec_decode_frame(drmp3dec *dec, const drmp3_uint8 *mp3, int mp3_bytes, void *pcm, drmp3dec_frame_info *info);
-DRMP3_API void drmp3dec_f32_to_s16(const float *in, drmp3_int16 *out, size_t num_samples);
-typedef enum
+ ma_resource_manager_data_source_config config;
+
+ config = ma_resource_manager_data_source_config_init();
+ config.pFilePath = pFilePath;
+ config.flags = flags;
+ config.pNotifications = pNotifications;
+
+ return ma_resource_manager_data_stream_init_ex(pResourceManager, &config, pDataStream);
+}
+
+MA_API ma_result ma_resource_manager_data_stream_init_w(ma_resource_manager* pResourceManager, const wchar_t* pFilePath, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_stream* pDataStream)
{
- drmp3_seek_origin_start,
- drmp3_seek_origin_current
-} drmp3_seek_origin;
-typedef struct
+ ma_resource_manager_data_source_config config;
+
+ config = ma_resource_manager_data_source_config_init();
+ config.pFilePathW = pFilePath;
+ config.flags = flags;
+ config.pNotifications = pNotifications;
+
+ return ma_resource_manager_data_stream_init_ex(pResourceManager, &config, pDataStream);
+}
+
+MA_API ma_result ma_resource_manager_data_stream_uninit(ma_resource_manager_data_stream* pDataStream)
{
- drmp3_uint64 seekPosInBytes;
- drmp3_uint64 pcmFrameIndex;
- drmp3_uint16 mp3FramesToDiscard;
- drmp3_uint16 pcmFramesToDiscard;
-} drmp3_seek_point;
-typedef size_t (* drmp3_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead);
-typedef drmp3_bool32 (* drmp3_seek_proc)(void* pUserData, int offset, drmp3_seek_origin origin);
-typedef struct
+ ma_resource_manager_inline_notification freeEvent;
+ ma_job job;
+
+ if (pDataStream == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ /* The first thing to do is set the result to unavailable. This will prevent future page decoding. */
+ c89atomic_exchange_i32(&pDataStream->result, MA_UNAVAILABLE);
+
+ /*
+ We need to post a job to ensure we're not in the middle or decoding or anything. Because the object is owned by the caller, we'll need
+ to wait for it to complete before returning which means we need an event.
+ */
+ ma_resource_manager_inline_notification_init(pDataStream->pResourceManager, &freeEvent);
+
+ job = ma_job_init(MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_STREAM);
+ job.order = ma_resource_manager_data_stream_next_execution_order(pDataStream);
+ job.data.resourceManager.freeDataStream.pDataStream = pDataStream;
+ job.data.resourceManager.freeDataStream.pDoneNotification = &freeEvent;
+ job.data.resourceManager.freeDataStream.pDoneFence = NULL;
+ ma_resource_manager_post_job(pDataStream->pResourceManager, &job);
+
+ /* We need to wait for the job to finish processing before we return. */
+ ma_resource_manager_inline_notification_wait_and_uninit(&freeEvent);
+
+ return MA_SUCCESS;
+}
+
+
+static ma_uint32 ma_resource_manager_data_stream_get_page_size_in_frames(ma_resource_manager_data_stream* pDataStream)
{
- void* pUserData;
- void* (* onMalloc)(size_t sz, void* pUserData);
- void* (* onRealloc)(void* p, size_t sz, void* pUserData);
- void (* onFree)(void* p, void* pUserData);
-} drmp3_allocation_callbacks;
-typedef struct
+ MA_ASSERT(pDataStream != NULL);
+ MA_ASSERT(pDataStream->isDecoderInitialized == MA_TRUE);
+
+ return MA_RESOURCE_MANAGER_PAGE_SIZE_IN_MILLISECONDS * (pDataStream->decoder.outputSampleRate/1000);
+}
+
+static void* ma_resource_manager_data_stream_get_page_data_pointer(ma_resource_manager_data_stream* pDataStream, ma_uint32 pageIndex, ma_uint32 relativeCursor)
{
- drmp3_uint32 channels;
- drmp3_uint32 sampleRate;
-} drmp3_config;
-typedef struct
+ MA_ASSERT(pDataStream != NULL);
+ MA_ASSERT(pDataStream->isDecoderInitialized == MA_TRUE);
+ MA_ASSERT(pageIndex == 0 || pageIndex == 1);
+
+ return ma_offset_ptr(pDataStream->pPageData, ((ma_resource_manager_data_stream_get_page_size_in_frames(pDataStream) * pageIndex) + relativeCursor) * ma_get_bytes_per_frame(pDataStream->decoder.outputFormat, pDataStream->decoder.outputChannels));
+}
+
+static void ma_resource_manager_data_stream_fill_page(ma_resource_manager_data_stream* pDataStream, ma_uint32 pageIndex)
{
- drmp3dec decoder;
- drmp3dec_frame_info frameInfo;
- drmp3_uint32 channels;
- drmp3_uint32 sampleRate;
- drmp3_read_proc onRead;
- drmp3_seek_proc onSeek;
- void* pUserData;
- drmp3_allocation_callbacks allocationCallbacks;
- drmp3_uint32 mp3FrameChannels;
- drmp3_uint32 mp3FrameSampleRate;
- drmp3_uint32 pcmFramesConsumedInMP3Frame;
- drmp3_uint32 pcmFramesRemainingInMP3Frame;
- drmp3_uint8 pcmFrames[sizeof(float)*DRMP3_MAX_SAMPLES_PER_FRAME];
- drmp3_uint64 currentPCMFrame;
- drmp3_uint64 streamCursor;
- drmp3_seek_point* pSeekPoints;
- drmp3_uint32 seekPointCount;
- size_t dataSize;
- size_t dataCapacity;
- size_t dataConsumed;
- drmp3_uint8* pData;
- drmp3_bool32 atEnd : 1;
- struct
+ ma_result result = MA_SUCCESS;
+ ma_uint64 pageSizeInFrames;
+ ma_uint64 totalFramesReadForThisPage = 0;
+ void* pPageData = ma_resource_manager_data_stream_get_page_data_pointer(pDataStream, pageIndex, 0);
+
+ pageSizeInFrames = ma_resource_manager_data_stream_get_page_size_in_frames(pDataStream);
+
+ /* The decoder needs to inherit the stream's looping and range state. */
{
- const drmp3_uint8* pData;
- size_t dataSize;
- size_t currentReadPos;
- } memory;
-} drmp3;
-DRMP3_API drmp3_bool32 drmp3_init(drmp3* pMP3, drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, const drmp3_allocation_callbacks* pAllocationCallbacks);
-DRMP3_API drmp3_bool32 drmp3_init_memory(drmp3* pMP3, const void* pData, size_t dataSize, const drmp3_allocation_callbacks* pAllocationCallbacks);
-#ifndef DR_MP3_NO_STDIO
-DRMP3_API drmp3_bool32 drmp3_init_file(drmp3* pMP3, const char* pFilePath, const drmp3_allocation_callbacks* pAllocationCallbacks);
-DRMP3_API drmp3_bool32 drmp3_init_file_w(drmp3* pMP3, const wchar_t* pFilePath, const drmp3_allocation_callbacks* pAllocationCallbacks);
-#endif
-DRMP3_API void drmp3_uninit(drmp3* pMP3);
-DRMP3_API drmp3_uint64 drmp3_read_pcm_frames_f32(drmp3* pMP3, drmp3_uint64 framesToRead, float* pBufferOut);
-DRMP3_API drmp3_uint64 drmp3_read_pcm_frames_s16(drmp3* pMP3, drmp3_uint64 framesToRead, drmp3_int16* pBufferOut);
-DRMP3_API drmp3_bool32 drmp3_seek_to_pcm_frame(drmp3* pMP3, drmp3_uint64 frameIndex);
-DRMP3_API drmp3_uint64 drmp3_get_pcm_frame_count(drmp3* pMP3);
-DRMP3_API drmp3_uint64 drmp3_get_mp3_frame_count(drmp3* pMP3);
-DRMP3_API drmp3_bool32 drmp3_get_mp3_and_pcm_frame_count(drmp3* pMP3, drmp3_uint64* pMP3FrameCount, drmp3_uint64* pPCMFrameCount);
-DRMP3_API drmp3_bool32 drmp3_calculate_seek_points(drmp3* pMP3, drmp3_uint32* pSeekPointCount, drmp3_seek_point* pSeekPoints);
-DRMP3_API drmp3_bool32 drmp3_bind_seek_table(drmp3* pMP3, drmp3_uint32 seekPointCount, drmp3_seek_point* pSeekPoints);
-DRMP3_API float* drmp3_open_and_read_pcm_frames_f32(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks);
-DRMP3_API drmp3_int16* drmp3_open_and_read_pcm_frames_s16(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks);
-DRMP3_API float* drmp3_open_memory_and_read_pcm_frames_f32(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks);
-DRMP3_API drmp3_int16* drmp3_open_memory_and_read_pcm_frames_s16(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks);
-#ifndef DR_MP3_NO_STDIO
-DRMP3_API float* drmp3_open_file_and_read_pcm_frames_f32(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks);
-DRMP3_API drmp3_int16* drmp3_open_file_and_read_pcm_frames_s16(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks);
-#endif
-DRMP3_API void* drmp3_malloc(size_t sz, const drmp3_allocation_callbacks* pAllocationCallbacks);
-DRMP3_API void drmp3_free(void* p, const drmp3_allocation_callbacks* pAllocationCallbacks);
-#ifdef __cplusplus
+ ma_uint64 rangeBeg;
+ ma_uint64 rangeEnd;
+ ma_uint64 loopPointBeg;
+ ma_uint64 loopPointEnd;
+
+ ma_data_source_set_looping(&pDataStream->decoder, ma_resource_manager_data_stream_is_looping(pDataStream));
+
+ ma_data_source_get_range_in_pcm_frames(pDataStream, &rangeBeg, &rangeEnd);
+ ma_data_source_set_range_in_pcm_frames(&pDataStream->decoder, rangeBeg, rangeEnd);
+
+ ma_data_source_get_loop_point_in_pcm_frames(pDataStream, &loopPointBeg, &loopPointEnd);
+ ma_data_source_set_loop_point_in_pcm_frames(&pDataStream->decoder, loopPointBeg, loopPointEnd);
+ }
+
+ /* Just read straight from the decoder. It will deal with ranges and looping for us. */
+ result = ma_data_source_read_pcm_frames(&pDataStream->decoder, pPageData, pageSizeInFrames, &totalFramesReadForThisPage);
+ if (result == MA_AT_END || totalFramesReadForThisPage < pageSizeInFrames) {
+ c89atomic_exchange_32(&pDataStream->isDecoderAtEnd, MA_TRUE);
+ }
+
+ c89atomic_exchange_32(&pDataStream->pageFrameCount[pageIndex], (ma_uint32)totalFramesReadForThisPage);
+ c89atomic_exchange_32(&pDataStream->isPageValid[pageIndex], MA_TRUE);
+}
+
+static void ma_resource_manager_data_stream_fill_pages(ma_resource_manager_data_stream* pDataStream)
+{
+ ma_uint32 iPage;
+
+ MA_ASSERT(pDataStream != NULL);
+
+ for (iPage = 0; iPage < 2; iPage += 1) {
+ ma_resource_manager_data_stream_fill_page(pDataStream, iPage);
+ }
+}
+
+
+static ma_result ma_resource_manager_data_stream_map(ma_resource_manager_data_stream* pDataStream, void** ppFramesOut, ma_uint64* pFrameCount)
+{
+ ma_uint64 framesAvailable;
+ ma_uint64 frameCount = 0;
+
+ /* We cannot be using the data source after it's been uninitialized. */
+ MA_ASSERT(ma_resource_manager_data_stream_result(pDataStream) != MA_UNAVAILABLE);
+
+ if (pFrameCount != NULL) {
+ frameCount = *pFrameCount;
+ *pFrameCount = 0;
+ }
+ if (ppFramesOut != NULL) {
+ *ppFramesOut = NULL;
+ }
+
+ if (pDataStream == NULL || ppFramesOut == NULL || pFrameCount == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (ma_resource_manager_data_stream_result(pDataStream) != MA_SUCCESS) {
+ return MA_INVALID_OPERATION;
+ }
+
+ /* Don't attempt to read while we're in the middle of seeking. Tell the caller that we're busy. */
+ if (ma_resource_manager_data_stream_seek_counter(pDataStream) > 0) {
+ return MA_BUSY;
+ }
+
+ /* If the page we're on is invalid it means we've caught up to the job thread. */
+ if (c89atomic_load_32(&pDataStream->isPageValid[pDataStream->currentPageIndex]) == MA_FALSE) {
+ framesAvailable = 0;
+ } else {
+ /*
+ The page we're on is valid so we must have some frames available. We need to make sure that we don't overflow into the next page, even if it's valid. The reason is
+ that the unmap process will only post an update for one page at a time. Keeping mapping tied to page boundaries makes this simpler.
+ */
+ ma_uint32 currentPageFrameCount = c89atomic_load_32(&pDataStream->pageFrameCount[pDataStream->currentPageIndex]);
+ MA_ASSERT(currentPageFrameCount >= pDataStream->relativeCursor);
+
+ framesAvailable = currentPageFrameCount - pDataStream->relativeCursor;
+ }
+
+ /* If there's no frames available and the result is set to MA_AT_END we need to return MA_AT_END. */
+ if (framesAvailable == 0) {
+ if (ma_resource_manager_data_stream_is_decoder_at_end(pDataStream)) {
+ return MA_AT_END;
+ } else {
+ return MA_BUSY; /* There are no frames available, but we're not marked as EOF so we might have caught up to the job thread. Need to return MA_BUSY and wait for more data. */
+ }
+ }
+
+ MA_ASSERT(framesAvailable > 0);
+
+ if (frameCount > framesAvailable) {
+ frameCount = framesAvailable;
+ }
+
+ *ppFramesOut = ma_resource_manager_data_stream_get_page_data_pointer(pDataStream, pDataStream->currentPageIndex, pDataStream->relativeCursor);
+ *pFrameCount = frameCount;
+
+ return MA_SUCCESS;
+}
+
+static ma_result ma_resource_manager_data_stream_unmap(ma_resource_manager_data_stream* pDataStream, ma_uint64 frameCount)
+{
+ ma_uint32 newRelativeCursor;
+ ma_uint32 pageSizeInFrames;
+ ma_job job;
+
+ /* We cannot be using the data source after it's been uninitialized. */
+ MA_ASSERT(ma_resource_manager_data_stream_result(pDataStream) != MA_UNAVAILABLE);
+
+ if (pDataStream == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (ma_resource_manager_data_stream_result(pDataStream) != MA_SUCCESS) {
+ return MA_INVALID_OPERATION;
+ }
+
+ /* The frame count should always fit inside a 32-bit integer. */
+ if (frameCount > 0xFFFFFFFF) {
+ return MA_INVALID_ARGS;
+ }
+
+ pageSizeInFrames = ma_resource_manager_data_stream_get_page_size_in_frames(pDataStream);
+
+ /* The absolute cursor needs to be updated for ma_resource_manager_data_stream_get_cursor_in_pcm_frames(). */
+ ma_resource_manager_data_stream_set_absolute_cursor(pDataStream, c89atomic_load_64(&pDataStream->absoluteCursor) + frameCount);
+
+ /* Here is where we need to check if we need to load a new page, and if so, post a job to load it. */
+ newRelativeCursor = pDataStream->relativeCursor + (ma_uint32)frameCount;
+
+ /* If the new cursor has flowed over to the next page we need to mark the old one as invalid and post an event for it. */
+ if (newRelativeCursor >= pageSizeInFrames) {
+ newRelativeCursor -= pageSizeInFrames;
+
+ /* Here is where we post the job start decoding. */
+ job = ma_job_init(MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_STREAM);
+ job.order = ma_resource_manager_data_stream_next_execution_order(pDataStream);
+ job.data.resourceManager.pageDataStream.pDataStream = pDataStream;
+ job.data.resourceManager.pageDataStream.pageIndex = pDataStream->currentPageIndex;
+
+ /* The page needs to be marked as invalid so that the public API doesn't try reading from it. */
+ c89atomic_exchange_32(&pDataStream->isPageValid[pDataStream->currentPageIndex], MA_FALSE);
+
+ /* Before posting the job we need to make sure we set some state. */
+ pDataStream->relativeCursor = newRelativeCursor;
+ pDataStream->currentPageIndex = (pDataStream->currentPageIndex + 1) & 0x01;
+ return ma_resource_manager_post_job(pDataStream->pResourceManager, &job);
+ } else {
+ /* We haven't moved into a new page so we can just move the cursor forward. */
+ pDataStream->relativeCursor = newRelativeCursor;
+ return MA_SUCCESS;
+ }
+}
+
+
+MA_API ma_result ma_resource_manager_data_stream_read_pcm_frames(ma_resource_manager_data_stream* pDataStream, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+{
+ ma_result result = MA_SUCCESS;
+ ma_uint64 totalFramesProcessed;
+ ma_format format;
+ ma_uint32 channels;
+
+ /* Safety. */
+ if (pFramesRead != NULL) {
+ *pFramesRead = 0;
+ }
+
+ if (frameCount == 0) {
+ return MA_INVALID_ARGS;
+ }
+
+ /* We cannot be using the data source after it's been uninitialized. */
+ MA_ASSERT(ma_resource_manager_data_stream_result(pDataStream) != MA_UNAVAILABLE);
+
+ if (pDataStream == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (ma_resource_manager_data_stream_result(pDataStream) != MA_SUCCESS) {
+ return MA_INVALID_OPERATION;
+ }
+
+ /* Don't attempt to read while we're in the middle of seeking. Tell the caller that we're busy. */
+ if (ma_resource_manager_data_stream_seek_counter(pDataStream) > 0) {
+ return MA_BUSY;
+ }
+
+ ma_resource_manager_data_stream_get_data_format(pDataStream, &format, &channels, NULL, NULL, 0);
+
+ /* Reading is implemented in terms of map/unmap. We need to run this in a loop because mapping is clamped against page boundaries. */
+ totalFramesProcessed = 0;
+ while (totalFramesProcessed < frameCount) {
+ void* pMappedFrames;
+ ma_uint64 mappedFrameCount;
+
+ mappedFrameCount = frameCount - totalFramesProcessed;
+ result = ma_resource_manager_data_stream_map(pDataStream, &pMappedFrames, &mappedFrameCount);
+ if (result != MA_SUCCESS) {
+ break;
+ }
+
+ /* Copy the mapped data to the output buffer if we have one. It's allowed for pFramesOut to be NULL in which case a relative forward seek is performed. */
+ if (pFramesOut != NULL) {
+ ma_copy_pcm_frames(ma_offset_pcm_frames_ptr(pFramesOut, totalFramesProcessed, format, channels), pMappedFrames, mappedFrameCount, format, channels);
+ }
+
+ totalFramesProcessed += mappedFrameCount;
+
+ result = ma_resource_manager_data_stream_unmap(pDataStream, mappedFrameCount);
+ if (result != MA_SUCCESS) {
+ break; /* This is really bad - will only get an error here if we failed to post a job to the queue for loading the next page. */
+ }
+ }
+
+ if (pFramesRead != NULL) {
+ *pFramesRead = totalFramesProcessed;
+ }
+
+ if (result == MA_SUCCESS && totalFramesProcessed == 0) {
+ result = MA_AT_END;
+ }
+
+ return result;
}
-#endif
-#endif
-/* dr_mp3_h end */
-#endif /* MA_NO_MP3 */
+MA_API ma_result ma_resource_manager_data_stream_seek_to_pcm_frame(ma_resource_manager_data_stream* pDataStream, ma_uint64 frameIndex)
+{
+ ma_job job;
+ ma_result streamResult;
-/**************************************************************************************************************************************************************
+ streamResult = ma_resource_manager_data_stream_result(pDataStream);
-Decoding
+ /* We cannot be using the data source after it's been uninitialized. */
+ MA_ASSERT(streamResult != MA_UNAVAILABLE);
-**************************************************************************************************************************************************************/
-#ifndef MA_NO_DECODING
+ if (pDataStream == NULL) {
+ return MA_INVALID_ARGS;
+ }
-static ma_result ma_decoder_read_bytes(ma_decoder* pDecoder, void* pBufferOut, size_t bytesToRead, size_t* pBytesRead)
+ if (streamResult != MA_SUCCESS && streamResult != MA_BUSY) {
+ return MA_INVALID_OPERATION;
+ }
+
+ /* Increment the seek counter first to indicate to read_paged_pcm_frames() and map_paged_pcm_frames() that we are in the middle of a seek and MA_BUSY should be returned. */
+ c89atomic_fetch_add_32(&pDataStream->seekCounter, 1);
+
+ /* Update the absolute cursor so that ma_resource_manager_data_stream_get_cursor_in_pcm_frames() returns the new position. */
+ ma_resource_manager_data_stream_set_absolute_cursor(pDataStream, frameIndex);
+
+ /*
+ We need to clear our currently loaded pages so that the stream starts playback from the new seek point as soon as possible. These are for the purpose of the public
+ API and will be ignored by the seek job. The seek job will operate on the assumption that both pages have been marked as invalid and the cursor is at the start of
+ the first page.
+ */
+ pDataStream->relativeCursor = 0;
+ pDataStream->currentPageIndex = 0;
+ c89atomic_exchange_32(&pDataStream->isPageValid[0], MA_FALSE);
+ c89atomic_exchange_32(&pDataStream->isPageValid[1], MA_FALSE);
+
+ /* Make sure the data stream is not marked as at the end or else if we seek in response to hitting the end, we won't be able to read any more data. */
+ c89atomic_exchange_32(&pDataStream->isDecoderAtEnd, MA_FALSE);
+
+ /*
+ The public API is not allowed to touch the internal decoder so we need to use a job to perform the seek. When seeking, the job thread will assume both pages
+ are invalid and any content contained within them will be discarded and replaced with newly decoded data.
+ */
+ job = ma_job_init(MA_JOB_TYPE_RESOURCE_MANAGER_SEEK_DATA_STREAM);
+ job.order = ma_resource_manager_data_stream_next_execution_order(pDataStream);
+ job.data.resourceManager.seekDataStream.pDataStream = pDataStream;
+ job.data.resourceManager.seekDataStream.frameIndex = frameIndex;
+ return ma_resource_manager_post_job(pDataStream->pResourceManager, &job);
+}
+
+MA_API ma_result ma_resource_manager_data_stream_get_data_format(ma_resource_manager_data_stream* pDataStream, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
- size_t bytesRead;
+ /* We cannot be using the data source after it's been uninitialized. */
+ MA_ASSERT(ma_resource_manager_data_stream_result(pDataStream) != MA_UNAVAILABLE);
- MA_ASSERT(pDecoder != NULL);
- MA_ASSERT(pBufferOut != NULL);
- MA_ASSERT(bytesToRead > 0); /* It's an error to call this with a byte count of zero. */
+ if (pFormat != NULL) {
+ *pFormat = ma_format_unknown;
+ }
- bytesRead = pDecoder->onRead(pDecoder, pBufferOut, bytesToRead);
+ if (pChannels != NULL) {
+ *pChannels = 0;
+ }
- if (pBytesRead != NULL) {
- *pBytesRead = bytesRead;
+ if (pSampleRate != NULL) {
+ *pSampleRate = 0;
}
- if (bytesRead == 0) {
- return MA_AT_END;
+ if (pChannelMap != NULL) {
+ MA_ZERO_MEMORY(pChannelMap, sizeof(*pChannelMap) * channelMapCap);
}
- return MA_SUCCESS;
+ if (pDataStream == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (ma_resource_manager_data_stream_result(pDataStream) != MA_SUCCESS) {
+ return MA_INVALID_OPERATION;
+ }
+
+ /*
+ We're being a little bit naughty here and accessing the internal decoder from the public API. The output data format is constant, and we've defined this function
+ such that the application is responsible for ensuring it's not called while uninitializing so it should be safe.
+ */
+ return ma_data_source_get_data_format(&pDataStream->decoder, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
}
-static ma_result ma_decoder_seek_bytes(ma_decoder* pDecoder, ma_int64 byteOffset, ma_seek_origin origin)
+MA_API ma_result ma_resource_manager_data_stream_get_cursor_in_pcm_frames(ma_resource_manager_data_stream* pDataStream, ma_uint64* pCursor)
{
- ma_bool32 wasSuccessful;
+ ma_result result;
- MA_ASSERT(pDecoder != NULL);
+ if (pCursor == NULL) {
+ return MA_INVALID_ARGS;
+ }
- wasSuccessful = pDecoder->onSeek(pDecoder, byteOffset, origin);
- if (wasSuccessful) {
- return MA_SUCCESS;
- } else {
- return MA_ERROR;
+ *pCursor = 0;
+
+ /* We cannot be using the data source after it's been uninitialized. */
+ MA_ASSERT(ma_resource_manager_data_stream_result(pDataStream) != MA_UNAVAILABLE);
+
+ if (pDataStream == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ /*
+ If the stream is in an erroneous state we need to return an invalid operation. We can allow
+ this to be called when the data stream is in a busy state because the caller may have asked
+ for an initial seek position and it's convenient to return that as the cursor position.
+ */
+ result = ma_resource_manager_data_stream_result(pDataStream);
+ if (result != MA_SUCCESS && result != MA_BUSY) {
+ return MA_INVALID_OPERATION;
}
+
+ *pCursor = c89atomic_load_64(&pDataStream->absoluteCursor);
+
+ return MA_SUCCESS;
}
-static ma_result ma_decoder_tell_bytes(ma_decoder* pDecoder, ma_int64* pCursor)
+MA_API ma_result ma_resource_manager_data_stream_get_length_in_pcm_frames(ma_resource_manager_data_stream* pDataStream, ma_uint64* pLength)
{
- MA_ASSERT(pDecoder != NULL);
-
- if (pDecoder->onTell == NULL) {
- return MA_NOT_IMPLEMENTED;
+ ma_result streamResult;
+
+ if (pLength == NULL) {
+ return MA_INVALID_ARGS;
}
- return pDecoder->onTell(pDecoder, pCursor);
+ *pLength = 0;
+
+ streamResult = ma_resource_manager_data_stream_result(pDataStream);
+
+ /* We cannot be using the data source after it's been uninitialized. */
+ MA_ASSERT(streamResult != MA_UNAVAILABLE);
+
+ if (pDataStream == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (streamResult != MA_SUCCESS) {
+ return streamResult;
+ }
+
+ /*
+ We most definitely do not want to be calling ma_decoder_get_length_in_pcm_frames() directly. Instead we want to use a cached value that we
+ calculated when we initialized it on the job thread.
+ */
+ *pLength = pDataStream->totalLengthInPCMFrames;
+ if (*pLength == 0) {
+ return MA_NOT_IMPLEMENTED; /* Some decoders may not have a known length. */
+ }
+
+ return MA_SUCCESS;
}
+MA_API ma_result ma_resource_manager_data_stream_result(const ma_resource_manager_data_stream* pDataStream)
+{
+ if (pDataStream == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ return (ma_result)c89atomic_load_i32(&pDataStream->result);
+}
-MA_API ma_decoding_backend_config ma_decoding_backend_config_init(ma_format preferredFormat)
+MA_API ma_result ma_resource_manager_data_stream_set_looping(ma_resource_manager_data_stream* pDataStream, ma_bool32 isLooping)
{
- ma_decoding_backend_config config;
+ if (pDataStream == NULL) {
+ return MA_INVALID_ARGS;
+ }
- MA_ZERO_OBJECT(&config);
- config.preferredFormat = preferredFormat;
+ c89atomic_exchange_32(&pDataStream->isLooping, isLooping);
- return config;
+ return MA_SUCCESS;
}
+MA_API ma_bool32 ma_resource_manager_data_stream_is_looping(const ma_resource_manager_data_stream* pDataStream)
+{
+ if (pDataStream == NULL) {
+ return MA_FALSE;
+ }
+
+ return c89atomic_load_32((ma_bool32*)&pDataStream->isLooping); /* Naughty const-cast. Value won't change from here in practice (maybe from another thread). */
+}
-MA_API ma_decoder_config ma_decoder_config_init(ma_format outputFormat, ma_uint32 outputChannels, ma_uint32 outputSampleRate)
+MA_API ma_result ma_resource_manager_data_stream_get_available_frames(ma_resource_manager_data_stream* pDataStream, ma_uint64* pAvailableFrames)
{
- ma_decoder_config config;
- MA_ZERO_OBJECT(&config);
- config.format = outputFormat;
- config.channels = ma_min(outputChannels, ma_countof(config.channelMap));
- config.sampleRate = outputSampleRate;
- config.resampling.algorithm = ma_resample_algorithm_linear;
- config.resampling.linear.lpfOrder = ma_min(MA_DEFAULT_RESAMPLER_LPF_ORDER, MA_MAX_FILTER_ORDER);
- config.resampling.speex.quality = 3;
- config.encodingFormat = ma_encoding_format_unknown;
+ ma_uint32 pageIndex0;
+ ma_uint32 pageIndex1;
+ ma_uint32 relativeCursor;
+ ma_uint64 availableFrames;
- /* Note that we are intentionally leaving the channel map empty here which will cause the default channel map to be used. */
+ if (pAvailableFrames == NULL) {
+ return MA_INVALID_ARGS;
+ }
- return config;
+ *pAvailableFrames = 0;
+
+ if (pDataStream == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ pageIndex0 = pDataStream->currentPageIndex;
+ pageIndex1 = (pDataStream->currentPageIndex + 1) & 0x01;
+ relativeCursor = pDataStream->relativeCursor;
+
+ availableFrames = 0;
+ if (c89atomic_load_32(&pDataStream->isPageValid[pageIndex0])) {
+ availableFrames += c89atomic_load_32(&pDataStream->pageFrameCount[pageIndex0]) - relativeCursor;
+ if (c89atomic_load_32(&pDataStream->isPageValid[pageIndex1])) {
+ availableFrames += c89atomic_load_32(&pDataStream->pageFrameCount[pageIndex1]);
+ }
+ }
+
+ *pAvailableFrames = availableFrames;
+ return MA_SUCCESS;
}
-MA_API ma_decoder_config ma_decoder_config_init_default()
+
+static ma_result ma_resource_manager_data_source_preinit(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source_config* pConfig, ma_resource_manager_data_source* pDataSource)
{
- return ma_decoder_config_init(ma_format_unknown, 0, 0);
+ if (pDataSource == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ MA_ZERO_OBJECT(pDataSource);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (pResourceManager == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ pDataSource->flags = pConfig->flags;
+
+ return MA_SUCCESS;
}
-MA_API ma_decoder_config ma_decoder_config_init_copy(const ma_decoder_config* pConfig)
+MA_API ma_result ma_resource_manager_data_source_init_ex(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source_config* pConfig, ma_resource_manager_data_source* pDataSource)
{
- ma_decoder_config config;
- if (pConfig != NULL) {
- config = *pConfig;
+ ma_result result;
+
+ result = ma_resource_manager_data_source_preinit(pResourceManager, pConfig, pDataSource);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ /* The data source itself is just a data stream or a data buffer. */
+ if ((pConfig->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
+ return ma_resource_manager_data_stream_init_ex(pResourceManager, pConfig, &pDataSource->backend.stream);
} else {
- MA_ZERO_OBJECT(&config);
+ return ma_resource_manager_data_buffer_init_ex(pResourceManager, pConfig, &pDataSource->backend.buffer);
}
+}
- return config;
+MA_API ma_result ma_resource_manager_data_source_init(ma_resource_manager* pResourceManager, const char* pName, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_source* pDataSource)
+{
+ ma_resource_manager_data_source_config config;
+
+ config = ma_resource_manager_data_source_config_init();
+ config.pFilePath = pName;
+ config.flags = flags;
+ config.pNotifications = pNotifications;
+
+ return ma_resource_manager_data_source_init_ex(pResourceManager, &config, pDataSource);
}
-static ma_result ma_decoder__init_data_converter(ma_decoder* pDecoder, const ma_decoder_config* pConfig)
+MA_API ma_result ma_resource_manager_data_source_init_w(ma_resource_manager* pResourceManager, const wchar_t* pName, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_source* pDataSource)
+{
+ ma_resource_manager_data_source_config config;
+
+ config = ma_resource_manager_data_source_config_init();
+ config.pFilePathW = pName;
+ config.flags = flags;
+ config.pNotifications = pNotifications;
+
+ return ma_resource_manager_data_source_init_ex(pResourceManager, &config, pDataSource);
+}
+
+MA_API ma_result ma_resource_manager_data_source_init_copy(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source* pExistingDataSource, ma_resource_manager_data_source* pDataSource)
{
ma_result result;
- ma_data_converter_config converterConfig;
- ma_format internalFormat;
- ma_uint32 internalChannels;
- ma_uint32 internalSampleRate;
- ma_channel internalChannelMap[MA_MAX_CHANNELS];
+ ma_resource_manager_data_source_config config;
- MA_ASSERT(pDecoder != NULL);
- MA_ASSERT(pConfig != NULL);
+ if (pExistingDataSource == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ config = ma_resource_manager_data_source_config_init();
+ config.flags = pExistingDataSource->flags;
- result = ma_data_source_get_data_format(pDecoder->pBackend, &internalFormat, &internalChannels, &internalSampleRate);
+ result = ma_resource_manager_data_source_preinit(pResourceManager, &config, pDataSource);
if (result != MA_SUCCESS) {
- return result; /* Failed to retrieve the internal data format. */
+ return result;
}
- /* Channel map needs to be retrieved separately. */
- if (pDecoder->pBackendVTable != NULL && pDecoder->pBackendVTable->onGetChannelMap != NULL) {
- pDecoder->pBackendVTable->onGetChannelMap(pDecoder->pBackendUserData, pDecoder->pBackend, internalChannelMap, ma_countof(internalChannelMap));
- } else {
- ma_get_standard_channel_map(ma_standard_channel_map_default, ma_min(internalChannels, ma_countof(internalChannelMap)), internalChannelMap);
+ /* Copying can only be done from data buffers. Streams cannot be copied. */
+ if ((pExistingDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
+ return MA_INVALID_OPERATION;
}
+ return ma_resource_manager_data_buffer_init_copy(pResourceManager, &pExistingDataSource->backend.buffer, &pDataSource->backend.buffer);
+}
-
- /* Make sure we're not asking for too many channels. */
- if (pConfig->channels > MA_MAX_CHANNELS) {
+MA_API ma_result ma_resource_manager_data_source_uninit(ma_resource_manager_data_source* pDataSource)
+{
+ if (pDataSource == NULL) {
return MA_INVALID_ARGS;
}
- /* The internal channels should have already been validated at a higher level, but we'll do it again explicitly here for safety. */
- if (internalChannels > MA_MAX_CHANNELS) {
- return MA_INVALID_ARGS;
+ /* All we need to is uninitialize the underlying data buffer or data stream. */
+ if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
+ return ma_resource_manager_data_stream_uninit(&pDataSource->backend.stream);
+ } else {
+ return ma_resource_manager_data_buffer_uninit(&pDataSource->backend.buffer);
}
+}
+MA_API ma_result ma_resource_manager_data_source_read_pcm_frames(ma_resource_manager_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+{
+ /* Safety. */
+ if (pFramesRead != NULL) {
+ *pFramesRead = 0;
+ }
- /* Output format. */
- if (pConfig->format == ma_format_unknown) {
- pDecoder->outputFormat = internalFormat;
- } else {
- pDecoder->outputFormat = pConfig->format;
+ if (pDataSource == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pConfig->channels == 0) {
- pDecoder->outputChannels = internalChannels;
+ if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
+ return ma_resource_manager_data_stream_read_pcm_frames(&pDataSource->backend.stream, pFramesOut, frameCount, pFramesRead);
} else {
- pDecoder->outputChannels = pConfig->channels;
+ return ma_resource_manager_data_buffer_read_pcm_frames(&pDataSource->backend.buffer, pFramesOut, frameCount, pFramesRead);
}
+}
- if (pConfig->sampleRate == 0) {
- pDecoder->outputSampleRate = internalSampleRate;
+MA_API ma_result ma_resource_manager_data_source_seek_to_pcm_frame(ma_resource_manager_data_source* pDataSource, ma_uint64 frameIndex)
+{
+ if (pDataSource == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
+ return ma_resource_manager_data_stream_seek_to_pcm_frame(&pDataSource->backend.stream, frameIndex);
} else {
- pDecoder->outputSampleRate = pConfig->sampleRate;
+ return ma_resource_manager_data_buffer_seek_to_pcm_frame(&pDataSource->backend.buffer, frameIndex);
+ }
+}
+
+MA_API ma_result ma_resource_manager_data_source_map(ma_resource_manager_data_source* pDataSource, void** ppFramesOut, ma_uint64* pFrameCount)
+{
+ if (pDataSource == NULL) {
+ return MA_INVALID_ARGS;
}
- if (ma_channel_map_blank(pDecoder->outputChannels, pConfig->channelMap)) {
- ma_get_standard_channel_map(ma_standard_channel_map_default, pDecoder->outputChannels, pDecoder->outputChannelMap);
+ if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
+ return ma_resource_manager_data_stream_map(&pDataSource->backend.stream, ppFramesOut, pFrameCount);
} else {
- MA_COPY_MEMORY(pDecoder->outputChannelMap, pConfig->channelMap, sizeof(pConfig->channelMap));
+ return MA_NOT_IMPLEMENTED; /* Mapping not supported with data buffers. */
+ }
+}
+
+MA_API ma_result ma_resource_manager_data_source_unmap(ma_resource_manager_data_source* pDataSource, ma_uint64 frameCount)
+{
+ if (pDataSource == NULL) {
+ return MA_INVALID_ARGS;
}
+ if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
+ return ma_resource_manager_data_stream_unmap(&pDataSource->backend.stream, frameCount);
+ } else {
+ return MA_NOT_IMPLEMENTED; /* Mapping not supported with data buffers. */
+ }
+}
- converterConfig = ma_data_converter_config_init(
- internalFormat, pDecoder->outputFormat,
- internalChannels, pDecoder->outputChannels,
- internalSampleRate, pDecoder->outputSampleRate
- );
- ma_channel_map_copy(converterConfig.channelMapIn, internalChannelMap, internalChannels);
- ma_channel_map_copy(converterConfig.channelMapOut, pDecoder->outputChannelMap, pDecoder->outputChannels);
- converterConfig.channelMixMode = pConfig->channelMixMode;
- converterConfig.ditherMode = pConfig->ditherMode;
- converterConfig.resampling.allowDynamicSampleRate = MA_FALSE; /* Never allow dynamic sample rate conversion. Setting this to true will disable passthrough optimizations. */
- converterConfig.resampling.algorithm = pConfig->resampling.algorithm;
- converterConfig.resampling.linear.lpfOrder = pConfig->resampling.linear.lpfOrder;
- converterConfig.resampling.speex.quality = pConfig->resampling.speex.quality;
+MA_API ma_result ma_resource_manager_data_source_get_data_format(ma_resource_manager_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
+{
+ if (pDataSource == NULL) {
+ return MA_INVALID_ARGS;
+ }
- return ma_data_converter_init(&converterConfig, &pDecoder->converter);
+ if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
+ return ma_resource_manager_data_stream_get_data_format(&pDataSource->backend.stream, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
+ } else {
+ return ma_resource_manager_data_buffer_get_data_format(&pDataSource->backend.buffer, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
+ }
}
+MA_API ma_result ma_resource_manager_data_source_get_cursor_in_pcm_frames(ma_resource_manager_data_source* pDataSource, ma_uint64* pCursor)
+{
+ if (pDataSource == NULL) {
+ return MA_INVALID_ARGS;
+ }
+ if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
+ return ma_resource_manager_data_stream_get_cursor_in_pcm_frames(&pDataSource->backend.stream, pCursor);
+ } else {
+ return ma_resource_manager_data_buffer_get_cursor_in_pcm_frames(&pDataSource->backend.buffer, pCursor);
+ }
+}
-static ma_result ma_decoder_internal_on_read__custom(void* pUserData, void* pBufferOut, size_t bytesToRead, size_t* pBytesRead)
+MA_API ma_result ma_resource_manager_data_source_get_length_in_pcm_frames(ma_resource_manager_data_source* pDataSource, ma_uint64* pLength)
{
- ma_decoder* pDecoder = (ma_decoder*)pUserData;
- MA_ASSERT(pDecoder != NULL);
+ if (pDataSource == NULL) {
+ return MA_INVALID_ARGS;
+ }
- return ma_decoder_read_bytes(pDecoder, pBufferOut, bytesToRead, pBytesRead);
+ if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
+ return ma_resource_manager_data_stream_get_length_in_pcm_frames(&pDataSource->backend.stream, pLength);
+ } else {
+ return ma_resource_manager_data_buffer_get_length_in_pcm_frames(&pDataSource->backend.buffer, pLength);
+ }
}
-static ma_result ma_decoder_internal_on_seek__custom(void* pUserData, ma_int64 offset, ma_seek_origin origin)
+MA_API ma_result ma_resource_manager_data_source_result(const ma_resource_manager_data_source* pDataSource)
{
- ma_decoder* pDecoder = (ma_decoder*)pUserData;
- MA_ASSERT(pDecoder != NULL);
+ if (pDataSource == NULL) {
+ return MA_INVALID_ARGS;
+ }
- return ma_decoder_seek_bytes(pDecoder, offset, origin);
+ if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
+ return ma_resource_manager_data_stream_result(&pDataSource->backend.stream);
+ } else {
+ return ma_resource_manager_data_buffer_result(&pDataSource->backend.buffer);
+ }
}
-static ma_result ma_decoder_internal_on_tell__custom(void* pUserData, ma_int64* pCursor)
+MA_API ma_result ma_resource_manager_data_source_set_looping(ma_resource_manager_data_source* pDataSource, ma_bool32 isLooping)
{
- ma_decoder* pDecoder = (ma_decoder*)pUserData;
- MA_ASSERT(pDecoder != NULL);
+ if (pDataSource == NULL) {
+ return MA_INVALID_ARGS;
+ }
- return ma_decoder_tell_bytes(pDecoder, pCursor);
+ if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
+ return ma_resource_manager_data_stream_set_looping(&pDataSource->backend.stream, isLooping);
+ } else {
+ return ma_resource_manager_data_buffer_set_looping(&pDataSource->backend.buffer, isLooping);
+ }
}
+MA_API ma_bool32 ma_resource_manager_data_source_is_looping(const ma_resource_manager_data_source* pDataSource)
+{
+ if (pDataSource == NULL) {
+ return MA_FALSE;
+ }
-static ma_result ma_decoder_init_from_vtable(const ma_decoding_backend_vtable* pVTable, void* pVTableUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+ if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
+ return ma_resource_manager_data_stream_is_looping(&pDataSource->backend.stream);
+ } else {
+ return ma_resource_manager_data_buffer_is_looping(&pDataSource->backend.buffer);
+ }
+}
+
+MA_API ma_result ma_resource_manager_data_source_get_available_frames(ma_resource_manager_data_source* pDataSource, ma_uint64* pAvailableFrames)
{
- ma_result result;
- ma_decoding_backend_config backendConfig;
- ma_data_source* pBackend;
+ if (pAvailableFrames == NULL) {
+ return MA_INVALID_ARGS;
+ }
- MA_ASSERT(pVTable != NULL);
- MA_ASSERT(pConfig != NULL);
- MA_ASSERT(pDecoder != NULL);
+ *pAvailableFrames = 0;
- if (pVTable->onInit == NULL) {
- return MA_NOT_IMPLEMENTED;
+ if (pDataSource == NULL) {
+ return MA_INVALID_ARGS;
}
- backendConfig = ma_decoding_backend_config_init(pConfig->format);
+ if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
+ return ma_resource_manager_data_stream_get_available_frames(&pDataSource->backend.stream, pAvailableFrames);
+ } else {
+ return ma_resource_manager_data_buffer_get_available_frames(&pDataSource->backend.buffer, pAvailableFrames);
+ }
+}
- result = pVTable->onInit(pVTableUserData, ma_decoder_internal_on_read__custom, ma_decoder_internal_on_seek__custom, ma_decoder_internal_on_tell__custom, pDecoder, &backendConfig, &pDecoder->allocationCallbacks, &pBackend);
- if (result != MA_SUCCESS) {
- return result; /* Failed to initialize the backend from this vtable. */
+
+MA_API ma_result ma_resource_manager_post_job(ma_resource_manager* pResourceManager, const ma_job* pJob)
+{
+ if (pResourceManager == NULL) {
+ return MA_INVALID_ARGS;
}
- /* Getting here means we were able to initialize the backend so we can now initialize the decoder. */
- pDecoder->pBackend = pBackend;
- pDecoder->pBackendVTable = pVTable;
- pDecoder->pBackendUserData = pConfig->pCustomBackendUserData;
+ return ma_job_queue_post(&pResourceManager->jobQueue, pJob);
+}
- return MA_SUCCESS;
+MA_API ma_result ma_resource_manager_post_job_quit(ma_resource_manager* pResourceManager)
+{
+ ma_job job = ma_job_init(MA_JOB_TYPE_QUIT);
+ return ma_resource_manager_post_job(pResourceManager, &job);
}
+MA_API ma_result ma_resource_manager_next_job(ma_resource_manager* pResourceManager, ma_job* pJob)
+{
+ if (pResourceManager == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ return ma_job_queue_next(&pResourceManager->jobQueue, pJob);
+}
-static ma_result ma_decoder_init_custom__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+static ma_result ma_job_process__resource_manager__load_data_buffer_node(ma_job* pJob)
{
- ma_result result = MA_NO_BACKEND;
- size_t ivtable;
+ ma_result result = MA_SUCCESS;
+ ma_resource_manager* pResourceManager;
+ ma_resource_manager_data_buffer_node* pDataBufferNode;
- MA_ASSERT(pConfig != NULL);
- MA_ASSERT(pDecoder != NULL);
+ MA_ASSERT(pJob != NULL);
- if (pConfig->ppCustomBackendVTables == NULL) {
- return MA_NO_BACKEND;
+ pResourceManager = (ma_resource_manager*)pJob->data.resourceManager.loadDataBufferNode.pResourceManager;
+ MA_ASSERT(pResourceManager != NULL);
+
+ pDataBufferNode = (ma_resource_manager_data_buffer_node*)pJob->data.resourceManager.loadDataBufferNode.pDataBufferNode;
+ MA_ASSERT(pDataBufferNode != NULL);
+ MA_ASSERT(pDataBufferNode->isDataOwnedByResourceManager == MA_TRUE); /* The data should always be owned by the resource manager. */
+
+ /* The data buffer is not getting deleted, but we may be getting executed out of order. If so, we need to push the job back onto the queue and return. */
+ if (pJob->order != c89atomic_load_32(&pDataBufferNode->executionPointer)) {
+ return ma_resource_manager_post_job(pResourceManager, pJob); /* Attempting to execute out of order. Probably interleaved with a MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_BUFFER job. */
}
- /* The order each backend is listed is what defines the priority. */
- for (ivtable = 0; ivtable < pConfig->customBackendCount; ivtable += 1) {
- const ma_decoding_backend_vtable* pVTable = pConfig->ppCustomBackendVTables[ivtable];
- if (pVTable != NULL && pVTable->onInit != NULL) {
- result = ma_decoder_init_from_vtable(pVTable, pConfig->pCustomBackendUserData, pConfig, pDecoder);
- if (result == MA_SUCCESS) {
- return MA_SUCCESS;
+ /* First thing we need to do is check whether or not the data buffer is getting deleted. If so we just abort. */
+ if (ma_resource_manager_data_buffer_node_result(pDataBufferNode) != MA_BUSY) {
+ result = ma_resource_manager_data_buffer_node_result(pDataBufferNode); /* The data buffer may be getting deleted before it's even been loaded. */
+ goto done;
+ }
+
+ /*
+ We're ready to start loading. Essentially what we're doing here is initializing the data supply
+ of the node. Once this is complete, data buffers can have their connectors initialized which
+ will allow then to have audio data read from them.
+
+ Note that when the data supply type has been moved away from "unknown", that is when other threads
+ will determine that the node is available for data delivery and the data buffer connectors can be
+ initialized. Therefore, it's important that it is set after the data supply has been initialized.
+ */
+ if (pJob->data.resourceManager.loadDataBufferNode.decode) {
+ /*
+ Decoding. This is the complex case because we're not going to be doing the entire decoding
+ process here. Instead it's going to be split of multiple jobs and loaded in pages. The
+ reason for this is to evenly distribute decoding time across multiple sounds, rather than
+ having one huge sound hog all the available processing resources.
+
+ The first thing we do is initialize a decoder. This is allocated on the heap and is passed
+ around to the paging jobs. When the last paging job has completed it's processing, it'll
+ free the decoder for us.
+
+ This job does not do any actual decoding. It instead just posts a PAGE_DATA_BUFFER_NODE job
+ which is where the actual decoding work will be done. However, once this job is complete,
+ the node will be in a state where data buffer connectors can be initialized.
+ */
+ ma_decoder* pDecoder; /* <-- Free'd on the last page decode. */
+ ma_job pageDataBufferNodeJob;
+
+ /* Allocate the decoder by initializing a decoded data supply. */
+ result = ma_resource_manager_data_buffer_node_init_supply_decoded(pResourceManager, pDataBufferNode, pJob->data.resourceManager.loadDataBufferNode.pFilePath, pJob->data.resourceManager.loadDataBufferNode.pFilePathW, &pDecoder);
+
+ /*
+ Don't ever propagate an MA_BUSY result code or else the resource manager will think the
+ node is just busy decoding rather than in an error state. This should never happen, but
+ including this logic for safety just in case.
+ */
+ if (result == MA_BUSY) {
+ result = MA_ERROR;
+ }
+
+ if (result != MA_SUCCESS) {
+ if (pJob->data.resourceManager.loadDataBufferNode.pFilePath != NULL) {
+ ma_log_postf(ma_resource_manager_get_log(pResourceManager), MA_LOG_LEVEL_WARNING, "Failed to initialize data supply for \"%s\". %s.\n", pJob->data.resourceManager.loadDataBufferNode.pFilePath, ma_result_description(result));
} else {
- /* Initialization failed. Move on to the next one, but seek back to the start first so the next vtable starts from the first byte of the file. */
- result = ma_decoder_seek_bytes(pDecoder, 0, ma_seek_origin_start);
- if (result != MA_SUCCESS) {
- return result; /* Failed to seek back to the start. */
- }
+ #if (defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L) || defined(_MSC_VER)
+ ma_log_postf(ma_resource_manager_get_log(pResourceManager), MA_LOG_LEVEL_WARNING, "Failed to initialize data supply for \"%ls\", %s.\n", pJob->data.resourceManager.loadDataBufferNode.pFilePathW, ma_result_description(result));
+ #endif
}
+
+ goto done;
+ }
+
+ /*
+ At this point the node's data supply is initialized and other threads can start initializing
+ their data buffer connectors. However, no data will actually be available until we start to
+ actually decode it. To do this, we need to post a paging job which is where the decoding
+ work is done.
+
+ Note that if an error occurred at an earlier point, this section will have been skipped.
+ */
+ pageDataBufferNodeJob = ma_job_init(MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_BUFFER_NODE);
+ pageDataBufferNodeJob.order = ma_resource_manager_data_buffer_node_next_execution_order(pDataBufferNode);
+ pageDataBufferNodeJob.data.resourceManager.pageDataBufferNode.pResourceManager = pResourceManager;
+ pageDataBufferNodeJob.data.resourceManager.pageDataBufferNode.pDataBufferNode = pDataBufferNode;
+ pageDataBufferNodeJob.data.resourceManager.pageDataBufferNode.pDecoder = pDecoder;
+ pageDataBufferNodeJob.data.resourceManager.pageDataBufferNode.pDoneNotification = pJob->data.resourceManager.loadDataBufferNode.pDoneNotification;
+ pageDataBufferNodeJob.data.resourceManager.pageDataBufferNode.pDoneFence = pJob->data.resourceManager.loadDataBufferNode.pDoneFence;
+
+ /* The job has been set up so it can now be posted. */
+ result = ma_resource_manager_post_job(pResourceManager, &pageDataBufferNodeJob);
+
+ /*
+ When we get here, we want to make sure the result code is set to MA_BUSY. The reason for
+ this is that the result will be copied over to the node's internal result variable. In
+ this case, since the decoding is still in-progress, we need to make sure the result code
+ is set to MA_BUSY.
+ */
+ if (result != MA_SUCCESS) {
+ ma_log_postf(ma_resource_manager_get_log(pResourceManager), MA_LOG_LEVEL_ERROR, "Failed to post MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_BUFFER_NODE job. %s\n", ma_result_description(result));
+ ma_decoder_uninit(pDecoder);
+ ma_free(pDecoder, &pResourceManager->config.allocationCallbacks);
} else {
- /* No vtable. */
+ result = MA_BUSY;
}
+ } else {
+ /* No decoding. This is the simple case. We need only read the file content into memory and we're done. */
+ result = ma_resource_manager_data_buffer_node_init_supply_encoded(pResourceManager, pDataBufferNode, pJob->data.resourceManager.loadDataBufferNode.pFilePath, pJob->data.resourceManager.loadDataBufferNode.pFilePathW);
}
- /* Getting here means we couldn't find a backend. */
- return MA_NO_BACKEND;
-}
-
-/* WAV */
-#ifdef dr_wav_h
-#define MA_HAS_WAV
+done:
+ /* File paths are no longer needed. */
+ ma_free(pJob->data.resourceManager.loadDataBufferNode.pFilePath, &pResourceManager->config.allocationCallbacks);
+ ma_free(pJob->data.resourceManager.loadDataBufferNode.pFilePathW, &pResourceManager->config.allocationCallbacks);
-typedef struct
-{
- ma_data_source_base ds;
- ma_read_proc onRead;
- ma_seek_proc onSeek;
- ma_tell_proc onTell;
- void* pReadSeekTellUserData;
- ma_format format; /* Can be f32, s16 or s32. */
-#if !defined(MA_NO_WAV)
- drwav dr;
-#endif
-} ma_wav;
+ /*
+ We need to set the result to at the very end to ensure no other threads try reading the data before we've fully initialized the object. Other threads
+ are going to be inspecting this variable to determine whether or not they're ready to read data. We can only change the result if it's set to MA_BUSY
+ because otherwise we may be changing away from an error code which would be bad. An example is if the application creates a data buffer, but then
+ immediately deletes it before we've got to this point. In this case, pDataBuffer->result will be MA_UNAVAILABLE, and setting it to MA_SUCCESS or any
+ other error code would cause the buffer to look like it's in a state that it's not.
+ */
+ c89atomic_compare_and_swap_i32(&pDataBufferNode->result, MA_BUSY, result);
-MA_API ma_result ma_wav_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav);
-MA_API ma_result ma_wav_init_file(const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav);
-MA_API ma_result ma_wav_init_file_w(const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav);
-MA_API ma_result ma_wav_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav);
-MA_API void ma_wav_uninit(ma_wav* pWav, const ma_allocation_callbacks* pAllocationCallbacks);
-MA_API ma_result ma_wav_read_pcm_frames(ma_wav* pWav, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
-MA_API ma_result ma_wav_seek_to_pcm_frame(ma_wav* pWav, ma_uint64 frameIndex);
-MA_API ma_result ma_wav_get_data_format(ma_wav* pWav, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
-MA_API ma_result ma_wav_get_cursor_in_pcm_frames(ma_wav* pWav, ma_uint64* pCursor);
-MA_API ma_result ma_wav_get_length_in_pcm_frames(ma_wav* pWav, ma_uint64* pLength);
+ /* At this point initialization is complete and we can signal the notification if any. */
+ if (pJob->data.resourceManager.loadDataBufferNode.pInitNotification != NULL) {
+ ma_async_notification_signal(pJob->data.resourceManager.loadDataBufferNode.pInitNotification);
+ }
+ if (pJob->data.resourceManager.loadDataBufferNode.pInitFence != NULL) {
+ ma_fence_release(pJob->data.resourceManager.loadDataBufferNode.pInitFence);
+ }
+ /* If we have a success result it means we've fully loaded the buffer. This will happen in the non-decoding case. */
+ if (result != MA_BUSY) {
+ if (pJob->data.resourceManager.loadDataBufferNode.pDoneNotification != NULL) {
+ ma_async_notification_signal(pJob->data.resourceManager.loadDataBufferNode.pDoneNotification);
+ }
+ if (pJob->data.resourceManager.loadDataBufferNode.pDoneFence != NULL) {
+ ma_fence_release(pJob->data.resourceManager.loadDataBufferNode.pDoneFence);
+ }
+ }
-static ma_result ma_wav_ds_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
-{
- return ma_wav_read_pcm_frames((ma_wav*)pDataSource, pFramesOut, frameCount, pFramesRead);
+ /* Increment the node's execution pointer so that the next jobs can be processed. This is how we keep decoding of pages in-order. */
+ c89atomic_fetch_add_32(&pDataBufferNode->executionPointer, 1);
+ return result;
}
-static ma_result ma_wav_ds_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
+static ma_result ma_job_process__resource_manager__free_data_buffer_node(ma_job* pJob)
{
- return ma_wav_seek_to_pcm_frame((ma_wav*)pDataSource, frameIndex);
-}
+ ma_resource_manager* pResourceManager;
+ ma_resource_manager_data_buffer_node* pDataBufferNode;
-static ma_result ma_wav_ds_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate)
-{
- return ma_wav_get_data_format((ma_wav*)pDataSource, pFormat, pChannels, pSampleRate, NULL, 0);
-}
+ MA_ASSERT(pJob != NULL);
-static ma_result ma_wav_ds_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
-{
- return ma_wav_get_cursor_in_pcm_frames((ma_wav*)pDataSource, pCursor);
+ pResourceManager = (ma_resource_manager*)pJob->data.resourceManager.freeDataBufferNode.pResourceManager;
+ MA_ASSERT(pResourceManager != NULL);
+
+ pDataBufferNode = (ma_resource_manager_data_buffer_node*)pJob->data.resourceManager.freeDataBufferNode.pDataBufferNode;
+ MA_ASSERT(pDataBufferNode != NULL);
+
+ if (pJob->order != c89atomic_load_32(&pDataBufferNode->executionPointer)) {
+ return ma_resource_manager_post_job(pResourceManager, pJob); /* Out of order. */
+ }
+
+ ma_resource_manager_data_buffer_node_free(pResourceManager, pDataBufferNode);
+
+ /* The event needs to be signalled last. */
+ if (pJob->data.resourceManager.freeDataBufferNode.pDoneNotification != NULL) {
+ ma_async_notification_signal(pJob->data.resourceManager.freeDataBufferNode.pDoneNotification);
+ }
+
+ if (pJob->data.resourceManager.freeDataBufferNode.pDoneFence != NULL) {
+ ma_fence_release(pJob->data.resourceManager.freeDataBufferNode.pDoneFence);
+ }
+
+ c89atomic_fetch_add_32(&pDataBufferNode->executionPointer, 1);
+ return MA_SUCCESS;
}
-static ma_result ma_wav_ds_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
+static ma_result ma_job_process__resource_manager__page_data_buffer_node(ma_job* pJob)
{
- return ma_wav_get_length_in_pcm_frames((ma_wav*)pDataSource, pLength);
+ ma_result result = MA_SUCCESS;
+ ma_resource_manager* pResourceManager;
+ ma_resource_manager_data_buffer_node* pDataBufferNode;
+
+ MA_ASSERT(pJob != NULL);
+
+ pResourceManager = (ma_resource_manager*)pJob->data.resourceManager.pageDataBufferNode.pResourceManager;
+ MA_ASSERT(pResourceManager != NULL);
+
+ pDataBufferNode = (ma_resource_manager_data_buffer_node*)pJob->data.resourceManager.pageDataBufferNode.pDataBufferNode;
+ MA_ASSERT(pDataBufferNode != NULL);
+
+ if (pJob->order != c89atomic_load_32(&pDataBufferNode->executionPointer)) {
+ return ma_resource_manager_post_job(pResourceManager, pJob); /* Out of order. */
+ }
+
+ /* Don't do any more decoding if the data buffer has started the uninitialization process. */
+ result = ma_resource_manager_data_buffer_node_result(pDataBufferNode);
+ if (result != MA_BUSY) {
+ goto done;
+ }
+
+ /* We're ready to decode the next page. */
+ result = ma_resource_manager_data_buffer_node_decode_next_page(pResourceManager, pDataBufferNode, (ma_decoder*)pJob->data.resourceManager.pageDataBufferNode.pDecoder);
+
+ /*
+ If we have a success code by this point, we want to post another job. We're going to set the
+ result back to MA_BUSY to make it clear that there's still more to load.
+ */
+ if (result == MA_SUCCESS) {
+ ma_job newJob;
+ newJob = *pJob; /* Everything is the same as the input job, except the execution order. */
+ newJob.order = ma_resource_manager_data_buffer_node_next_execution_order(pDataBufferNode); /* We need a fresh execution order. */
+
+ result = ma_resource_manager_post_job(pResourceManager, &newJob);
+
+ /* Since the sound isn't yet fully decoded we want the status to be set to busy. */
+ if (result == MA_SUCCESS) {
+ result = MA_BUSY;
+ }
+ }
+
+done:
+ /* If there's still more to decode the result will be set to MA_BUSY. Otherwise we can free the decoder. */
+ if (result != MA_BUSY) {
+ ma_decoder_uninit((ma_decoder*)pJob->data.resourceManager.pageDataBufferNode.pDecoder);
+ ma_free(pJob->data.resourceManager.pageDataBufferNode.pDecoder, &pResourceManager->config.allocationCallbacks);
+ }
+
+ /* If we reached the end we need to treat it as successful. */
+ if (result == MA_AT_END) {
+ result = MA_SUCCESS;
+ }
+
+ /* Make sure we set the result of node in case some error occurred. */
+ c89atomic_compare_and_swap_i32(&pDataBufferNode->result, MA_BUSY, result);
+
+ /* Signal the notification after setting the result in case the notification callback wants to inspect the result code. */
+ if (result != MA_BUSY) {
+ if (pJob->data.resourceManager.pageDataBufferNode.pDoneNotification != NULL) {
+ ma_async_notification_signal(pJob->data.resourceManager.pageDataBufferNode.pDoneNotification);
+ }
+
+ if (pJob->data.resourceManager.pageDataBufferNode.pDoneFence != NULL) {
+ ma_fence_release(pJob->data.resourceManager.pageDataBufferNode.pDoneFence);
+ }
+ }
+
+ c89atomic_fetch_add_32(&pDataBufferNode->executionPointer, 1);
+ return result;
}
-static ma_data_source_vtable g_ma_wav_ds_vtable =
+
+static ma_result ma_job_process__resource_manager__load_data_buffer(ma_job* pJob)
{
- ma_wav_ds_read,
- ma_wav_ds_seek,
- NULL, /* onMap() */
- NULL, /* onUnmap() */
- ma_wav_ds_get_data_format,
- ma_wav_ds_get_cursor,
- ma_wav_ds_get_length
-};
+ ma_result result = MA_SUCCESS;
+ ma_resource_manager* pResourceManager;
+ ma_resource_manager_data_buffer* pDataBuffer;
+ ma_resource_manager_data_supply_type dataSupplyType = ma_resource_manager_data_supply_type_unknown;
+ ma_bool32 isConnectorInitialized = MA_FALSE;
+ /*
+ All we're doing here is checking if the node has finished loading. If not, we just re-post the job
+ and keep waiting. Otherwise we increment the execution counter and set the buffer's result code.
+ */
+ MA_ASSERT(pJob != NULL);
-#if !defined(MA_NO_WAV)
-static drwav_allocation_callbacks drwav_allocation_callbacks_from_miniaudio(const ma_allocation_callbacks* pAllocationCallbacks)
-{
- drwav_allocation_callbacks callbacks;
+ pDataBuffer = (ma_resource_manager_data_buffer*)pJob->data.resourceManager.loadDataBuffer.pDataBuffer;
+ MA_ASSERT(pDataBuffer != NULL);
- if (pAllocationCallbacks != NULL) {
- callbacks.onMalloc = pAllocationCallbacks->onMalloc;
- callbacks.onRealloc = pAllocationCallbacks->onRealloc;
- callbacks.onFree = pAllocationCallbacks->onFree;
- callbacks.pUserData = pAllocationCallbacks->pUserData;
+ pResourceManager = pDataBuffer->pResourceManager;
+
+ if (pJob->order != c89atomic_load_32(&pDataBuffer->executionPointer)) {
+ return ma_resource_manager_post_job(pResourceManager, pJob); /* Attempting to execute out of order. Probably interleaved with a MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_BUFFER job. */
+ }
+
+ /*
+ First thing we need to do is check whether or not the data buffer is getting deleted. If so we
+ just abort, but making sure we increment the execution pointer.
+ */
+ result = ma_resource_manager_data_buffer_result(pDataBuffer);
+ if (result != MA_BUSY) {
+ goto done; /* <-- This will ensure the exucution pointer is incremented. */
} else {
- callbacks.onMalloc = ma__malloc_default;
- callbacks.onRealloc = ma__realloc_default;
- callbacks.onFree = ma__free_default;
- callbacks.pUserData = NULL;
+ result = MA_SUCCESS; /* <-- Make sure this is reset. */
}
- return callbacks;
-}
+ /* Try initializing the connector if we haven't already. */
+ isConnectorInitialized = pDataBuffer->isConnectorInitialized;
+ if (isConnectorInitialized == MA_FALSE) {
+ dataSupplyType = ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode);
-static size_t ma_wav_dr_callback__read(void* pUserData, void* pBufferOut, size_t bytesToRead)
-{
- ma_wav* pWav = (ma_wav*)pUserData;
- ma_result result;
- size_t bytesRead;
+ if (dataSupplyType != ma_resource_manager_data_supply_type_unknown) {
+ /* We can now initialize the connector. If this fails, we need to abort. It's very rare for this to fail. */
+ result = ma_resource_manager_data_buffer_init_connector(pDataBuffer, pJob->data.resourceManager.loadDataBuffer.pInitNotification, pJob->data.resourceManager.loadDataBuffer.pInitFence);
+ if (result != MA_SUCCESS) {
+ ma_log_postf(ma_resource_manager_get_log(pResourceManager), MA_LOG_LEVEL_ERROR, "Failed to initialize connector for data buffer. %s.\n", ma_result_description(result));
+ goto done;
+ }
+ } else {
+ /* Don't have a known data supply type. Most likely the data buffer node is still loading, but it could be that an error occurred. */
+ }
+ } else {
+ /* The connector is already initialized. Nothing to do here. */
+ }
- MA_ASSERT(pWav != NULL);
+ /*
+ If the data node is still loading, we need to repost the job and *not* increment the execution
+ pointer (i.e. we need to not fall through to the "done" label).
- result = pWav->onRead(pWav->pReadSeekTellUserData, pBufferOut, bytesToRead, &bytesRead);
- (void)result;
+ There is a hole between here and the where the data connector is initialized where the data
+ buffer node may have finished initializing. We need to check for this by checking the result of
+ the data buffer node and whether or not we had an unknown data supply type at the time of
+ trying to initialize the data connector.
+ */
+ result = ma_resource_manager_data_buffer_node_result(pDataBuffer->pNode);
+ if (result == MA_BUSY || (result == MA_SUCCESS && isConnectorInitialized == MA_FALSE && dataSupplyType == ma_resource_manager_data_supply_type_unknown)) {
+ return ma_resource_manager_post_job(pResourceManager, pJob);
+ }
- return bytesRead;
+done:
+ /* Only move away from a busy code so that we don't trash any existing error codes. */
+ c89atomic_compare_and_swap_i32(&pDataBuffer->result, MA_BUSY, result);
+
+ /* Only signal the other threads after the result has been set just for cleanliness sake. */
+ if (pJob->data.resourceManager.loadDataBuffer.pDoneNotification != NULL) {
+ ma_async_notification_signal(pJob->data.resourceManager.loadDataBuffer.pDoneNotification);
+ }
+ if (pJob->data.resourceManager.loadDataBuffer.pDoneFence != NULL) {
+ ma_fence_release(pJob->data.resourceManager.loadDataBuffer.pDoneFence);
+ }
+
+ /*
+ If at this point the data buffer has not had it's connector initialized, it means the
+ notification event was never signalled which means we need to signal it here.
+ */
+ if (pDataBuffer->isConnectorInitialized == MA_FALSE && result != MA_SUCCESS) {
+ if (pJob->data.resourceManager.loadDataBuffer.pInitNotification != NULL) {
+ ma_async_notification_signal(pJob->data.resourceManager.loadDataBuffer.pInitNotification);
+ }
+ if (pJob->data.resourceManager.loadDataBuffer.pInitFence != NULL) {
+ ma_fence_release(pJob->data.resourceManager.loadDataBuffer.pInitFence);
+ }
+ }
+
+ c89atomic_fetch_add_32(&pDataBuffer->executionPointer, 1);
+ return result;
}
-static drwav_bool32 ma_wav_dr_callback__seek(void* pUserData, int offset, drwav_seek_origin origin)
+static ma_result ma_job_process__resource_manager__free_data_buffer(ma_job* pJob)
{
- ma_wav* pWav = (ma_wav*)pUserData;
- ma_result result;
- ma_seek_origin maSeekOrigin;
+ ma_resource_manager* pResourceManager;
+ ma_resource_manager_data_buffer* pDataBuffer;
- MA_ASSERT(pWav != NULL);
+ MA_ASSERT(pJob != NULL);
- maSeekOrigin = ma_seek_origin_start;
- if (origin == drwav_seek_origin_current) {
- maSeekOrigin = ma_seek_origin_current;
+ pDataBuffer = (ma_resource_manager_data_buffer*)pJob->data.resourceManager.freeDataBuffer.pDataBuffer;
+ MA_ASSERT(pDataBuffer != NULL);
+
+ pResourceManager = pDataBuffer->pResourceManager;
+
+ if (pJob->order != c89atomic_load_32(&pDataBuffer->executionPointer)) {
+ return ma_resource_manager_post_job(pResourceManager, pJob); /* Out of order. */
}
- result = pWav->onSeek(pWav->pReadSeekTellUserData, offset, maSeekOrigin);
- if (result != MA_SUCCESS) {
- return MA_FALSE;
+ ma_resource_manager_data_buffer_uninit_internal(pDataBuffer);
+
+ /* The event needs to be signalled last. */
+ if (pJob->data.resourceManager.freeDataBuffer.pDoneNotification != NULL) {
+ ma_async_notification_signal(pJob->data.resourceManager.freeDataBuffer.pDoneNotification);
}
- return MA_TRUE;
+ if (pJob->data.resourceManager.freeDataBuffer.pDoneFence != NULL) {
+ ma_fence_release(pJob->data.resourceManager.freeDataBuffer.pDoneFence);
+ }
+
+ c89atomic_fetch_add_32(&pDataBuffer->executionPointer, 1);
+ return MA_SUCCESS;
}
-#endif
-static ma_result ma_wav_init_internal(const ma_decoding_backend_config* pConfig, ma_wav* pWav)
+static ma_result ma_job_process__resource_manager__load_data_stream(ma_job* pJob)
{
- ma_result result;
- ma_data_source_config dataSourceConfig;
+ ma_result result = MA_SUCCESS;
+ ma_decoder_config decoderConfig;
+ ma_uint32 pageBufferSizeInBytes;
+ ma_resource_manager* pResourceManager;
+ ma_resource_manager_data_stream* pDataStream;
- if (pWav == NULL) {
- return MA_INVALID_ARGS;
+ MA_ASSERT(pJob != NULL);
+
+ pDataStream = (ma_resource_manager_data_stream*)pJob->data.resourceManager.loadDataStream.pDataStream;
+ MA_ASSERT(pDataStream != NULL);
+
+ pResourceManager = pDataStream->pResourceManager;
+
+ if (pJob->order != c89atomic_load_32(&pDataStream->executionPointer)) {
+ return ma_resource_manager_post_job(pResourceManager, pJob); /* Out of order. */
}
- MA_ZERO_OBJECT(pWav);
- pWav->format = ma_format_f32; /* f32 by default. */
+ if (ma_resource_manager_data_stream_result(pDataStream) != MA_BUSY) {
+ result = MA_INVALID_OPERATION; /* Most likely the data stream is being uninitialized. */
+ goto done;
+ }
- if (pConfig != NULL && (pConfig->preferredFormat == ma_format_f32 || pConfig->preferredFormat == ma_format_s16 || pConfig->preferredFormat == ma_format_s32)) {
- pWav->format = pConfig->preferredFormat;
+ /* We need to initialize the decoder first so we can determine the size of the pages. */
+ decoderConfig = ma_resource_manager__init_decoder_config(pResourceManager);
+
+ if (pJob->data.resourceManager.loadDataStream.pFilePath != NULL) {
+ result = ma_decoder_init_vfs(pResourceManager->config.pVFS, pJob->data.resourceManager.loadDataStream.pFilePath, &decoderConfig, &pDataStream->decoder);
} else {
- /* Getting here means something other than f32 and s16 was specified. Just leave this unset to use the default format. */
+ result = ma_decoder_init_vfs_w(pResourceManager->config.pVFS, pJob->data.resourceManager.loadDataStream.pFilePathW, &decoderConfig, &pDataStream->decoder);
+ }
+ if (result != MA_SUCCESS) {
+ goto done;
}
- dataSourceConfig = ma_data_source_config_init();
- dataSourceConfig.vtable = &g_ma_wav_ds_vtable;
-
- result = ma_data_source_init(&dataSourceConfig, &pWav->ds);
+ /* Retrieve the total length of the file before marking the decoder are loaded. */
+ result = ma_decoder_get_length_in_pcm_frames(&pDataStream->decoder, &pDataStream->totalLengthInPCMFrames);
if (result != MA_SUCCESS) {
- return result; /* Failed to initialize the base data source. */
+ goto done; /* Failed to retrieve the length. */
}
- return MA_SUCCESS;
-}
+ /*
+ Only mark the decoder as initialized when the length of the decoder has been retrieved because that can possibly require a scan over the whole file
+ and we don't want to have another thread trying to access the decoder while it's scanning.
+ */
+ pDataStream->isDecoderInitialized = MA_TRUE;
-MA_API ma_result ma_wav_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav)
-{
- ma_result result;
+ /* We have the decoder so we can now initialize our page buffer. */
+ pageBufferSizeInBytes = ma_resource_manager_data_stream_get_page_size_in_frames(pDataStream) * 2 * ma_get_bytes_per_frame(pDataStream->decoder.outputFormat, pDataStream->decoder.outputChannels);
- result = ma_wav_init_internal(pConfig, pWav);
- if (result != MA_SUCCESS) {
- return result;
+ pDataStream->pPageData = ma_malloc(pageBufferSizeInBytes, &pResourceManager->config.allocationCallbacks);
+ if (pDataStream->pPageData == NULL) {
+ ma_decoder_uninit(&pDataStream->decoder);
+ result = MA_OUT_OF_MEMORY;
+ goto done;
}
- if (onRead == NULL || onSeek == NULL) {
- return MA_INVALID_ARGS; /* onRead and onSeek are mandatory. */
- }
+ /* Seek to our initial seek point before filling the initial pages. */
+ ma_decoder_seek_to_pcm_frame(&pDataStream->decoder, pJob->data.resourceManager.loadDataStream.initialSeekPoint);
- pWav->onRead = onRead;
- pWav->onSeek = onSeek;
- pWav->onTell = onTell;
- pWav->pReadSeekTellUserData = pReadSeekTellUserData;
+ /* We have our decoder and our page buffer, so now we need to fill our pages. */
+ ma_resource_manager_data_stream_fill_pages(pDataStream);
- #if !defined(MA_NO_WAV)
- {
- drwav_allocation_callbacks wavAllocationCallbacks = drwav_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
- drwav_bool32 wavResult;
+ /* And now we're done. We want to make sure the result is MA_SUCCESS. */
+ result = MA_SUCCESS;
- wavResult = drwav_init(&pWav->dr, ma_wav_dr_callback__read, ma_wav_dr_callback__seek, pWav, &wavAllocationCallbacks);
- if (wavResult != MA_TRUE) {
- return MA_INVALID_FILE;
- }
+done:
+ ma_free(pJob->data.resourceManager.loadDataStream.pFilePath, &pResourceManager->config.allocationCallbacks);
+ ma_free(pJob->data.resourceManager.loadDataStream.pFilePathW, &pResourceManager->config.allocationCallbacks);
- return MA_SUCCESS;
+ /* We can only change the status away from MA_BUSY. If it's set to anything else it means an error has occurred somewhere or the uninitialization process has started (most likely). */
+ c89atomic_compare_and_swap_i32(&pDataStream->result, MA_BUSY, result);
+
+ /* Only signal the other threads after the result has been set just for cleanliness sake. */
+ if (pJob->data.resourceManager.loadDataStream.pInitNotification != NULL) {
+ ma_async_notification_signal(pJob->data.resourceManager.loadDataStream.pInitNotification);
}
- #else
- {
- /* wav is disabled. */
- (void)pAllocationCallbacks;
- return MA_NOT_IMPLEMENTED;
+ if (pJob->data.resourceManager.loadDataStream.pInitFence != NULL) {
+ ma_fence_release(pJob->data.resourceManager.loadDataStream.pInitFence);
}
- #endif
+
+ c89atomic_fetch_add_32(&pDataStream->executionPointer, 1);
+ return result;
}
-MA_API ma_result ma_wav_init_file(const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav)
+static ma_result ma_job_process__resource_manager__free_data_stream(ma_job* pJob)
{
- ma_result result;
+ ma_resource_manager* pResourceManager;
+ ma_resource_manager_data_stream* pDataStream;
- result = ma_wav_init_internal(pConfig, pWav);
- if (result != MA_SUCCESS) {
- return result;
- }
+ MA_ASSERT(pJob != NULL);
- #if !defined(MA_NO_WAV)
- {
- drwav_allocation_callbacks wavAllocationCallbacks = drwav_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
- drwav_bool32 wavResult;
+ pDataStream = (ma_resource_manager_data_stream*)pJob->data.resourceManager.freeDataStream.pDataStream;
+ MA_ASSERT(pDataStream != NULL);
- wavResult = drwav_init_file(&pWav->dr, pFilePath, &wavAllocationCallbacks);
- if (wavResult != MA_TRUE) {
- return MA_INVALID_FILE;
- }
+ pResourceManager = pDataStream->pResourceManager;
- return MA_SUCCESS;
- }
- #else
- {
- /* wav is disabled. */
- (void)pFilePath;
- (void)pAllocationCallbacks;
- return MA_NOT_IMPLEMENTED;
+ if (pJob->order != c89atomic_load_32(&pDataStream->executionPointer)) {
+ return ma_resource_manager_post_job(pResourceManager, pJob); /* Out of order. */
}
- #endif
-}
-MA_API ma_result ma_wav_init_file_w(const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav)
-{
- ma_result result;
+ /* If our status is not MA_UNAVAILABLE we have a bug somewhere. */
+ MA_ASSERT(ma_resource_manager_data_stream_result(pDataStream) == MA_UNAVAILABLE);
- result = ma_wav_init_internal(pConfig, pWav);
- if (result != MA_SUCCESS) {
- return result;
+ if (pDataStream->isDecoderInitialized) {
+ ma_decoder_uninit(&pDataStream->decoder);
}
- #if !defined(MA_NO_WAV)
- {
- drwav_allocation_callbacks wavAllocationCallbacks = drwav_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
- drwav_bool32 wavResult;
+ if (pDataStream->pPageData != NULL) {
+ ma_free(pDataStream->pPageData, &pResourceManager->config.allocationCallbacks);
+ pDataStream->pPageData = NULL; /* Just in case... */
+ }
- wavResult = drwav_init_file_w(&pWav->dr, pFilePath, &wavAllocationCallbacks);
- if (wavResult != MA_TRUE) {
- return MA_INVALID_FILE;
- }
+ ma_data_source_uninit(&pDataStream->ds);
- return MA_SUCCESS;
+ /* The event needs to be signalled last. */
+ if (pJob->data.resourceManager.freeDataStream.pDoneNotification != NULL) {
+ ma_async_notification_signal(pJob->data.resourceManager.freeDataStream.pDoneNotification);
}
- #else
- {
- /* wav is disabled. */
- (void)pFilePath;
- (void)pAllocationCallbacks;
- return MA_NOT_IMPLEMENTED;
+ if (pJob->data.resourceManager.freeDataStream.pDoneFence != NULL) {
+ ma_fence_release(pJob->data.resourceManager.freeDataStream.pDoneFence);
}
- #endif
+
+ /*c89atomic_fetch_add_32(&pDataStream->executionPointer, 1);*/
+ return MA_SUCCESS;
}
-MA_API ma_result ma_wav_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav)
+static ma_result ma_job_process__resource_manager__page_data_stream(ma_job* pJob)
{
- ma_result result;
+ ma_result result = MA_SUCCESS;
+ ma_resource_manager* pResourceManager;
+ ma_resource_manager_data_stream* pDataStream;
- result = ma_wav_init_internal(pConfig, pWav);
- if (result != MA_SUCCESS) {
- return result;
- }
+ MA_ASSERT(pJob != NULL);
- #if !defined(MA_NO_WAV)
- {
- drwav_allocation_callbacks wavAllocationCallbacks = drwav_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
- drwav_bool32 wavResult;
+ pDataStream = (ma_resource_manager_data_stream*)pJob->data.resourceManager.pageDataStream.pDataStream;
+ MA_ASSERT(pDataStream != NULL);
- wavResult = drwav_init_memory(&pWav->dr, pData, dataSize, &wavAllocationCallbacks);
- if (wavResult != MA_TRUE) {
- return MA_INVALID_FILE;
- }
+ pResourceManager = pDataStream->pResourceManager;
- return MA_SUCCESS;
+ if (pJob->order != c89atomic_load_32(&pDataStream->executionPointer)) {
+ return ma_resource_manager_post_job(pResourceManager, pJob); /* Out of order. */
}
- #else
- {
- /* wav is disabled. */
- (void)pData;
- (void)dataSize;
- (void)pAllocationCallbacks;
- return MA_NOT_IMPLEMENTED;
+
+ /* For streams, the status should be MA_SUCCESS. */
+ if (ma_resource_manager_data_stream_result(pDataStream) != MA_SUCCESS) {
+ result = MA_INVALID_OPERATION;
+ goto done;
}
- #endif
+
+ ma_resource_manager_data_stream_fill_page(pDataStream, pJob->data.resourceManager.pageDataStream.pageIndex);
+
+done:
+ c89atomic_fetch_add_32(&pDataStream->executionPointer, 1);
+ return result;
}
-MA_API void ma_wav_uninit(ma_wav* pWav, const ma_allocation_callbacks* pAllocationCallbacks)
+static ma_result ma_job_process__resource_manager__seek_data_stream(ma_job* pJob)
{
- if (pWav == NULL) {
- return;
- }
+ ma_result result = MA_SUCCESS;
+ ma_resource_manager* pResourceManager;
+ ma_resource_manager_data_stream* pDataStream;
- (void)pAllocationCallbacks;
+ MA_ASSERT(pJob != NULL);
- #if !defined(MA_NO_WAV)
- {
- drwav_uninit(&pWav->dr);
+ pDataStream = (ma_resource_manager_data_stream*)pJob->data.resourceManager.seekDataStream.pDataStream;
+ MA_ASSERT(pDataStream != NULL);
+
+ pResourceManager = pDataStream->pResourceManager;
+
+ if (pJob->order != c89atomic_load_32(&pDataStream->executionPointer)) {
+ return ma_resource_manager_post_job(pResourceManager, pJob); /* Out of order. */
}
- #else
- {
- /* wav is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
+
+ /* For streams the status should be MA_SUCCESS for this to do anything. */
+ if (ma_resource_manager_data_stream_result(pDataStream) != MA_SUCCESS || pDataStream->isDecoderInitialized == MA_FALSE) {
+ result = MA_INVALID_OPERATION;
+ goto done;
}
- #endif
- ma_data_source_uninit(&pWav->ds);
+ /*
+ With seeking we just assume both pages are invalid and the relative frame cursor at position 0. This is basically exactly the same as loading, except
+ instead of initializing the decoder, we seek to a frame.
+ */
+ ma_decoder_seek_to_pcm_frame(&pDataStream->decoder, pJob->data.resourceManager.seekDataStream.frameIndex);
+
+ /* After seeking we'll need to reload the pages. */
+ ma_resource_manager_data_stream_fill_pages(pDataStream);
+
+ /* We need to let the public API know that we're done seeking. */
+ c89atomic_fetch_sub_32(&pDataStream->seekCounter, 1);
+
+done:
+ c89atomic_fetch_add_32(&pDataStream->executionPointer, 1);
+ return result;
}
-MA_API ma_result ma_wav_read_pcm_frames(ma_wav* pWav, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+MA_API ma_result ma_resource_manager_process_job(ma_resource_manager* pResourceManager, ma_job* pJob)
{
- if (pWav == NULL) {
+ if (pResourceManager == NULL || pJob == NULL) {
return MA_INVALID_ARGS;
}
- #if !defined(MA_NO_WAV)
- {
- /* We always use floating point format. */
- ma_result result = MA_SUCCESS; /* Must be initialized to MA_SUCCESS. */
- ma_uint64 totalFramesRead = 0;
- ma_format format;
+ return ma_job_process(pJob);
+}
- ma_wav_get_data_format(pWav, &format, NULL, NULL, NULL, 0);
+MA_API ma_result ma_resource_manager_process_next_job(ma_resource_manager* pResourceManager)
+{
+ ma_result result;
+ ma_job job;
- switch (format)
- {
- case ma_format_f32:
- {
- totalFramesRead = drwav_read_pcm_frames_f32(&pWav->dr, frameCount, (float*)pFramesOut);
- } break;
+ if (pResourceManager == NULL) {
+ return MA_INVALID_ARGS;
+ }
- case ma_format_s16:
- {
- totalFramesRead = drwav_read_pcm_frames_s16(&pWav->dr, frameCount, (drwav_int16*)pFramesOut);
- } break;
+ /* This will return MA_CANCELLED if the next job is a quit job. */
+ result = ma_resource_manager_next_job(pResourceManager, &job);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- case ma_format_s32:
- {
- totalFramesRead = drwav_read_pcm_frames_s32(&pWav->dr, frameCount, (drwav_int32*)pFramesOut);
- } break;
+ return ma_job_process(&job);
+}
+#else
+/* We'll get here if the resource manager is being excluded from the build. We need to define the job processing callbacks as no-ops. */
+static ma_result ma_job_process__resource_manager__load_data_buffer_node(ma_job* pJob) { return ma_job_process__noop(pJob); }
+static ma_result ma_job_process__resource_manager__free_data_buffer_node(ma_job* pJob) { return ma_job_process__noop(pJob); }
+static ma_result ma_job_process__resource_manager__page_data_buffer_node(ma_job* pJob) { return ma_job_process__noop(pJob); }
+static ma_result ma_job_process__resource_manager__load_data_buffer(ma_job* pJob) { return ma_job_process__noop(pJob); }
+static ma_result ma_job_process__resource_manager__free_data_buffer(ma_job* pJob) { return ma_job_process__noop(pJob); }
+static ma_result ma_job_process__resource_manager__load_data_stream(ma_job* pJob) { return ma_job_process__noop(pJob); }
+static ma_result ma_job_process__resource_manager__free_data_stream(ma_job* pJob) { return ma_job_process__noop(pJob); }
+static ma_result ma_job_process__resource_manager__page_data_stream(ma_job* pJob) { return ma_job_process__noop(pJob); }
+static ma_result ma_job_process__resource_manager__seek_data_stream(ma_job* pJob) { return ma_job_process__noop(pJob); }
+#endif /* MA_NO_RESOURCE_MANAGER */
- /* Fallback to a raw read. */
- case ma_format_unknown: return MA_INVALID_OPERATION; /* <-- this should never be hit because initialization would just fall back to supported format. */
- default:
- {
- totalFramesRead = drwav_read_pcm_frames(&pWav->dr, frameCount, pFramesOut);
- } break;
- }
- /* In the future we'll update dr_wav to return MA_AT_END for us. */
- if (totalFramesRead == 0) {
- result = MA_AT_END;
- }
+#ifndef MA_NO_NODE_GRAPH
+/* 10ms @ 48K = 480. Must never exceed 65535. */
+#ifndef MA_DEFAULT_NODE_CACHE_CAP_IN_FRAMES_PER_BUS
+#define MA_DEFAULT_NODE_CACHE_CAP_IN_FRAMES_PER_BUS 480
+#endif
- if (pFramesRead != NULL) {
- *pFramesRead = totalFramesRead;
- }
- return result;
+static ma_result ma_node_read_pcm_frames(ma_node* pNode, ma_uint32 outputBusIndex, float* pFramesOut, ma_uint32 frameCount, ma_uint32* pFramesRead, ma_uint64 globalTime);
+
+MA_API void ma_debug_fill_pcm_frames_with_sine_wave(float* pFramesOut, ma_uint32 frameCount, ma_format format, ma_uint32 channels, ma_uint32 sampleRate)
+{
+ #ifndef MA_NO_GENERATION
+ {
+ ma_waveform_config waveformConfig;
+ ma_waveform waveform;
+
+ waveformConfig = ma_waveform_config_init(format, channels, sampleRate, ma_waveform_type_sine, 1.0, 400);
+ ma_waveform_init(&waveformConfig, &waveform);
+ ma_waveform_read_pcm_frames(&waveform, pFramesOut, frameCount, NULL);
}
#else
{
- /* wav is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
-
(void)pFramesOut;
(void)frameCount;
- (void)pFramesRead;
-
- return MA_NOT_IMPLEMENTED;
+ (void)format;
+ (void)channels;
+ (void)sampleRate;
+ #if defined(MA_DEBUG_OUTPUT)
+ {
+ #if _MSC_VER
+ #pragma message ("ma_debug_fill_pcm_frames_with_sine_wave() will do nothing because MA_NO_GENERATION is enabled.")
+ #endif
+ }
+ #endif
}
#endif
}
-MA_API ma_result ma_wav_seek_to_pcm_frame(ma_wav* pWav, ma_uint64 frameIndex)
+
+
+static ma_result ma_mix_pcm_frames_f32(float* pDst, const float* pSrc, ma_uint64 frameCount, ma_uint32 channels, float volume)
{
- if (pWav == NULL) {
+ ma_uint64 iSample;
+ ma_uint64 sampleCount;
+
+ if (pDst == NULL || pSrc == NULL || channels == 0) {
return MA_INVALID_ARGS;
}
- #if !defined(MA_NO_WAV)
- {
- drwav_bool32 wavResult;
-
- wavResult = drwav_seek_to_pcm_frame(&pWav->dr, frameIndex);
- if (wavResult != DRWAV_TRUE) {
- return MA_ERROR;
- }
-
- return MA_SUCCESS;
+ if (volume == 0) {
+ return MA_SUCCESS; /* No changes if the volume is 0. */
}
- #else
- {
- /* wav is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
- (void)frameIndex;
+ sampleCount = frameCount * channels;
- return MA_NOT_IMPLEMENTED;
+ if (volume == 1) {
+ for (iSample = 0; iSample < sampleCount; iSample += 1) {
+ pDst[iSample] += pSrc[iSample];
+ }
+ } else {
+ for (iSample = 0; iSample < sampleCount; iSample += 1) {
+ pDst[iSample] += ma_apply_volume_unclipped_f32(pSrc[iSample], volume);
+ }
}
- #endif
-}
-MA_API ma_result ma_wav_get_data_format(ma_wav* pWav, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
-{
- /* Defaults for safety. */
- if (pFormat != NULL) {
- *pFormat = ma_format_unknown;
- }
- if (pChannels != NULL) {
- *pChannels = 0;
- }
- if (pSampleRate != NULL) {
- *pSampleRate = 0;
- }
- if (pChannelMap != NULL) {
- MA_ZERO_MEMORY(pChannelMap, sizeof(*pChannelMap) * channelMapCap);
- }
+ return MA_SUCCESS;
+}
- if (pWav == NULL) {
- return MA_INVALID_OPERATION;
- }
- if (pFormat != NULL) {
- *pFormat = pWav->format;
- }
+MA_API ma_node_graph_config ma_node_graph_config_init(ma_uint32 channels)
+{
+ ma_node_graph_config config;
- #if !defined(MA_NO_WAV)
- {
- if (pChannels != NULL) {
- *pChannels = pWav->dr.channels;
- }
+ MA_ZERO_OBJECT(&config);
+ config.channels = channels;
+ config.nodeCacheCapInFrames = MA_DEFAULT_NODE_CACHE_CAP_IN_FRAMES_PER_BUS;
- if (pSampleRate != NULL) {
- *pSampleRate = pWav->dr.sampleRate;
- }
+ return config;
+}
- if (pChannelMap != NULL) {
- ma_get_standard_channel_map(ma_standard_channel_map_microsoft, (ma_uint32)ma_min(pWav->dr.channels, channelMapCap), pChannelMap);
- }
- return MA_SUCCESS;
- }
- #else
- {
- /* wav is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
- return MA_NOT_IMPLEMENTED;
- }
- #endif
+static void ma_node_graph_set_is_reading(ma_node_graph* pNodeGraph, ma_bool32 isReading)
+{
+ MA_ASSERT(pNodeGraph != NULL);
+ c89atomic_exchange_32(&pNodeGraph->isReading, isReading);
}
-MA_API ma_result ma_wav_get_cursor_in_pcm_frames(ma_wav* pWav, ma_uint64* pCursor)
+#if 0
+static ma_bool32 ma_node_graph_is_reading(ma_node_graph* pNodeGraph)
{
- if (pCursor == NULL) {
- return MA_INVALID_ARGS;
- }
+ MA_ASSERT(pNodeGraph != NULL);
+ return c89atomic_load_32(&pNodeGraph->isReading);
+}
+#endif
- *pCursor = 0; /* Safety. */
- if (pWav == NULL) {
- return MA_INVALID_ARGS;
- }
+static void ma_node_graph_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
+{
+ ma_node_graph* pNodeGraph = (ma_node_graph*)pNode;
+ ma_uint64 framesRead;
- #if !defined(MA_NO_WAV)
- {
- drwav_result wavResult = drwav_get_cursor_in_pcm_frames(&pWav->dr, pCursor);
- if (wavResult != DRWAV_SUCCESS) {
- return (ma_result)wavResult; /* dr_wav result codes map to miniaudio's. */
- }
+ ma_node_graph_read_pcm_frames(pNodeGraph, ppFramesOut[0], *pFrameCountOut, &framesRead);
- return MA_SUCCESS;
- }
- #else
- {
- /* wav is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
- return MA_NOT_IMPLEMENTED;
- }
- #endif
+ *pFrameCountOut = (ma_uint32)framesRead; /* Safe cast. */
+
+ (void)ppFramesIn;
+ (void)pFrameCountIn;
}
-MA_API ma_result ma_wav_get_length_in_pcm_frames(ma_wav* pWav, ma_uint64* pLength)
+static ma_node_vtable g_node_graph_node_vtable =
{
- if (pLength == NULL) {
- return MA_INVALID_ARGS;
- }
+ ma_node_graph_node_process_pcm_frames,
+ NULL, /* onGetRequiredInputFrameCount */
+ 0, /* 0 input buses. */
+ 1, /* 1 output bus. */
+ 0 /* Flags. */
+};
- *pLength = 0; /* Safety. */
+static void ma_node_graph_endpoint_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
+{
+ MA_ASSERT(pNode != NULL);
+ MA_ASSERT(ma_node_get_input_bus_count(pNode) == 1);
+ MA_ASSERT(ma_node_get_output_bus_count(pNode) == 1);
- if (pWav == NULL) {
- return MA_INVALID_ARGS;
- }
+ /* Input channel count needs to be the same as the output channel count. */
+ MA_ASSERT(ma_node_get_input_channels(pNode, 0) == ma_node_get_output_channels(pNode, 0));
- #if !defined(MA_NO_WAV)
- {
- drwav_result wavResult = drwav_get_length_in_pcm_frames(&pWav->dr, pLength);
- if (wavResult != DRWAV_SUCCESS) {
- return (ma_result)wavResult; /* dr_wav result codes map to miniaudio's. */
- }
+ /* We don't need to do anything here because it's a passthrough. */
+ (void)pNode;
+ (void)ppFramesIn;
+ (void)pFrameCountIn;
+ (void)ppFramesOut;
+ (void)pFrameCountOut;
- return MA_SUCCESS;
- }
- #else
- {
- /* wav is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
- return MA_NOT_IMPLEMENTED;
+#if 0
+ /* The data has already been mixed. We just need to move it to the output buffer. */
+ if (ppFramesIn != NULL) {
+ ma_copy_pcm_frames(ppFramesOut[0], ppFramesIn[0], *pFrameCountOut, ma_format_f32, ma_node_get_output_channels(pNode, 0));
}
- #endif
+#endif
}
+static ma_node_vtable g_node_graph_endpoint_vtable =
+{
+ ma_node_graph_endpoint_process_pcm_frames,
+ NULL, /* onGetRequiredInputFrameCount */
+ 1, /* 1 input bus. */
+ 1, /* 1 output bus. */
+ MA_NODE_FLAG_PASSTHROUGH /* Flags. The endpoint is a passthrough. */
+};
-static ma_result ma_decoding_backend_init__wav(void* pUserData, ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
+MA_API ma_result ma_node_graph_init(const ma_node_graph_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_node_graph* pNodeGraph)
{
ma_result result;
- ma_wav* pWav;
+ ma_node_config baseConfig;
+ ma_node_config endpointConfig;
- (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
+ if (pNodeGraph == NULL) {
+ return MA_INVALID_ARGS;
+ }
- /* For now we're just allocating the decoder backend on the heap. */
- pWav = (ma_wav*)ma_malloc(sizeof(*pWav), pAllocationCallbacks);
- if (pWav == NULL) {
- return MA_OUT_OF_MEMORY;
+ MA_ZERO_OBJECT(pNodeGraph);
+ pNodeGraph->nodeCacheCapInFrames = pConfig->nodeCacheCapInFrames;
+ if (pNodeGraph->nodeCacheCapInFrames == 0) {
+ pNodeGraph->nodeCacheCapInFrames = MA_DEFAULT_NODE_CACHE_CAP_IN_FRAMES_PER_BUS;
}
- result = ma_wav_init(onRead, onSeek, onTell, pReadSeekTellUserData, pConfig, pAllocationCallbacks, pWav);
+
+ /* Base node so we can use the node graph as a node into another graph. */
+ baseConfig = ma_node_config_init();
+ baseConfig.vtable = &g_node_graph_node_vtable;
+ baseConfig.pOutputChannels = &pConfig->channels;
+
+ result = ma_node_init(pNodeGraph, &baseConfig, pAllocationCallbacks, &pNodeGraph->base);
if (result != MA_SUCCESS) {
- ma_free(pWav, pAllocationCallbacks);
return result;
}
- *ppBackend = pWav;
+
+ /* Endpoint. */
+ endpointConfig = ma_node_config_init();
+ endpointConfig.vtable = &g_node_graph_endpoint_vtable;
+ endpointConfig.pInputChannels = &pConfig->channels;
+ endpointConfig.pOutputChannels = &pConfig->channels;
+
+ result = ma_node_init(pNodeGraph, &endpointConfig, pAllocationCallbacks, &pNodeGraph->endpoint);
+ if (result != MA_SUCCESS) {
+ ma_node_uninit(&pNodeGraph->base, pAllocationCallbacks);
+ return result;
+ }
return MA_SUCCESS;
}
-static ma_result ma_decoding_backend_init_file__wav(void* pUserData, const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
+MA_API void ma_node_graph_uninit(ma_node_graph* pNodeGraph, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_result result;
- ma_wav* pWav;
+ if (pNodeGraph == NULL) {
+ return;
+ }
- (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
+ ma_node_uninit(&pNodeGraph->endpoint, pAllocationCallbacks);
+}
- /* For now we're just allocating the decoder backend on the heap. */
- pWav = (ma_wav*)ma_malloc(sizeof(*pWav), pAllocationCallbacks);
- if (pWav == NULL) {
- return MA_OUT_OF_MEMORY;
+MA_API ma_node* ma_node_graph_get_endpoint(ma_node_graph* pNodeGraph)
+{
+ if (pNodeGraph == NULL) {
+ return NULL;
}
- result = ma_wav_init_file(pFilePath, pConfig, pAllocationCallbacks, pWav);
- if (result != MA_SUCCESS) {
- ma_free(pWav, pAllocationCallbacks);
- return result;
+ return &pNodeGraph->endpoint;
+}
+
+MA_API ma_result ma_node_graph_read_pcm_frames(ma_node_graph* pNodeGraph, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+{
+ ma_result result = MA_SUCCESS;
+ ma_uint64 totalFramesRead;
+ ma_uint32 channels;
+
+ if (pFramesRead != NULL) {
+ *pFramesRead = 0; /* Safety. */
}
- *ppBackend = pWav;
+ if (pNodeGraph == NULL) {
+ return MA_INVALID_ARGS;
+ }
- return MA_SUCCESS;
-}
+ channels = ma_node_get_output_channels(&pNodeGraph->endpoint, 0);
-static ma_result ma_decoding_backend_init_file_w__wav(void* pUserData, const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
-{
- ma_result result;
- ma_wav* pWav;
- (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
+ /* We'll be nice and try to do a full read of all frameCount frames. */
+ totalFramesRead = 0;
+ while (totalFramesRead < frameCount) {
+ ma_uint32 framesJustRead;
+ ma_uint64 framesToRead = frameCount - totalFramesRead;
- /* For now we're just allocating the decoder backend on the heap. */
- pWav = (ma_wav*)ma_malloc(sizeof(*pWav), pAllocationCallbacks);
- if (pWav == NULL) {
- return MA_OUT_OF_MEMORY;
+ if (framesToRead > 0xFFFFFFFF) {
+ framesToRead = 0xFFFFFFFF;
+ }
+
+ ma_node_graph_set_is_reading(pNodeGraph, MA_TRUE);
+ {
+ result = ma_node_read_pcm_frames(&pNodeGraph->endpoint, 0, (float*)ma_offset_pcm_frames_ptr(pFramesOut, totalFramesRead, ma_format_f32, channels), (ma_uint32)framesToRead, &framesJustRead, ma_node_get_time(&pNodeGraph->endpoint));
+ }
+ ma_node_graph_set_is_reading(pNodeGraph, MA_FALSE);
+
+ totalFramesRead += framesJustRead;
+
+ if (result != MA_SUCCESS) {
+ break;
+ }
+
+ /* Abort if we weren't able to read any frames or else we risk getting stuck in a loop. */
+ if (framesJustRead == 0) {
+ break;
+ }
}
- result = ma_wav_init_file_w(pFilePath, pConfig, pAllocationCallbacks, pWav);
- if (result != MA_SUCCESS) {
- ma_free(pWav, pAllocationCallbacks);
- return result;
+ /* Let's go ahead and silence any leftover frames just for some added safety to ensure the caller doesn't try emitting garbage out of the speakers. */
+ if (totalFramesRead < frameCount) {
+ ma_silence_pcm_frames(ma_offset_pcm_frames_ptr(pFramesOut, totalFramesRead, ma_format_f32, channels), (frameCount - totalFramesRead), ma_format_f32, channels);
}
- *ppBackend = pWav;
+ if (pFramesRead != NULL) {
+ *pFramesRead = totalFramesRead;
+ }
- return MA_SUCCESS;
+ return result;
}
-static ma_result ma_decoding_backend_init_memory__wav(void* pUserData, const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
+MA_API ma_uint32 ma_node_graph_get_channels(const ma_node_graph* pNodeGraph)
{
- ma_result result;
- ma_wav* pWav;
-
- (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
-
- /* For now we're just allocating the decoder backend on the heap. */
- pWav = (ma_wav*)ma_malloc(sizeof(*pWav), pAllocationCallbacks);
- if (pWav == NULL) {
- return MA_OUT_OF_MEMORY;
+ if (pNodeGraph == NULL) {
+ return 0;
}
- result = ma_wav_init_memory(pData, dataSize, pConfig, pAllocationCallbacks, pWav);
- if (result != MA_SUCCESS) {
- ma_free(pWav, pAllocationCallbacks);
- return result;
- }
+ return ma_node_get_output_channels(&pNodeGraph->endpoint, 0);
+}
- *ppBackend = pWav;
+MA_API ma_uint64 ma_node_graph_get_time(const ma_node_graph* pNodeGraph)
+{
+ if (pNodeGraph == NULL) {
+ return 0;
+ }
- return MA_SUCCESS;
+ return ma_node_get_time(&pNodeGraph->endpoint); /* Global time is just the local time of the endpoint. */
}
-static void ma_decoding_backend_uninit__wav(void* pUserData, ma_data_source* pBackend, const ma_allocation_callbacks* pAllocationCallbacks)
+MA_API ma_result ma_node_graph_set_time(ma_node_graph* pNodeGraph, ma_uint64 globalTime)
{
- ma_wav* pWav = (ma_wav*)pBackend;
-
- (void)pUserData;
+ if (pNodeGraph == NULL) {
+ return MA_INVALID_ARGS;
+ }
- ma_wav_uninit(pWav, pAllocationCallbacks);
- ma_free(pWav, pAllocationCallbacks);
+ return ma_node_set_time(&pNodeGraph->endpoint, globalTime); /* Global time is just the local time of the endpoint. */
}
-static ma_result ma_decoding_backend_get_channel_map__wav(void* pUserData, ma_data_source* pBackend, ma_channel* pChannelMap, size_t channelMapCap)
+
+#define MA_NODE_OUTPUT_BUS_FLAG_HAS_READ 0x01 /* Whether or not this bus ready to read more data. Only used on nodes with multiple output buses. */
+
+static ma_result ma_node_output_bus_init(ma_node* pNode, ma_uint32 outputBusIndex, ma_uint32 channels, ma_node_output_bus* pOutputBus)
{
- ma_wav* pWav = (ma_wav*)pBackend;
+ MA_ASSERT(pOutputBus != NULL);
+ MA_ASSERT(outputBusIndex < MA_MAX_NODE_BUS_COUNT);
+ MA_ASSERT(outputBusIndex < ma_node_get_output_bus_count(pNode));
+ MA_ASSERT(channels < 256);
- (void)pUserData;
+ MA_ZERO_OBJECT(pOutputBus);
+
+ if (channels == 0) {
+ return MA_INVALID_ARGS;
+ }
+
+ pOutputBus->pNode = pNode;
+ pOutputBus->outputBusIndex = (ma_uint8)outputBusIndex;
+ pOutputBus->channels = (ma_uint8)channels;
+ pOutputBus->flags = MA_NODE_OUTPUT_BUS_FLAG_HAS_READ; /* <-- Important that this flag is set by default. */
+ pOutputBus->volume = 1;
- return ma_wav_get_data_format(pWav, NULL, NULL, NULL, pChannelMap, channelMapCap);
+ return MA_SUCCESS;
}
-static ma_decoding_backend_vtable g_ma_decoding_backend_vtable_wav =
+static void ma_node_output_bus_lock(ma_node_output_bus* pOutputBus)
{
- ma_decoding_backend_init__wav,
- ma_decoding_backend_init_file__wav,
- ma_decoding_backend_init_file_w__wav,
- ma_decoding_backend_init_memory__wav,
- ma_decoding_backend_uninit__wav,
- ma_decoding_backend_get_channel_map__wav
-};
+ ma_spinlock_lock(&pOutputBus->lock);
+}
-static ma_result ma_decoder_init_wav__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+static void ma_node_output_bus_unlock(ma_node_output_bus* pOutputBus)
{
- return ma_decoder_init_from_vtable(&g_ma_decoding_backend_vtable_wav, NULL, pConfig, pDecoder);
+ ma_spinlock_unlock(&pOutputBus->lock);
}
-#endif /* dr_wav_h */
-/* FLAC */
-#ifdef dr_flac_h
-#define MA_HAS_FLAC
-typedef struct
+static ma_uint32 ma_node_output_bus_get_channels(const ma_node_output_bus* pOutputBus)
{
- ma_data_source_base ds;
- ma_read_proc onRead;
- ma_seek_proc onSeek;
- ma_tell_proc onTell;
- void* pReadSeekTellUserData;
- ma_format format; /* Can be f32, s16 or s32. */
-#if !defined(MA_NO_FLAC)
- drflac* dr;
-#endif
-} ma_flac;
-
-MA_API ma_result ma_flac_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac);
-MA_API ma_result ma_flac_init_file(const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac);
-MA_API ma_result ma_flac_init_file_w(const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac);
-MA_API ma_result ma_flac_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac);
-MA_API void ma_flac_uninit(ma_flac* pFlac, const ma_allocation_callbacks* pAllocationCallbacks);
-MA_API ma_result ma_flac_read_pcm_frames(ma_flac* pFlac, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
-MA_API ma_result ma_flac_seek_to_pcm_frame(ma_flac* pFlac, ma_uint64 frameIndex);
-MA_API ma_result ma_flac_get_data_format(ma_flac* pFlac, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
-MA_API ma_result ma_flac_get_cursor_in_pcm_frames(ma_flac* pFlac, ma_uint64* pCursor);
-MA_API ma_result ma_flac_get_length_in_pcm_frames(ma_flac* pFlac, ma_uint64* pLength);
+ return pOutputBus->channels;
+}
-static ma_result ma_flac_ds_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+static void ma_node_output_bus_set_has_read(ma_node_output_bus* pOutputBus, ma_bool32 hasRead)
{
- return ma_flac_read_pcm_frames((ma_flac*)pDataSource, pFramesOut, frameCount, pFramesRead);
+ if (hasRead) {
+ c89atomic_fetch_or_32(&pOutputBus->flags, MA_NODE_OUTPUT_BUS_FLAG_HAS_READ);
+ } else {
+ c89atomic_fetch_and_32(&pOutputBus->flags, (ma_uint32)~MA_NODE_OUTPUT_BUS_FLAG_HAS_READ);
+ }
}
-static ma_result ma_flac_ds_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
+static ma_bool32 ma_node_output_bus_has_read(ma_node_output_bus* pOutputBus)
{
- return ma_flac_seek_to_pcm_frame((ma_flac*)pDataSource, frameIndex);
+ return (c89atomic_load_32(&pOutputBus->flags) & MA_NODE_OUTPUT_BUS_FLAG_HAS_READ) != 0;
}
-static ma_result ma_flac_ds_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate)
+
+static void ma_node_output_bus_set_is_attached(ma_node_output_bus* pOutputBus, ma_bool32 isAttached)
{
- return ma_flac_get_data_format((ma_flac*)pDataSource, pFormat, pChannels, pSampleRate, NULL, 0);
+ c89atomic_exchange_32(&pOutputBus->isAttached, isAttached);
}
-static ma_result ma_flac_ds_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
+static ma_bool32 ma_node_output_bus_is_attached(ma_node_output_bus* pOutputBus)
{
- return ma_flac_get_cursor_in_pcm_frames((ma_flac*)pDataSource, pCursor);
+ return c89atomic_load_32(&pOutputBus->isAttached);
}
-static ma_result ma_flac_ds_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
+
+static ma_result ma_node_output_bus_set_volume(ma_node_output_bus* pOutputBus, float volume)
{
- return ma_flac_get_length_in_pcm_frames((ma_flac*)pDataSource, pLength);
+ MA_ASSERT(pOutputBus != NULL);
+
+ if (volume < 0.0f) {
+ volume = 0.0f;
+ }
+
+ c89atomic_exchange_f32(&pOutputBus->volume, volume);
+
+ return MA_SUCCESS;
}
-static ma_data_source_vtable g_ma_flac_ds_vtable =
+static float ma_node_output_bus_get_volume(const ma_node_output_bus* pOutputBus)
{
- ma_flac_ds_read,
- ma_flac_ds_seek,
- NULL, /* onMap() */
- NULL, /* onUnmap() */
- ma_flac_ds_get_data_format,
- ma_flac_ds_get_cursor,
- ma_flac_ds_get_length
-};
+ return c89atomic_load_f32((float*)&pOutputBus->volume);
+}
-#if !defined(MA_NO_FLAC)
-static drflac_allocation_callbacks drflac_allocation_callbacks_from_miniaudio(const ma_allocation_callbacks* pAllocationCallbacks)
+static ma_result ma_node_input_bus_init(ma_uint32 channels, ma_node_input_bus* pInputBus)
{
- drflac_allocation_callbacks callbacks;
+ MA_ASSERT(pInputBus != NULL);
+ MA_ASSERT(channels < 256);
- if (pAllocationCallbacks != NULL) {
- callbacks.onMalloc = pAllocationCallbacks->onMalloc;
- callbacks.onRealloc = pAllocationCallbacks->onRealloc;
- callbacks.onFree = pAllocationCallbacks->onFree;
- callbacks.pUserData = pAllocationCallbacks->pUserData;
- } else {
- callbacks.onMalloc = ma__malloc_default;
- callbacks.onRealloc = ma__realloc_default;
- callbacks.onFree = ma__free_default;
- callbacks.pUserData = NULL;
+ MA_ZERO_OBJECT(pInputBus);
+
+ if (channels == 0) {
+ return MA_INVALID_ARGS;
}
- return callbacks;
+ pInputBus->channels = (ma_uint8)channels;
+
+ return MA_SUCCESS;
}
-static size_t ma_flac_dr_callback__read(void* pUserData, void* pBufferOut, size_t bytesToRead)
+static void ma_node_input_bus_lock(ma_node_input_bus* pInputBus)
{
- ma_flac* pFlac = (ma_flac*)pUserData;
- ma_result result;
- size_t bytesRead;
+ ma_spinlock_lock(&pInputBus->lock);
+}
- MA_ASSERT(pFlac != NULL);
+static void ma_node_input_bus_unlock(ma_node_input_bus* pInputBus)
+{
+ ma_spinlock_unlock(&pInputBus->lock);
+}
- result = pFlac->onRead(pFlac->pReadSeekTellUserData, pBufferOut, bytesToRead, &bytesRead);
- (void)result;
- return bytesRead;
+static void ma_node_input_bus_next_begin(ma_node_input_bus* pInputBus)
+{
+ c89atomic_fetch_add_32(&pInputBus->nextCounter, 1);
}
-static drflac_bool32 ma_flac_dr_callback__seek(void* pUserData, int offset, drflac_seek_origin origin)
+static void ma_node_input_bus_next_end(ma_node_input_bus* pInputBus)
{
- ma_flac* pFlac = (ma_flac*)pUserData;
- ma_result result;
- ma_seek_origin maSeekOrigin;
-
- MA_ASSERT(pFlac != NULL);
+ c89atomic_fetch_sub_32(&pInputBus->nextCounter, 1);
+}
- maSeekOrigin = ma_seek_origin_start;
- if (origin == drflac_seek_origin_current) {
- maSeekOrigin = ma_seek_origin_current;
- }
+static ma_uint32 ma_node_input_bus_get_next_counter(ma_node_input_bus* pInputBus)
+{
+ return c89atomic_load_32(&pInputBus->nextCounter);
+}
- result = pFlac->onSeek(pFlac->pReadSeekTellUserData, offset, maSeekOrigin);
- if (result != MA_SUCCESS) {
- return MA_FALSE;
- }
- return MA_TRUE;
+static ma_uint32 ma_node_input_bus_get_channels(const ma_node_input_bus* pInputBus)
+{
+ return pInputBus->channels;
}
-#endif
-static ma_result ma_flac_init_internal(const ma_decoding_backend_config* pConfig, ma_flac* pFlac)
+
+static void ma_node_input_bus_detach__no_output_bus_lock(ma_node_input_bus* pInputBus, ma_node_output_bus* pOutputBus)
{
- ma_result result;
- ma_data_source_config dataSourceConfig;
+ MA_ASSERT(pInputBus != NULL);
+ MA_ASSERT(pOutputBus != NULL);
- if (pFlac == NULL) {
- return MA_INVALID_ARGS;
- }
+ /*
+ Mark the output bus as detached first. This will prevent future iterations on the audio thread
+ from iterating this output bus.
+ */
+ ma_node_output_bus_set_is_attached(pOutputBus, MA_FALSE);
- MA_ZERO_OBJECT(pFlac);
- pFlac->format = ma_format_f32; /* f32 by default. */
+ /*
+ We cannot use the output bus lock here since it'll be getting used at a higher level, but we do
+ still need to use the input bus lock since we'll be updating pointers on two different output
+ buses. The same rules apply here as the attaching case. Although we're using a lock here, we're
+ *not* using a lock when iterating over the list in the audio thread. We therefore need to craft
+ this in a way such that the iteration on the audio thread doesn't break.
+
+ The the first thing to do is swap out the "next" pointer of the previous output bus with the
+ new "next" output bus. This is the operation that matters for iteration on the audio thread.
+ After that, the previous pointer on the new "next" pointer needs to be updated, after which
+ point the linked list will be in a good state.
+ */
+ ma_node_input_bus_lock(pInputBus);
+ {
+ ma_node_output_bus* pOldPrev = (ma_node_output_bus*)c89atomic_load_ptr(&pOutputBus->pPrev);
+ ma_node_output_bus* pOldNext = (ma_node_output_bus*)c89atomic_load_ptr(&pOutputBus->pNext);
- if (pConfig != NULL && (pConfig->preferredFormat == ma_format_f32 || pConfig->preferredFormat == ma_format_s16 || pConfig->preferredFormat == ma_format_s32)) {
- pFlac->format = pConfig->preferredFormat;
- } else {
- /* Getting here means something other than f32 and s16 was specified. Just leave this unset to use the default format. */
+ if (pOldPrev != NULL) {
+ c89atomic_exchange_ptr(&pOldPrev->pNext, pOldNext); /* <-- This is where the output bus is detached from the list. */
+ }
+ if (pOldNext != NULL) {
+ c89atomic_exchange_ptr(&pOldNext->pPrev, pOldPrev); /* <-- This is required for detachment. */
+ }
}
+ ma_node_input_bus_unlock(pInputBus);
- dataSourceConfig = ma_data_source_config_init();
- dataSourceConfig.vtable = &g_ma_flac_ds_vtable;
-
- result = ma_data_source_init(&dataSourceConfig, &pFlac->ds);
- if (result != MA_SUCCESS) {
- return result; /* Failed to initialize the base data source. */
- }
+ /* At this point the output bus is detached and the linked list is completely unaware of it. Reset some data for safety. */
+ c89atomic_exchange_ptr(&pOutputBus->pNext, NULL); /* Using atomic exchanges here, mainly for the benefit of analysis tools which don't always recognize spinlocks. */
+ c89atomic_exchange_ptr(&pOutputBus->pPrev, NULL); /* As above. */
+ pOutputBus->pInputNode = NULL;
+ pOutputBus->inputNodeInputBusIndex = 0;
- return MA_SUCCESS;
-}
-MA_API ma_result ma_flac_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac)
-{
- ma_result result;
+ /*
+ For thread-safety reasons, we don't want to be returning from this straight away. We need to
+ wait for the audio thread to finish with the output bus. There's two things we need to wait
+ for. The first is the part that selects the next output bus in the list, and the other is the
+ part that reads from the output bus. Basically all we're doing is waiting for the input bus
+ to stop referencing the output bus.
+
+ We're doing this part last because we want the section above to run while the audio thread
+ is finishing up with the output bus, just for efficiency reasons. We marked the output bus as
+ detached right at the top of this function which is going to prevent the audio thread from
+ iterating the output bus again.
+ */
- result = ma_flac_init_internal(pConfig, pFlac);
- if (result != MA_SUCCESS) {
- return result;
+ /* Part 1: Wait for the current iteration to complete. */
+ while (ma_node_input_bus_get_next_counter(pInputBus) > 0) {
+ ma_yield();
}
- if (onRead == NULL || onSeek == NULL) {
- return MA_INVALID_ARGS; /* onRead and onSeek are mandatory. */
+ /* Part 2: Wait for any reads to complete. */
+ while (c89atomic_load_32(&pOutputBus->refCount) > 0) {
+ ma_yield();
}
- pFlac->onRead = onRead;
- pFlac->onSeek = onSeek;
- pFlac->onTell = onTell;
- pFlac->pReadSeekTellUserData = pReadSeekTellUserData;
-
- #if !defined(MA_NO_FLAC)
- {
- drflac_allocation_callbacks flacAllocationCallbacks = drflac_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
+ /*
+ At this point we're done detaching and we can be guaranteed that the audio thread is not going
+ to attempt to reference this output bus again (until attached again).
+ */
+}
- pFlac->dr = drflac_open(ma_flac_dr_callback__read, ma_flac_dr_callback__seek, pFlac, &flacAllocationCallbacks);
- if (pFlac->dr == NULL) {
- return MA_INVALID_FILE;
- }
+#if 0 /* Not used at the moment, but leaving here in case I need it later. */
+static void ma_node_input_bus_detach(ma_node_input_bus* pInputBus, ma_node_output_bus* pOutputBus)
+{
+ MA_ASSERT(pInputBus != NULL);
+ MA_ASSERT(pOutputBus != NULL);
- return MA_SUCCESS;
- }
- #else
+ ma_node_output_bus_lock(pOutputBus);
{
- /* flac is disabled. */
- (void)pAllocationCallbacks;
- return MA_NOT_IMPLEMENTED;
+ ma_node_input_bus_detach__no_output_bus_lock(pInputBus, pOutputBus);
}
- #endif
+ ma_node_output_bus_unlock(pOutputBus);
}
+#endif
-MA_API ma_result ma_flac_init_file(const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac)
+static void ma_node_input_bus_attach(ma_node_input_bus* pInputBus, ma_node_output_bus* pOutputBus, ma_node* pNewInputNode, ma_uint32 inputNodeInputBusIndex)
{
- ma_result result;
-
- result = ma_flac_init_internal(pConfig, pFlac);
- if (result != MA_SUCCESS) {
- return result;
- }
+ MA_ASSERT(pInputBus != NULL);
+ MA_ASSERT(pOutputBus != NULL);
- #if !defined(MA_NO_FLAC)
+ ma_node_output_bus_lock(pOutputBus);
{
- drflac_allocation_callbacks flacAllocationCallbacks = drflac_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
+ ma_node_output_bus* pOldInputNode = (ma_node_output_bus*)c89atomic_load_ptr(&pOutputBus->pInputNode);
- pFlac->dr = drflac_open_file(pFilePath, &flacAllocationCallbacks);
- if (pFlac->dr == NULL) {
- return MA_INVALID_FILE;
+ /* Detach from any existing attachment first if necessary. */
+ if (pOldInputNode != NULL) {
+ ma_node_input_bus_detach__no_output_bus_lock(pInputBus, pOutputBus);
}
- return MA_SUCCESS;
- }
- #else
- {
- /* flac is disabled. */
- (void)pFilePath;
- (void)pAllocationCallbacks;
- return MA_NOT_IMPLEMENTED;
- }
- #endif
-}
+ /*
+ At this point we can be sure the output bus is not attached to anything. The linked list in the
+ old input bus has been updated so that pOutputBus will not get iterated again.
+ */
+ pOutputBus->pInputNode = pNewInputNode; /* No need for an atomic assignment here because modification of this variable always happens within a lock. */
+ pOutputBus->inputNodeInputBusIndex = (ma_uint8)inputNodeInputBusIndex; /* As above. */
-MA_API ma_result ma_flac_init_file_w(const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac)
-{
- ma_result result;
+ /*
+ Now we need to attach the output bus to the linked list. This involves updating two pointers on
+ two different output buses so I'm going to go ahead and keep this simple and just use a lock.
+ There are ways to do this without a lock, but it's just too hard to maintain for it's value.
+
+ Although we're locking here, it's important to remember that we're *not* locking when iterating
+ and reading audio data since that'll be running on the audio thread. As a result we need to be
+ careful how we craft this so that we don't break iteration. What we're going to do is always
+ attach the new item so that it becomes the first item in the list. That way, as we're iterating
+ we won't break any links in the list and iteration will continue safely. The detaching case will
+ also be crafted in a way as to not break list iteration. It's important to remember to use
+ atomic exchanges here since no locking is happening on the audio thread during iteration.
+ */
+ ma_node_input_bus_lock(pInputBus);
+ {
+ ma_node_output_bus* pNewPrev = &pInputBus->head;
+ ma_node_output_bus* pNewNext = (ma_node_output_bus*)c89atomic_load_ptr(&pInputBus->head.pNext);
- result = ma_flac_init_internal(pConfig, pFlac);
- if (result != MA_SUCCESS) {
- return result;
- }
+ /* Update the local output bus. */
+ c89atomic_exchange_ptr(&pOutputBus->pPrev, pNewPrev);
+ c89atomic_exchange_ptr(&pOutputBus->pNext, pNewNext);
- #if !defined(MA_NO_FLAC)
- {
- drflac_allocation_callbacks flacAllocationCallbacks = drflac_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
+ /* Update the other output buses to point back to the local output bus. */
+ c89atomic_exchange_ptr(&pInputBus->head.pNext, pOutputBus); /* <-- This is where the output bus is actually attached to the input bus. */
- pFlac->dr = drflac_open_file_w(pFilePath, &flacAllocationCallbacks);
- if (pFlac->dr == NULL) {
- return MA_INVALID_FILE;
+ /* Do the previous pointer last. This is only used for detachment. */
+ if (pNewNext != NULL) {
+ c89atomic_exchange_ptr(&pNewNext->pPrev, pOutputBus);
+ }
}
+ ma_node_input_bus_unlock(pInputBus);
- return MA_SUCCESS;
- }
- #else
- {
- /* flac is disabled. */
- (void)pFilePath;
- (void)pAllocationCallbacks;
- return MA_NOT_IMPLEMENTED;
+ /*
+ Mark the node as attached last. This is used to controlling whether or the output bus will be
+ iterated on the audio thread. Mainly required for detachment purposes.
+ */
+ ma_node_output_bus_set_is_attached(pOutputBus, MA_TRUE);
}
- #endif
+ ma_node_output_bus_unlock(pOutputBus);
}
-MA_API ma_result ma_flac_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac)
+static ma_node_output_bus* ma_node_input_bus_next(ma_node_input_bus* pInputBus, ma_node_output_bus* pOutputBus)
{
- ma_result result;
+ ma_node_output_bus* pNext;
- result = ma_flac_init_internal(pConfig, pFlac);
- if (result != MA_SUCCESS) {
- return result;
+ MA_ASSERT(pInputBus != NULL);
+
+ if (pOutputBus == NULL) {
+ return NULL;
}
- #if !defined(MA_NO_FLAC)
+ ma_node_input_bus_next_begin(pInputBus);
{
- drflac_allocation_callbacks flacAllocationCallbacks = drflac_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
+ pNext = pOutputBus;
+ for (;;) {
+ pNext = (ma_node_output_bus*)c89atomic_load_ptr(&pNext->pNext);
+ if (pNext == NULL) {
+ break; /* Reached the end. */
+ }
- pFlac->dr = drflac_open_memory(pData, dataSize, &flacAllocationCallbacks);
- if (pFlac->dr == NULL) {
- return MA_INVALID_FILE;
+ if (ma_node_output_bus_is_attached(pNext) == MA_FALSE) {
+ continue; /* The node is not attached. Keep checking. */
+ }
+
+ /* The next node has been selected. */
+ break;
}
- return MA_SUCCESS;
- }
- #else
- {
- /* flac is disabled. */
- (void)pData;
- (void)dataSize;
- (void)pAllocationCallbacks;
- return MA_NOT_IMPLEMENTED;
+ /* We need to increment the reference count of the selected node. */
+ if (pNext != NULL) {
+ c89atomic_fetch_add_32(&pNext->refCount, 1);
+ }
+
+ /* The previous node is no longer being referenced. */
+ c89atomic_fetch_sub_32(&pOutputBus->refCount, 1);
}
- #endif
+ ma_node_input_bus_next_end(pInputBus);
+
+ return pNext;
}
-MA_API void ma_flac_uninit(ma_flac* pFlac, const ma_allocation_callbacks* pAllocationCallbacks)
+static ma_node_output_bus* ma_node_input_bus_first(ma_node_input_bus* pInputBus)
{
- if (pFlac == NULL) {
- return;
- }
-
- (void)pAllocationCallbacks;
+ return ma_node_input_bus_next(pInputBus, &pInputBus->head);
+}
- #if !defined(MA_NO_FLAC)
- {
- drflac_close(pFlac->dr);
- }
- #else
- {
- /* flac is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
- }
- #endif
- ma_data_source_uninit(&pFlac->ds);
-}
-MA_API ma_result ma_flac_read_pcm_frames(ma_flac* pFlac, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+static ma_result ma_node_input_bus_read_pcm_frames(ma_node* pInputNode, ma_node_input_bus* pInputBus, float* pFramesOut, ma_uint32 frameCount, ma_uint32* pFramesRead, ma_uint64 globalTime)
{
- if (pFlac == NULL) {
- return MA_INVALID_ARGS;
+ ma_result result = MA_SUCCESS;
+ ma_node_output_bus* pOutputBus;
+ ma_node_output_bus* pFirst;
+ ma_uint32 inputChannels;
+ ma_bool32 doesOutputBufferHaveContent = MA_FALSE;
+
+ /*
+ This will be called from the audio thread which means we can't be doing any locking. Basically,
+ this function will not perfom any locking, whereas attaching and detaching will, but crafted in
+ such a way that we don't need to perform any locking here. The important thing to remember is
+ to always iterate in a forward direction.
+
+ In order to process any data we need to first read from all input buses. That's where this
+ function comes in. This iterates over each of the attachments and accumulates/mixes them. We
+ also convert the channels to the nodes output channel count before mixing. We want to do this
+ channel conversion so that the caller of this function can invoke the processing callback
+ without having to do it themselves.
+
+ When we iterate over each of the attachments on the input bus, we need to read as much data as
+ we can from each of them so that we don't end up with holes between each of the attachments. To
+ do this, we need to read from each attachment in a loop and read as many frames as we can, up
+ to `frameCount`.
+ */
+ MA_ASSERT(pInputNode != NULL);
+ MA_ASSERT(pFramesRead != NULL); /* pFramesRead is critical and must always be specified. On input it's undefined and on output it'll be set to the number of frames actually read. */
+
+ *pFramesRead = 0; /* Safety. */
+
+ inputChannels = ma_node_input_bus_get_channels(pInputBus);
+
+ /*
+ We need to be careful with how we call ma_node_input_bus_first() and ma_node_input_bus_next(). They
+ are both critical to our lock-free thread-safety system. We can only call ma_node_input_bus_first()
+ once per iteration, however we have an optimization to checks whether or not it's the first item in
+ the list. We therefore need to store a pointer to the first item rather than repeatedly calling
+ ma_node_input_bus_first(). It's safe to keep hold of this pointer, so long as we don't dereference it
+ after calling ma_node_input_bus_next(), which we won't be.
+ */
+ pFirst = ma_node_input_bus_first(pInputBus);
+ if (pFirst == NULL) {
+ return MA_SUCCESS; /* No attachments. Read nothing. */
}
- #if !defined(MA_NO_FLAC)
- {
- /* We always use floating point format. */
- ma_result result = MA_SUCCESS; /* Must be initialized to MA_SUCCESS. */
- ma_uint64 totalFramesRead = 0;
- ma_format format;
+ for (pOutputBus = pFirst; pOutputBus != NULL; pOutputBus = ma_node_input_bus_next(pInputBus, pOutputBus)) {
+ ma_uint32 framesProcessed = 0;
+ ma_bool32 isSilentOutput = MA_FALSE;
- ma_flac_get_data_format(pFlac, &format, NULL, NULL, NULL, 0);
+ MA_ASSERT(pOutputBus->pNode != NULL);
- switch (format)
- {
- case ma_format_f32:
- {
- totalFramesRead = drflac_read_pcm_frames_f32(pFlac->dr, frameCount, (float*)pFramesOut);
- } break;
+ isSilentOutput = (((ma_node_base*)pOutputBus->pNode)->vtable->flags & MA_NODE_FLAG_SILENT_OUTPUT) != 0;
- case ma_format_s16:
- {
- totalFramesRead = drflac_read_pcm_frames_s16(pFlac->dr, frameCount, (drflac_int16*)pFramesOut);
- } break;
+ if (pFramesOut != NULL) {
+ /* Read. */
+ float temp[MA_DATA_CONVERTER_STACK_BUFFER_SIZE / sizeof(float)];
+ ma_uint32 tempCapInFrames = ma_countof(temp) / inputChannels;
+
+ while (framesProcessed < frameCount) {
+ float* pRunningFramesOut;
+ ma_uint32 framesToRead;
+ ma_uint32 framesJustRead;
+
+ framesToRead = frameCount - framesProcessed;
+ if (framesToRead > tempCapInFrames) {
+ framesToRead = tempCapInFrames;
+ }
- case ma_format_s32:
- {
- totalFramesRead = drflac_read_pcm_frames_s32(pFlac->dr, frameCount, (drflac_int32*)pFramesOut);
- } break;
+ pRunningFramesOut = ma_offset_pcm_frames_ptr_f32(pFramesOut, framesProcessed, inputChannels);
- case ma_format_u8:
- case ma_format_s24:
- case ma_format_unknown:
- default:
- {
- return MA_INVALID_OPERATION;
- };
- }
+ if (doesOutputBufferHaveContent == MA_FALSE) {
+ /* Fast path. First attachment. We just read straight into the output buffer (no mixing required). */
+ result = ma_node_read_pcm_frames(pOutputBus->pNode, pOutputBus->outputBusIndex, pRunningFramesOut, framesToRead, &framesJustRead, globalTime + framesProcessed);
+ } else {
+ /* Slow path. Not the first attachment. Mixing required. */
+ result = ma_node_read_pcm_frames(pOutputBus->pNode, pOutputBus->outputBusIndex, temp, framesToRead, &framesJustRead, globalTime + framesProcessed);
+ if (result == MA_SUCCESS || result == MA_AT_END) {
+ if (isSilentOutput == MA_FALSE) { /* Don't mix if the node outputs silence. */
+ ma_mix_pcm_frames_f32(pRunningFramesOut, temp, framesJustRead, inputChannels, /*volume*/1);
+ }
+ }
+ }
- /* In the future we'll update dr_flac to return MA_AT_END for us. */
- if (totalFramesRead == 0) {
- result = MA_AT_END;
- }
+ framesProcessed += framesJustRead;
- if (pFramesRead != NULL) {
- *pFramesRead = totalFramesRead;
+ /* If we reached the end or otherwise failed to read any data we need to finish up with this output node. */
+ if (result != MA_SUCCESS) {
+ break;
+ }
+
+ /* If we didn't read anything, abort so we don't get stuck in a loop. */
+ if (framesJustRead == 0) {
+ break;
+ }
+ }
+
+ /* If it's the first attachment we didn't do any mixing. Any leftover samples need to be silenced. */
+ if (pOutputBus == pFirst && framesProcessed < frameCount) {
+ ma_silence_pcm_frames(ma_offset_pcm_frames_ptr(pFramesOut, framesProcessed, ma_format_f32, inputChannels), (frameCount - framesProcessed), ma_format_f32, inputChannels);
+ }
+
+ if (isSilentOutput == MA_FALSE) {
+ doesOutputBufferHaveContent = MA_TRUE;
+ }
+ } else {
+ /* Seek. */
+ ma_node_read_pcm_frames(pOutputBus->pNode, pOutputBus->outputBusIndex, NULL, frameCount, &framesProcessed, globalTime);
}
+ }
- return result;
+ /* If we didn't output anything, output silence. */
+ if (doesOutputBufferHaveContent == MA_FALSE && pFramesOut != NULL) {
+ ma_silence_pcm_frames(pFramesOut, frameCount, ma_format_f32, inputChannels);
}
- #else
- {
- /* flac is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
- (void)pFramesOut;
- (void)frameCount;
- (void)pFramesRead;
+ /* In this path we always "process" the entire amount. */
+ *pFramesRead = frameCount;
- return MA_NOT_IMPLEMENTED;
- }
- #endif
+ return result;
}
-MA_API ma_result ma_flac_seek_to_pcm_frame(ma_flac* pFlac, ma_uint64 frameIndex)
+
+MA_API ma_node_config ma_node_config_init(void)
{
- if (pFlac == NULL) {
- return MA_INVALID_ARGS;
- }
+ ma_node_config config;
- #if !defined(MA_NO_FLAC)
- {
- drflac_bool32 flacResult;
-
- flacResult = drflac_seek_to_pcm_frame(pFlac->dr, frameIndex);
- if (flacResult != DRFLAC_TRUE) {
- return MA_ERROR;
- }
+ MA_ZERO_OBJECT(&config);
+ config.initialState = ma_node_state_started; /* Nodes are started by default. */
+ config.inputBusCount = MA_NODE_BUS_COUNT_UNKNOWN;
+ config.outputBusCount = MA_NODE_BUS_COUNT_UNKNOWN;
- return MA_SUCCESS;
- }
- #else
- {
- /* flac is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
+ return config;
+}
- (void)frameIndex;
- return MA_NOT_IMPLEMENTED;
- }
- #endif
-}
-MA_API ma_result ma_flac_get_data_format(ma_flac* pFlac, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
+static ma_result ma_node_detach_full(ma_node* pNode);
+
+static float* ma_node_get_cached_input_ptr(ma_node* pNode, ma_uint32 inputBusIndex)
{
- /* Defaults for safety. */
- if (pFormat != NULL) {
- *pFormat = ma_format_unknown;
- }
- if (pChannels != NULL) {
- *pChannels = 0;
- }
- if (pSampleRate != NULL) {
- *pSampleRate = 0;
- }
- if (pChannelMap != NULL) {
- MA_ZERO_MEMORY(pChannelMap, sizeof(*pChannelMap) * channelMapCap);
- }
+ ma_node_base* pNodeBase = (ma_node_base*)pNode;
+ ma_uint32 iInputBus;
+ float* pBasePtr;
- if (pFlac == NULL) {
- return MA_INVALID_OPERATION;
- }
+ MA_ASSERT(pNodeBase != NULL);
- if (pFormat != NULL) {
- *pFormat = pFlac->format;
+ /* Input data is stored at the front of the buffer. */
+ pBasePtr = pNodeBase->pCachedData;
+ for (iInputBus = 0; iInputBus < inputBusIndex; iInputBus += 1) {
+ pBasePtr += pNodeBase->cachedDataCapInFramesPerBus * ma_node_input_bus_get_channels(&pNodeBase->pInputBuses[iInputBus]);
}
- #if !defined(MA_NO_FLAC)
- {
- if (pChannels != NULL) {
- *pChannels = pFlac->dr->channels;
- }
+ return pBasePtr;
+}
- if (pSampleRate != NULL) {
- *pSampleRate = pFlac->dr->sampleRate;
- }
+static float* ma_node_get_cached_output_ptr(ma_node* pNode, ma_uint32 outputBusIndex)
+{
+ ma_node_base* pNodeBase = (ma_node_base*)pNode;
+ ma_uint32 iInputBus;
+ ma_uint32 iOutputBus;
+ float* pBasePtr;
- if (pChannelMap != NULL) {
- ma_get_standard_channel_map(ma_standard_channel_map_microsoft, (ma_uint32)ma_min(pFlac->dr->channels, channelMapCap), pChannelMap);
- }
+ MA_ASSERT(pNodeBase != NULL);
- return MA_SUCCESS;
+ /* Cached output data starts after the input data. */
+ pBasePtr = pNodeBase->pCachedData;
+ for (iInputBus = 0; iInputBus < ma_node_get_input_bus_count(pNodeBase); iInputBus += 1) {
+ pBasePtr += pNodeBase->cachedDataCapInFramesPerBus * ma_node_input_bus_get_channels(&pNodeBase->pInputBuses[iInputBus]);
}
- #else
- {
- /* flac is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
- return MA_NOT_IMPLEMENTED;
+
+ for (iOutputBus = 0; iOutputBus < outputBusIndex; iOutputBus += 1) {
+ pBasePtr += pNodeBase->cachedDataCapInFramesPerBus * ma_node_output_bus_get_channels(&pNodeBase->pOutputBuses[iOutputBus]);
}
- #endif
+
+ return pBasePtr;
}
-MA_API ma_result ma_flac_get_cursor_in_pcm_frames(ma_flac* pFlac, ma_uint64* pCursor)
+
+typedef struct
{
- if (pCursor == NULL) {
- return MA_INVALID_ARGS;
+ size_t sizeInBytes;
+ size_t inputBusOffset;
+ size_t outputBusOffset;
+ size_t cachedDataOffset;
+ ma_uint32 inputBusCount; /* So it doesn't have to be calculated twice. */
+ ma_uint32 outputBusCount; /* So it doesn't have to be calculated twice. */
+} ma_node_heap_layout;
+
+static ma_result ma_node_translate_bus_counts(const ma_node_config* pConfig, ma_uint32* pInputBusCount, ma_uint32* pOutputBusCount)
+{
+ ma_uint32 inputBusCount;
+ ma_uint32 outputBusCount;
+
+ MA_ASSERT(pConfig != NULL);
+ MA_ASSERT(pInputBusCount != NULL);
+ MA_ASSERT(pOutputBusCount != NULL);
+
+ /* Bus counts are determined by the vtable, unless they're set to `MA_NODE_BUS_COUNT_UNKNWON`, in which case they're taken from the config. */
+ if (pConfig->vtable->inputBusCount == MA_NODE_BUS_COUNT_UNKNOWN) {
+ inputBusCount = pConfig->inputBusCount;
+ } else {
+ inputBusCount = pConfig->vtable->inputBusCount;
+
+ if (pConfig->inputBusCount != MA_NODE_BUS_COUNT_UNKNOWN && pConfig->inputBusCount != pConfig->vtable->inputBusCount) {
+ return MA_INVALID_ARGS; /* Invalid configuration. You must not specify a conflicting bus count between the node's config and the vtable. */
+ }
}
- *pCursor = 0; /* Safety. */
+ if (pConfig->vtable->outputBusCount == MA_NODE_BUS_COUNT_UNKNOWN) {
+ outputBusCount = pConfig->outputBusCount;
+ } else {
+ outputBusCount = pConfig->vtable->outputBusCount;
- if (pFlac == NULL) {
+ if (pConfig->outputBusCount != MA_NODE_BUS_COUNT_UNKNOWN && pConfig->outputBusCount != pConfig->vtable->outputBusCount) {
+ return MA_INVALID_ARGS; /* Invalid configuration. You must not specify a conflicting bus count between the node's config and the vtable. */
+ }
+ }
+
+ /* Bus counts must be within limits. */
+ if (inputBusCount > MA_MAX_NODE_BUS_COUNT || outputBusCount > MA_MAX_NODE_BUS_COUNT) {
return MA_INVALID_ARGS;
}
- #if !defined(MA_NO_FLAC)
- {
- *pCursor = pFlac->dr->currentPCMFrame;
- return MA_SUCCESS;
+ /* We must have channel counts for each bus. */
+ if ((inputBusCount > 0 && pConfig->pInputChannels == NULL) || (outputBusCount > 0 && pConfig->pOutputChannels == NULL)) {
+ return MA_INVALID_ARGS; /* You must specify channel counts for each input and output bus. */
}
- #else
- {
- /* flac is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
- return MA_NOT_IMPLEMENTED;
+
+
+ /* Some special rules for passthrough nodes. */
+ if ((pConfig->vtable->flags & MA_NODE_FLAG_PASSTHROUGH) != 0) {
+ if (pConfig->vtable->inputBusCount != 1 || pConfig->vtable->outputBusCount != 1) {
+ return MA_INVALID_ARGS; /* Passthrough nodes must have exactly 1 input bus and 1 output bus. */
+ }
+
+ if (pConfig->pInputChannels[0] != pConfig->pOutputChannels[0]) {
+ return MA_INVALID_ARGS; /* Passthrough nodes must have the same number of channels between input and output nodes. */
+ }
}
- #endif
+
+
+ *pInputBusCount = inputBusCount;
+ *pOutputBusCount = outputBusCount;
+
+ return MA_SUCCESS;
}
-MA_API ma_result ma_flac_get_length_in_pcm_frames(ma_flac* pFlac, ma_uint64* pLength)
+static ma_result ma_node_get_heap_layout(ma_node_graph* pNodeGraph, const ma_node_config* pConfig, ma_node_heap_layout* pHeapLayout)
{
- if (pLength == NULL) {
- return MA_INVALID_ARGS;
- }
+ ma_result result;
+ ma_uint32 inputBusCount;
+ ma_uint32 outputBusCount;
- *pLength = 0; /* Safety. */
+ MA_ASSERT(pHeapLayout != NULL);
- if (pFlac == NULL) {
+ MA_ZERO_OBJECT(pHeapLayout);
+
+ if (pConfig == NULL || pConfig->vtable == NULL || pConfig->vtable->onProcess == NULL) {
return MA_INVALID_ARGS;
}
- #if !defined(MA_NO_FLAC)
- {
- *pLength = pFlac->dr->totalPCMFrameCount;
+ result = ma_node_translate_bus_counts(pConfig, &inputBusCount, &outputBusCount);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- return MA_SUCCESS;
+ pHeapLayout->sizeInBytes = 0;
+
+ /* Input buses. */
+ if (inputBusCount > MA_MAX_NODE_LOCAL_BUS_COUNT) {
+ pHeapLayout->inputBusOffset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += ma_align_64(sizeof(ma_node_input_bus) * inputBusCount);
+ } else {
+ pHeapLayout->inputBusOffset = MA_SIZE_MAX; /* MA_SIZE_MAX indicates that no heap allocation is required for the input bus. */
}
- #else
- {
- /* flac is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
- return MA_NOT_IMPLEMENTED;
+
+ /* Output buses. */
+ if (outputBusCount > MA_MAX_NODE_LOCAL_BUS_COUNT) {
+ pHeapLayout->outputBusOffset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += ma_align_64(sizeof(ma_node_output_bus) * outputBusCount);
+ } else {
+ pHeapLayout->outputBusOffset = MA_SIZE_MAX;
}
- #endif
-}
+ /*
+ Cached audio data.
-static ma_result ma_decoding_backend_init__flac(void* pUserData, ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
-{
- ma_result result;
- ma_flac* pFlac;
+ We need to allocate memory for a caching both input and output data. We have an optimization
+ where no caching is necessary for specific conditions:
- (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
+ - The node has 0 inputs and 1 output.
- /* For now we're just allocating the decoder backend on the heap. */
- pFlac = (ma_flac*)ma_malloc(sizeof(*pFlac), pAllocationCallbacks);
- if (pFlac == NULL) {
- return MA_OUT_OF_MEMORY;
- }
+ When a node meets the above conditions, no cache is allocated.
- result = ma_flac_init(onRead, onSeek, onTell, pReadSeekTellUserData, pConfig, pAllocationCallbacks, pFlac);
- if (result != MA_SUCCESS) {
- ma_free(pFlac, pAllocationCallbacks);
- return result;
+ The size choice for this buffer is a little bit finicky. We don't want to be too wasteful by
+ allocating too much, but at the same time we want it be large enough so that enough frames can
+ be processed for each call to ma_node_read_pcm_frames() so that it keeps things efficient. For
+ now I'm going with 10ms @ 48K which is 480 frames per bus. This is configurable at compile
+ time. It might also be worth investigating whether or not this can be configured at run time.
+ */
+ if (inputBusCount == 0 && outputBusCount == 1) {
+ /* Fast path. No cache needed. */
+ pHeapLayout->cachedDataOffset = MA_SIZE_MAX;
+ } else {
+ /* Slow path. Cache needed. */
+ size_t cachedDataSizeInBytes = 0;
+ ma_uint32 iBus;
+
+ for (iBus = 0; iBus < inputBusCount; iBus += 1) {
+ cachedDataSizeInBytes += pNodeGraph->nodeCacheCapInFrames * ma_get_bytes_per_frame(ma_format_f32, pConfig->pInputChannels[iBus]);
+ }
+
+ for (iBus = 0; iBus < outputBusCount; iBus += 1) {
+ cachedDataSizeInBytes += pNodeGraph->nodeCacheCapInFrames * ma_get_bytes_per_frame(ma_format_f32, pConfig->pOutputChannels[iBus]);
+ }
+
+ pHeapLayout->cachedDataOffset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += ma_align_64(cachedDataSizeInBytes);
}
- *ppBackend = pFlac;
+
+ /*
+ Not technically part of the heap, but we can output the input and output bus counts so we can
+ avoid a redundant call to ma_node_translate_bus_counts().
+ */
+ pHeapLayout->inputBusCount = inputBusCount;
+ pHeapLayout->outputBusCount = outputBusCount;
+
+ /* Make sure allocation size is aligned. */
+ pHeapLayout->sizeInBytes = ma_align_64(pHeapLayout->sizeInBytes);
return MA_SUCCESS;
}
-static ma_result ma_decoding_backend_init_file__flac(void* pUserData, const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
+MA_API ma_result ma_node_get_heap_size(ma_node_graph* pNodeGraph, const ma_node_config* pConfig, size_t* pHeapSizeInBytes)
{
ma_result result;
- ma_flac* pFlac;
-
- (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
+ ma_node_heap_layout heapLayout;
- /* For now we're just allocating the decoder backend on the heap. */
- pFlac = (ma_flac*)ma_malloc(sizeof(*pFlac), pAllocationCallbacks);
- if (pFlac == NULL) {
- return MA_OUT_OF_MEMORY;
+ if (pHeapSizeInBytes == NULL) {
+ return MA_INVALID_ARGS;
}
- result = ma_flac_init_file(pFilePath, pConfig, pAllocationCallbacks, pFlac);
+ *pHeapSizeInBytes = 0;
+
+ result = ma_node_get_heap_layout(pNodeGraph, pConfig, &heapLayout);
if (result != MA_SUCCESS) {
- ma_free(pFlac, pAllocationCallbacks);
return result;
}
- *ppBackend = pFlac;
+ *pHeapSizeInBytes = heapLayout.sizeInBytes;
return MA_SUCCESS;
}
-static ma_result ma_decoding_backend_init_file_w__flac(void* pUserData, const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
+MA_API ma_result ma_node_init_preallocated(ma_node_graph* pNodeGraph, const ma_node_config* pConfig, void* pHeap, ma_node* pNode)
{
+ ma_node_base* pNodeBase = (ma_node_base*)pNode;
ma_result result;
- ma_flac* pFlac;
+ ma_node_heap_layout heapLayout;
+ ma_uint32 iInputBus;
+ ma_uint32 iOutputBus;
- (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
-
- /* For now we're just allocating the decoder backend on the heap. */
- pFlac = (ma_flac*)ma_malloc(sizeof(*pFlac), pAllocationCallbacks);
- if (pFlac == NULL) {
- return MA_OUT_OF_MEMORY;
+ if (pNodeBase == NULL) {
+ return MA_INVALID_ARGS;
}
- result = ma_flac_init_file_w(pFilePath, pConfig, pAllocationCallbacks, pFlac);
+ MA_ZERO_OBJECT(pNodeBase);
+
+ result = ma_node_get_heap_layout(pNodeGraph, pConfig, &heapLayout);
if (result != MA_SUCCESS) {
- ma_free(pFlac, pAllocationCallbacks);
return result;
}
- *ppBackend = pFlac;
+ pNodeBase->_pHeap = pHeap;
+ MA_ZERO_MEMORY(pHeap, heapLayout.sizeInBytes);
- return MA_SUCCESS;
-}
+ pNodeBase->pNodeGraph = pNodeGraph;
+ pNodeBase->vtable = pConfig->vtable;
+ pNodeBase->state = pConfig->initialState;
+ pNodeBase->stateTimes[ma_node_state_started] = 0;
+ pNodeBase->stateTimes[ma_node_state_stopped] = (ma_uint64)(ma_int64)-1; /* Weird casting for VC6 compatibility. */
+ pNodeBase->inputBusCount = heapLayout.inputBusCount;
+ pNodeBase->outputBusCount = heapLayout.outputBusCount;
-static ma_result ma_decoding_backend_init_memory__flac(void* pUserData, const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
-{
- ma_result result;
- ma_flac* pFlac;
+ if (heapLayout.inputBusOffset != MA_SIZE_MAX) {
+ pNodeBase->pInputBuses = (ma_node_input_bus*)ma_offset_ptr(pHeap, heapLayout.inputBusOffset);
+ } else {
+ pNodeBase->pInputBuses = pNodeBase->_inputBuses;
+ }
- (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
+ if (heapLayout.outputBusOffset != MA_SIZE_MAX) {
+ pNodeBase->pOutputBuses = (ma_node_output_bus*)ma_offset_ptr(pHeap, heapLayout.inputBusOffset);
+ } else {
+ pNodeBase->pOutputBuses = pNodeBase->_outputBuses;
+ }
- /* For now we're just allocating the decoder backend on the heap. */
- pFlac = (ma_flac*)ma_malloc(sizeof(*pFlac), pAllocationCallbacks);
- if (pFlac == NULL) {
- return MA_OUT_OF_MEMORY;
+ if (heapLayout.cachedDataOffset != MA_SIZE_MAX) {
+ pNodeBase->pCachedData = (float*)ma_offset_ptr(pHeap, heapLayout.cachedDataOffset);
+ pNodeBase->cachedDataCapInFramesPerBus = pNodeGraph->nodeCacheCapInFrames;
+ } else {
+ pNodeBase->pCachedData = NULL;
}
- result = ma_flac_init_memory(pData, dataSize, pConfig, pAllocationCallbacks, pFlac);
- if (result != MA_SUCCESS) {
- ma_free(pFlac, pAllocationCallbacks);
- return result;
+
+
+ /* We need to run an initialization step for each input and output bus. */
+ for (iInputBus = 0; iInputBus < ma_node_get_input_bus_count(pNodeBase); iInputBus += 1) {
+ result = ma_node_input_bus_init(pConfig->pInputChannels[iInputBus], &pNodeBase->pInputBuses[iInputBus]);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
}
- *ppBackend = pFlac;
+ for (iOutputBus = 0; iOutputBus < ma_node_get_output_bus_count(pNodeBase); iOutputBus += 1) {
+ result = ma_node_output_bus_init(pNodeBase, iOutputBus, pConfig->pOutputChannels[iOutputBus], &pNodeBase->pOutputBuses[iOutputBus]);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ }
- return MA_SUCCESS;
-}
-static void ma_decoding_backend_uninit__flac(void* pUserData, ma_data_source* pBackend, const ma_allocation_callbacks* pAllocationCallbacks)
-{
- ma_flac* pFlac = (ma_flac*)pBackend;
+ /* The cached data needs to be initialized to silence (or a sine wave tone if we're debugging). */
+ if (pNodeBase->pCachedData != NULL) {
+ ma_uint32 iBus;
- (void)pUserData;
+ #if 1 /* Toggle this between 0 and 1 to turn debugging on or off. 1 = fill with a sine wave for debugging; 0 = fill with silence. */
+ /* For safety we'll go ahead and default the buffer to silence. */
+ for (iBus = 0; iBus < ma_node_get_input_bus_count(pNodeBase); iBus += 1) {
+ ma_silence_pcm_frames(ma_node_get_cached_input_ptr(pNode, iBus), pNodeBase->cachedDataCapInFramesPerBus, ma_format_f32, ma_node_input_bus_get_channels(&pNodeBase->pInputBuses[iBus]));
+ }
+ for (iBus = 0; iBus < ma_node_get_output_bus_count(pNodeBase); iBus += 1) {
+ ma_silence_pcm_frames(ma_node_get_cached_output_ptr(pNode, iBus), pNodeBase->cachedDataCapInFramesPerBus, ma_format_f32, ma_node_output_bus_get_channels(&pNodeBase->pOutputBuses[iBus]));
+ }
+ #else
+ /* For debugging. Default to a sine wave. */
+ for (iBus = 0; iBus < ma_node_get_input_bus_count(pNodeBase); iBus += 1) {
+ ma_debug_fill_pcm_frames_with_sine_wave(ma_node_get_cached_input_ptr(pNode, iBus), pNodeBase->cachedDataCapInFramesPerBus, ma_format_f32, ma_node_input_bus_get_channels(&pNodeBase->pInputBuses[iBus]), 48000);
+ }
+ for (iBus = 0; iBus < ma_node_get_output_bus_count(pNodeBase); iBus += 1) {
+ ma_debug_fill_pcm_frames_with_sine_wave(ma_node_get_cached_output_ptr(pNode, iBus), pNodeBase->cachedDataCapInFramesPerBus, ma_format_f32, ma_node_output_bus_get_channels(&pNodeBase->pOutputBuses[iBus]), 48000);
+ }
+ #endif
+ }
- ma_flac_uninit(pFlac, pAllocationCallbacks);
- ma_free(pFlac, pAllocationCallbacks);
+ return MA_SUCCESS;
}
-static ma_result ma_decoding_backend_get_channel_map__flac(void* pUserData, ma_data_source* pBackend, ma_channel* pChannelMap, size_t channelMapCap)
+MA_API ma_result ma_node_init(ma_node_graph* pNodeGraph, const ma_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_node* pNode)
{
- ma_flac* pFlac = (ma_flac*)pBackend;
+ ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
- (void)pUserData;
+ result = ma_node_get_heap_size(pNodeGraph, pConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- return ma_flac_get_data_format(pFlac, NULL, NULL, NULL, pChannelMap, channelMapCap);
-}
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+ } else {
+ pHeap = NULL;
+ }
-static ma_decoding_backend_vtable g_ma_decoding_backend_vtable_flac =
-{
- ma_decoding_backend_init__flac,
- ma_decoding_backend_init_file__flac,
- ma_decoding_backend_init_file_w__flac,
- ma_decoding_backend_init_memory__flac,
- ma_decoding_backend_uninit__flac,
- ma_decoding_backend_get_channel_map__flac
-};
+ result = ma_node_init_preallocated(pNodeGraph, pConfig, pHeap, pNode);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
+ return result;
+ }
-static ma_result ma_decoder_init_flac__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
-{
- return ma_decoder_init_from_vtable(&g_ma_decoding_backend_vtable_flac, NULL, pConfig, pDecoder);
+ ((ma_node_base*)pNode)->_ownsHeap = MA_TRUE;
+ return MA_SUCCESS;
}
-#endif /* dr_flac_h */
-
-/* MP3 */
-#ifdef dr_mp3_h
-#define MA_HAS_MP3
-typedef struct
+MA_API void ma_node_uninit(ma_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_data_source_base ds;
- ma_read_proc onRead;
- ma_seek_proc onSeek;
- ma_tell_proc onTell;
- void* pReadSeekTellUserData;
- ma_format format; /* Can be f32 or s16. */
-#if !defined(MA_NO_MP3)
- drmp3 dr;
-#endif
-} ma_mp3;
+ ma_node_base* pNodeBase = (ma_node_base*)pNode;
-MA_API ma_result ma_mp3_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3);
-MA_API ma_result ma_mp3_init_file(const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3);
-MA_API ma_result ma_mp3_init_file_w(const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3);
-MA_API ma_result ma_mp3_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3);
-MA_API void ma_mp3_uninit(ma_mp3* pMP3, const ma_allocation_callbacks* pAllocationCallbacks);
-MA_API ma_result ma_mp3_read_pcm_frames(ma_mp3* pMP3, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
-MA_API ma_result ma_mp3_seek_to_pcm_frame(ma_mp3* pMP3, ma_uint64 frameIndex);
-MA_API ma_result ma_mp3_get_data_format(ma_mp3* pMP3, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
-MA_API ma_result ma_mp3_get_cursor_in_pcm_frames(ma_mp3* pMP3, ma_uint64* pCursor);
-MA_API ma_result ma_mp3_get_length_in_pcm_frames(ma_mp3* pMP3, ma_uint64* pLength);
+ if (pNodeBase == NULL) {
+ return;
+ }
+ /*
+ The first thing we need to do is fully detach the node. This will detach all inputs and
+ outputs. We need to do this first because it will sever the connection with the node graph and
+ allow us to complete uninitialization without needing to worry about thread-safety with the
+ audio thread. The detachment process will wait for any local processing of the node to finish.
+ */
+ ma_node_detach_full(pNode);
-static ma_result ma_mp3_ds_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
-{
- return ma_mp3_read_pcm_frames((ma_mp3*)pDataSource, pFramesOut, frameCount, pFramesRead);
+ /*
+ At this point the node should be completely unreferenced by the node graph and we can finish up
+ the uninitialization process without needing to worry about thread-safety.
+ */
+ if (pNodeBase->_ownsHeap) {
+ ma_free(pNodeBase->_pHeap, pAllocationCallbacks);
+ }
}
-static ma_result ma_mp3_ds_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
+MA_API ma_node_graph* ma_node_get_node_graph(const ma_node* pNode)
{
- return ma_mp3_seek_to_pcm_frame((ma_mp3*)pDataSource, frameIndex);
-}
+ if (pNode == NULL) {
+ return NULL;
+ }
-static ma_result ma_mp3_ds_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate)
-{
- return ma_mp3_get_data_format((ma_mp3*)pDataSource, pFormat, pChannels, pSampleRate, NULL, 0);
+ return ((const ma_node_base*)pNode)->pNodeGraph;
}
-static ma_result ma_mp3_ds_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
+MA_API ma_uint32 ma_node_get_input_bus_count(const ma_node* pNode)
{
- return ma_mp3_get_cursor_in_pcm_frames((ma_mp3*)pDataSource, pCursor);
-}
+ if (pNode == NULL) {
+ return 0;
+ }
-static ma_result ma_mp3_ds_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
-{
- return ma_mp3_get_length_in_pcm_frames((ma_mp3*)pDataSource, pLength);
+ return ((ma_node_base*)pNode)->inputBusCount;
}
-static ma_data_source_vtable g_ma_mp3_ds_vtable =
-{
- ma_mp3_ds_read,
- ma_mp3_ds_seek,
- NULL, /* onMap() */
- NULL, /* onUnmap() */
- ma_mp3_ds_get_data_format,
- ma_mp3_ds_get_cursor,
- ma_mp3_ds_get_length
-};
-
-
-#if !defined(MA_NO_MP3)
-static drmp3_allocation_callbacks drmp3_allocation_callbacks_from_miniaudio(const ma_allocation_callbacks* pAllocationCallbacks)
+MA_API ma_uint32 ma_node_get_output_bus_count(const ma_node* pNode)
{
- drmp3_allocation_callbacks callbacks;
-
- if (pAllocationCallbacks != NULL) {
- callbacks.onMalloc = pAllocationCallbacks->onMalloc;
- callbacks.onRealloc = pAllocationCallbacks->onRealloc;
- callbacks.onFree = pAllocationCallbacks->onFree;
- callbacks.pUserData = pAllocationCallbacks->pUserData;
- } else {
- callbacks.onMalloc = ma__malloc_default;
- callbacks.onRealloc = ma__realloc_default;
- callbacks.onFree = ma__free_default;
- callbacks.pUserData = NULL;
+ if (pNode == NULL) {
+ return 0;
}
- return callbacks;
+ return ((ma_node_base*)pNode)->outputBusCount;
}
-static size_t ma_mp3_dr_callback__read(void* pUserData, void* pBufferOut, size_t bytesToRead)
+
+MA_API ma_uint32 ma_node_get_input_channels(const ma_node* pNode, ma_uint32 inputBusIndex)
{
- ma_mp3* pMP3 = (ma_mp3*)pUserData;
- ma_result result;
- size_t bytesRead;
+ const ma_node_base* pNodeBase = (const ma_node_base*)pNode;
- MA_ASSERT(pMP3 != NULL);
+ if (pNode == NULL) {
+ return 0;
+ }
- result = pMP3->onRead(pMP3->pReadSeekTellUserData, pBufferOut, bytesToRead, &bytesRead);
- (void)result;
+ if (inputBusIndex >= ma_node_get_input_bus_count(pNode)) {
+ return 0; /* Invalid bus index. */
+ }
- return bytesRead;
+ return ma_node_input_bus_get_channels(&pNodeBase->pInputBuses[inputBusIndex]);
}
-static drmp3_bool32 ma_mp3_dr_callback__seek(void* pUserData, int offset, drmp3_seek_origin origin)
+MA_API ma_uint32 ma_node_get_output_channels(const ma_node* pNode, ma_uint32 outputBusIndex)
{
- ma_mp3* pMP3 = (ma_mp3*)pUserData;
- ma_result result;
- ma_seek_origin maSeekOrigin;
-
- MA_ASSERT(pMP3 != NULL);
+ const ma_node_base* pNodeBase = (const ma_node_base*)pNode;
- maSeekOrigin = ma_seek_origin_start;
- if (origin == drmp3_seek_origin_current) {
- maSeekOrigin = ma_seek_origin_current;
+ if (pNode == NULL) {
+ return 0;
}
- result = pMP3->onSeek(pMP3->pReadSeekTellUserData, offset, maSeekOrigin);
- if (result != MA_SUCCESS) {
- return MA_FALSE;
+ if (outputBusIndex >= ma_node_get_output_bus_count(pNode)) {
+ return 0; /* Invalid bus index. */
}
- return MA_TRUE;
+ return ma_node_output_bus_get_channels(&pNodeBase->pOutputBuses[outputBusIndex]);
}
-#endif
-static ma_result ma_mp3_init_internal(const ma_decoding_backend_config* pConfig, ma_mp3* pMP3)
+
+static ma_result ma_node_detach_full(ma_node* pNode)
{
- ma_result result;
- ma_data_source_config dataSourceConfig;
+ ma_node_base* pNodeBase = (ma_node_base*)pNode;
+ ma_uint32 iInputBus;
- if (pMP3 == NULL) {
+ if (pNodeBase == NULL) {
return MA_INVALID_ARGS;
}
- MA_ZERO_OBJECT(pMP3);
- pMP3->format = ma_format_f32; /* f32 by default. */
+ /*
+ Make sure the node is completely detached first. This will not return until the output bus is
+ guaranteed to no longer be referenced by the audio thread.
+ */
+ ma_node_detach_all_output_buses(pNode);
- if (pConfig != NULL && (pConfig->preferredFormat == ma_format_f32 || pConfig->preferredFormat == ma_format_s16)) {
- pMP3->format = pConfig->preferredFormat;
- } else {
- /* Getting here means something other than f32 and s16 was specified. Just leave this unset to use the default format. */
- }
+ /*
+ At this point all output buses will have been detached from the graph and we can be guaranteed
+ that none of it's input nodes will be getting processed by the graph. We can detach these
+ without needing to worry about the audio thread touching them.
+ */
+ for (iInputBus = 0; iInputBus < ma_node_get_input_bus_count(pNode); iInputBus += 1) {
+ ma_node_input_bus* pInputBus;
+ ma_node_output_bus* pOutputBus;
- dataSourceConfig = ma_data_source_config_init();
- dataSourceConfig.vtable = &g_ma_mp3_ds_vtable;
+ pInputBus = &pNodeBase->pInputBuses[iInputBus];
- result = ma_data_source_init(&dataSourceConfig, &pMP3->ds);
- if (result != MA_SUCCESS) {
- return result; /* Failed to initialize the base data source. */
+ /*
+ This is important. We cannot be using ma_node_input_bus_first() or ma_node_input_bus_next(). Those
+ functions are specifically for the audio thread. We'll instead just manually iterate using standard
+ linked list logic. We don't need to worry about the audio thread referencing these because the step
+ above severed the connection to the graph.
+ */
+ for (pOutputBus = (ma_node_output_bus*)c89atomic_load_ptr(&pInputBus->head.pNext); pOutputBus != NULL; pOutputBus = (ma_node_output_bus*)c89atomic_load_ptr(&pOutputBus->pNext)) {
+ ma_node_detach_output_bus(pOutputBus->pNode, pOutputBus->outputBusIndex); /* This won't do any waiting in practice and should be efficient. */
+ }
}
return MA_SUCCESS;
}
-MA_API ma_result ma_mp3_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3)
+MA_API ma_result ma_node_detach_output_bus(ma_node* pNode, ma_uint32 outputBusIndex)
{
- ma_result result;
+ ma_result result = MA_SUCCESS;
+ ma_node_base* pNodeBase = (ma_node_base*)pNode;
+ ma_node_base* pInputNodeBase;
- result = ma_mp3_init_internal(pConfig, pMP3);
- if (result != MA_SUCCESS) {
- return result;
+ if (pNode == NULL) {
+ return MA_INVALID_ARGS;
}
- if (onRead == NULL || onSeek == NULL) {
- return MA_INVALID_ARGS; /* onRead and onSeek are mandatory. */
+ if (outputBusIndex >= ma_node_get_output_bus_count(pNode)) {
+ return MA_INVALID_ARGS; /* Invalid output bus index. */
}
- pMP3->onRead = onRead;
- pMP3->onSeek = onSeek;
- pMP3->onTell = onTell;
- pMP3->pReadSeekTellUserData = pReadSeekTellUserData;
-
- #if !defined(MA_NO_MP3)
+ /* We need to lock the output bus because we need to inspect the input node and grab it's input bus. */
+ ma_node_output_bus_lock(&pNodeBase->pOutputBuses[outputBusIndex]);
{
- drmp3_allocation_callbacks mp3AllocationCallbacks = drmp3_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
- drmp3_bool32 mp3Result;
-
- mp3Result = drmp3_init(&pMP3->dr, ma_mp3_dr_callback__read, ma_mp3_dr_callback__seek, pMP3, &mp3AllocationCallbacks);
- if (mp3Result != MA_TRUE) {
- return MA_INVALID_FILE;
+ pInputNodeBase = (ma_node_base*)pNodeBase->pOutputBuses[outputBusIndex].pInputNode;
+ if (pInputNodeBase != NULL) {
+ ma_node_input_bus_detach__no_output_bus_lock(&pInputNodeBase->pInputBuses[pNodeBase->pOutputBuses[outputBusIndex].inputNodeInputBusIndex], &pNodeBase->pOutputBuses[outputBusIndex]);
}
-
- return MA_SUCCESS;
- }
- #else
- {
- /* mp3 is disabled. */
- (void)pAllocationCallbacks;
- return MA_NOT_IMPLEMENTED;
}
- #endif
+ ma_node_output_bus_unlock(&pNodeBase->pOutputBuses[outputBusIndex]);
+
+ return result;
}
-MA_API ma_result ma_mp3_init_file(const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3)
+MA_API ma_result ma_node_detach_all_output_buses(ma_node* pNode)
{
- ma_result result;
+ ma_uint32 iOutputBus;
- result = ma_mp3_init_internal(pConfig, pMP3);
- if (result != MA_SUCCESS) {
- return result;
+ if (pNode == NULL) {
+ return MA_INVALID_ARGS;
}
- #if !defined(MA_NO_MP3)
- {
- drmp3_allocation_callbacks mp3AllocationCallbacks = drmp3_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
- drmp3_bool32 mp3Result;
-
- mp3Result = drmp3_init_file(&pMP3->dr, pFilePath, &mp3AllocationCallbacks);
- if (mp3Result != MA_TRUE) {
- return MA_INVALID_FILE;
- }
-
- return MA_SUCCESS;
- }
- #else
- {
- /* mp3 is disabled. */
- (void)pFilePath;
- (void)pAllocationCallbacks;
- return MA_NOT_IMPLEMENTED;
+ for (iOutputBus = 0; iOutputBus < ma_node_get_output_bus_count(pNode); iOutputBus += 1) {
+ ma_node_detach_output_bus(pNode, iOutputBus);
}
- #endif
+
+ return MA_SUCCESS;
}
-MA_API ma_result ma_mp3_init_file_w(const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3)
+MA_API ma_result ma_node_attach_output_bus(ma_node* pNode, ma_uint32 outputBusIndex, ma_node* pOtherNode, ma_uint32 otherNodeInputBusIndex)
{
- ma_result result;
+ ma_node_base* pNodeBase = (ma_node_base*)pNode;
+ ma_node_base* pOtherNodeBase = (ma_node_base*)pOtherNode;
- result = ma_mp3_init_internal(pConfig, pMP3);
- if (result != MA_SUCCESS) {
- return result;
+ if (pNodeBase == NULL || pOtherNodeBase == NULL) {
+ return MA_INVALID_ARGS;
}
- #if !defined(MA_NO_MP3)
- {
- drmp3_allocation_callbacks mp3AllocationCallbacks = drmp3_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
- drmp3_bool32 mp3Result;
-
- mp3Result = drmp3_init_file_w(&pMP3->dr, pFilePath, &mp3AllocationCallbacks);
- if (mp3Result != MA_TRUE) {
- return MA_INVALID_FILE;
- }
+ if (pNodeBase == pOtherNodeBase) {
+ return MA_INVALID_OPERATION; /* Cannot attach a node to itself. */
+ }
- return MA_SUCCESS;
+ if (outputBusIndex >= ma_node_get_output_bus_count(pNode) || otherNodeInputBusIndex >= ma_node_get_input_bus_count(pOtherNode)) {
+ return MA_INVALID_OPERATION; /* Invalid bus index. */
}
- #else
- {
- /* mp3 is disabled. */
- (void)pFilePath;
- (void)pAllocationCallbacks;
- return MA_NOT_IMPLEMENTED;
+
+ /* The output channel count of the output node must be the same as the input channel count of the input node. */
+ if (ma_node_get_output_channels(pNode, outputBusIndex) != ma_node_get_input_channels(pOtherNode, otherNodeInputBusIndex)) {
+ return MA_INVALID_OPERATION; /* Channel count is incompatible. */
}
- #endif
+
+ /* This will deal with detaching if the output bus is already attached to something. */
+ ma_node_input_bus_attach(&pOtherNodeBase->pInputBuses[otherNodeInputBusIndex], &pNodeBase->pOutputBuses[outputBusIndex], pOtherNode, otherNodeInputBusIndex);
+
+ return MA_SUCCESS;
}
-MA_API ma_result ma_mp3_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3)
+MA_API ma_result ma_node_set_output_bus_volume(ma_node* pNode, ma_uint32 outputBusIndex, float volume)
{
- ma_result result;
+ ma_node_base* pNodeBase = (ma_node_base*)pNode;
- result = ma_mp3_init_internal(pConfig, pMP3);
- if (result != MA_SUCCESS) {
- return result;
+ if (pNodeBase == NULL) {
+ return MA_INVALID_ARGS;
}
- #if !defined(MA_NO_MP3)
- {
- drmp3_allocation_callbacks mp3AllocationCallbacks = drmp3_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
- drmp3_bool32 mp3Result;
-
- mp3Result = drmp3_init_memory(&pMP3->dr, pData, dataSize, &mp3AllocationCallbacks);
- if (mp3Result != MA_TRUE) {
- return MA_INVALID_FILE;
- }
-
- return MA_SUCCESS;
- }
- #else
- {
- /* mp3 is disabled. */
- (void)pData;
- (void)dataSize;
- (void)pAllocationCallbacks;
- return MA_NOT_IMPLEMENTED;
+ if (outputBusIndex >= ma_node_get_output_bus_count(pNode)) {
+ return MA_INVALID_ARGS; /* Invalid bus index. */
}
- #endif
+
+ return ma_node_output_bus_set_volume(&pNodeBase->pOutputBuses[outputBusIndex], volume);
}
-MA_API void ma_mp3_uninit(ma_mp3* pMP3, const ma_allocation_callbacks* pAllocationCallbacks)
+MA_API float ma_node_get_output_bus_volume(const ma_node* pNode, ma_uint32 outputBusIndex)
{
- if (pMP3 == NULL) {
- return;
- }
-
- (void)pAllocationCallbacks;
+ const ma_node_base* pNodeBase = (const ma_node_base*)pNode;
- #if !defined(MA_NO_MP3)
- {
- drmp3_uninit(&pMP3->dr);
+ if (pNodeBase == NULL) {
+ return 0;
}
- #else
- {
- /* mp3 is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
+
+ if (outputBusIndex >= ma_node_get_output_bus_count(pNode)) {
+ return 0; /* Invalid bus index. */
}
- #endif
- ma_data_source_uninit(&pMP3->ds);
+ return ma_node_output_bus_get_volume(&pNodeBase->pOutputBuses[outputBusIndex]);
}
-MA_API ma_result ma_mp3_read_pcm_frames(ma_mp3* pMP3, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+MA_API ma_result ma_node_set_state(ma_node* pNode, ma_node_state state)
{
- if (pMP3 == NULL) {
+ ma_node_base* pNodeBase = (ma_node_base*)pNode;
+
+ if (pNodeBase == NULL) {
return MA_INVALID_ARGS;
}
- #if !defined(MA_NO_MP3)
- {
- /* We always use floating point format. */
- ma_result result = MA_SUCCESS; /* Must be initialized to MA_SUCCESS. */
- ma_uint64 totalFramesRead = 0;
- ma_format format;
-
- ma_mp3_get_data_format(pMP3, &format, NULL, NULL, NULL, 0);
-
- switch (format)
- {
- case ma_format_f32:
- {
- totalFramesRead = drmp3_read_pcm_frames_f32(&pMP3->dr, frameCount, (float*)pFramesOut);
- } break;
-
- case ma_format_s16:
- {
- totalFramesRead = drmp3_read_pcm_frames_s16(&pMP3->dr, frameCount, (drmp3_int16*)pFramesOut);
- } break;
-
- case ma_format_u8:
- case ma_format_s24:
- case ma_format_s32:
- case ma_format_unknown:
- default:
- {
- return MA_INVALID_OPERATION;
- };
- }
+ c89atomic_exchange_i32(&pNodeBase->state, state);
- /* In the future we'll update dr_mp3 to return MA_AT_END for us. */
- if (totalFramesRead == 0) {
- result = MA_AT_END;
- }
+ return MA_SUCCESS;
+}
- if (pFramesRead != NULL) {
- *pFramesRead = totalFramesRead;
- }
+MA_API ma_node_state ma_node_get_state(const ma_node* pNode)
+{
+ const ma_node_base* pNodeBase = (const ma_node_base*)pNode;
- return result;
+ if (pNodeBase == NULL) {
+ return ma_node_state_stopped;
}
- #else
- {
- /* mp3 is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
-
- (void)pFramesOut;
- (void)frameCount;
- (void)pFramesRead;
- return MA_NOT_IMPLEMENTED;
- }
- #endif
+ return (ma_node_state)c89atomic_load_i32(&pNodeBase->state);
}
-MA_API ma_result ma_mp3_seek_to_pcm_frame(ma_mp3* pMP3, ma_uint64 frameIndex)
+MA_API ma_result ma_node_set_state_time(ma_node* pNode, ma_node_state state, ma_uint64 globalTime)
{
- if (pMP3 == NULL) {
+ if (pNode == NULL) {
return MA_INVALID_ARGS;
}
- #if !defined(MA_NO_MP3)
- {
- drmp3_bool32 mp3Result;
-
- mp3Result = drmp3_seek_to_pcm_frame(&pMP3->dr, frameIndex);
- if (mp3Result != DRMP3_TRUE) {
- return MA_ERROR;
- }
-
- return MA_SUCCESS;
+ /* Validation check for safety since we'll be using this as an index into stateTimes[]. */
+ if (state != ma_node_state_started && state != ma_node_state_stopped) {
+ return MA_INVALID_ARGS;
}
- #else
- {
- /* mp3 is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
- (void)frameIndex;
+ c89atomic_exchange_64(&((ma_node_base*)pNode)->stateTimes[state], globalTime);
- return MA_NOT_IMPLEMENTED;
- }
- #endif
+ return MA_SUCCESS;
}
-MA_API ma_result ma_mp3_get_data_format(ma_mp3* pMP3, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
+MA_API ma_uint64 ma_node_get_state_time(const ma_node* pNode, ma_node_state state)
{
- /* Defaults for safety. */
- if (pFormat != NULL) {
- *pFormat = ma_format_unknown;
- }
- if (pChannels != NULL) {
- *pChannels = 0;
- }
- if (pSampleRate != NULL) {
- *pSampleRate = 0;
- }
- if (pChannelMap != NULL) {
- MA_ZERO_MEMORY(pChannelMap, sizeof(*pChannelMap) * channelMapCap);
+ if (pNode == NULL) {
+ return 0;
}
- if (pMP3 == NULL) {
- return MA_INVALID_OPERATION;
+ /* Validation check for safety since we'll be using this as an index into stateTimes[]. */
+ if (state != ma_node_state_started && state != ma_node_state_stopped) {
+ return 0;
}
- if (pFormat != NULL) {
- *pFormat = pMP3->format;
+ return c89atomic_load_64(&((ma_node_base*)pNode)->stateTimes[state]);
+}
+
+MA_API ma_node_state ma_node_get_state_by_time(const ma_node* pNode, ma_uint64 globalTime)
+{
+ if (pNode == NULL) {
+ return ma_node_state_stopped;
}
- #if !defined(MA_NO_MP3)
- {
- if (pChannels != NULL) {
- *pChannels = pMP3->dr.channels;
- }
+ return ma_node_get_state_by_time_range(pNode, globalTime, globalTime);
+}
- if (pSampleRate != NULL) {
- *pSampleRate = pMP3->dr.sampleRate;
- }
+MA_API ma_node_state ma_node_get_state_by_time_range(const ma_node* pNode, ma_uint64 globalTimeBeg, ma_uint64 globalTimeEnd)
+{
+ ma_node_state state;
- if (pChannelMap != NULL) {
- ma_get_standard_channel_map(ma_standard_channel_map_default, (ma_uint32)ma_min(pMP3->dr.channels, channelMapCap), pChannelMap);
- }
+ if (pNode == NULL) {
+ return ma_node_state_stopped;
+ }
- return MA_SUCCESS;
+ state = ma_node_get_state(pNode);
+
+ /* An explicitly stopped node is always stopped. */
+ if (state == ma_node_state_stopped) {
+ return ma_node_state_stopped;
}
- #else
- {
- /* mp3 is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
- return MA_NOT_IMPLEMENTED;
+
+ /*
+ Getting here means the node is marked as started, but it may still not be truly started due to
+ it's start time not having been reached yet. Also, the stop time may have also been reached in
+ which case it'll be considered stopped.
+ */
+ if (ma_node_get_state_time(pNode, ma_node_state_started) > globalTimeBeg) {
+ return ma_node_state_stopped; /* Start time has not yet been reached. */
}
- #endif
+
+ if (ma_node_get_state_time(pNode, ma_node_state_stopped) <= globalTimeEnd) {
+ return ma_node_state_stopped; /* Stop time has been reached. */
+ }
+
+ /* Getting here means the node is marked as started and is within it's start/stop times. */
+ return ma_node_state_started;
}
-MA_API ma_result ma_mp3_get_cursor_in_pcm_frames(ma_mp3* pMP3, ma_uint64* pCursor)
+MA_API ma_uint64 ma_node_get_time(const ma_node* pNode)
{
- if (pCursor == NULL) {
- return MA_INVALID_ARGS;
+ if (pNode == NULL) {
+ return 0;
}
- *pCursor = 0; /* Safety. */
+ return c89atomic_load_64(&((ma_node_base*)pNode)->localTime);
+}
- if (pMP3 == NULL) {
+MA_API ma_result ma_node_set_time(ma_node* pNode, ma_uint64 localTime)
+{
+ if (pNode == NULL) {
return MA_INVALID_ARGS;
}
- #if !defined(MA_NO_MP3)
- {
- *pCursor = pMP3->dr.currentPCMFrame;
+ c89atomic_exchange_64(&((ma_node_base*)pNode)->localTime, localTime);
- return MA_SUCCESS;
- }
- #else
- {
- /* mp3 is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
- return MA_NOT_IMPLEMENTED;
+ return MA_SUCCESS;
+}
+
+
+
+static void ma_node_process_pcm_frames_internal(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
+{
+ ma_node_base* pNodeBase = (ma_node_base*)pNode;
+
+ MA_ASSERT(pNode != NULL);
+
+ if (pNodeBase->vtable->onProcess) {
+ pNodeBase->vtable->onProcess(pNode, ppFramesIn, pFrameCountIn, ppFramesOut, pFrameCountOut);
}
- #endif
}
-MA_API ma_result ma_mp3_get_length_in_pcm_frames(ma_mp3* pMP3, ma_uint64* pLength)
+static ma_result ma_node_read_pcm_frames(ma_node* pNode, ma_uint32 outputBusIndex, float* pFramesOut, ma_uint32 frameCount, ma_uint32* pFramesRead, ma_uint64 globalTime)
{
- if (pLength == NULL) {
+ ma_node_base* pNodeBase = (ma_node_base*)pNode;
+ ma_result result = MA_SUCCESS;
+ ma_uint32 iInputBus;
+ ma_uint32 iOutputBus;
+ ma_uint32 inputBusCount;
+ ma_uint32 outputBusCount;
+ ma_uint32 totalFramesRead = 0;
+ float* ppFramesIn[MA_MAX_NODE_BUS_COUNT];
+ float* ppFramesOut[MA_MAX_NODE_BUS_COUNT];
+ ma_uint64 globalTimeBeg;
+ ma_uint64 globalTimeEnd;
+ ma_uint64 startTime;
+ ma_uint64 stopTime;
+ ma_uint32 timeOffsetBeg;
+ ma_uint32 timeOffsetEnd;
+ ma_uint32 frameCountIn;
+ ma_uint32 frameCountOut;
+
+ /*
+ pFramesRead is mandatory. It must be used to determine how many frames were read. It's normal and
+ expected that the number of frames read may be different to that requested. Therefore, the caller
+ must look at this value to correctly determine how many frames were read.
+ */
+ MA_ASSERT(pFramesRead != NULL); /* <-- If you've triggered this assert, you're using this function wrong. You *must* use this variable and inspect it after the call returns. */
+ if (pFramesRead == NULL) {
return MA_INVALID_ARGS;
}
- *pLength = 0; /* Safety. */
+ *pFramesRead = 0; /* Safety. */
- if (pMP3 == NULL) {
+ if (pNodeBase == NULL) {
return MA_INVALID_ARGS;
}
- #if !defined(MA_NO_MP3)
- {
- *pLength = drmp3_get_pcm_frame_count(&pMP3->dr);
-
- return MA_SUCCESS;
+ if (outputBusIndex >= ma_node_get_output_bus_count(pNodeBase)) {
+ return MA_INVALID_ARGS; /* Invalid output bus index. */
}
- #else
- {
- /* mp3 is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
- return MA_NOT_IMPLEMENTED;
+
+ /* Don't do anything if we're in a stopped state. */
+ if (ma_node_get_state_by_time_range(pNode, globalTime, globalTime + frameCount) != ma_node_state_started) {
+ return MA_SUCCESS; /* We're in a stopped state. This is not an error - we just need to not read anything. */
}
- #endif
-}
-static ma_result ma_decoding_backend_init__mp3(void* pUserData, ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
-{
- ma_result result;
- ma_mp3* pMP3;
+ globalTimeBeg = globalTime;
+ globalTimeEnd = globalTime + frameCount;
+ startTime = ma_node_get_state_time(pNode, ma_node_state_started);
+ stopTime = ma_node_get_state_time(pNode, ma_node_state_stopped);
- (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
+ /*
+ At this point we know that we are inside our start/stop times. However, we may need to adjust
+ our frame count and output pointer to accomodate since we could be straddling the time period
+ that this function is getting called for.
- /* For now we're just allocating the decoder backend on the heap. */
- pMP3 = (ma_mp3*)ma_malloc(sizeof(*pMP3), pAllocationCallbacks);
- if (pMP3 == NULL) {
- return MA_OUT_OF_MEMORY;
+ It's possible (and likely) that the start time does not line up with the output buffer. We
+ therefore need to offset it by a number of frames to accomodate. The same thing applies for
+ the stop time.
+ */
+ timeOffsetBeg = (globalTimeBeg < startTime) ? (ma_uint32)(globalTimeEnd - startTime) : 0;
+ timeOffsetEnd = (globalTimeEnd > stopTime) ? (ma_uint32)(globalTimeEnd - stopTime) : 0;
+
+ /* Trim based on the start offset. We need to silence the start of the buffer. */
+ if (timeOffsetBeg > 0) {
+ ma_silence_pcm_frames(pFramesOut, timeOffsetBeg, ma_format_f32, ma_node_get_output_channels(pNode, outputBusIndex));
+ pFramesOut += timeOffsetBeg * ma_node_get_output_channels(pNode, outputBusIndex);
+ frameCount -= timeOffsetBeg;
}
- result = ma_mp3_init(onRead, onSeek, onTell, pReadSeekTellUserData, pConfig, pAllocationCallbacks, pMP3);
- if (result != MA_SUCCESS) {
- ma_free(pMP3, pAllocationCallbacks);
- return result;
+ /* Trim based on the end offset. We don't need to silence the tail section because we'll just have a reduced value written to pFramesRead. */
+ if (timeOffsetEnd > 0) {
+ frameCount -= timeOffsetEnd;
}
- *ppBackend = pMP3;
- return MA_SUCCESS;
-}
+ /* We run on different paths depending on the bus counts. */
+ inputBusCount = ma_node_get_input_bus_count(pNode);
+ outputBusCount = ma_node_get_output_bus_count(pNode);
-static ma_result ma_decoding_backend_init_file__mp3(void* pUserData, const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
-{
- ma_result result;
- ma_mp3* pMP3;
+ /*
+ Run a simplified path when there are no inputs and one output. In this case there's nothing to
+ actually read and we can go straight to output. This is a very common scenario because the vast
+ majority of data source nodes will use this setup so this optimization I think is worthwhile.
+ */
+ if (inputBusCount == 0 && outputBusCount == 1) {
+ /* Fast path. No need to read from input and no need for any caching. */
+ frameCountIn = 0;
+ frameCountOut = frameCount; /* Just read as much as we can. The callback will return what was actually read. */
+
+ ppFramesOut[0] = pFramesOut;
+ ma_node_process_pcm_frames_internal(pNode, NULL, &frameCountIn, ppFramesOut, &frameCountOut);
+ totalFramesRead = frameCountOut;
+ } else {
+ /* Slow path. Need to read input data. */
+ if ((pNodeBase->vtable->flags & MA_NODE_FLAG_PASSTHROUGH) != 0) {
+ /*
+ Fast path. We're running a passthrough. We need to read directly into the output buffer, but
+ still fire the callback so that event handling and trigger nodes can do their thing. Since
+ it's a passthrough there's no need for any kind of caching logic.
+ */
+ MA_ASSERT(outputBusCount == inputBusCount);
+ MA_ASSERT(outputBusCount == 1);
+ MA_ASSERT(outputBusIndex == 0);
- (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
+ /* We just read directly from input bus to output buffer, and then afterwards fire the callback. */
+ ppFramesOut[0] = pFramesOut;
+ ppFramesIn[0] = ppFramesOut[0];
- /* For now we're just allocating the decoder backend on the heap. */
- pMP3 = (ma_mp3*)ma_malloc(sizeof(*pMP3), pAllocationCallbacks);
- if (pMP3 == NULL) {
- return MA_OUT_OF_MEMORY;
- }
+ result = ma_node_input_bus_read_pcm_frames(pNodeBase, &pNodeBase->pInputBuses[0], ppFramesIn[0], frameCount, &totalFramesRead, globalTime);
+ if (result == MA_SUCCESS) {
+ /* Even though it's a passthrough, we still need to fire the callback. */
+ frameCountIn = totalFramesRead;
+ frameCountOut = totalFramesRead;
- result = ma_mp3_init_file(pFilePath, pConfig, pAllocationCallbacks, pMP3);
- if (result != MA_SUCCESS) {
- ma_free(pMP3, pAllocationCallbacks);
- return result;
- }
+ if (totalFramesRead > 0) {
+ ma_node_process_pcm_frames_internal(pNode, (const float**)ppFramesIn, &frameCountIn, ppFramesOut, &frameCountOut); /* From GCC: expected 'const float **' but argument is of type 'float **'. Shouldn't this be implicit? Excplicit cast to silence the warning. */
+ }
- *ppBackend = pMP3;
+ /*
+ A passthrough should never have modified the input and output frame counts. If you're
+ triggering these assers you need to fix your processing callback.
+ */
+ MA_ASSERT(frameCountIn == totalFramesRead);
+ MA_ASSERT(frameCountOut == totalFramesRead);
+ }
+ } else {
+ /* Slow path. Need to do caching. */
+ ma_uint32 framesToProcessIn;
+ ma_uint32 framesToProcessOut;
+ ma_bool32 consumeNullInput = MA_FALSE;
- return MA_SUCCESS;
-}
+ /*
+ We use frameCount as a basis for the number of frames to read since that's what's being
+ requested, however we still need to clamp it to whatever can fit in the cache.
+
+ This will also be used as the basis for determining how many input frames to read. This is
+ not ideal because it can result in too many input frames being read which introduces latency.
+ To solve this, nodes can implement an optional callback called onGetRequiredInputFrameCount
+ which is used as hint to miniaudio as to how many input frames it needs to read at a time. This
+ callback is completely optional, and if it's not set, miniaudio will assume `frameCount`.
+
+ This function will be called multiple times for each period of time, once for each output node.
+ We cannot read from each input node each time this function is called. Instead we need to check
+ whether or not this is first output bus to be read from for this time period, and if so, read
+ from our input data.
+
+ To determine whether or not we're ready to read data, we check a flag. There will be one flag
+ for each output. When the flag is set, it means data has been read previously and that we're
+ ready to advance time forward for our input nodes by reading fresh data.
+ */
+ framesToProcessOut = frameCount;
+ if (framesToProcessOut > pNodeBase->cachedDataCapInFramesPerBus) {
+ framesToProcessOut = pNodeBase->cachedDataCapInFramesPerBus;
+ }
-static ma_result ma_decoding_backend_init_file_w__mp3(void* pUserData, const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
-{
- ma_result result;
- ma_mp3* pMP3;
+ framesToProcessIn = frameCount;
+ if (pNodeBase->vtable->onGetRequiredInputFrameCount) {
+ pNodeBase->vtable->onGetRequiredInputFrameCount(pNode, framesToProcessOut, &framesToProcessIn); /* <-- It does not matter if this fails. */
+ }
+ if (framesToProcessIn > pNodeBase->cachedDataCapInFramesPerBus) {
+ framesToProcessIn = pNodeBase->cachedDataCapInFramesPerBus;
+ }
- (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
- /* For now we're just allocating the decoder backend on the heap. */
- pMP3 = (ma_mp3*)ma_malloc(sizeof(*pMP3), pAllocationCallbacks);
- if (pMP3 == NULL) {
- return MA_OUT_OF_MEMORY;
- }
+ MA_ASSERT(framesToProcessIn <= 0xFFFF);
+ MA_ASSERT(framesToProcessOut <= 0xFFFF);
- result = ma_mp3_init_file_w(pFilePath, pConfig, pAllocationCallbacks, pMP3);
- if (result != MA_SUCCESS) {
- ma_free(pMP3, pAllocationCallbacks);
- return result;
- }
+ if (ma_node_output_bus_has_read(&pNodeBase->pOutputBuses[outputBusIndex])) {
+ /* Getting here means we need to do another round of processing. */
+ pNodeBase->cachedFrameCountOut = 0;
- *ppBackend = pMP3;
+ for (;;) {
+ frameCountOut = 0;
- return MA_SUCCESS;
-}
+ /*
+ We need to prepare our output frame pointers for processing. In the same iteration we need
+ to mark every output bus as unread so that future calls to this function for different buses
+ for the current time period don't pull in data when they should instead be reading from cache.
+ */
+ for (iOutputBus = 0; iOutputBus < outputBusCount; iOutputBus += 1) {
+ ma_node_output_bus_set_has_read(&pNodeBase->pOutputBuses[iOutputBus], MA_FALSE); /* <-- This is what tells the next calls to this function for other output buses for this time period to read from cache instead of pulling in more data. */
+ ppFramesOut[iOutputBus] = ma_node_get_cached_output_ptr(pNode, iOutputBus);
+ }
-static ma_result ma_decoding_backend_init_memory__mp3(void* pUserData, const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
-{
- ma_result result;
- ma_mp3* pMP3;
+ /* We only need to read from input buses if there isn't already some data in the cache. */
+ if (pNodeBase->cachedFrameCountIn == 0) {
+ ma_uint32 maxFramesReadIn = 0;
- (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
+ /* Here is where we pull in data from the input buses. This is what will trigger an advance in time. */
+ for (iInputBus = 0; iInputBus < inputBusCount; iInputBus += 1) {
+ ma_uint32 framesRead;
- /* For now we're just allocating the decoder backend on the heap. */
- pMP3 = (ma_mp3*)ma_malloc(sizeof(*pMP3), pAllocationCallbacks);
- if (pMP3 == NULL) {
- return MA_OUT_OF_MEMORY;
- }
+ /* The first thing to do is get the offset within our bulk allocation to store this input data. */
+ ppFramesIn[iInputBus] = ma_node_get_cached_input_ptr(pNode, iInputBus);
- result = ma_mp3_init_memory(pData, dataSize, pConfig, pAllocationCallbacks, pMP3);
- if (result != MA_SUCCESS) {
- ma_free(pMP3, pAllocationCallbacks);
- return result;
- }
+ /* Once we've determined our destination pointer we can read. Note that we must inspect the number of frames read and fill any leftovers with silence for safety. */
+ result = ma_node_input_bus_read_pcm_frames(pNodeBase, &pNodeBase->pInputBuses[iInputBus], ppFramesIn[iInputBus], framesToProcessIn, &framesRead, globalTime);
+ if (result != MA_SUCCESS) {
+ /* It doesn't really matter if we fail because we'll just fill with silence. */
+ framesRead = 0; /* Just for safety, but I don't think it's really needed. */
+ }
- *ppBackend = pMP3;
+ /* TODO: Minor optimization opportunity here. If no frames were read and the buffer is already filled with silence, no need to re-silence it. */
+ /* Any leftover frames need to silenced for safety. */
+ if (framesRead < framesToProcessIn) {
+ ma_silence_pcm_frames(ppFramesIn[iInputBus] + (framesRead * ma_node_get_input_channels(pNodeBase, iInputBus)), (framesToProcessIn - framesRead), ma_format_f32, ma_node_get_input_channels(pNodeBase, iInputBus));
+ }
- return MA_SUCCESS;
-}
+ maxFramesReadIn = ma_max(maxFramesReadIn, framesRead);
+ }
-static void ma_decoding_backend_uninit__mp3(void* pUserData, ma_data_source* pBackend, const ma_allocation_callbacks* pAllocationCallbacks)
-{
- ma_mp3* pMP3 = (ma_mp3*)pBackend;
+ /* This was a fresh load of input data so reset our consumption counter. */
+ pNodeBase->consumedFrameCountIn = 0;
- (void)pUserData;
+ /*
+ We don't want to keep processing if there's nothing to process, so set the number of cached
+ input frames to the maximum number we read from each attachment (the lesser will be padded
+ with silence). If we didn't read anything, this will be set to 0 and the entire buffer will
+ have been assigned to silence. This being equal to 0 is an important property for us because
+ it allows us to detect when NULL can be passed into the processing callback for the input
+ buffer for the purpose of continuous processing.
+ */
+ pNodeBase->cachedFrameCountIn = (ma_uint16)maxFramesReadIn;
+ } else {
+ /* We don't need to read anything, but we do need to prepare our input frame pointers. */
+ for (iInputBus = 0; iInputBus < inputBusCount; iInputBus += 1) {
+ ppFramesIn[iInputBus] = ma_node_get_cached_input_ptr(pNode, iInputBus) + (pNodeBase->consumedFrameCountIn * ma_node_get_input_channels(pNodeBase, iInputBus));
+ }
+ }
- ma_mp3_uninit(pMP3, pAllocationCallbacks);
- ma_free(pMP3, pAllocationCallbacks);
-}
+ /*
+ At this point we have our input data so now we need to do some processing. Sneaky little
+ optimization here - we can set the pointer to the output buffer for this output bus so
+ that the final copy into the output buffer is done directly by onProcess().
+ */
+ if (pFramesOut != NULL) {
+ ppFramesOut[outputBusIndex] = ma_offset_pcm_frames_ptr_f32(pFramesOut, pNodeBase->cachedFrameCountOut, ma_node_get_output_channels(pNode, outputBusIndex));
+ }
-static ma_result ma_decoding_backend_get_channel_map__mp3(void* pUserData, ma_data_source* pBackend, ma_channel* pChannelMap, size_t channelMapCap)
-{
- ma_mp3* pMP3 = (ma_mp3*)pBackend;
- (void)pUserData;
+ /* Give the processing function the entire capacity of the output buffer. */
+ frameCountOut = (framesToProcessOut - pNodeBase->cachedFrameCountOut);
- return ma_mp3_get_data_format(pMP3, NULL, NULL, NULL, pChannelMap, channelMapCap);
-}
+ /*
+ We need to treat nodes with continuous processing a little differently. For these ones,
+ we always want to fire the callback with the requested number of frames, regardless of
+ pNodeBase->cachedFrameCountIn, which could be 0. Also, we want to check if we can pass
+ in NULL for the input buffer to the callback.
+ */
+ if ((pNodeBase->vtable->flags & MA_NODE_FLAG_CONTINUOUS_PROCESSING) != 0) {
+ /* We're using continuous processing. Make sure we specify the whole frame count at all times. */
+ frameCountIn = framesToProcessIn; /* Give the processing function as much input data as we've got in the buffer, including any silenced padding from short reads. */
-static ma_decoding_backend_vtable g_ma_decoding_backend_vtable_mp3 =
-{
- ma_decoding_backend_init__mp3,
- ma_decoding_backend_init_file__mp3,
- ma_decoding_backend_init_file_w__mp3,
- ma_decoding_backend_init_memory__mp3,
- ma_decoding_backend_uninit__mp3,
- ma_decoding_backend_get_channel_map__mp3
-};
+ if ((pNodeBase->vtable->flags & MA_NODE_FLAG_ALLOW_NULL_INPUT) != 0 && pNodeBase->consumedFrameCountIn == 0 && pNodeBase->cachedFrameCountIn == 0) {
+ consumeNullInput = MA_TRUE;
+ } else {
+ consumeNullInput = MA_FALSE;
+ }
-static ma_result ma_decoder_init_mp3__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
-{
- return ma_decoder_init_from_vtable(&g_ma_decoding_backend_vtable_mp3, NULL, pConfig, pDecoder);
-}
-#endif /* dr_mp3_h */
+ /*
+ Since we're using continuous processing we're always passing in a full frame count
+ regardless of how much input data was read. If this is greater than what we read as
+ input, we'll end up with an underflow. We instead need to make sure our cached frame
+ count is set to the number of frames we'll be passing to the data callback. Not
+ doing this will result in an underflow when we "consume" the cached data later on.
+
+ Note that this check needs to be done after the "consumeNullInput" check above because
+ we use the property of cachedFrameCountIn being 0 to determine whether or not we
+ should be passing in a null pointer to the processing callback for when the node is
+ configured with MA_NODE_FLAG_ALLOW_NULL_INPUT.
+ */
+ if (pNodeBase->cachedFrameCountIn < (ma_uint16)frameCountIn) {
+ pNodeBase->cachedFrameCountIn = (ma_uint16)frameCountIn;
+ }
+ } else {
+ frameCountIn = pNodeBase->cachedFrameCountIn; /* Give the processing function as much valid input data as we've got. */
+ consumeNullInput = MA_FALSE;
+ }
-/* Vorbis */
-#ifdef STB_VORBIS_INCLUDE_STB_VORBIS_H
-#define MA_HAS_VORBIS
+ /*
+ Process data slightly differently depending on whether or not we're consuming NULL
+ input (checked just above).
+ */
+ if (consumeNullInput) {
+ ma_node_process_pcm_frames_internal(pNode, NULL, &frameCountIn, ppFramesOut, &frameCountOut);
+ } else {
+ /*
+ We want to skip processing if there's no input data, but we can only do that safely if
+ we know that there is no chance of any output frames being produced. If continuous
+ processing is being used, this won't be a problem because the input frame count will
+ always be non-0. However, if continuous processing is *not* enabled and input and output
+ data is processed at different rates, we still need to process that last input frame
+ because there could be a few excess output frames needing to be produced from cached
+ data. The `MA_NODE_FLAG_DIFFERENT_PROCESSING_RATES` flag is used as the indicator for
+ determining whether or not we need to process the node even when there are no input
+ frames available right now.
+ */
+ if (frameCountIn > 0 || (pNodeBase->vtable->flags & MA_NODE_FLAG_DIFFERENT_PROCESSING_RATES) != 0) {
+ ma_node_process_pcm_frames_internal(pNode, (const float**)ppFramesIn, &frameCountIn, ppFramesOut, &frameCountOut); /* From GCC: expected 'const float **' but argument is of type 'float **'. Shouldn't this be implicit? Excplicit cast to silence the warning. */
+ } else {
+ frameCountOut = 0; /* No data was processed. */
+ }
+ }
-/* The size in bytes of each chunk of data to read from the Vorbis stream. */
-#define MA_VORBIS_DATA_CHUNK_SIZE 4096
+ /*
+ Thanks to our sneaky optimization above we don't need to do any data copying directly into
+ the output buffer - the onProcess() callback just did that for us. We do, however, need to
+ apply the number of input and output frames that were processed. Note that due to continuous
+ processing above, we need to do explicit checks here. If we just consumed a NULL input
+ buffer it means that no actual input data was processed from the internal buffers and we
+ don't want to be modifying any counters.
+ */
+ if (consumeNullInput == MA_FALSE) {
+ pNodeBase->consumedFrameCountIn += (ma_uint16)frameCountIn;
+ pNodeBase->cachedFrameCountIn -= (ma_uint16)frameCountIn;
+ }
-typedef struct
-{
- ma_data_source_base ds;
- ma_read_proc onRead;
- ma_seek_proc onSeek;
- ma_tell_proc onTell;
- void* pReadSeekTellUserData;
- ma_allocation_callbacks allocationCallbacks; /* Store the allocation callbacks within the structure because we may need to dynamically expand a buffer in ma_stbvorbis_read_pcm_frames() when using push mode. */
- ma_format format; /* Only f32 is allowed with stb_vorbis. */
- ma_uint32 channels;
- ma_uint32 sampleRate;
- ma_uint64 cursor;
-#if !defined(MA_NO_VORBIS)
- stb_vorbis* stb;
- ma_bool32 usingPushMode;
- struct
- {
- ma_uint8* pData;
- size_t dataSize;
- size_t dataCapacity;
- ma_uint32 framesConsumed; /* The number of frames consumed in ppPacketData. */
- ma_uint32 framesRemaining; /* The number of frames remaining in ppPacketData. */
- float** ppPacketData;
- } push;
-#endif
-} ma_stbvorbis;
+ /* The cached output frame count is always equal to what we just read. */
+ pNodeBase->cachedFrameCountOut += (ma_uint16)frameCountOut;
-MA_API ma_result ma_stbvorbis_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_stbvorbis* pVorbis);
-MA_API ma_result ma_stbvorbis_init_file(const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_stbvorbis* pVorbis);
-MA_API ma_result ma_stbvorbis_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_stbvorbis* pVorbis);
-MA_API void ma_stbvorbis_uninit(ma_stbvorbis* pVorbis, const ma_allocation_callbacks* pAllocationCallbacks);
-MA_API ma_result ma_stbvorbis_read_pcm_frames(ma_stbvorbis* pVorbis, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
-MA_API ma_result ma_stbvorbis_seek_to_pcm_frame(ma_stbvorbis* pVorbis, ma_uint64 frameIndex);
-MA_API ma_result ma_stbvorbis_get_data_format(ma_stbvorbis* pVorbis, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
-MA_API ma_result ma_stbvorbis_get_cursor_in_pcm_frames(ma_stbvorbis* pVorbis, ma_uint64* pCursor);
-MA_API ma_result ma_stbvorbis_get_length_in_pcm_frames(ma_stbvorbis* pVorbis, ma_uint64* pLength);
+ /* If we couldn't process any data, we're done. The loop needs to be terminated here or else we'll get stuck in a loop. */
+ if (pNodeBase->cachedFrameCountOut == framesToProcessOut || (frameCountOut == 0 && frameCountIn == 0)) {
+ break;
+ }
+ }
+ } else {
+ /*
+ We're not needing to read anything from the input buffer so just read directly from our
+ already-processed data.
+ */
+ if (pFramesOut != NULL) {
+ ma_copy_pcm_frames(pFramesOut, ma_node_get_cached_output_ptr(pNodeBase, outputBusIndex), pNodeBase->cachedFrameCountOut, ma_format_f32, ma_node_get_output_channels(pNodeBase, outputBusIndex));
+ }
+ }
+ /* The number of frames read is always equal to the number of cached output frames. */
+ totalFramesRead = pNodeBase->cachedFrameCountOut;
-static ma_result ma_stbvorbis_ds_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
-{
- return ma_stbvorbis_read_pcm_frames((ma_stbvorbis*)pDataSource, pFramesOut, frameCount, pFramesRead);
-}
+ /* Now that we've read the data, make sure our read flag is set. */
+ ma_node_output_bus_set_has_read(&pNodeBase->pOutputBuses[outputBusIndex], MA_TRUE);
+ }
+ }
+
+ /* Apply volume, if necessary. */
+ ma_apply_volume_factor_f32(pFramesOut, totalFramesRead * ma_node_get_output_channels(pNodeBase, outputBusIndex), ma_node_output_bus_get_volume(&pNodeBase->pOutputBuses[outputBusIndex]));
-static ma_result ma_stbvorbis_ds_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
-{
- return ma_stbvorbis_seek_to_pcm_frame((ma_stbvorbis*)pDataSource, frameIndex);
-}
+ /* Advance our local time forward. */
+ c89atomic_fetch_add_64(&pNodeBase->localTime, (ma_uint64)totalFramesRead);
-static ma_result ma_stbvorbis_ds_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate)
-{
- return ma_stbvorbis_get_data_format((ma_stbvorbis*)pDataSource, pFormat, pChannels, pSampleRate, NULL, 0);
+ *pFramesRead = totalFramesRead + timeOffsetBeg; /* Must include the silenced section at the start of the buffer. */
+ return result;
}
-static ma_result ma_stbvorbis_ds_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
+
+
+
+/* Data source node. */
+MA_API ma_data_source_node_config ma_data_source_node_config_init(ma_data_source* pDataSource)
{
- return ma_stbvorbis_get_cursor_in_pcm_frames((ma_stbvorbis*)pDataSource, pCursor);
+ ma_data_source_node_config config;
+
+ MA_ZERO_OBJECT(&config);
+ config.nodeConfig = ma_node_config_init();
+ config.pDataSource = pDataSource;
+
+ return config;
}
-static ma_result ma_stbvorbis_ds_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
+
+static void ma_data_source_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
- return ma_stbvorbis_get_length_in_pcm_frames((ma_stbvorbis*)pDataSource, pLength);
+ ma_data_source_node* pDataSourceNode = (ma_data_source_node*)pNode;
+ ma_format format;
+ ma_uint32 channels;
+ ma_uint32 frameCount;
+ ma_uint64 framesRead = 0;
+
+ MA_ASSERT(pDataSourceNode != NULL);
+ MA_ASSERT(pDataSourceNode->pDataSource != NULL);
+ MA_ASSERT(ma_node_get_input_bus_count(pDataSourceNode) == 0);
+ MA_ASSERT(ma_node_get_output_bus_count(pDataSourceNode) == 1);
+
+ /* We don't want to read from ppFramesIn at all. Instead we read from the data source. */
+ (void)ppFramesIn;
+ (void)pFrameCountIn;
+
+ frameCount = *pFrameCountOut;
+
+ /* miniaudio should never be calling this with a frame count of zero. */
+ MA_ASSERT(frameCount > 0);
+
+ if (ma_data_source_get_data_format(pDataSourceNode->pDataSource, &format, &channels, NULL, NULL, 0) == MA_SUCCESS) { /* <-- Don't care about sample rate here. */
+ /* The node graph system requires samples be in floating point format. This is checked in ma_data_source_node_init(). */
+ MA_ASSERT(format == ma_format_f32);
+ (void)format; /* Just to silence some static analysis tools. */
+
+ ma_data_source_read_pcm_frames(pDataSourceNode->pDataSource, ppFramesOut[0], frameCount, &framesRead);
+ }
+
+ *pFrameCountOut = (ma_uint32)framesRead;
}
-static ma_data_source_vtable g_ma_stbvorbis_ds_vtable =
+static ma_node_vtable g_ma_data_source_node_vtable =
{
- ma_stbvorbis_ds_read,
- ma_stbvorbis_ds_seek,
- NULL, /* onMap() */
- NULL, /* onUnmap() */
- ma_stbvorbis_ds_get_data_format,
- ma_stbvorbis_ds_get_cursor,
- ma_stbvorbis_ds_get_length
+ ma_data_source_node_process_pcm_frames,
+ NULL, /* onGetRequiredInputFrameCount */
+ 0, /* 0 input buses. */
+ 1, /* 1 output bus. */
+ 0
};
-
-static ma_result ma_stbvorbis_init_internal(const ma_decoding_backend_config* pConfig, ma_stbvorbis* pVorbis)
+MA_API ma_result ma_data_source_node_init(ma_node_graph* pNodeGraph, const ma_data_source_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source_node* pDataSourceNode)
{
ma_result result;
- ma_data_source_config dataSourceConfig;
-
- (void)pConfig;
+ ma_format format; /* For validating the format, which must be ma_format_f32. */
+ ma_uint32 channels; /* For specifying the channel count of the output bus. */
+ ma_node_config baseConfig;
- if (pVorbis == NULL) {
+ if (pDataSourceNode == NULL) {
return MA_INVALID_ARGS;
}
- MA_ZERO_OBJECT(pVorbis);
- pVorbis->format = ma_format_f32; /* Only supporting f32. */
+ MA_ZERO_OBJECT(pDataSourceNode);
- dataSourceConfig = ma_data_source_config_init();
- dataSourceConfig.vtable = &g_ma_stbvorbis_ds_vtable;
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
- result = ma_data_source_init(&dataSourceConfig, &pVorbis->ds);
+ result = ma_data_source_get_data_format(pConfig->pDataSource, &format, &channels, NULL, NULL, 0); /* Don't care about sample rate. This will check pDataSource for NULL. */
if (result != MA_SUCCESS) {
- return result; /* Failed to initialize the base data source. */
+ return result;
}
- return MA_SUCCESS;
-}
+ MA_ASSERT(format == ma_format_f32); /* <-- If you've triggered this it means your data source is not outputting floating-point samples. You must configure your data source to use ma_format_f32. */
+ if (format != ma_format_f32) {
+ return MA_INVALID_ARGS; /* Invalid format. */
+ }
-#if !defined(MA_NO_VORBIS)
-static ma_result ma_stbvorbis_post_init(ma_stbvorbis* pVorbis)
-{
- stb_vorbis_info info;
+ /* The channel count is defined by the data source. If the caller has manually changed the channels we just ignore it. */
+ baseConfig = pConfig->nodeConfig;
+ baseConfig.vtable = &g_ma_data_source_node_vtable; /* Explicitly set the vtable here to prevent callers from setting it incorrectly. */
- MA_ASSERT(pVorbis != NULL);
+ /*
+ The channel count is defined by the data source. It is invalid for the caller to manually set
+ the channel counts in the config. `ma_data_source_node_config_init()` will have defaulted the
+ channel count pointer to NULL which is how it must remain. If you trigger any of these asserts
+ it means you're explicitly setting the channel count. Instead, configure the output channel
+ count of your data source to be the necessary channel count.
+ */
+ if (baseConfig.pOutputChannels != NULL) {
+ return MA_INVALID_ARGS;
+ }
- info = stb_vorbis_get_info(pVorbis->stb);
+ baseConfig.pOutputChannels = &channels;
- pVorbis->channels = info.channels;
- pVorbis->sampleRate = info.sample_rate;
+ result = ma_node_init(pNodeGraph, &baseConfig, pAllocationCallbacks, &pDataSourceNode->base);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ pDataSourceNode->pDataSource = pConfig->pDataSource;
return MA_SUCCESS;
}
-#endif
-MA_API ma_result ma_stbvorbis_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_stbvorbis* pVorbis)
+MA_API void ma_data_source_node_uninit(ma_data_source_node* pDataSourceNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_result result;
+ ma_node_uninit(&pDataSourceNode->base, pAllocationCallbacks);
+}
- result = ma_stbvorbis_init_internal(pConfig, pVorbis);
- if (result != MA_SUCCESS) {
- return result;
+MA_API ma_result ma_data_source_node_set_looping(ma_data_source_node* pDataSourceNode, ma_bool32 isLooping)
+{
+ if (pDataSourceNode == NULL) {
+ return MA_INVALID_ARGS;
}
- if (onRead == NULL || onSeek == NULL) {
- return MA_INVALID_ARGS; /* onRead and onSeek are mandatory. */
- }
+ return ma_data_source_set_looping(pDataSourceNode->pDataSource, isLooping);
+}
- pVorbis->onRead = onRead;
- pVorbis->onSeek = onSeek;
- pVorbis->onTell = onTell;
- pVorbis->pReadSeekTellUserData = pReadSeekTellUserData;
- ma_allocation_callbacks_init_copy(&pVorbis->allocationCallbacks, pAllocationCallbacks);
+MA_API ma_bool32 ma_data_source_node_is_looping(ma_data_source_node* pDataSourceNode)
+{
+ if (pDataSourceNode == NULL) {
+ return MA_FALSE;
+ }
- #if !defined(MA_NO_VORBIS)
- {
- /*
- stb_vorbis lacks a callback based API for it's pulling API which means we're stuck with the
- pushing API. In order for us to be able to successfully initialize the decoder we need to
- supply it with enough data. We need to keep loading data until we have enough.
- */
- stb_vorbis* stb;
- size_t dataSize = 0;
- size_t dataCapacity = 0;
- ma_uint8* pData = NULL; /* <-- Must be initialized to NULL. */
+ return ma_data_source_is_looping(pDataSourceNode->pDataSource);
+}
- for (;;) {
- int vorbisError;
- int consumedDataSize; /* <-- Fill by stb_vorbis_open_pushdata(). */
- size_t bytesRead;
- ma_uint8* pNewData;
- /* Allocate memory for the new chunk. */
- dataCapacity += MA_VORBIS_DATA_CHUNK_SIZE;
- pNewData = (ma_uint8*)ma_realloc(pData, dataCapacity, pAllocationCallbacks);
- if (pNewData == NULL) {
- ma_free(pData, pAllocationCallbacks);
- return MA_OUT_OF_MEMORY;
- }
- pData = pNewData;
+/* Splitter Node. */
+MA_API ma_splitter_node_config ma_splitter_node_config_init(ma_uint32 channels)
+{
+ ma_splitter_node_config config;
- /* Read in the next chunk. */
- result = pVorbis->onRead(pVorbis->pReadSeekTellUserData, ma_offset_ptr(pData, dataSize), (dataCapacity - dataSize), &bytesRead);
- dataSize += bytesRead;
+ MA_ZERO_OBJECT(&config);
+ config.nodeConfig = ma_node_config_init();
+ config.channels = channels;
- if (result != MA_SUCCESS) {
- ma_free(pData, pAllocationCallbacks);
- return result;
- }
+ return config;
+}
- /* We have a maximum of 31 bits with stb_vorbis. */
- if (dataSize > INT_MAX) {
- ma_free(pData, pAllocationCallbacks);
- return MA_TOO_BIG;
- }
- stb = stb_vorbis_open_pushdata(pData, (int)dataSize, &consumedDataSize, &vorbisError, NULL);
- if (stb != NULL) {
- /*
- Successfully opened the Vorbis decoder. We might have some leftover unprocessed
- data so we'll need to move that down to the front.
- */
- dataSize -= (size_t)consumedDataSize; /* Consume the data. */
- MA_MOVE_MEMORY(pData, ma_offset_ptr(pData, consumedDataSize), dataSize);
- break;
- } else {
- /* Failed to open the decoder. */
- if (vorbisError == VORBIS_need_more_data) {
- continue;
- } else {
- ma_free(pData, pAllocationCallbacks);
- return MA_ERROR; /* Failed to open the stb_vorbis decoder. */
- }
- }
- }
+static void ma_splitter_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
+{
+ ma_node_base* pNodeBase = (ma_node_base*)pNode;
+ ma_uint32 iOutputBus;
+ ma_uint32 channels;
- MA_ASSERT(stb != NULL);
- pVorbis->stb = stb;
- pVorbis->push.pData = pData;
- pVorbis->push.dataSize = dataSize;
- pVorbis->push.dataCapacity = dataCapacity;
+ MA_ASSERT(pNodeBase != NULL);
+ MA_ASSERT(ma_node_get_input_bus_count(pNodeBase) == 1);
+ MA_ASSERT(ma_node_get_output_bus_count(pNodeBase) >= 2);
- pVorbis->usingPushMode = MA_TRUE;
+ /* We don't need to consider the input frame count - it'll be the same as the output frame count and we process everything. */
+ (void)pFrameCountIn;
- result = ma_stbvorbis_post_init(pVorbis);
- if (result != MA_SUCCESS) {
- stb_vorbis_close(pVorbis->stb);
- ma_free(pData, pAllocationCallbacks);
- return result;
- }
+ /* NOTE: This assumes the same number of channels for all inputs and outputs. This was checked in ma_splitter_node_init(). */
+ channels = ma_node_get_input_channels(pNodeBase, 0);
- return MA_SUCCESS;
- }
- #else
- {
- /* vorbis is disabled. */
- (void)pAllocationCallbacks;
- return MA_NOT_IMPLEMENTED;
+ /* Splitting is just copying the first input bus and copying it over to each output bus. */
+ for (iOutputBus = 0; iOutputBus < ma_node_get_output_bus_count(pNodeBase); iOutputBus += 1) {
+ ma_copy_pcm_frames(ppFramesOut[iOutputBus], ppFramesIn[0], *pFrameCountOut, ma_format_f32, channels);
}
- #endif
}
-MA_API ma_result ma_stbvorbis_init_file(const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_stbvorbis* pVorbis)
+static ma_node_vtable g_ma_splitter_node_vtable =
+{
+ ma_splitter_node_process_pcm_frames,
+ NULL, /* onGetRequiredInputFrameCount */
+ 1, /* 1 input bus. */
+ 2, /* 2 output buses. */
+ 0
+};
+
+MA_API ma_result ma_splitter_node_init(ma_node_graph* pNodeGraph, const ma_splitter_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_splitter_node* pSplitterNode)
{
ma_result result;
+ ma_node_config baseConfig;
+ ma_uint32 pInputChannels[1];
+ ma_uint32 pOutputChannels[2];
- result = ma_stbvorbis_init_internal(pConfig, pVorbis);
+ if (pSplitterNode == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ MA_ZERO_OBJECT(pSplitterNode);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ /* Splitters require the same number of channels between inputs and outputs. */
+ pInputChannels[0] = pConfig->channels;
+ pOutputChannels[0] = pConfig->channels;
+ pOutputChannels[1] = pConfig->channels;
+
+ baseConfig = pConfig->nodeConfig;
+ baseConfig.vtable = &g_ma_splitter_node_vtable;
+ baseConfig.pInputChannels = pInputChannels;
+ baseConfig.pOutputChannels = pOutputChannels;
+
+ result = ma_node_init(pNodeGraph, &baseConfig, pAllocationCallbacks, &pSplitterNode->base);
if (result != MA_SUCCESS) {
- return result;
+ return result; /* Failed to initialize the base node. */
}
- #if !defined(MA_NO_VORBIS)
- {
- (void)pAllocationCallbacks; /* Don't know how to make use of this with stb_vorbis. */
+ return MA_SUCCESS;
+}
- /* We can use stb_vorbis' pull mode for file based streams. */
- pVorbis->stb = stb_vorbis_open_filename(pFilePath, NULL, NULL);
- if (pVorbis->stb == NULL) {
- return MA_INVALID_FILE;
- }
+MA_API void ma_splitter_node_uninit(ma_splitter_node* pSplitterNode, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ ma_node_uninit(pSplitterNode, pAllocationCallbacks);
+}
- pVorbis->usingPushMode = MA_FALSE;
- result = ma_stbvorbis_post_init(pVorbis);
- if (result != MA_SUCCESS) {
- stb_vorbis_close(pVorbis->stb);
- return result;
- }
+/*
+Biquad Node
+*/
+MA_API ma_biquad_node_config ma_biquad_node_config_init(ma_uint32 channels, float b0, float b1, float b2, float a0, float a1, float a2)
+{
+ ma_biquad_node_config config;
- return MA_SUCCESS;
+ config.nodeConfig = ma_node_config_init();
+ config.biquad = ma_biquad_config_init(ma_format_f32, channels, b0, b1, b2, a0, a1, a2);
+
+ return config;
+}
+
+static void ma_biquad_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
+{
+ ma_biquad_node* pLPFNode = (ma_biquad_node*)pNode;
+
+ MA_ASSERT(pNode != NULL);
+ (void)pFrameCountIn;
+
+ ma_biquad_process_pcm_frames(&pLPFNode->biquad, ppFramesOut[0], ppFramesIn[0], *pFrameCountOut);
+}
+
+static ma_node_vtable g_ma_biquad_node_vtable =
+{
+ ma_biquad_node_process_pcm_frames,
+ NULL, /* onGetRequiredInputFrameCount */
+ 1, /* One input. */
+ 1, /* One output. */
+ 0 /* Default flags. */
+};
+
+MA_API ma_result ma_biquad_node_init(ma_node_graph* pNodeGraph, const ma_biquad_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_biquad_node* pNode)
+{
+ ma_result result;
+ ma_node_config baseNodeConfig;
+
+ if (pNode == NULL) {
+ return MA_INVALID_ARGS;
}
- #else
- {
- /* vorbis is disabled. */
- (void)pFilePath;
- (void)pAllocationCallbacks;
- return MA_NOT_IMPLEMENTED;
+
+ MA_ZERO_OBJECT(pNode);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- #endif
-}
-MA_API ma_result ma_stbvorbis_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_stbvorbis* pVorbis)
-{
- ma_result result;
+ if (pConfig->biquad.format != ma_format_f32) {
+ return MA_INVALID_ARGS; /* The format must be f32. */
+ }
- result = ma_stbvorbis_init_internal(pConfig, pVorbis);
+ result = ma_biquad_init(&pConfig->biquad, pAllocationCallbacks, &pNode->biquad);
if (result != MA_SUCCESS) {
return result;
}
- #if !defined(MA_NO_VORBIS)
- {
- (void)pAllocationCallbacks;
+ baseNodeConfig = ma_node_config_init();
+ baseNodeConfig.vtable = &g_ma_biquad_node_vtable;
+ baseNodeConfig.pInputChannels = &pConfig->biquad.channels;
+ baseNodeConfig.pOutputChannels = &pConfig->biquad.channels;
- /* stb_vorbis uses an int as it's size specifier, restricting it to 32-bit even on 64-bit systems. *sigh*. */
- if (dataSize > INT_MAX) {
- return MA_TOO_BIG;
- }
+ result = ma_node_init(pNodeGraph, &baseNodeConfig, pAllocationCallbacks, pNode);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- pVorbis->stb = stb_vorbis_open_memory((const unsigned char*)pData, (int)dataSize, NULL, NULL);
- if (pVorbis->stb == NULL) {
- return MA_INVALID_FILE;
- }
+ return result;
+}
- pVorbis->usingPushMode = MA_FALSE;
+MA_API ma_result ma_biquad_node_reinit(const ma_biquad_config* pConfig, ma_biquad_node* pNode)
+{
+ ma_biquad_node* pLPFNode = (ma_biquad_node*)pNode;
- result = ma_stbvorbis_post_init(pVorbis);
- if (result != MA_SUCCESS) {
- stb_vorbis_close(pVorbis->stb);
- return result;
- }
+ MA_ASSERT(pNode != NULL);
- return MA_SUCCESS;
- }
- #else
- {
- /* vorbis is disabled. */
- (void)pData;
- (void)dataSize;
- (void)pAllocationCallbacks;
- return MA_NOT_IMPLEMENTED;
- }
- #endif
+ return ma_biquad_reinit(pConfig, &pLPFNode->biquad);
}
-MA_API void ma_stbvorbis_uninit(ma_stbvorbis* pVorbis, const ma_allocation_callbacks* pAllocationCallbacks)
+MA_API void ma_biquad_node_uninit(ma_biquad_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
- if (pVorbis == NULL) {
+ ma_biquad_node* pLPFNode = (ma_biquad_node*)pNode;
+
+ if (pNode == NULL) {
return;
}
- #if !defined(MA_NO_VORBIS)
- {
- stb_vorbis_close(pVorbis->stb);
+ ma_node_uninit(pNode, pAllocationCallbacks);
+ ma_biquad_uninit(&pLPFNode->biquad, pAllocationCallbacks);
+}
- /* We'll have to clear some memory if we're using push mode. */
- if (pVorbis->usingPushMode) {
- ma_free(pVorbis->push.pData, pAllocationCallbacks);
- }
- }
- #else
- {
- /* vorbis is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
- }
- #endif
- ma_data_source_uninit(&pVorbis->ds);
+
+/*
+Low Pass Filter Node
+*/
+MA_API ma_lpf_node_config ma_lpf_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order)
+{
+ ma_lpf_node_config config;
+
+ config.nodeConfig = ma_node_config_init();
+ config.lpf = ma_lpf_config_init(ma_format_f32, channels, sampleRate, cutoffFrequency, order);
+
+ return config;
}
-MA_API ma_result ma_stbvorbis_read_pcm_frames(ma_stbvorbis* pVorbis, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+static void ma_lpf_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
- if (pVorbis == NULL) {
+ ma_lpf_node* pLPFNode = (ma_lpf_node*)pNode;
+
+ MA_ASSERT(pNode != NULL);
+ (void)pFrameCountIn;
+
+ ma_lpf_process_pcm_frames(&pLPFNode->lpf, ppFramesOut[0], ppFramesIn[0], *pFrameCountOut);
+}
+
+static ma_node_vtable g_ma_lpf_node_vtable =
+{
+ ma_lpf_node_process_pcm_frames,
+ NULL, /* onGetRequiredInputFrameCount */
+ 1, /* One input. */
+ 1, /* One output. */
+ 0 /* Default flags. */
+};
+
+MA_API ma_result ma_lpf_node_init(ma_node_graph* pNodeGraph, const ma_lpf_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_lpf_node* pNode)
+{
+ ma_result result;
+ ma_node_config baseNodeConfig;
+
+ if (pNode == NULL) {
return MA_INVALID_ARGS;
}
- #if !defined(MA_NO_VORBIS)
- {
- /* We always use floating point format. */
- ma_result result = MA_SUCCESS; /* Must be initialized to MA_SUCCESS. */
- ma_uint64 totalFramesRead = 0;
- ma_format format;
- ma_uint32 channels;
+ MA_ZERO_OBJECT(pNode);
- ma_stbvorbis_get_data_format(pVorbis, &format, &channels, NULL, NULL, 0);
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
- if (format == ma_format_f32) {
- /* We read differently depending on whether or not we're using push mode. */
- if (pVorbis->usingPushMode) {
- /* Push mode. This is the complex case. */
- float* pFramesOutF32 = (float*)pFramesOut;
+ if (pConfig->lpf.format != ma_format_f32) {
+ return MA_INVALID_ARGS; /* The format must be f32. */
+ }
- while (totalFramesRead < frameCount) {
- /* The first thing to do is read from any already-cached frames. */
- ma_uint32 framesToReadFromCache = (ma_uint32)ma_min(pVorbis->push.framesRemaining, (frameCount - totalFramesRead)); /* Safe cast because pVorbis->framesRemaining is 32-bit. */
+ result = ma_lpf_init(&pConfig->lpf, pAllocationCallbacks, &pNode->lpf);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- /* The output pointer can be null in which case we just treate it as a seek. */
- if (pFramesOut != NULL) {
- ma_uint64 iFrame;
- for (iFrame = 0; iFrame < framesToReadFromCache; iFrame += 1) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < pVorbis->channels; iChannel += 1) {
- pFramesOutF32[iChannel] = pVorbis->push.ppPacketData[iChannel][pVorbis->push.framesConsumed + iFrame];
- }
+ baseNodeConfig = ma_node_config_init();
+ baseNodeConfig.vtable = &g_ma_lpf_node_vtable;
+ baseNodeConfig.pInputChannels = &pConfig->lpf.channels;
+ baseNodeConfig.pOutputChannels = &pConfig->lpf.channels;
- pFramesOutF32 += pVorbis->channels;
- }
- }
+ result = ma_node_init(pNodeGraph, &baseNodeConfig, pAllocationCallbacks, pNode);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- /* Update pointers and counters. */
- pVorbis->push.framesConsumed += framesToReadFromCache;
- pVorbis->push.framesRemaining -= framesToReadFromCache;
- totalFramesRead += framesToReadFromCache;
+ return result;
+}
- /* Don't bother reading any more frames right now if we've just finished loading. */
- if (totalFramesRead == frameCount) {
- break;
- }
+MA_API ma_result ma_lpf_node_reinit(const ma_lpf_config* pConfig, ma_lpf_node* pNode)
+{
+ ma_lpf_node* pLPFNode = (ma_lpf_node*)pNode;
- MA_ASSERT(pVorbis->push.framesRemaining == 0);
+ if (pNode == NULL) {
+ return MA_INVALID_ARGS;
+ }
- /* Getting here means we've run out of cached frames. We'll need to load some more. */
- for (;;) {
- int samplesRead = 0;
- int consumedDataSize;
+ return ma_lpf_reinit(pConfig, &pLPFNode->lpf);
+}
- /* We need to case dataSize to an int, so make sure we can do it safely. */
- if (pVorbis->push.dataSize > INT_MAX) {
- break; /* Too big. */
- }
+MA_API void ma_lpf_node_uninit(ma_lpf_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ ma_lpf_node* pLPFNode = (ma_lpf_node*)pNode;
- consumedDataSize = stb_vorbis_decode_frame_pushdata(pVorbis->stb, pVorbis->push.pData, (int)pVorbis->push.dataSize, NULL, &pVorbis->push.ppPacketData, &samplesRead);
- if (consumedDataSize != 0) {
- /* Successfully decoded a Vorbis frame. Consume the data. */
- pVorbis->push.dataSize -= (size_t)consumedDataSize;
- MA_MOVE_MEMORY(pVorbis->push.pData, ma_offset_ptr(pVorbis->push.pData, consumedDataSize), pVorbis->push.dataSize);
+ if (pNode == NULL) {
+ return;
+ }
- pVorbis->push.framesConsumed = 0;
- pVorbis->push.framesRemaining = samplesRead;
+ ma_node_uninit(pNode, pAllocationCallbacks);
+ ma_lpf_uninit(&pLPFNode->lpf, pAllocationCallbacks);
+}
- break;
- } else {
- /* Not enough data. Read more. */
- size_t bytesRead;
- /* Expand the data buffer if necessary. */
- if (pVorbis->push.dataCapacity == pVorbis->push.dataSize) {
- size_t newCap = pVorbis->push.dataCapacity + MA_VORBIS_DATA_CHUNK_SIZE;
- ma_uint8* pNewData;
- pNewData = (ma_uint8*)ma_realloc(pVorbis->push.pData, newCap, &pVorbis->allocationCallbacks);
- if (pNewData == NULL) {
- result = MA_OUT_OF_MEMORY;
- break;
- }
+/*
+High Pass Filter Node
+*/
+MA_API ma_hpf_node_config ma_hpf_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order)
+{
+ ma_hpf_node_config config;
- pVorbis->push.pData = pNewData;
- pVorbis->push.dataCapacity = newCap;
- }
+ config.nodeConfig = ma_node_config_init();
+ config.hpf = ma_hpf_config_init(ma_format_f32, channels, sampleRate, cutoffFrequency, order);
- /* We should have enough room to load some data. */
- result = pVorbis->onRead(pVorbis->pReadSeekTellUserData, ma_offset_ptr(pVorbis->push.pData, pVorbis->push.dataSize), (pVorbis->push.dataCapacity - pVorbis->push.dataSize), &bytesRead);
- pVorbis->push.dataSize += bytesRead;
+ return config;
+}
- if (result != MA_SUCCESS) {
- break; /* Failed to read any data. Get out. */
- }
- }
- }
+static void ma_hpf_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
+{
+ ma_hpf_node* pHPFNode = (ma_hpf_node*)pNode;
- /* If we don't have a success code at this point it means we've encounted an error or the end of the file has been reached (probably the latter). */
- if (result != MA_SUCCESS) {
- break;
- }
- }
- } else {
- /* Pull mode. This is the simple case, but we still need to run in a loop because stb_vorbis loves using 32-bit instead of 64-bit. */
- while (totalFramesRead < frameCount) {
- ma_uint64 framesRemaining = (frameCount - totalFramesRead);
- int framesRead;
-
- if (framesRemaining > INT_MAX) {
- framesRemaining = INT_MAX;
- }
+ MA_ASSERT(pNode != NULL);
+ (void)pFrameCountIn;
- framesRead = stb_vorbis_get_samples_float_interleaved(pVorbis->stb, channels, (float*)ma_offset_pcm_frames_ptr(pFramesOut, totalFramesRead, format, channels), (int)framesRemaining * channels); /* Safe cast. */
- totalFramesRead += framesRead;
+ ma_hpf_process_pcm_frames(&pHPFNode->hpf, ppFramesOut[0], ppFramesIn[0], *pFrameCountOut);
+}
- if (framesRead < framesRemaining) {
- break; /* Nothing left to read. Get out. */
- }
- }
- }
- } else {
- result = MA_INVALID_ARGS;
- }
+static ma_node_vtable g_ma_hpf_node_vtable =
+{
+ ma_hpf_node_process_pcm_frames,
+ NULL, /* onGetRequiredInputFrameCount */
+ 1, /* One input. */
+ 1, /* One output. */
+ 0 /* Default flags. */
+};
- pVorbis->cursor += totalFramesRead;
+MA_API ma_result ma_hpf_node_init(ma_node_graph* pNodeGraph, const ma_hpf_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hpf_node* pNode)
+{
+ ma_result result;
+ ma_node_config baseNodeConfig;
- if (totalFramesRead == 0) {
- result = MA_AT_END;
- }
+ if (pNode == NULL) {
+ return MA_INVALID_ARGS;
+ }
- if (pFramesRead != NULL) {
- *pFramesRead = totalFramesRead;
- }
+ MA_ZERO_OBJECT(pNode);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ if (pConfig->hpf.format != ma_format_f32) {
+ return MA_INVALID_ARGS; /* The format must be f32. */
+ }
+ result = ma_hpf_init(&pConfig->hpf, pAllocationCallbacks, &pNode->hpf);
+ if (result != MA_SUCCESS) {
return result;
}
- #else
- {
- /* vorbis is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
- (void)pFramesOut;
- (void)frameCount;
- (void)pFramesRead;
+ baseNodeConfig = ma_node_config_init();
+ baseNodeConfig.vtable = &g_ma_hpf_node_vtable;
+ baseNodeConfig.pInputChannels = &pConfig->hpf.channels;
+ baseNodeConfig.pOutputChannels = &pConfig->hpf.channels;
- return MA_NOT_IMPLEMENTED;
+ result = ma_node_init(pNodeGraph, &baseNodeConfig, pAllocationCallbacks, pNode);
+ if (result != MA_SUCCESS) {
+ return result;
}
- #endif
+
+ return result;
}
-MA_API ma_result ma_stbvorbis_seek_to_pcm_frame(ma_stbvorbis* pVorbis, ma_uint64 frameIndex)
+MA_API ma_result ma_hpf_node_reinit(const ma_hpf_config* pConfig, ma_hpf_node* pNode)
{
- if (pVorbis == NULL) {
+ ma_hpf_node* pHPFNode = (ma_hpf_node*)pNode;
+
+ if (pNode == NULL) {
return MA_INVALID_ARGS;
}
- #if !defined(MA_NO_VORBIS)
- {
- /* Different seeking methods depending on whether or not we're using push mode. */
- if (pVorbis->usingPushMode) {
- /* Push mode. This is the complex case. */
- ma_result result;
- float buffer[4096];
+ return ma_hpf_reinit(pConfig, &pHPFNode->hpf);
+}
- /*
- This is terribly inefficient because stb_vorbis does not have a good seeking solution with it's push API. Currently this just performs
- a full decode right from the start of the stream. Later on I'll need to write a layer that goes through all of the Ogg pages until we
- find the one containing the sample we need. Then we know exactly where to seek for stb_vorbis.
+MA_API void ma_hpf_node_uninit(ma_hpf_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ ma_hpf_node* pHPFNode = (ma_hpf_node*)pNode;
- TODO: Use seeking logic documented for stb_vorbis_flush_pushdata().
- */
+ if (pNode == NULL) {
+ return;
+ }
- /* Seek to the start of the file to begin with. */
- result = pVorbis->onSeek(pVorbis->pReadSeekTellUserData, 0, ma_seek_origin_start);
- if (result != MA_SUCCESS) {
- return result;
- }
+ ma_node_uninit(pNode, pAllocationCallbacks);
+ ma_hpf_uninit(&pHPFNode->hpf, pAllocationCallbacks);
+}
- stb_vorbis_flush_pushdata(pVorbis->stb);
- pVorbis->push.framesRemaining = 0;
- pVorbis->push.dataSize = 0;
- /* Move the cursor back to the start. We'll increment this in the loop below. */
- pVorbis->cursor = 0;
- while (pVorbis->cursor < frameIndex) {
- ma_uint64 framesRead;
- ma_uint64 framesToRead = ma_countof(buffer)/pVorbis->channels;
- if (framesToRead > (frameIndex - pVorbis->cursor)) {
- framesToRead = (frameIndex - pVorbis->cursor);
- }
- result = ma_stbvorbis_read_pcm_frames(pVorbis, buffer, framesToRead, &framesRead);
- pVorbis->cursor += framesRead;
+/*
+Band Pass Filter Node
+*/
+MA_API ma_bpf_node_config ma_bpf_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order)
+{
+ ma_bpf_node_config config;
- if (result != MA_SUCCESS) {
- return result;
- }
- }
- } else {
- /* Pull mode. This is the simple case. */
- int vorbisResult;
+ config.nodeConfig = ma_node_config_init();
+ config.bpf = ma_bpf_config_init(ma_format_f32, channels, sampleRate, cutoffFrequency, order);
- if (frameIndex > UINT_MAX) {
- return MA_INVALID_ARGS; /* Trying to seek beyond the 32-bit maximum of stb_vorbis. */
- }
+ return config;
+}
- vorbisResult = stb_vorbis_seek(pVorbis->stb, (unsigned int)frameIndex); /* Safe cast. */
- if (vorbisResult == 0) {
- return MA_ERROR; /* See failed. */
- }
+static void ma_bpf_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
+{
+ ma_bpf_node* pBPFNode = (ma_bpf_node*)pNode;
- pVorbis->cursor = frameIndex;
- }
+ MA_ASSERT(pNode != NULL);
+ (void)pFrameCountIn;
- return MA_SUCCESS;
- }
- #else
- {
- /* vorbis is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
+ ma_bpf_process_pcm_frames(&pBPFNode->bpf, ppFramesOut[0], ppFramesIn[0], *pFrameCountOut);
+}
- (void)frameIndex;
+static ma_node_vtable g_ma_bpf_node_vtable =
+{
+ ma_bpf_node_process_pcm_frames,
+ NULL, /* onGetRequiredInputFrameCount */
+ 1, /* One input. */
+ 1, /* One output. */
+ 0 /* Default flags. */
+};
- return MA_NOT_IMPLEMENTED;
+MA_API ma_result ma_bpf_node_init(ma_node_graph* pNodeGraph, const ma_bpf_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_bpf_node* pNode)
+{
+ ma_result result;
+ ma_node_config baseNodeConfig;
+
+ if (pNode == NULL) {
+ return MA_INVALID_ARGS;
}
- #endif
-}
-MA_API ma_result ma_stbvorbis_get_data_format(ma_stbvorbis* pVorbis, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
-{
- /* Defaults for safety. */
- if (pFormat != NULL) {
- *pFormat = ma_format_unknown;
+ MA_ZERO_OBJECT(pNode);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pChannels != NULL) {
- *pChannels = 0;
+
+ if (pConfig->bpf.format != ma_format_f32) {
+ return MA_INVALID_ARGS; /* The format must be f32. */
}
- if (pSampleRate != NULL) {
- *pSampleRate = 0;
+
+ result = ma_bpf_init(&pConfig->bpf, pAllocationCallbacks, &pNode->bpf);
+ if (result != MA_SUCCESS) {
+ return result;
}
- if (pChannelMap != NULL) {
- MA_ZERO_MEMORY(pChannelMap, sizeof(*pChannelMap) * channelMapCap);
+
+ baseNodeConfig = ma_node_config_init();
+ baseNodeConfig.vtable = &g_ma_bpf_node_vtable;
+ baseNodeConfig.pInputChannels = &pConfig->bpf.channels;
+ baseNodeConfig.pOutputChannels = &pConfig->bpf.channels;
+
+ result = ma_node_init(pNodeGraph, &baseNodeConfig, pAllocationCallbacks, pNode);
+ if (result != MA_SUCCESS) {
+ return result;
}
- if (pVorbis == NULL) {
- return MA_INVALID_OPERATION;
+ return result;
+}
+
+MA_API ma_result ma_bpf_node_reinit(const ma_bpf_config* pConfig, ma_bpf_node* pNode)
+{
+ ma_bpf_node* pBPFNode = (ma_bpf_node*)pNode;
+
+ if (pNode == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pFormat != NULL) {
- *pFormat = pVorbis->format;
+ return ma_bpf_reinit(pConfig, &pBPFNode->bpf);
+}
+
+MA_API void ma_bpf_node_uninit(ma_bpf_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ ma_bpf_node* pBPFNode = (ma_bpf_node*)pNode;
+
+ if (pNode == NULL) {
+ return;
}
- #if !defined(MA_NO_VORBIS)
- {
- if (pChannels != NULL) {
- *pChannels = pVorbis->channels;
- }
+ ma_node_uninit(pNode, pAllocationCallbacks);
+ ma_bpf_uninit(&pBPFNode->bpf, pAllocationCallbacks);
+}
- if (pSampleRate != NULL) {
- *pSampleRate = pVorbis->sampleRate;
- }
- if (pChannelMap != NULL) {
- ma_get_standard_channel_map(ma_standard_channel_map_vorbis, (ma_uint32)ma_min(pVorbis->channels, channelMapCap), pChannelMap);
- }
- return MA_SUCCESS;
- }
- #else
- {
- /* vorbis is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
- return MA_NOT_IMPLEMENTED;
- }
- #endif
+/*
+Notching Filter Node
+*/
+MA_API ma_notch_node_config ma_notch_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double q, double frequency)
+{
+ ma_notch_node_config config;
+
+ config.nodeConfig = ma_node_config_init();
+ config.notch = ma_notch2_config_init(ma_format_f32, channels, sampleRate, q, frequency);
+
+ return config;
}
-MA_API ma_result ma_stbvorbis_get_cursor_in_pcm_frames(ma_stbvorbis* pVorbis, ma_uint64* pCursor)
+static void ma_notch_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
- if (pCursor == NULL) {
+ ma_notch_node* pBPFNode = (ma_notch_node*)pNode;
+
+ MA_ASSERT(pNode != NULL);
+ (void)pFrameCountIn;
+
+ ma_notch2_process_pcm_frames(&pBPFNode->notch, ppFramesOut[0], ppFramesIn[0], *pFrameCountOut);
+}
+
+static ma_node_vtable g_ma_notch_node_vtable =
+{
+ ma_notch_node_process_pcm_frames,
+ NULL, /* onGetRequiredInputFrameCount */
+ 1, /* One input. */
+ 1, /* One output. */
+ 0 /* Default flags. */
+};
+
+MA_API ma_result ma_notch_node_init(ma_node_graph* pNodeGraph, const ma_notch_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_notch_node* pNode)
+{
+ ma_result result;
+ ma_node_config baseNodeConfig;
+
+ if (pNode == NULL) {
return MA_INVALID_ARGS;
}
- *pCursor = 0; /* Safety. */
+ MA_ZERO_OBJECT(pNode);
- if (pVorbis == NULL) {
+ if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
- #if !defined(MA_NO_VORBIS)
- {
- *pCursor = pVorbis->cursor;
+ if (pConfig->notch.format != ma_format_f32) {
+ return MA_INVALID_ARGS; /* The format must be f32. */
+ }
- return MA_SUCCESS;
+ result = ma_notch2_init(&pConfig->notch, pAllocationCallbacks, &pNode->notch);
+ if (result != MA_SUCCESS) {
+ return result;
}
- #else
- {
- /* vorbis is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
- return MA_NOT_IMPLEMENTED;
+
+ baseNodeConfig = ma_node_config_init();
+ baseNodeConfig.vtable = &g_ma_notch_node_vtable;
+ baseNodeConfig.pInputChannels = &pConfig->notch.channels;
+ baseNodeConfig.pOutputChannels = &pConfig->notch.channels;
+
+ result = ma_node_init(pNodeGraph, &baseNodeConfig, pAllocationCallbacks, pNode);
+ if (result != MA_SUCCESS) {
+ return result;
}
- #endif
+
+ return result;
}
-MA_API ma_result ma_stbvorbis_get_length_in_pcm_frames(ma_stbvorbis* pVorbis, ma_uint64* pLength)
+MA_API ma_result ma_notch_node_reinit(const ma_notch_config* pConfig, ma_notch_node* pNode)
{
- if (pLength == NULL) {
+ ma_notch_node* pNotchNode = (ma_notch_node*)pNode;
+
+ if (pNode == NULL) {
return MA_INVALID_ARGS;
}
- *pLength = 0; /* Safety. */
+ return ma_notch2_reinit(pConfig, &pNotchNode->notch);
+}
- if (pVorbis == NULL) {
- return MA_INVALID_ARGS;
- }
+MA_API void ma_notch_node_uninit(ma_notch_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ ma_notch_node* pNotchNode = (ma_notch_node*)pNode;
- #if !defined(MA_NO_VORBIS)
- {
- if (pVorbis->usingPushMode) {
- *pLength = 0; /* I don't know of a good way to determine this reliably with stb_vorbis and push mode. */
- } else {
- *pLength = stb_vorbis_stream_length_in_samples(pVorbis->stb);
- }
-
- return MA_SUCCESS;
- }
- #else
- {
- /* vorbis is disabled. Should never hit this since initialization would have failed. */
- MA_ASSERT(MA_FALSE);
- return MA_NOT_IMPLEMENTED;
+ if (pNode == NULL) {
+ return;
}
- #endif
+
+ ma_node_uninit(pNode, pAllocationCallbacks);
+ ma_notch2_uninit(&pNotchNode->notch, pAllocationCallbacks);
}
-static ma_result ma_decoding_backend_init__stbvorbis(void* pUserData, ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
+
+/*
+Peaking Filter Node
+*/
+MA_API ma_peak_node_config ma_peak_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double q, double frequency)
{
- ma_result result;
- ma_stbvorbis* pVorbis;
+ ma_peak_node_config config;
- (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
+ config.nodeConfig = ma_node_config_init();
+ config.peak = ma_peak2_config_init(ma_format_f32, channels, sampleRate, gainDB, q, frequency);
- /* For now we're just allocating the decoder backend on the heap. */
- pVorbis = (ma_stbvorbis*)ma_malloc(sizeof(*pVorbis), pAllocationCallbacks);
- if (pVorbis == NULL) {
- return MA_OUT_OF_MEMORY;
- }
+ return config;
+}
- result = ma_stbvorbis_init(onRead, onSeek, onTell, pReadSeekTellUserData, pConfig, pAllocationCallbacks, pVorbis);
- if (result != MA_SUCCESS) {
- ma_free(pVorbis, pAllocationCallbacks);
- return result;
- }
+static void ma_peak_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
+{
+ ma_peak_node* pBPFNode = (ma_peak_node*)pNode;
- *ppBackend = pVorbis;
+ MA_ASSERT(pNode != NULL);
+ (void)pFrameCountIn;
- return MA_SUCCESS;
+ ma_peak2_process_pcm_frames(&pBPFNode->peak, ppFramesOut[0], ppFramesIn[0], *pFrameCountOut);
}
-static ma_result ma_decoding_backend_init_file__stbvorbis(void* pUserData, const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
+static ma_node_vtable g_ma_peak_node_vtable =
+{
+ ma_peak_node_process_pcm_frames,
+ NULL, /* onGetRequiredInputFrameCount */
+ 1, /* One input. */
+ 1, /* One output. */
+ 0 /* Default flags. */
+};
+
+MA_API ma_result ma_peak_node_init(ma_node_graph* pNodeGraph, const ma_peak_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_peak_node* pNode)
{
ma_result result;
- ma_stbvorbis* pVorbis;
+ ma_node_config baseNodeConfig;
- (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
+ if (pNode == NULL) {
+ return MA_INVALID_ARGS;
+ }
- /* For now we're just allocating the decoder backend on the heap. */
- pVorbis = (ma_stbvorbis*)ma_malloc(sizeof(*pVorbis), pAllocationCallbacks);
- if (pVorbis == NULL) {
- return MA_OUT_OF_MEMORY;
+ MA_ZERO_OBJECT(pNode);
+
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
- result = ma_stbvorbis_init_file(pFilePath, pConfig, pAllocationCallbacks, pVorbis);
+ if (pConfig->peak.format != ma_format_f32) {
+ return MA_INVALID_ARGS; /* The format must be f32. */
+ }
+
+ result = ma_peak2_init(&pConfig->peak, pAllocationCallbacks, &pNode->peak);
if (result != MA_SUCCESS) {
- ma_free(pVorbis, pAllocationCallbacks);
+ ma_node_uninit(pNode, pAllocationCallbacks);
return result;
}
- *ppBackend = pVorbis;
+ baseNodeConfig = ma_node_config_init();
+ baseNodeConfig.vtable = &g_ma_peak_node_vtable;
+ baseNodeConfig.pInputChannels = &pConfig->peak.channels;
+ baseNodeConfig.pOutputChannels = &pConfig->peak.channels;
- return MA_SUCCESS;
+ result = ma_node_init(pNodeGraph, &baseNodeConfig, pAllocationCallbacks, pNode);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ return result;
}
-static ma_result ma_decoding_backend_init_memory__stbvorbis(void* pUserData, const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
+MA_API ma_result ma_peak_node_reinit(const ma_peak_config* pConfig, ma_peak_node* pNode)
{
- ma_result result;
- ma_stbvorbis* pVorbis;
-
- (void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
+ ma_peak_node* pPeakNode = (ma_peak_node*)pNode;
- /* For now we're just allocating the decoder backend on the heap. */
- pVorbis = (ma_stbvorbis*)ma_malloc(sizeof(*pVorbis), pAllocationCallbacks);
- if (pVorbis == NULL) {
- return MA_OUT_OF_MEMORY;
+ if (pNode == NULL) {
+ return MA_INVALID_ARGS;
}
- result = ma_stbvorbis_init_memory(pData, dataSize, pConfig, pAllocationCallbacks, pVorbis);
- if (result != MA_SUCCESS) {
- ma_free(pVorbis, pAllocationCallbacks);
- return result;
- }
+ return ma_peak2_reinit(pConfig, &pPeakNode->peak);
+}
- *ppBackend = pVorbis;
+MA_API void ma_peak_node_uninit(ma_peak_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+ ma_peak_node* pPeakNode = (ma_peak_node*)pNode;
- return MA_SUCCESS;
+ if (pNode == NULL) {
+ return;
+ }
+
+ ma_node_uninit(pNode, pAllocationCallbacks);
+ ma_peak2_uninit(&pPeakNode->peak, pAllocationCallbacks);
}
-static void ma_decoding_backend_uninit__stbvorbis(void* pUserData, ma_data_source* pBackend, const ma_allocation_callbacks* pAllocationCallbacks)
+
+
+/*
+Low Shelf Filter Node
+*/
+MA_API ma_loshelf_node_config ma_loshelf_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double q, double frequency)
{
- ma_stbvorbis* pVorbis = (ma_stbvorbis*)pBackend;
+ ma_loshelf_node_config config;
- (void)pUserData;
+ config.nodeConfig = ma_node_config_init();
+ config.loshelf = ma_loshelf2_config_init(ma_format_f32, channels, sampleRate, gainDB, q, frequency);
- ma_stbvorbis_uninit(pVorbis, pAllocationCallbacks);
- ma_free(pVorbis, pAllocationCallbacks);
+ return config;
}
-static ma_result ma_decoding_backend_get_channel_map__stbvorbis(void* pUserData, ma_data_source* pBackend, ma_channel* pChannelMap, size_t channelMapCap)
+static void ma_loshelf_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
- ma_stbvorbis* pVorbis = (ma_stbvorbis*)pBackend;
+ ma_loshelf_node* pBPFNode = (ma_loshelf_node*)pNode;
- (void)pUserData;
+ MA_ASSERT(pNode != NULL);
+ (void)pFrameCountIn;
- return ma_stbvorbis_get_data_format(pVorbis, NULL, NULL, NULL, pChannelMap, channelMapCap);
+ ma_loshelf2_process_pcm_frames(&pBPFNode->loshelf, ppFramesOut[0], ppFramesIn[0], *pFrameCountOut);
}
-static ma_decoding_backend_vtable g_ma_decoding_backend_vtable_stbvorbis =
+static ma_node_vtable g_ma_loshelf_node_vtable =
{
- ma_decoding_backend_init__stbvorbis,
- ma_decoding_backend_init_file__stbvorbis,
- NULL, /* onInitFileW() */
- ma_decoding_backend_init_memory__stbvorbis,
- ma_decoding_backend_uninit__stbvorbis,
- ma_decoding_backend_get_channel_map__stbvorbis
+ ma_loshelf_node_process_pcm_frames,
+ NULL, /* onGetRequiredInputFrameCount */
+ 1, /* One input. */
+ 1, /* One output. */
+ 0 /* Default flags. */
};
-static ma_result ma_decoder_init_vorbis__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API ma_result ma_loshelf_node_init(ma_node_graph* pNodeGraph, const ma_loshelf_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_loshelf_node* pNode)
{
- return ma_decoder_init_from_vtable(&g_ma_decoding_backend_vtable_stbvorbis, NULL, pConfig, pDecoder);
-}
-#endif /* STB_VORBIS_INCLUDE_STB_VORBIS_H */
-
+ ma_result result;
+ ma_node_config baseNodeConfig;
+ if (pNode == NULL) {
+ return MA_INVALID_ARGS;
+ }
-static ma_result ma_decoder__init_allocation_callbacks(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
-{
- MA_ASSERT(pDecoder != NULL);
+ MA_ZERO_OBJECT(pNode);
- if (pConfig != NULL) {
- return ma_allocation_callbacks_init_copy(&pDecoder->allocationCallbacks, &pConfig->allocationCallbacks);
- } else {
- pDecoder->allocationCallbacks = ma_allocation_callbacks_init_default();
- return MA_SUCCESS;
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
}
-}
-static ma_result ma_decoder__data_source_on_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
-{
- ma_uint64 framesRead = ma_decoder_read_pcm_frames((ma_decoder*)pDataSource, pFramesOut, frameCount);
+ if (pConfig->loshelf.format != ma_format_f32) {
+ return MA_INVALID_ARGS; /* The format must be f32. */
+ }
- if (pFramesRead != NULL) {
- *pFramesRead = framesRead;
+ result = ma_loshelf2_init(&pConfig->loshelf, pAllocationCallbacks, &pNode->loshelf);
+ if (result != MA_SUCCESS) {
+ return result;
}
- if (framesRead == 0) {
- return MA_AT_END;
+ baseNodeConfig = ma_node_config_init();
+ baseNodeConfig.vtable = &g_ma_loshelf_node_vtable;
+ baseNodeConfig.pInputChannels = &pConfig->loshelf.channels;
+ baseNodeConfig.pOutputChannels = &pConfig->loshelf.channels;
+
+ result = ma_node_init(pNodeGraph, &baseNodeConfig, pAllocationCallbacks, pNode);
+ if (result != MA_SUCCESS) {
+ return result;
}
- return MA_SUCCESS;
+ return result;
}
-static ma_result ma_decoder__data_source_on_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
+MA_API ma_result ma_loshelf_node_reinit(const ma_loshelf_config* pConfig, ma_loshelf_node* pNode)
{
- return ma_decoder_seek_to_pcm_frame((ma_decoder*)pDataSource, frameIndex);
+ ma_loshelf_node* pLoshelfNode = (ma_loshelf_node*)pNode;
+
+ if (pNode == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ return ma_loshelf2_reinit(pConfig, &pLoshelfNode->loshelf);
}
-static ma_result ma_decoder__data_source_on_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate)
+MA_API void ma_loshelf_node_uninit(ma_loshelf_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_decoder* pDecoder = (ma_decoder*)pDataSource;
+ ma_loshelf_node* pLoshelfNode = (ma_loshelf_node*)pNode;
- *pFormat = pDecoder->outputFormat;
- *pChannels = pDecoder->outputChannels;
- *pSampleRate = pDecoder->outputSampleRate;
+ if (pNode == NULL) {
+ return;
+ }
- return MA_SUCCESS;
+ ma_node_uninit(pNode, pAllocationCallbacks);
+ ma_loshelf2_uninit(&pLoshelfNode->loshelf, pAllocationCallbacks);
}
-static ma_result ma_decoder__data_source_on_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
+
+
+/*
+High Shelf Filter Node
+*/
+MA_API ma_hishelf_node_config ma_hishelf_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double q, double frequency)
{
- ma_decoder* pDecoder = (ma_decoder*)pDataSource;
+ ma_hishelf_node_config config;
+
+ config.nodeConfig = ma_node_config_init();
+ config.hishelf = ma_hishelf2_config_init(ma_format_f32, channels, sampleRate, gainDB, q, frequency);
- return ma_decoder_get_cursor_in_pcm_frames(pDecoder, pCursor);
+ return config;
}
-static ma_result ma_decoder__data_source_on_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
+static void ma_hishelf_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
- ma_decoder* pDecoder = (ma_decoder*)pDataSource;
+ ma_hishelf_node* pBPFNode = (ma_hishelf_node*)pNode;
- *pLength = ma_decoder_get_length_in_pcm_frames(pDecoder);
- if (*pLength == 0) {
- return MA_NOT_IMPLEMENTED;
- }
+ MA_ASSERT(pNode != NULL);
+ (void)pFrameCountIn;
- return MA_SUCCESS;
+ ma_hishelf2_process_pcm_frames(&pBPFNode->hishelf, ppFramesOut[0], ppFramesIn[0], *pFrameCountOut);
}
-static ma_data_source_vtable g_ma_decoder_data_source_vtable =
+static ma_node_vtable g_ma_hishelf_node_vtable =
{
- ma_decoder__data_source_on_read,
- ma_decoder__data_source_on_seek,
- NULL, /* onMap */
- NULL, /* onUnmap */
- ma_decoder__data_source_on_get_data_format,
- ma_decoder__data_source_on_get_cursor,
- ma_decoder__data_source_on_get_length
+ ma_hishelf_node_process_pcm_frames,
+ NULL, /* onGetRequiredInputFrameCount */
+ 1, /* One input. */
+ 1, /* One output. */
+ 0 /* Default flags. */
};
-static ma_result ma_decoder__preinit(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, ma_decoder_tell_proc onTell, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API ma_result ma_hishelf_node_init(ma_node_graph* pNodeGraph, const ma_hishelf_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hishelf_node* pNode)
{
ma_result result;
- ma_data_source_config dataSourceConfig;
-
- MA_ASSERT(pConfig != NULL);
+ ma_node_config baseNodeConfig;
- if (pDecoder == NULL) {
+ if (pNode == NULL) {
return MA_INVALID_ARGS;
}
- MA_ZERO_OBJECT(pDecoder);
+ MA_ZERO_OBJECT(pNode);
- if (onRead == NULL || onSeek == NULL) {
+ if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
- dataSourceConfig = ma_data_source_config_init();
- dataSourceConfig.vtable = &g_ma_decoder_data_source_vtable;
-
- result = ma_data_source_init(&dataSourceConfig, &pDecoder->ds);
- if (result != MA_SUCCESS) {
- return result;
+ if (pConfig->hishelf.format != ma_format_f32) {
+ return MA_INVALID_ARGS; /* The format must be f32. */
}
- pDecoder->onRead = onRead;
- pDecoder->onSeek = onSeek;
- pDecoder->onTell = onTell;
- pDecoder->pUserData = pUserData;
-
- result = ma_decoder__init_allocation_callbacks(pConfig, pDecoder);
+ result = ma_hishelf2_init(&pConfig->hishelf, pAllocationCallbacks, &pNode->hishelf);
if (result != MA_SUCCESS) {
- ma_data_source_uninit(&pDecoder->ds);
return result;
}
- return MA_SUCCESS;
-}
-
-static ma_result ma_decoder__postinit(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
-{
- ma_result result = MA_SUCCESS;
-
- /* Basic validation in case the internal decoder supports different limits to miniaudio. */
- {
- /* TODO: Remove this block once we remove MA_MIN_CHANNELS and MA_MAX_CHANNELS. */
- ma_uint32 internalChannels;
- ma_data_source_get_data_format(pDecoder->pBackend, NULL, &internalChannels, NULL);
-
- if (internalChannels < MA_MIN_CHANNELS || internalChannels > MA_MAX_CHANNELS) {
- result = MA_INVALID_DATA;
- }
- }
-
- if (result == MA_SUCCESS) {
- result = ma_decoder__init_data_converter(pDecoder, pConfig);
- }
+ baseNodeConfig = ma_node_config_init();
+ baseNodeConfig.vtable = &g_ma_hishelf_node_vtable;
+ baseNodeConfig.pInputChannels = &pConfig->hishelf.channels;
+ baseNodeConfig.pOutputChannels = &pConfig->hishelf.channels;
- /* If we failed post initialization we need to uninitialize the decoder before returning to prevent a memory leak. */
+ result = ma_node_init(pNodeGraph, &baseNodeConfig, pAllocationCallbacks, pNode);
if (result != MA_SUCCESS) {
- ma_decoder_uninit(pDecoder);
return result;
}
return result;
}
-MA_API ma_result ma_decoder_init_wav(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API ma_result ma_hishelf_node_reinit(const ma_hishelf_config* pConfig, ma_hishelf_node* pNode)
{
-#ifdef MA_HAS_WAV
- ma_decoder_config config;
+ ma_hishelf_node* pHishelfNode = (ma_hishelf_node*)pNode;
- config = ma_decoder_config_init_copy(pConfig);
- config.encodingFormat = ma_encoding_format_wav;
+ if (pNode == NULL) {
+ return MA_INVALID_ARGS;
+ }
- return ma_decoder_init(onRead, onSeek, pUserData, &config, pDecoder);
-#else
- (void)onRead;
- (void)onSeek;
- (void)pUserData;
- (void)pConfig;
- (void)pDecoder;
- return MA_NO_BACKEND;
-#endif
+ return ma_hishelf2_reinit(pConfig, &pHishelfNode->hishelf);
}
-MA_API ma_result ma_decoder_init_flac(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API void ma_hishelf_node_uninit(ma_hishelf_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
-#ifdef MA_HAS_FLAC
- ma_decoder_config config;
+ ma_hishelf_node* pHishelfNode = (ma_hishelf_node*)pNode;
- config = ma_decoder_config_init_copy(pConfig);
- config.encodingFormat = ma_encoding_format_flac;
+ if (pNode == NULL) {
+ return;
+ }
- return ma_decoder_init(onRead, onSeek, pUserData, &config, pDecoder);
-#else
- (void)onRead;
- (void)onSeek;
- (void)pUserData;
- (void)pConfig;
- (void)pDecoder;
- return MA_NO_BACKEND;
-#endif
+ ma_node_uninit(pNode, pAllocationCallbacks);
+ ma_hishelf2_uninit(&pHishelfNode->hishelf, pAllocationCallbacks);
}
-MA_API ma_result ma_decoder_init_mp3(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
-{
-#ifdef MA_HAS_MP3
- ma_decoder_config config;
- config = ma_decoder_config_init_copy(pConfig);
- config.encodingFormat = ma_encoding_format_mp3;
- return ma_decoder_init(onRead, onSeek, pUserData, &config, pDecoder);
-#else
- (void)onRead;
- (void)onSeek;
- (void)pUserData;
- (void)pConfig;
- (void)pDecoder;
- return MA_NO_BACKEND;
-#endif
-}
-MA_API ma_result ma_decoder_init_vorbis(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API ma_delay_node_config ma_delay_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 delayInFrames, float decay)
{
-#ifdef MA_HAS_VORBIS
- ma_decoder_config config;
+ ma_delay_node_config config;
- config = ma_decoder_config_init_copy(pConfig);
- config.encodingFormat = ma_encoding_format_vorbis;
+ config.nodeConfig = ma_node_config_init();
+ config.delay = ma_delay_config_init(channels, sampleRate, delayInFrames, decay);
- return ma_decoder_init(onRead, onSeek, pUserData, &config, pDecoder);
-#else
- (void)onRead;
- (void)onSeek;
- (void)pUserData;
- (void)pConfig;
- (void)pDecoder;
- return MA_NO_BACKEND;
-#endif
+ return config;
}
-
-static ma_result ma_decoder_init__internal(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+static void ma_delay_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
- ma_result result = MA_NO_BACKEND;
+ ma_delay_node* pDelayNode = (ma_delay_node*)pNode;
- MA_ASSERT(pConfig != NULL);
- MA_ASSERT(pDecoder != NULL);
+ (void)pFrameCountIn;
- /* Silence some warnings in the case that we don't have any decoder backends enabled. */
- (void)onRead;
- (void)onSeek;
- (void)pUserData;
+ ma_delay_process_pcm_frames(&pDelayNode->delay, ppFramesOut[0], ppFramesIn[0], *pFrameCountOut);
+}
+static ma_node_vtable g_ma_delay_node_vtable =
+{
+ ma_delay_node_process_pcm_frames,
+ NULL,
+ 1, /* 1 input channels. */
+ 1, /* 1 output channel. */
+ MA_NODE_FLAG_CONTINUOUS_PROCESSING /* Delay requires continuous processing to ensure the tail get's processed. */
+};
- /* If we've specified a specific encoding type, try that first. */
- if (pConfig->encodingFormat != ma_encoding_format_unknown) {
- #ifdef MA_HAS_WAV
- if (pConfig->encodingFormat == ma_encoding_format_wav) {
- result = ma_decoder_init_wav__internal(pConfig, pDecoder);
- }
- #endif
- #ifdef MA_HAS_FLAC
- if (pConfig->encodingFormat == ma_encoding_format_flac) {
- result = ma_decoder_init_flac__internal(pConfig, pDecoder);
- }
- #endif
- #ifdef MA_HAS_MP3
- if (pConfig->encodingFormat == ma_encoding_format_mp3) {
- result = ma_decoder_init_mp3__internal(pConfig, pDecoder);
- }
- #endif
- #ifdef MA_HAS_VORBIS
- if (pConfig->encodingFormat == ma_encoding_format_vorbis) {
- result = ma_decoder_init_vorbis__internal(pConfig, pDecoder);
- }
- #endif
+MA_API ma_result ma_delay_node_init(ma_node_graph* pNodeGraph, const ma_delay_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_delay_node* pDelayNode)
+{
+ ma_result result;
+ ma_node_config baseConfig;
- /* If we weren't able to initialize the decoder, seek back to the start to give the next attempts a clean start. */
- if (result != MA_SUCCESS) {
- onSeek(pDecoder, 0, ma_seek_origin_start);
- }
+ if (pDelayNode == NULL) {
+ return MA_INVALID_ARGS;
}
- if (result != MA_SUCCESS) {
- /* Getting here means we couldn't load a specific decoding backend based on the encoding format. */
-
- /*
- We use trial and error to open a decoder. We prioritize custom decoders so that if they
- implement the same encoding format they take priority over the built-in decoders.
- */
- if (result != MA_SUCCESS) {
- result = ma_decoder_init_custom__internal(pConfig, pDecoder);
- if (result != MA_SUCCESS) {
- onSeek(pDecoder, 0, ma_seek_origin_start);
- }
- }
-
- /*
- If we get to this point and we still haven't found a decoder, and the caller has requested a
- specific encoding format, there's no hope for it. Abort.
- */
- if (pConfig->encodingFormat != ma_encoding_format_unknown) {
- return MA_NO_BACKEND;
- }
+ MA_ZERO_OBJECT(pDelayNode);
- #ifdef MA_HAS_WAV
- if (result != MA_SUCCESS) {
- result = ma_decoder_init_wav__internal(pConfig, pDecoder);
- if (result != MA_SUCCESS) {
- onSeek(pDecoder, 0, ma_seek_origin_start);
- }
- }
- #endif
- #ifdef MA_HAS_FLAC
- if (result != MA_SUCCESS) {
- result = ma_decoder_init_flac__internal(pConfig, pDecoder);
- if (result != MA_SUCCESS) {
- onSeek(pDecoder, 0, ma_seek_origin_start);
- }
- }
- #endif
- #ifdef MA_HAS_MP3
- if (result != MA_SUCCESS) {
- result = ma_decoder_init_mp3__internal(pConfig, pDecoder);
- if (result != MA_SUCCESS) {
- onSeek(pDecoder, 0, ma_seek_origin_start);
- }
- }
- #endif
- #ifdef MA_HAS_VORBIS
- if (result != MA_SUCCESS) {
- result = ma_decoder_init_vorbis__internal(pConfig, pDecoder);
- if (result != MA_SUCCESS) {
- onSeek(pDecoder, 0, ma_seek_origin_start);
- }
- }
- #endif
+ result = ma_delay_init(&pConfig->delay, pAllocationCallbacks, &pDelayNode->delay);
+ if (result != MA_SUCCESS) {
+ return result;
}
+ baseConfig = pConfig->nodeConfig;
+ baseConfig.vtable = &g_ma_delay_node_vtable;
+ baseConfig.pInputChannels = &pConfig->delay.channels;
+ baseConfig.pOutputChannels = &pConfig->delay.channels;
+
+ result = ma_node_init(pNodeGraph, &baseConfig, pAllocationCallbacks, &pDelayNode->baseNode);
if (result != MA_SUCCESS) {
+ ma_delay_uninit(&pDelayNode->delay, pAllocationCallbacks);
return result;
}
- return ma_decoder__postinit(pConfig, pDecoder);
+ return result;
}
-MA_API ma_result ma_decoder_init(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API void ma_delay_node_uninit(ma_delay_node* pDelayNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
- ma_decoder_config config;
- ma_result result;
+ if (pDelayNode == NULL) {
+ return;
+ }
- config = ma_decoder_config_init_copy(pConfig);
+ /* The base node is always uninitialized first. */
+ ma_node_uninit(pDelayNode, pAllocationCallbacks);
+ ma_delay_uninit(&pDelayNode->delay, pAllocationCallbacks);
+}
- result = ma_decoder__preinit(onRead, onSeek, NULL, pUserData, &config, pDecoder);
- if (result != MA_SUCCESS) {
- return result;
+MA_API void ma_delay_node_set_wet(ma_delay_node* pDelayNode, float value)
+{
+ if (pDelayNode == NULL) {
+ return;
}
- return ma_decoder_init__internal(onRead, onSeek, pUserData, &config, pDecoder);
+ ma_delay_set_wet(&pDelayNode->delay, value);
}
+MA_API float ma_delay_node_get_wet(const ma_delay_node* pDelayNode)
+{
+ if (pDelayNode == NULL) {
+ return 0;
+ }
+
+ return ma_delay_get_wet(&pDelayNode->delay);
+}
-static size_t ma_decoder__on_read_memory(ma_decoder* pDecoder, void* pBufferOut, size_t bytesToRead)
+MA_API void ma_delay_node_set_dry(ma_delay_node* pDelayNode, float value)
{
- size_t bytesRemaining;
+ if (pDelayNode == NULL) {
+ return;
+ }
- MA_ASSERT(pDecoder->data.memory.dataSize >= pDecoder->data.memory.currentReadPos);
+ ma_delay_set_dry(&pDelayNode->delay, value);
+}
- bytesRemaining = pDecoder->data.memory.dataSize - pDecoder->data.memory.currentReadPos;
- if (bytesToRead > bytesRemaining) {
- bytesToRead = bytesRemaining;
+MA_API float ma_delay_node_get_dry(const ma_delay_node* pDelayNode)
+{
+ if (pDelayNode == NULL) {
+ return 0;
}
- if (bytesToRead > 0) {
- MA_COPY_MEMORY(pBufferOut, pDecoder->data.memory.pData + pDecoder->data.memory.currentReadPos, bytesToRead);
- pDecoder->data.memory.currentReadPos += bytesToRead;
+ return ma_delay_get_dry(&pDelayNode->delay);
+}
+
+MA_API void ma_delay_node_set_decay(ma_delay_node* pDelayNode, float value)
+{
+ if (pDelayNode == NULL) {
+ return;
}
- return bytesToRead;
+ ma_delay_set_decay(&pDelayNode->delay, value);
}
-static ma_bool32 ma_decoder__on_seek_memory(ma_decoder* pDecoder, ma_int64 byteOffset, ma_seek_origin origin)
+MA_API float ma_delay_node_get_decay(const ma_delay_node* pDelayNode)
{
- if (byteOffset > 0 && (ma_uint64)byteOffset > MA_SIZE_MAX) {
- return MA_FALSE; /* Too far. */
+ if (pDelayNode == NULL) {
+ return 0;
}
- if (origin == ma_seek_origin_current) {
- if (byteOffset > 0) {
- if (pDecoder->data.memory.currentReadPos + byteOffset > pDecoder->data.memory.dataSize) {
- byteOffset = (ma_int64)(pDecoder->data.memory.dataSize - pDecoder->data.memory.currentReadPos); /* Trying to seek too far forward. */
- }
+ return ma_delay_get_decay(&pDelayNode->delay);
+}
+#endif /* MA_NO_NODE_GRAPH */
- pDecoder->data.memory.currentReadPos += (size_t)byteOffset;
- } else {
- if (pDecoder->data.memory.currentReadPos < (size_t)-byteOffset) {
- byteOffset = -(ma_int64)pDecoder->data.memory.currentReadPos; /* Trying to seek too far backwards. */
- }
- pDecoder->data.memory.currentReadPos -= (size_t)-byteOffset;
- }
- } else {
- if (origin == ma_seek_origin_end) {
- if (byteOffset < 0) {
- byteOffset = -byteOffset;
- }
+#if !defined(MA_NO_ENGINE) && !defined(MA_NO_NODE_GRAPH)
+/**************************************************************************************************************************************************************
- if (byteOffset > (ma_int64)pDecoder->data.memory.dataSize) {
- pDecoder->data.memory.currentReadPos = 0; /* Trying to seek too far back. */
- } else {
- pDecoder->data.memory.currentReadPos = pDecoder->data.memory.dataSize - (size_t)byteOffset;
- }
- } else {
- if ((size_t)byteOffset <= pDecoder->data.memory.dataSize) {
- pDecoder->data.memory.currentReadPos = (size_t)byteOffset;
- } else {
- pDecoder->data.memory.currentReadPos = pDecoder->data.memory.dataSize; /* Trying to seek too far forward. */
- }
- }
- }
+Engine
- return MA_TRUE;
-}
+**************************************************************************************************************************************************************/
+#define MA_SEEK_TARGET_NONE (~(ma_uint64)0)
-static ma_result ma_decoder__on_tell_memory(ma_decoder* pDecoder, ma_int64* pCursor)
+MA_API ma_engine_node_config ma_engine_node_config_init(ma_engine* pEngine, ma_engine_node_type type, ma_uint32 flags)
{
- MA_ASSERT(pDecoder != NULL);
- MA_ASSERT(pCursor != NULL);
+ ma_engine_node_config config;
- *pCursor = (ma_int64)pDecoder->data.memory.currentReadPos;
+ MA_ZERO_OBJECT(&config);
+ config.pEngine = pEngine;
+ config.type = type;
+ config.isPitchDisabled = (flags & MA_SOUND_FLAG_NO_PITCH) != 0;
+ config.isSpatializationDisabled = (flags & MA_SOUND_FLAG_NO_SPATIALIZATION) != 0;
- return MA_SUCCESS;
+ return config;
}
-static ma_result ma_decoder__preinit_memory(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+
+static void ma_engine_node_update_pitch_if_required(ma_engine_node* pEngineNode)
{
- ma_result result = ma_decoder__preinit(ma_decoder__on_read_memory, ma_decoder__on_seek_memory, ma_decoder__on_tell_memory, NULL, pConfig, pDecoder);
- if (result != MA_SUCCESS) {
- return result;
- }
+ ma_bool32 isUpdateRequired = MA_FALSE;
+ float newPitch;
- if (pData == NULL || dataSize == 0) {
- return MA_INVALID_ARGS;
+ MA_ASSERT(pEngineNode != NULL);
+
+ newPitch = c89atomic_load_explicit_f32(&pEngineNode->pitch, c89atomic_memory_order_acquire);
+
+ if (pEngineNode->oldPitch != newPitch) {
+ pEngineNode->oldPitch = newPitch;
+ isUpdateRequired = MA_TRUE;
}
- pDecoder->data.memory.pData = (const ma_uint8*)pData;
- pDecoder->data.memory.dataSize = dataSize;
- pDecoder->data.memory.currentReadPos = 0;
+ if (pEngineNode->oldDopplerPitch != pEngineNode->spatializer.dopplerPitch) {
+ pEngineNode->oldDopplerPitch = pEngineNode->spatializer.dopplerPitch;
+ isUpdateRequired = MA_TRUE;
+ }
- (void)pConfig;
- return MA_SUCCESS;
+ if (isUpdateRequired) {
+ float basePitch = (float)pEngineNode->sampleRate / ma_engine_get_sample_rate(pEngineNode->pEngine);
+ ma_linear_resampler_set_rate_ratio(&pEngineNode->resampler, basePitch * pEngineNode->oldPitch * pEngineNode->oldDopplerPitch);
+ }
}
-MA_API ma_result ma_decoder_init_memory(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+static ma_bool32 ma_engine_node_is_pitching_enabled(const ma_engine_node* pEngineNode)
{
- ma_decoder_config config;
- ma_result result;
+ MA_ASSERT(pEngineNode != NULL);
- config = ma_decoder_config_init_copy(pConfig); /* Make sure the config is not NULL. */
+ /* Don't try to be clever by skiping resampling in the pitch=1 case or else you'll glitch when moving away from 1. */
+ return !c89atomic_load_explicit_32(&pEngineNode->isPitchDisabled, c89atomic_memory_order_acquire);
+}
- result = ma_decoder__preinit_memory(pData, dataSize, &config, pDecoder);
- if (result != MA_SUCCESS) {
- return result;
- }
+static ma_bool32 ma_engine_node_is_spatialization_enabled(const ma_engine_node* pEngineNode)
+{
+ MA_ASSERT(pEngineNode != NULL);
- return ma_decoder_init__internal(ma_decoder__on_read_memory, ma_decoder__on_seek_memory, NULL, &config, pDecoder);
+ return !c89atomic_load_explicit_32(&pEngineNode->isSpatializationDisabled, c89atomic_memory_order_acquire);
}
-MA_API ma_result ma_decoder_init_memory_wav(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+static ma_uint64 ma_engine_node_get_required_input_frame_count(const ma_engine_node* pEngineNode, ma_uint64 outputFrameCount)
{
-#ifdef MA_HAS_WAV
- ma_decoder_config config;
+ ma_uint64 inputFrameCount = 0;
- config = ma_decoder_config_init_copy(pConfig); /* Make sure the config is not NULL. */
- config.encodingFormat = ma_encoding_format_wav;
+ if (ma_engine_node_is_pitching_enabled(pEngineNode)) {
+ ma_result result = ma_linear_resampler_get_required_input_frame_count(&pEngineNode->resampler, outputFrameCount, &inputFrameCount);
+ if (result != MA_SUCCESS) {
+ inputFrameCount = 0;
+ }
+ } else {
+ inputFrameCount = outputFrameCount; /* No resampling, so 1:1. */
+ }
- return ma_decoder_init_memory(pData, dataSize, &config, pDecoder);
-#else
- (void)pData;
- (void)dataSize;
- (void)pConfig;
- (void)pDecoder;
- return MA_NO_BACKEND;
-#endif
+ return inputFrameCount;
}
-MA_API ma_result ma_decoder_init_memory_flac(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+static void ma_engine_node_process_pcm_frames__general(ma_engine_node* pEngineNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
-#ifdef MA_HAS_FLAC
- ma_decoder_config config;
+ ma_uint32 frameCountIn;
+ ma_uint32 frameCountOut;
+ ma_uint32 totalFramesProcessedIn;
+ ma_uint32 totalFramesProcessedOut;
+ ma_uint32 channelsIn;
+ ma_uint32 channelsOut;
+ ma_bool32 isPitchingEnabled;
+ ma_bool32 isFadingEnabled;
+ ma_bool32 isSpatializationEnabled;
+ ma_bool32 isPanningEnabled;
- config = ma_decoder_config_init_copy(pConfig); /* Make sure the config is not NULL. */
- config.encodingFormat = ma_encoding_format_flac;
+ frameCountIn = *pFrameCountIn;
+ frameCountOut = *pFrameCountOut;
- return ma_decoder_init_memory(pData, dataSize, &config, pDecoder);
-#else
- (void)pData;
- (void)dataSize;
- (void)pConfig;
- (void)pDecoder;
- return MA_NO_BACKEND;
-#endif
-}
+ channelsIn = ma_spatializer_get_input_channels(&pEngineNode->spatializer);
+ channelsOut = ma_spatializer_get_output_channels(&pEngineNode->spatializer);
-MA_API ma_result ma_decoder_init_memory_mp3(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
-{
-#ifdef MA_HAS_MP3
- ma_decoder_config config;
+ totalFramesProcessedIn = 0;
+ totalFramesProcessedOut = 0;
- config = ma_decoder_config_init_copy(pConfig); /* Make sure the config is not NULL. */
- config.encodingFormat = ma_encoding_format_mp3;
+ isPitchingEnabled = ma_engine_node_is_pitching_enabled(pEngineNode);
+ isFadingEnabled = pEngineNode->fader.volumeBeg != 1 || pEngineNode->fader.volumeEnd != 1;
+ isSpatializationEnabled = ma_engine_node_is_spatialization_enabled(pEngineNode);
+ isPanningEnabled = pEngineNode->panner.pan != 0 && channelsOut != 1;
- return ma_decoder_init_memory(pData, dataSize, &config, pDecoder);
-#else
- (void)pData;
- (void)dataSize;
- (void)pConfig;
- (void)pDecoder;
- return MA_NO_BACKEND;
-#endif
-}
+ /* Keep going while we've still got data available for processing. */
+ while (totalFramesProcessedOut < frameCountOut) {
+ /*
+ We need to process in a specific order. We always do resampling first because it's likely
+ we're going to be increasing the channel count after spatialization. Also, I want to do
+ fading based on the output sample rate.
+
+ We'll first read into a buffer from the resampler. Then we'll do all processing that
+ operates on the on the input channel count. We'll then get the spatializer to output to
+ the output buffer and then do all effects from that point directly in the output buffer
+ in-place.
+
+ Note that we're always running the resampler. If we try to be clever and skip resampling
+ when the pitch is 1, we'll get a glitch when we move away from 1, back to 1, and then
+ away from 1 again. We'll want to implement any pitch=1 optimizations in the resampler
+ itself.
+
+ There's a small optimization here that we'll utilize since it might be a fairly common
+ case. When the input and output channel counts are the same, we'll read straight into the
+ output buffer from the resampler and do everything in-place.
+ */
+ const float* pRunningFramesIn;
+ float* pRunningFramesOut;
+ float* pWorkingBuffer; /* This is the buffer that we'll be processing frames in. This is in input channels. */
+ float temp[MA_DATA_CONVERTER_STACK_BUFFER_SIZE / sizeof(float)];
+ ma_uint32 tempCapInFrames = ma_countof(temp) / channelsIn;
+ ma_uint32 framesAvailableIn;
+ ma_uint32 framesAvailableOut;
+ ma_uint32 framesJustProcessedIn;
+ ma_uint32 framesJustProcessedOut;
+ ma_bool32 isWorkingBufferValid = MA_FALSE;
+
+ framesAvailableIn = frameCountIn - totalFramesProcessedIn;
+ framesAvailableOut = frameCountOut - totalFramesProcessedOut;
+
+ pRunningFramesIn = ma_offset_pcm_frames_const_ptr_f32(ppFramesIn[0], totalFramesProcessedIn, channelsIn);
+ pRunningFramesOut = ma_offset_pcm_frames_ptr_f32(ppFramesOut[0], totalFramesProcessedOut, channelsOut);
+
+ if (channelsIn == channelsOut) {
+ /* Fast path. Channel counts are the same. No need for an intermediary input buffer. */
+ pWorkingBuffer = pRunningFramesOut;
+ } else {
+ /* Slow path. Channel counts are different. Need to use an intermediary input buffer. */
+ pWorkingBuffer = temp;
+ if (framesAvailableOut > tempCapInFrames) {
+ framesAvailableOut = tempCapInFrames;
+ }
+ }
-MA_API ma_result ma_decoder_init_memory_vorbis(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
-{
-#ifdef MA_HAS_VORBIS
- ma_decoder_config config;
+ /* First is resampler. */
+ if (isPitchingEnabled) {
+ ma_uint64 resampleFrameCountIn = framesAvailableIn;
+ ma_uint64 resampleFrameCountOut = framesAvailableOut;
- config = ma_decoder_config_init_copy(pConfig); /* Make sure the config is not NULL. */
- config.encodingFormat = ma_encoding_format_vorbis;
+ ma_linear_resampler_process_pcm_frames(&pEngineNode->resampler, pRunningFramesIn, &resampleFrameCountIn, pWorkingBuffer, &resampleFrameCountOut);
+ isWorkingBufferValid = MA_TRUE;
- return ma_decoder_init_memory(pData, dataSize, &config, pDecoder);
-#else
- (void)pData;
- (void)dataSize;
- (void)pConfig;
- (void)pDecoder;
- return MA_NO_BACKEND;
-#endif
-}
+ framesJustProcessedIn = (ma_uint32)resampleFrameCountIn;
+ framesJustProcessedOut = (ma_uint32)resampleFrameCountOut;
+ } else {
+ framesJustProcessedIn = ma_min(framesAvailableIn, framesAvailableOut);
+ framesJustProcessedOut = framesJustProcessedIn; /* When no resampling is being performed, the number of output frames is the same as input frames. */
+ }
+ /* Fading. */
+ if (isFadingEnabled) {
+ if (isWorkingBufferValid) {
+ ma_fader_process_pcm_frames(&pEngineNode->fader, pWorkingBuffer, pWorkingBuffer, framesJustProcessedOut); /* In-place processing. */
+ } else {
+ ma_fader_process_pcm_frames(&pEngineNode->fader, pWorkingBuffer, pRunningFramesIn, framesJustProcessedOut);
+ isWorkingBufferValid = MA_TRUE;
+ }
+ }
+ /*
+ If at this point we still haven't actually done anything with the working buffer we need
+ to just read straight from the input buffer.
+ */
+ if (isWorkingBufferValid == MA_FALSE) {
+ pWorkingBuffer = (float*)pRunningFramesIn; /* Naughty const cast, but it's safe at this point because we won't ever be writing to it from this point out. */
+ }
-#if defined(MA_HAS_WAV) || \
- defined(MA_HAS_MP3) || \
- defined(MA_HAS_FLAC) || \
- defined(MA_HAS_VORBIS) || \
- defined(MA_HAS_OPUS)
-#define MA_HAS_PATH_API
-#endif
+ /* Spatialization. */
+ if (isSpatializationEnabled) {
+ ma_uint32 iListener;
-#if defined(MA_HAS_PATH_API)
-static const char* ma_path_file_name(const char* path)
-{
- const char* fileName;
+ /*
+ When determining the listener to use, we first check to see if the sound is pinned to a
+ specific listener. If so, we use that. Otherwise we just use the closest listener.
+ */
+ if (pEngineNode->pinnedListenerIndex != MA_LISTENER_INDEX_CLOSEST && pEngineNode->pinnedListenerIndex < ma_engine_get_listener_count(pEngineNode->pEngine)) {
+ iListener = pEngineNode->pinnedListenerIndex;
+ } else {
+ iListener = ma_engine_find_closest_listener(pEngineNode->pEngine, pEngineNode->spatializer.position.x, pEngineNode->spatializer.position.y, pEngineNode->spatializer.position.z);
+ }
- if (path == NULL) {
- return NULL;
- }
+ ma_spatializer_process_pcm_frames(&pEngineNode->spatializer, &pEngineNode->pEngine->listeners[iListener], pRunningFramesOut, pWorkingBuffer, framesJustProcessedOut);
+ } else {
+ /* No spatialization, but we still need to do channel conversion. */
+ if (channelsIn == channelsOut) {
+ /* No channel conversion required. Just copy straight to the output buffer. */
+ ma_copy_pcm_frames(pRunningFramesOut, pWorkingBuffer, framesJustProcessedOut, ma_format_f32, channelsOut);
+ } else {
+ /* Channel conversion required. TODO: Add support for channel maps here. */
+ ma_channel_map_apply_f32(pRunningFramesOut, NULL, channelsOut, pWorkingBuffer, NULL, channelsIn, framesJustProcessedOut, ma_channel_mix_mode_simple, pEngineNode->pEngine->monoExpansionMode);
+ }
+ }
- fileName = path;
+ /* At this point we can guarantee that the output buffer contains valid data. We can process everything in place now. */
- /* We just loop through the path until we find the last slash. */
- while (path[0] != '\0') {
- if (path[0] == '/' || path[0] == '\\') {
- fileName = path;
+ /* Panning. */
+ if (isPanningEnabled) {
+ ma_panner_process_pcm_frames(&pEngineNode->panner, pRunningFramesOut, pRunningFramesOut, framesJustProcessedOut); /* In-place processing. */
}
- path += 1;
- }
+ /* We're done for this chunk. */
+ totalFramesProcessedIn += framesJustProcessedIn;
+ totalFramesProcessedOut += framesJustProcessedOut;
- /* At this point the file name is sitting on a slash, so just move forward. */
- while (fileName[0] != '\0' && (fileName[0] == '/' || fileName[0] == '\\')) {
- fileName += 1;
+ /* If we didn't process any output frames this iteration it means we've either run out of input data, or run out of room in the output buffer. */
+ if (framesJustProcessedOut == 0) {
+ break;
+ }
}
- return fileName;
+ /* At this point we're done processing. */
+ *pFrameCountIn = totalFramesProcessedIn;
+ *pFrameCountOut = totalFramesProcessedOut;
}
-static const wchar_t* ma_path_file_name_w(const wchar_t* path)
+static void ma_engine_node_process_pcm_frames__sound(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
- const wchar_t* fileName;
-
- if (path == NULL) {
- return NULL;
+ /* For sounds, we need to first read from the data source. Then we need to apply the engine effects (pan, pitch, fades, etc.). */
+ ma_result result = MA_SUCCESS;
+ ma_sound* pSound = (ma_sound*)pNode;
+ ma_uint32 frameCount = *pFrameCountOut;
+ ma_uint32 totalFramesRead = 0;
+ ma_format dataSourceFormat;
+ ma_uint32 dataSourceChannels;
+ ma_uint8 temp[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
+ ma_uint32 tempCapInFrames;
+
+ /* This is a data source node which means no input buses. */
+ (void)ppFramesIn;
+ (void)pFrameCountIn;
+
+ /* If we're marked at the end we need to stop the sound and do nothing. */
+ if (ma_sound_at_end(pSound)) {
+ ma_sound_stop(pSound);
+ *pFrameCountOut = 0;
+ return;
}
- fileName = path;
+ /* If we're seeking, do so now before reading. */
+ if (pSound->seekTarget != MA_SEEK_TARGET_NONE) {
+ ma_data_source_seek_to_pcm_frame(pSound->pDataSource, pSound->seekTarget);
- /* We just loop through the path until we find the last slash. */
- while (path[0] != '\0') {
- if (path[0] == '/' || path[0] == '\\') {
- fileName = path;
- }
+ /* Any time-dependant effects need to have their times updated. */
+ ma_node_set_time(pSound, pSound->seekTarget);
- path += 1;
+ pSound->seekTarget = MA_SEEK_TARGET_NONE;
}
- /* At this point the file name is sitting on a slash, so just move forward. */
- while (fileName[0] != '\0' && (fileName[0] == '/' || fileName[0] == '\\')) {
- fileName += 1;
+ /*
+ We want to update the pitch once. For sounds, this can be either at the start or at the end. If
+ we don't force this to only ever be updating once, we could end up in a situation where
+ retrieving the required input frame count ends up being different to what we actually retrieve.
+ What could happen is that the required input frame count is calculated, the pitch is update,
+ and then this processing function is called resulting in a different number of input frames
+ being processed. Do not call this in ma_engine_node_process_pcm_frames__general() or else
+ you'll hit the aforementioned bug.
+ */
+ ma_engine_node_update_pitch_if_required(&pSound->engineNode);
+
+ /*
+ For the convenience of the caller, we're doing to allow data sources to use non-floating-point formats and channel counts that differ
+ from the main engine.
+ */
+ result = ma_data_source_get_data_format(pSound->pDataSource, &dataSourceFormat, &dataSourceChannels, NULL, NULL, 0);
+ if (result == MA_SUCCESS) {
+ tempCapInFrames = sizeof(temp) / ma_get_bytes_per_frame(dataSourceFormat, dataSourceChannels);
+
+ /* Keep reading until we've read as much as was requested or we reach the end of the data source. */
+ while (totalFramesRead < frameCount) {
+ ma_uint32 framesRemaining = frameCount - totalFramesRead;
+ ma_uint32 framesToRead;
+ ma_uint64 framesJustRead;
+ ma_uint32 frameCountIn;
+ ma_uint32 frameCountOut;
+ const float* pRunningFramesIn;
+ float* pRunningFramesOut;
+
+ /*
+ The first thing we need to do is read into the temporary buffer. We can calculate exactly
+ how many input frames we'll need after resampling.
+ */
+ framesToRead = (ma_uint32)ma_engine_node_get_required_input_frame_count(&pSound->engineNode, framesRemaining);
+ if (framesToRead > tempCapInFrames) {
+ framesToRead = tempCapInFrames;
+ }
+
+ result = ma_data_source_read_pcm_frames(pSound->pDataSource, temp, framesToRead, &framesJustRead);
+
+ /* If we reached the end of the sound we'll want to mark it as at the end and stop it. This should never be returned for looping sounds. */
+ if (result == MA_AT_END) {
+ c89atomic_exchange_32(&pSound->atEnd, MA_TRUE); /* This will be set to false in ma_sound_start(). */
+ }
+
+ pRunningFramesOut = ma_offset_pcm_frames_ptr_f32(ppFramesOut[0], totalFramesRead, ma_engine_get_channels(ma_sound_get_engine(pSound)));
+
+ frameCountIn = (ma_uint32)framesJustRead;
+ frameCountOut = framesRemaining;
+
+ /* Convert if necessary. */
+ if (dataSourceFormat == ma_format_f32) {
+ /* Fast path. No data conversion necessary. */
+ pRunningFramesIn = (float*)temp;
+ ma_engine_node_process_pcm_frames__general(&pSound->engineNode, &pRunningFramesIn, &frameCountIn, &pRunningFramesOut, &frameCountOut);
+ } else {
+ /* Slow path. Need to do sample format conversion to f32. If we give the f32 buffer the same count as the first temp buffer, we're guaranteed it'll be large enough. */
+ float tempf32[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* Do not do `MA_DATA_CONVERTER_STACK_BUFFER_SIZE/sizeof(float)` here like we've done in other places. */
+ ma_convert_pcm_frames_format(tempf32, ma_format_f32, temp, dataSourceFormat, framesJustRead, dataSourceChannels, ma_dither_mode_none);
+
+ /* Now that we have our samples in f32 format we can process like normal. */
+ pRunningFramesIn = tempf32;
+ ma_engine_node_process_pcm_frames__general(&pSound->engineNode, &pRunningFramesIn, &frameCountIn, &pRunningFramesOut, &frameCountOut);
+ }
+
+ /* We should have processed all of our input frames since we calculated the required number of input frames at the top. */
+ MA_ASSERT(frameCountIn == framesJustRead);
+ totalFramesRead += (ma_uint32)frameCountOut; /* Safe cast. */
+
+ if (result != MA_SUCCESS || ma_sound_at_end(pSound)) {
+ break; /* Might have reached the end. */
+ }
+ }
}
- return fileName;
+ *pFrameCountOut = totalFramesRead;
}
-
-static const char* ma_path_extension(const char* path)
+static void ma_engine_node_process_pcm_frames__group(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
- const char* extension;
- const char* lastOccurance;
+ /*
+ Make sure the pitch is updated before trying to read anything. It's important that this is done
+ only once and not in ma_engine_node_process_pcm_frames__general(). The reason for this is that
+ ma_engine_node_process_pcm_frames__general() will call ma_engine_node_get_required_input_frame_count(),
+ and if another thread modifies the pitch just after that call it can result in a glitch due to
+ the input rate changing.
+ */
+ ma_engine_node_update_pitch_if_required((ma_engine_node*)pNode);
- if (path == NULL) {
- path = "";
- }
+ /* For groups, the input data has already been read and we just need to apply the effect. */
+ ma_engine_node_process_pcm_frames__general((ma_engine_node*)pNode, ppFramesIn, pFrameCountIn, ppFramesOut, pFrameCountOut);
+}
- extension = ma_path_file_name(path);
- lastOccurance = NULL;
+static ma_result ma_engine_node_get_required_input_frame_count__group(ma_node* pNode, ma_uint32 outputFrameCount, ma_uint32* pInputFrameCount)
+{
+ ma_uint64 inputFrameCount;
- /* Just find the last '.' and return. */
- while (extension[0] != '\0') {
- if (extension[0] == '.') {
- extension += 1;
- lastOccurance = extension;
- }
+ MA_ASSERT(pInputFrameCount != NULL);
- extension += 1;
+ /* Our pitch will affect this calculation. We need to update it. */
+ ma_engine_node_update_pitch_if_required((ma_engine_node*)pNode);
+
+ inputFrameCount = ma_engine_node_get_required_input_frame_count((ma_engine_node*)pNode, outputFrameCount);
+ if (inputFrameCount > 0xFFFFFFFF) {
+ inputFrameCount = 0xFFFFFFFF; /* Will never happen because miniaudio will only ever process in relatively small chunks. */
}
- return (lastOccurance != NULL) ? lastOccurance : extension;
+ *pInputFrameCount = (ma_uint32)inputFrameCount;
+
+ return MA_SUCCESS;
}
-static const wchar_t* ma_path_extension_w(const wchar_t* path)
+
+static ma_node_vtable g_ma_engine_node_vtable__sound =
+{
+ ma_engine_node_process_pcm_frames__sound,
+ NULL, /* onGetRequiredInputFrameCount */
+ 0, /* Sounds are data source nodes which means they have zero inputs (their input is drawn from the data source itself). */
+ 1, /* Sounds have one output bus. */
+ 0 /* Default flags. */
+};
+
+static ma_node_vtable g_ma_engine_node_vtable__group =
{
- const wchar_t* extension;
- const wchar_t* lastOccurance;
+ ma_engine_node_process_pcm_frames__group,
+ ma_engine_node_get_required_input_frame_count__group,
+ 1, /* Groups have one input bus. */
+ 1, /* Groups have one output bus. */
+ MA_NODE_FLAG_DIFFERENT_PROCESSING_RATES /* The engine node does resampling so should let miniaudio know about it. */
+};
- if (path == NULL) {
- path = L"";
- }
- extension = ma_path_file_name_w(path);
- lastOccurance = NULL;
- /* Just find the last '.' and return. */
- while (extension[0] != '\0') {
- if (extension[0] == '.') {
- extension += 1;
- lastOccurance = extension;
- }
+static ma_node_config ma_engine_node_base_node_config_init(const ma_engine_node_config* pConfig)
+{
+ ma_node_config baseNodeConfig;
- extension += 1;
+ if (pConfig->type == ma_engine_node_type_sound) {
+ /* Sound. */
+ baseNodeConfig = ma_node_config_init();
+ baseNodeConfig.vtable = &g_ma_engine_node_vtable__sound;
+ baseNodeConfig.initialState = ma_node_state_stopped; /* Sounds are stopped by default. */
+ } else {
+ /* Group. */
+ baseNodeConfig = ma_node_config_init();
+ baseNodeConfig.vtable = &g_ma_engine_node_vtable__group;
+ baseNodeConfig.initialState = ma_node_state_started; /* Groups are started by default. */
}
- return (lastOccurance != NULL) ? lastOccurance : extension;
+ return baseNodeConfig;
}
+static ma_spatializer_config ma_engine_node_spatializer_config_init(const ma_node_config* pBaseNodeConfig)
+{
+ return ma_spatializer_config_init(pBaseNodeConfig->pInputChannels[0], pBaseNodeConfig->pOutputChannels[0]);
+}
-static ma_bool32 ma_path_extension_equal(const char* path, const char* extension)
+typedef struct
{
- const char* ext1;
- const char* ext2;
+ size_t sizeInBytes;
+ size_t baseNodeOffset;
+ size_t resamplerOffset;
+ size_t spatializerOffset;
+} ma_engine_node_heap_layout;
- if (path == NULL || extension == NULL) {
- return MA_FALSE;
- }
+static ma_result ma_engine_node_get_heap_layout(const ma_engine_node_config* pConfig, ma_engine_node_heap_layout* pHeapLayout)
+{
+ ma_result result;
+ size_t tempHeapSize;
+ ma_node_config baseNodeConfig;
+ ma_linear_resampler_config resamplerConfig;
+ ma_spatializer_config spatializerConfig;
+ ma_uint32 channelsIn;
+ ma_uint32 channelsOut;
- ext1 = extension;
- ext2 = ma_path_extension(path);
+ MA_ASSERT(pHeapLayout);
-#if defined(_MSC_VER) || defined(__DMC__)
- return _stricmp(ext1, ext2) == 0;
-#else
- return strcasecmp(ext1, ext2) == 0;
-#endif
-}
+ MA_ZERO_OBJECT(pHeapLayout);
-static ma_bool32 ma_path_extension_equal_w(const wchar_t* path, const wchar_t* extension)
-{
- const wchar_t* ext1;
- const wchar_t* ext2;
+ if (pConfig == NULL) {
+ return MA_INVALID_ARGS;
+ }
- if (path == NULL || extension == NULL) {
- return MA_FALSE;
+ if (pConfig->pEngine == NULL) {
+ return MA_INVALID_ARGS; /* An engine must be specified. */
}
- ext1 = extension;
- ext2 = ma_path_extension_w(path);
+ pHeapLayout->sizeInBytes = 0;
-#if defined(_MSC_VER) || defined(__WATCOMC__) || defined(__DMC__)
- return _wcsicmp(ext1, ext2) == 0;
-#else
- /*
- I'm not aware of a wide character version of strcasecmp(). I'm therefore converting the extensions to multibyte strings and comparing those. This
- isn't the most efficient way to do it, but it should work OK.
- */
- {
- char ext1MB[4096];
- char ext2MB[4096];
- const wchar_t* pext1 = ext1;
- const wchar_t* pext2 = ext2;
- mbstate_t mbs1;
- mbstate_t mbs2;
+ channelsIn = (pConfig->channelsIn != 0) ? pConfig->channelsIn : ma_engine_get_channels(pConfig->pEngine);
+ channelsOut = (pConfig->channelsOut != 0) ? pConfig->channelsOut : ma_engine_get_channels(pConfig->pEngine);
- MA_ZERO_OBJECT(&mbs1);
- MA_ZERO_OBJECT(&mbs2);
- if (wcsrtombs(ext1MB, &pext1, sizeof(ext1MB), &mbs1) == (size_t)-1) {
- return MA_FALSE;
- }
- if (wcsrtombs(ext2MB, &pext2, sizeof(ext2MB), &mbs2) == (size_t)-1) {
- return MA_FALSE;
- }
+ /* Base node. */
+ baseNodeConfig = ma_engine_node_base_node_config_init(pConfig);
+ baseNodeConfig.pInputChannels = &channelsIn;
+ baseNodeConfig.pOutputChannels = &channelsOut;
- return strcasecmp(ext1MB, ext2MB) == 0;
+ result = ma_node_get_heap_size(ma_engine_get_node_graph(pConfig->pEngine), &baseNodeConfig, &tempHeapSize);
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to retrieve the size of the heap for the base node. */
}
-#endif
-}
-#endif /* MA_HAS_PATH_API */
-
+ pHeapLayout->baseNodeOffset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += ma_align_64(tempHeapSize);
-static size_t ma_decoder__on_read_vfs(ma_decoder* pDecoder, void* pBufferOut, size_t bytesToRead)
-{
- size_t bytesRead;
-
- MA_ASSERT(pDecoder != NULL);
- MA_ASSERT(pBufferOut != NULL);
- ma_vfs_or_default_read(pDecoder->data.vfs.pVFS, pDecoder->data.vfs.file, pBufferOut, bytesToRead, &bytesRead);
+ /* Resmapler. */
+ resamplerConfig = ma_linear_resampler_config_init(ma_format_f32, channelsIn, 1, 1); /* Input and output sample rates don't affect the calculation of the heap size. */
+ resamplerConfig.lpfOrder = 0;
+
+ result = ma_linear_resampler_get_heap_size(&resamplerConfig, &tempHeapSize);
+ if (result != MA_SUCCESS) {
+ return result; /* Failed to retrieve the size of the heap for the resampler. */
+ }
- return bytesRead;
-}
+ pHeapLayout->resamplerOffset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += ma_align_64(tempHeapSize);
-static ma_bool32 ma_decoder__on_seek_vfs(ma_decoder* pDecoder, ma_int64 offset, ma_seek_origin origin)
-{
- ma_result result;
- MA_ASSERT(pDecoder != NULL);
+ /* Spatializer. */
+ spatializerConfig = ma_engine_node_spatializer_config_init(&baseNodeConfig);
- result = ma_vfs_or_default_seek(pDecoder->data.vfs.pVFS, pDecoder->data.vfs.file, offset, origin);
+ result = ma_spatializer_get_heap_size(&spatializerConfig, &tempHeapSize);
if (result != MA_SUCCESS) {
- return MA_FALSE;
+ return result; /* Failed to retrieve the size of the heap for the spatializer. */
}
- return MA_TRUE;
-}
+ pHeapLayout->spatializerOffset = pHeapLayout->sizeInBytes;
+ pHeapLayout->sizeInBytes += ma_align_64(tempHeapSize);
-static ma_result ma_decoder__on_tell_vfs(ma_decoder* pDecoder, ma_int64* pCursor)
-{
- MA_ASSERT(pDecoder != NULL);
- return ma_vfs_or_default_tell(pDecoder->data.vfs.pVFS, pDecoder->data.vfs.file, pCursor);
+ return MA_SUCCESS;
}
-static ma_result ma_decoder__preinit_vfs(ma_vfs* pVFS, const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API ma_result ma_engine_node_get_heap_size(const ma_engine_node_config* pConfig, size_t* pHeapSizeInBytes)
{
ma_result result;
- ma_vfs_file file;
-
- result = ma_decoder__preinit(ma_decoder__on_read_vfs, ma_decoder__on_seek_vfs, ma_decoder__on_tell_vfs, NULL, pConfig, pDecoder);
- if (result != MA_SUCCESS) {
- return result;
- }
+ ma_engine_node_heap_layout heapLayout;
- if (pFilePath == NULL || pFilePath[0] == '\0') {
+ if (pHeapSizeInBytes == NULL) {
return MA_INVALID_ARGS;
}
- result = ma_vfs_or_default_open(pVFS, pFilePath, MA_OPEN_MODE_READ, &file);
+ *pHeapSizeInBytes = 0;
+
+ result = ma_engine_node_get_heap_layout(pConfig, &heapLayout);
if (result != MA_SUCCESS) {
return result;
}
- pDecoder->data.vfs.pVFS = pVFS;
- pDecoder->data.vfs.file = file;
+ *pHeapSizeInBytes = heapLayout.sizeInBytes;
return MA_SUCCESS;
}
-MA_API ma_result ma_decoder_init_vfs(ma_vfs* pVFS, const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API ma_result ma_engine_node_init_preallocated(const ma_engine_node_config* pConfig, void* pHeap, ma_engine_node* pEngineNode)
{
ma_result result;
- ma_decoder_config config;
+ ma_engine_node_heap_layout heapLayout;
+ ma_node_config baseNodeConfig;
+ ma_linear_resampler_config resamplerConfig;
+ ma_fader_config faderConfig;
+ ma_spatializer_config spatializerConfig;
+ ma_panner_config pannerConfig;
+ ma_uint32 channelsIn;
+ ma_uint32 channelsOut;
- config = ma_decoder_config_init_copy(pConfig);
- result = ma_decoder__preinit_vfs(pVFS, pFilePath, &config, pDecoder);
+ if (pEngineNode == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ MA_ZERO_OBJECT(pEngineNode);
+
+ result = ma_engine_node_get_heap_layout(pConfig, &heapLayout);
if (result != MA_SUCCESS) {
return result;
}
- result = MA_NO_BACKEND;
+ if (pConfig->pinnedListenerIndex != MA_LISTENER_INDEX_CLOSEST && pConfig->pinnedListenerIndex >= ma_engine_get_listener_count(pConfig->pEngine)) {
+ return MA_INVALID_ARGS; /* Invalid listener. */
+ }
- if (config.encodingFormat != ma_encoding_format_unknown) {
- #ifdef MA_HAS_WAV
- if (config.encodingFormat == ma_encoding_format_wav) {
- result = ma_decoder_init_wav__internal(&config, pDecoder);
- }
- #endif
- #ifdef MA_HAS_FLAC
- if (config.encodingFormat == ma_encoding_format_flac) {
- result = ma_decoder_init_flac__internal(&config, pDecoder);
- }
- #endif
- #ifdef MA_HAS_MP3
- if (config.encodingFormat == ma_encoding_format_mp3) {
- result = ma_decoder_init_mp3__internal(&config, pDecoder);
- }
- #endif
- #ifdef MA_HAS_VORBIS
- if (config.encodingFormat == ma_encoding_format_vorbis) {
- result = ma_decoder_init_vorbis__internal(&config, pDecoder);
- }
- #endif
+ pEngineNode->_pHeap = pHeap;
+ MA_ZERO_MEMORY(pHeap, heapLayout.sizeInBytes);
- /* Make sure we seek back to the start if we didn't initialize a decoder successfully so the next attempts have a fresh start. */
- if (result != MA_SUCCESS) {
- ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
- }
+ pEngineNode->pEngine = pConfig->pEngine;
+ pEngineNode->sampleRate = (pConfig->sampleRate > 0) ? pConfig->sampleRate : ma_engine_get_sample_rate(pEngineNode->pEngine);
+ pEngineNode->pitch = 1;
+ pEngineNode->oldPitch = 1;
+ pEngineNode->oldDopplerPitch = 1;
+ pEngineNode->isPitchDisabled = pConfig->isPitchDisabled;
+ pEngineNode->isSpatializationDisabled = pConfig->isSpatializationDisabled;
+ pEngineNode->pinnedListenerIndex = pConfig->pinnedListenerIndex;
+
+
+ channelsIn = (pConfig->channelsIn != 0) ? pConfig->channelsIn : ma_engine_get_channels(pConfig->pEngine);
+ channelsOut = (pConfig->channelsOut != 0) ? pConfig->channelsOut : ma_engine_get_channels(pConfig->pEngine);
+
+
+ /* Base node. */
+ baseNodeConfig = ma_engine_node_base_node_config_init(pConfig);
+ baseNodeConfig.pInputChannels = &channelsIn;
+ baseNodeConfig.pOutputChannels = &channelsOut;
+
+ result = ma_node_init_preallocated(&pConfig->pEngine->nodeGraph, &baseNodeConfig, ma_offset_ptr(pHeap, heapLayout.baseNodeOffset), &pEngineNode->baseNode);
+ if (result != MA_SUCCESS) {
+ goto error0;
}
+
+ /*
+ We can now initialize the effects we need in order to implement the engine node. There's a
+ defined order of operations here, mainly centered around when we convert our channels from the
+ data source's native channel count to the engine's channel count. As a rule, we want to do as
+ much computation as possible before spatialization because there's a chance that will increase
+ the channel count, thereby increasing the amount of work needing to be done to process.
+ */
+
+ /* We'll always do resampling first. */
+ resamplerConfig = ma_linear_resampler_config_init(ma_format_f32, baseNodeConfig.pInputChannels[0], pEngineNode->sampleRate, ma_engine_get_sample_rate(pEngineNode->pEngine));
+ resamplerConfig.lpfOrder = 0; /* <-- Need to disable low-pass filtering for pitch shifting for now because there's cases where the biquads are becoming unstable. Need to figure out a better fix for this. */
+
+ result = ma_linear_resampler_init_preallocated(&resamplerConfig, ma_offset_ptr(pHeap, heapLayout.resamplerOffset), &pEngineNode->resampler);
if (result != MA_SUCCESS) {
- /* Getting here means we weren't able to initialize a decoder of a specific encoding format. */
+ goto error1;
+ }
- /*
- We use trial and error to open a decoder. We prioritize custom decoders so that if they
- implement the same encoding format they take priority over the built-in decoders.
- */
- if (result != MA_SUCCESS) {
- result = ma_decoder_init_custom__internal(&config, pDecoder);
- if (result != MA_SUCCESS) {
- ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
- }
- }
- /*
- If we get to this point and we still haven't found a decoder, and the caller has requested a
- specific encoding format, there's no hope for it. Abort.
- */
- if (config.encodingFormat != ma_encoding_format_unknown) {
- return MA_NO_BACKEND;
- }
+ /* After resampling will come the fader. */
+ faderConfig = ma_fader_config_init(ma_format_f32, baseNodeConfig.pInputChannels[0], ma_engine_get_sample_rate(pEngineNode->pEngine));
- #ifdef MA_HAS_WAV
- if (result != MA_SUCCESS && ma_path_extension_equal(pFilePath, "wav")) {
- result = ma_decoder_init_wav__internal(&config, pDecoder);
- if (result != MA_SUCCESS) {
- ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
- }
- }
- #endif
- #ifdef MA_HAS_FLAC
- if (result != MA_SUCCESS && ma_path_extension_equal(pFilePath, "flac")) {
- result = ma_decoder_init_flac__internal(&config, pDecoder);
- if (result != MA_SUCCESS) {
- ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
- }
- }
- #endif
- #ifdef MA_HAS_MP3
- if (result != MA_SUCCESS && ma_path_extension_equal(pFilePath, "mp3")) {
- result = ma_decoder_init_mp3__internal(&config, pDecoder);
- if (result != MA_SUCCESS) {
- ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
- }
- }
- #endif
+ result = ma_fader_init(&faderConfig, &pEngineNode->fader);
+ if (result != MA_SUCCESS) {
+ goto error2;
}
- /* If we still haven't got a result just use trial and error. Otherwise we can finish up. */
+
+ /*
+ Spatialization comes next. We spatialize based ont he node's output channel count. It's up the caller to
+ ensure channels counts link up correctly in the node graph.
+ */
+ spatializerConfig = ma_engine_node_spatializer_config_init(&baseNodeConfig);
+ spatializerConfig.gainSmoothTimeInFrames = pEngineNode->pEngine->gainSmoothTimeInFrames;
+
+ result = ma_spatializer_init_preallocated(&spatializerConfig, ma_offset_ptr(pHeap, heapLayout.spatializerOffset), &pEngineNode->spatializer);
if (result != MA_SUCCESS) {
- result = ma_decoder_init__internal(ma_decoder__on_read_vfs, ma_decoder__on_seek_vfs, NULL, &config, pDecoder);
- } else {
- result = ma_decoder__postinit(&config, pDecoder);
+ goto error2;
}
+
+ /*
+ After spatialization comes panning. We need to do this after spatialization because otherwise we wouldn't
+ be able to pan mono sounds.
+ */
+ pannerConfig = ma_panner_config_init(ma_format_f32, baseNodeConfig.pOutputChannels[0]);
+
+ result = ma_panner_init(&pannerConfig, &pEngineNode->panner);
if (result != MA_SUCCESS) {
- if (pDecoder->data.vfs.file != NULL) { /* <-- Will be reset to NULL if ma_decoder_uninit() is called in one of the steps above which allows us to avoid a double close of the file. */
- ma_vfs_or_default_close(pVFS, pDecoder->data.vfs.file);
+ goto error3;
+ }
+
+ return MA_SUCCESS;
+
+ /* No need for allocation callbacks here because we use a preallocated heap. */
+error3: ma_spatializer_uninit(&pEngineNode->spatializer, NULL);
+error2: ma_linear_resampler_uninit(&pEngineNode->resampler, NULL);
+error1: ma_node_uninit(&pEngineNode->baseNode, NULL);
+error0: return result;
+}
+
+MA_API ma_result ma_engine_node_init(const ma_engine_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_engine_node* pEngineNode)
+{
+ ma_result result;
+ size_t heapSizeInBytes;
+ void* pHeap;
+
+ result = ma_engine_node_get_heap_size(pConfig, &heapSizeInBytes);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ if (heapSizeInBytes > 0) {
+ pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+ if (pHeap == NULL) {
+ return MA_OUT_OF_MEMORY;
}
+ } else {
+ pHeap = NULL;
+ }
+ result = ma_engine_node_init_preallocated(pConfig, pHeap, pEngineNode);
+ if (result != MA_SUCCESS) {
+ ma_free(pHeap, pAllocationCallbacks);
return result;
}
+ pEngineNode->_ownsHeap = MA_TRUE;
return MA_SUCCESS;
}
-MA_API ma_result ma_decoder_init_vfs_wav(ma_vfs* pVFS, const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API void ma_engine_node_uninit(ma_engine_node* pEngineNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
-#ifdef MA_HAS_WAV
- ma_decoder_config config;
+ /*
+ The base node always needs to be uninitialized first to ensure it's detached from the graph completely before we
+ destroy anything that might be in the middle of being used by the processing function.
+ */
+ ma_node_uninit(&pEngineNode->baseNode, pAllocationCallbacks);
- config = ma_decoder_config_init_copy(pConfig);
- config.encodingFormat = ma_encoding_format_wav;
+ /* Now that the node has been uninitialized we can safely uninitialize the rest. */
+ ma_spatializer_uninit(&pEngineNode->spatializer, pAllocationCallbacks);
+ ma_linear_resampler_uninit(&pEngineNode->resampler, pAllocationCallbacks);
- return ma_decoder_init_vfs(pVFS, pFilePath, &config, pDecoder);
-#else
- (void)pVFS;
- (void)pFilePath;
- (void)pConfig;
- (void)pDecoder;
- return MA_NO_BACKEND;
-#endif
+ /* Free the heap last. */
+ if (pEngineNode->_ownsHeap) {
+ ma_free(pEngineNode->_pHeap, pAllocationCallbacks);
+ }
}
-MA_API ma_result ma_decoder_init_vfs_flac(ma_vfs* pVFS, const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+
+MA_API ma_sound_config ma_sound_config_init(void)
{
-#ifdef MA_HAS_FLAC
- ma_decoder_config config;
+ ma_sound_config config;
- config = ma_decoder_config_init_copy(pConfig);
- config.encodingFormat = ma_encoding_format_flac;
+ MA_ZERO_OBJECT(&config);
+ config.rangeEndInPCMFrames = ~((ma_uint64)0);
+ config.loopPointEndInPCMFrames = ~((ma_uint64)0);
- return ma_decoder_init_vfs(pVFS, pFilePath, &config, pDecoder);
-#else
- (void)pVFS;
- (void)pFilePath;
- (void)pConfig;
- (void)pDecoder;
- return MA_NO_BACKEND;
-#endif
+ return config;
}
-MA_API ma_result ma_decoder_init_vfs_mp3(ma_vfs* pVFS, const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API ma_sound_group_config ma_sound_group_config_init(void)
{
-#ifdef MA_HAS_MP3
- ma_decoder_config config;
+ ma_sound_group_config config;
- config = ma_decoder_config_init_copy(pConfig);
- config.encodingFormat = ma_encoding_format_mp3;
+ MA_ZERO_OBJECT(&config);
- return ma_decoder_init_vfs(pVFS, pFilePath, &config, pDecoder);
-#else
- (void)pVFS;
- (void)pFilePath;
- (void)pConfig;
- (void)pDecoder;
- return MA_NO_BACKEND;
-#endif
+ return config;
}
-MA_API ma_result ma_decoder_init_vfs_vorbis(ma_vfs* pVFS, const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+
+MA_API ma_engine_config ma_engine_config_init(void)
{
-#ifdef MA_HAS_VORBIS
- ma_decoder_config config;
+ ma_engine_config config;
- config = ma_decoder_config_init_copy(pConfig);
- config.encodingFormat = ma_encoding_format_vorbis;
+ MA_ZERO_OBJECT(&config);
+ config.listenerCount = 1; /* Always want at least one listener. */
+ config.monoExpansionMode = ma_mono_expansion_mode_default;
- return ma_decoder_init_vfs(pVFS, pFilePath, &config, pDecoder);
-#else
- (void)pVFS;
- (void)pFilePath;
- (void)pConfig;
- (void)pDecoder;
- return MA_NO_BACKEND;
-#endif
+ return config;
}
-
-static ma_result ma_decoder__preinit_vfs_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+#if !defined(MA_NO_DEVICE_IO)
+static void ma_engine_data_callback_internal(ma_device* pDevice, void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount)
{
- ma_result result;
- ma_vfs_file file;
+ ma_engine* pEngine = (ma_engine*)pDevice->pUserData;
- result = ma_decoder__preinit(ma_decoder__on_read_vfs, ma_decoder__on_seek_vfs, ma_decoder__on_tell_vfs, NULL, pConfig, pDecoder);
- if (result != MA_SUCCESS) {
- return result;
- }
+ (void)pFramesIn;
- if (pFilePath == NULL || pFilePath[0] == '\0') {
- return MA_INVALID_ARGS;
- }
+ /*
+ Experiment: Try processing a resource manager job if we're on the Emscripten build.
- result = ma_vfs_or_default_open_w(pVFS, pFilePath, MA_OPEN_MODE_READ, &file);
- if (result != MA_SUCCESS) {
- return result;
- }
+ This serves two purposes:
- pDecoder->data.vfs.pVFS = pVFS;
- pDecoder->data.vfs.file = file;
+ 1) It ensures jobs are actually processed at some point since we cannot guarantee that the
+ caller is doing the right thing and calling ma_resource_manager_process_next_job(); and
- return MA_SUCCESS;
+ 2) It's an attempt at working around an issue where processing jobs on the Emscripten main
+ loop doesn't work as well as it should. When trying to load sounds without the `DECODE`
+ flag or with the `ASYNC` flag, the sound data is just not able to be loaded in time
+ before the callback is processed. I think it's got something to do with the single-
+ threaded nature of Web, but I'm not entirely sure.
+ */
+ #if !defined(MA_NO_RESOURCE_MANAGER) && defined(MA_EMSCRIPTEN)
+ {
+ if (pEngine->pResourceManager != NULL) {
+ if ((pEngine->pResourceManager->config.flags & MA_RESOURCE_MANAGER_FLAG_NO_THREADING) != 0) {
+ ma_resource_manager_process_next_job(pEngine->pResourceManager);
+ }
+ }
+ }
+ #endif
+
+ ma_engine_read_pcm_frames(pEngine, pFramesOut, frameCount, NULL);
}
+#endif
-MA_API ma_result ma_decoder_init_vfs_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API ma_result ma_engine_init(const ma_engine_config* pConfig, ma_engine* pEngine)
{
ma_result result;
- ma_decoder_config config;
+ ma_node_graph_config nodeGraphConfig;
+ ma_engine_config engineConfig;
+ ma_spatializer_listener_config listenerConfig;
+ ma_uint32 iListener;
- config = ma_decoder_config_init_copy(pConfig);
- result = ma_decoder__preinit_vfs_w(pVFS, pFilePath, &config, pDecoder);
- if (result != MA_SUCCESS) {
- return result;
+ if (pEngine == NULL) {
+ return MA_INVALID_ARGS;
}
- result = MA_NO_BACKEND;
+ MA_ZERO_OBJECT(pEngine);
- if (config.encodingFormat != ma_encoding_format_unknown) {
- #ifdef MA_HAS_WAV
- if (config.encodingFormat == ma_encoding_format_wav) {
- result = ma_decoder_init_wav__internal(&config, pDecoder);
- }
- #endif
- #ifdef MA_HAS_FLAC
- if (config.encodingFormat == ma_encoding_format_flac) {
- result = ma_decoder_init_flac__internal(&config, pDecoder);
- }
+ /* The config is allowed to be NULL in which case we use defaults for everything. */
+ if (pConfig != NULL) {
+ engineConfig = *pConfig;
+ } else {
+ engineConfig = ma_engine_config_init();
+ }
+
+ pEngine->monoExpansionMode = engineConfig.monoExpansionMode;
+ ma_allocation_callbacks_init_copy(&pEngine->allocationCallbacks, &engineConfig.allocationCallbacks);
+
+ #if !defined(MA_NO_RESOURCE_MANAGER)
+ {
+ pEngine->pResourceManager = engineConfig.pResourceManager;
+ }
#endif
- #ifdef MA_HAS_MP3
- if (config.encodingFormat == ma_encoding_format_mp3) {
- result = ma_decoder_init_mp3__internal(&config, pDecoder);
+
+ #if !defined(MA_NO_DEVICE_IO)
+ {
+ pEngine->pDevice = engineConfig.pDevice;
+
+ /* If we don't have a device, we need one. */
+ if (pEngine->pDevice == NULL && engineConfig.noDevice == MA_FALSE) {
+ ma_device_config deviceConfig;
+
+ pEngine->pDevice = (ma_device*)ma_malloc(sizeof(*pEngine->pDevice), &pEngine->allocationCallbacks);
+ if (pEngine->pDevice == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+
+ deviceConfig = ma_device_config_init(ma_device_type_playback);
+ deviceConfig.playback.pDeviceID = engineConfig.pPlaybackDeviceID;
+ deviceConfig.playback.format = ma_format_f32;
+ deviceConfig.playback.channels = engineConfig.channels;
+ deviceConfig.sampleRate = engineConfig.sampleRate;
+ deviceConfig.dataCallback = ma_engine_data_callback_internal;
+ deviceConfig.pUserData = pEngine;
+ deviceConfig.periodSizeInFrames = engineConfig.periodSizeInFrames;
+ deviceConfig.periodSizeInMilliseconds = engineConfig.periodSizeInMilliseconds;
+ deviceConfig.noPreSilencedOutputBuffer = MA_TRUE; /* We'll always be outputting to every frame in the callback so there's no need for a pre-silenced buffer. */
+ deviceConfig.noClip = MA_TRUE; /* The engine will do clipping itself. */
+
+ if (engineConfig.pContext == NULL) {
+ ma_context_config contextConfig = ma_context_config_init();
+ contextConfig.allocationCallbacks = pEngine->allocationCallbacks;
+ contextConfig.pLog = engineConfig.pLog;
+
+ /* If the engine config does not specify a log, use the resource manager's if we have one. */
+ #ifndef MA_NO_RESOURCE_MANAGER
+ {
+ if (contextConfig.pLog == NULL && engineConfig.pResourceManager != NULL) {
+ contextConfig.pLog = ma_resource_manager_get_log(engineConfig.pResourceManager);
+ }
+ }
+ #endif
+
+ result = ma_device_init_ex(NULL, 0, &contextConfig, &deviceConfig, pEngine->pDevice);
+ } else {
+ result = ma_device_init(engineConfig.pContext, &deviceConfig, pEngine->pDevice);
+ }
+
+ if (result != MA_SUCCESS) {
+ ma_free(pEngine->pDevice, &pEngine->allocationCallbacks);
+ pEngine->pDevice = NULL;
+ return result;
+ }
+
+ pEngine->ownsDevice = MA_TRUE;
}
- #endif
- #ifdef MA_HAS_VORBIS
- if (config.encodingFormat == ma_encoding_format_vorbis) {
- result = ma_decoder_init_vorbis__internal(&config, pDecoder);
+
+ /* Update the channel count and sample rate of the engine config so we can reference it below. */
+ if (pEngine->pDevice != NULL) {
+ engineConfig.channels = pEngine->pDevice->playback.channels;
+ engineConfig.sampleRate = pEngine->pDevice->sampleRate;
}
+ }
#endif
- /* Make sure we seek back to the start if we didn't initialize a decoder successfully so the next attempts have a fresh start. */
- if (result != MA_SUCCESS) {
- ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
+ if (engineConfig.channels == 0 || engineConfig.sampleRate == 0) {
+ return MA_INVALID_ARGS;
+ }
+
+ pEngine->sampleRate = engineConfig.sampleRate;
+
+ /* The engine always uses either the log that was passed into the config, or the context's log is available. */
+ if (engineConfig.pLog != NULL) {
+ pEngine->pLog = engineConfig.pLog;
+ } else {
+ #if !defined(MA_NO_DEVICE_IO)
+ {
+ pEngine->pLog = ma_device_get_log(pEngine->pDevice);
}
+ #else
+ {
+ pEngine->pLog = NULL;
+ }
+ #endif
}
+
+ /* The engine is a node graph. This needs to be initialized after we have the device so we can can determine the channel count. */
+ nodeGraphConfig = ma_node_graph_config_init(engineConfig.channels);
+ nodeGraphConfig.nodeCacheCapInFrames = (engineConfig.periodSizeInFrames > 0xFFFF) ? 0xFFFF : (ma_uint16)engineConfig.periodSizeInFrames;
+
+ result = ma_node_graph_init(&nodeGraphConfig, &pEngine->allocationCallbacks, &pEngine->nodeGraph);
if (result != MA_SUCCESS) {
- /* Getting here means we weren't able to initialize a decoder of a specific encoding format. */
+ goto on_error_1;
+ }
+
+
+ /* We need at least one listener. */
+ if (engineConfig.listenerCount == 0) {
+ engineConfig.listenerCount = 1;
+ }
+
+ if (engineConfig.listenerCount > MA_ENGINE_MAX_LISTENERS) {
+ result = MA_INVALID_ARGS; /* Too many listeners. */
+ goto on_error_1;
+ }
+
+ for (iListener = 0; iListener < engineConfig.listenerCount; iListener += 1) {
+ listenerConfig = ma_spatializer_listener_config_init(ma_node_graph_get_channels(&pEngine->nodeGraph));
/*
- We use trial and error to open a decoder. We prioritize custom decoders so that if they
- implement the same encoding format they take priority over the built-in decoders.
+ If we're using a device, use the device's channel map for the listener. Otherwise just use
+ miniaudio's default channel map.
*/
- if (result != MA_SUCCESS) {
- result = ma_decoder_init_custom__internal(&config, pDecoder);
- if (result != MA_SUCCESS) {
- ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
+ #if !defined(MA_NO_DEVICE_IO)
+ {
+ if (pEngine->pDevice != NULL) {
+ /*
+ Temporarily disabled. There is a subtle bug here where front-left and front-right
+ will be used by the device's channel map, but this is not what we want to use for
+ spatialization. Instead we want to use side-left and side-right. I need to figure
+ out a better solution for this. For now, disabling the user of device channel maps.
+ */
+ /*listenerConfig.pChannelMapOut = pEngine->pDevice->playback.channelMap;*/
}
}
+ #endif
- /*
- If we get to this point and we still haven't found a decoder, and the caller has requested a
- specific encoding format, there's no hope for it. Abort.
- */
- if (config.encodingFormat != ma_encoding_format_unknown) {
- return MA_NO_BACKEND;
+ result = ma_spatializer_listener_init(&listenerConfig, &pEngine->allocationCallbacks, &pEngine->listeners[iListener]); /* TODO: Change this to a pre-allocated heap. */
+ if (result != MA_SUCCESS) {
+ goto on_error_2;
}
- #ifdef MA_HAS_WAV
- if (result != MA_SUCCESS && ma_path_extension_equal_w(pFilePath, L"wav")) {
- result = ma_decoder_init_wav__internal(&config, pDecoder);
+ pEngine->listenerCount += 1;
+ }
+
+
+ /* Gain smoothing for spatialized sounds. */
+ pEngine->gainSmoothTimeInFrames = engineConfig.gainSmoothTimeInFrames;
+ if (pEngine->gainSmoothTimeInFrames == 0) {
+ ma_uint32 gainSmoothTimeInMilliseconds = engineConfig.gainSmoothTimeInMilliseconds;
+ if (gainSmoothTimeInMilliseconds == 0) {
+ gainSmoothTimeInMilliseconds = 8;
+ }
+
+ pEngine->gainSmoothTimeInFrames = (gainSmoothTimeInMilliseconds * ma_engine_get_sample_rate(pEngine)) / 1000; /* 8ms by default. */
+ }
+
+
+ /* We need a resource manager. */
+ #ifndef MA_NO_RESOURCE_MANAGER
+ {
+ if (pEngine->pResourceManager == NULL) {
+ ma_resource_manager_config resourceManagerConfig;
+
+ pEngine->pResourceManager = (ma_resource_manager*)ma_malloc(sizeof(*pEngine->pResourceManager), &pEngine->allocationCallbacks);
+ if (pEngine->pResourceManager == NULL) {
+ result = MA_OUT_OF_MEMORY;
+ goto on_error_2;
+ }
+
+ resourceManagerConfig = ma_resource_manager_config_init();
+ resourceManagerConfig.pLog = pEngine->pLog; /* Always use the engine's log for internally-managed resource managers. */
+ resourceManagerConfig.decodedFormat = ma_format_f32;
+ resourceManagerConfig.decodedChannels = 0; /* Leave the decoded channel count as 0 so we can get good spatialization. */
+ resourceManagerConfig.decodedSampleRate = ma_engine_get_sample_rate(pEngine);
+ ma_allocation_callbacks_init_copy(&resourceManagerConfig.allocationCallbacks, &pEngine->allocationCallbacks);
+ resourceManagerConfig.pVFS = engineConfig.pResourceManagerVFS;
+
+ /* The Emscripten build cannot use threads. */
+ #if defined(MA_EMSCRIPTEN)
+ {
+ resourceManagerConfig.jobThreadCount = 0;
+ resourceManagerConfig.flags |= MA_RESOURCE_MANAGER_FLAG_NO_THREADING;
+ }
+ #endif
+
+ result = ma_resource_manager_init(&resourceManagerConfig, pEngine->pResourceManager);
if (result != MA_SUCCESS) {
- ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
+ goto on_error_3;
}
+
+ pEngine->ownsResourceManager = MA_TRUE;
}
+ }
#endif
- #ifdef MA_HAS_FLAC
- if (result != MA_SUCCESS && ma_path_extension_equal_w(pFilePath, L"flac")) {
- result = ma_decoder_init_flac__internal(&config, pDecoder);
+
+ /* Setup some stuff for inlined sounds. That is sounds played with ma_engine_play_sound(). */
+ pEngine->inlinedSoundLock = 0;
+ pEngine->pInlinedSoundHead = NULL;
+
+ /* Start the engine if required. This should always be the last step. */
+ #if !defined(MA_NO_DEVICE_IO)
+ {
+ if (engineConfig.noAutoStart == MA_FALSE && pEngine->pDevice != NULL) {
+ result = ma_engine_start(pEngine);
if (result != MA_SUCCESS) {
- ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
+ goto on_error_4; /* Failed to start the engine. */
}
}
+ }
#endif
- #ifdef MA_HAS_MP3
- if (result != MA_SUCCESS && ma_path_extension_equal_w(pFilePath, L"mp3")) {
- result = ma_decoder_init_mp3__internal(&config, pDecoder);
- if (result != MA_SUCCESS) {
- ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
+
+ return MA_SUCCESS;
+
+#if !defined(MA_NO_DEVICE_IO)
+on_error_4:
+#endif
+#if !defined(MA_NO_RESOURCE_MANAGER)
+on_error_3:
+ if (pEngine->ownsResourceManager) {
+ ma_free(pEngine->pResourceManager, &pEngine->allocationCallbacks);
+ }
+#endif /* MA_NO_RESOURCE_MANAGER */
+on_error_2:
+ for (iListener = 0; iListener < pEngine->listenerCount; iListener += 1) {
+ ma_spatializer_listener_uninit(&pEngine->listeners[iListener], &pEngine->allocationCallbacks);
+ }
+
+ ma_node_graph_uninit(&pEngine->nodeGraph, &pEngine->allocationCallbacks);
+on_error_1:
+ #if !defined(MA_NO_DEVICE_IO)
+ {
+ if (pEngine->ownsDevice) {
+ ma_device_uninit(pEngine->pDevice);
+ ma_free(pEngine->pDevice, &pEngine->allocationCallbacks);
+ }
+ }
+ #endif
+
+ return result;
+}
+
+MA_API void ma_engine_uninit(ma_engine* pEngine)
+{
+ ma_uint32 iListener;
+
+ if (pEngine == NULL) {
+ return;
+ }
+
+ /* The device must be uninitialized before the node graph to ensure the audio thread doesn't try accessing it. */
+ #if !defined(MA_NO_DEVICE_IO)
+ {
+ if (pEngine->ownsDevice) {
+ ma_device_uninit(pEngine->pDevice);
+ ma_free(pEngine->pDevice, &pEngine->allocationCallbacks);
+ } else {
+ if (pEngine->pDevice != NULL) {
+ ma_device_stop(pEngine->pDevice);
}
}
+ }
#endif
+
+ /*
+ All inlined sounds need to be deleted. I'm going to use a lock here just to future proof in case
+ I want to do some kind of garbage collection later on.
+ */
+ ma_spinlock_lock(&pEngine->inlinedSoundLock);
+ {
+ for (;;) {
+ ma_sound_inlined* pSoundToDelete = pEngine->pInlinedSoundHead;
+ if (pSoundToDelete == NULL) {
+ break; /* Done. */
+ }
+
+ pEngine->pInlinedSoundHead = pSoundToDelete->pNext;
+
+ ma_sound_uninit(&pSoundToDelete->sound);
+ ma_free(pSoundToDelete, &pEngine->allocationCallbacks);
+ }
}
+ ma_spinlock_unlock(&pEngine->inlinedSoundLock);
- /* If we still haven't got a result just use trial and error. Otherwise we can finish up. */
- if (result != MA_SUCCESS) {
- result = ma_decoder_init__internal(ma_decoder__on_read_vfs, ma_decoder__on_seek_vfs, NULL, &config, pDecoder);
- } else {
- result = ma_decoder__postinit(&config, pDecoder);
+ for (iListener = 0; iListener < pEngine->listenerCount; iListener += 1) {
+ ma_spatializer_listener_uninit(&pEngine->listeners[iListener], &pEngine->allocationCallbacks);
+ }
+
+ /* Make sure the node graph is uninitialized after the audio thread has been shutdown to prevent accessing of the node graph after being uninitialized. */
+ ma_node_graph_uninit(&pEngine->nodeGraph, &pEngine->allocationCallbacks);
+
+ /* Uninitialize the resource manager last to ensure we don't have a thread still trying to access it. */
+#ifndef MA_NO_RESOURCE_MANAGER
+ if (pEngine->ownsResourceManager) {
+ ma_resource_manager_uninit(pEngine->pResourceManager);
+ ma_free(pEngine->pResourceManager, &pEngine->allocationCallbacks);
+ }
+#endif
+}
+
+MA_API ma_result ma_engine_read_pcm_frames(ma_engine* pEngine, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+{
+ return ma_node_graph_read_pcm_frames(&pEngine->nodeGraph, pFramesOut, frameCount, pFramesRead);
+}
+
+MA_API ma_node_graph* ma_engine_get_node_graph(ma_engine* pEngine)
+{
+ if (pEngine == NULL) {
+ return NULL;
+ }
+
+ return &pEngine->nodeGraph;
+}
+
+#if !defined(MA_NO_RESOURCE_MANAGER)
+MA_API ma_resource_manager* ma_engine_get_resource_manager(ma_engine* pEngine)
+{
+ if (pEngine == NULL) {
+ return NULL;
+ }
+
+ #if !defined(MA_NO_RESOURCE_MANAGER)
+ {
+ return pEngine->pResourceManager;
+ }
+ #else
+ {
+ return NULL;
+ }
+ #endif
+}
+#endif
+
+MA_API ma_device* ma_engine_get_device(ma_engine* pEngine)
+{
+ if (pEngine == NULL) {
+ return NULL;
+ }
+
+ #if !defined(MA_NO_DEVICE_IO)
+ {
+ return pEngine->pDevice;
+ }
+ #else
+ {
+ return NULL;
+ }
+ #endif
+}
+
+MA_API ma_log* ma_engine_get_log(ma_engine* pEngine)
+{
+ if (pEngine == NULL) {
+ return NULL;
}
- if (result != MA_SUCCESS) {
- ma_vfs_or_default_close(pVFS, pDecoder->data.vfs.file);
- return result;
+ if (pEngine->pLog != NULL) {
+ return pEngine->pLog;
+ } else {
+ #if !defined(MA_NO_DEVICE_IO)
+ {
+ return ma_device_get_log(ma_engine_get_device(pEngine));
+ }
+ #else
+ {
+ return NULL;
+ }
+ #endif
}
-
- return MA_SUCCESS;
}
-MA_API ma_result ma_decoder_init_vfs_wav_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API ma_node* ma_engine_get_endpoint(ma_engine* pEngine)
{
-#ifdef MA_HAS_WAV
- ma_decoder_config config;
+ return ma_node_graph_get_endpoint(&pEngine->nodeGraph);
+}
- config = ma_decoder_config_init_copy(pConfig);
- config.encodingFormat = ma_encoding_format_wav;
+MA_API ma_uint64 ma_engine_get_time(const ma_engine* pEngine)
+{
+ return ma_node_graph_get_time(&pEngine->nodeGraph);
+}
- return ma_decoder_init_vfs_w(pVFS, pFilePath, &config, pDecoder);
-#else
- (void)pVFS;
- (void)pFilePath;
- (void)pConfig;
- (void)pDecoder;
- return MA_NO_BACKEND;
-#endif
+MA_API ma_result ma_engine_set_time(ma_engine* pEngine, ma_uint64 globalTime)
+{
+ return ma_node_graph_set_time(&pEngine->nodeGraph, globalTime);
}
-MA_API ma_result ma_decoder_init_vfs_flac_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API ma_uint32 ma_engine_get_channels(const ma_engine* pEngine)
{
-#ifdef MA_HAS_FLAC
- ma_decoder_config config;
+ return ma_node_graph_get_channels(&pEngine->nodeGraph);
+}
- config = ma_decoder_config_init_copy(pConfig);
- config.encodingFormat = ma_encoding_format_flac;
+MA_API ma_uint32 ma_engine_get_sample_rate(const ma_engine* pEngine)
+{
+ if (pEngine == NULL) {
+ return 0;
+ }
- return ma_decoder_init_vfs_w(pVFS, pFilePath, &config, pDecoder);
-#else
- (void)pVFS;
- (void)pFilePath;
- (void)pConfig;
- (void)pDecoder;
- return MA_NO_BACKEND;
-#endif
+ return pEngine->sampleRate;
}
-MA_API ma_result ma_decoder_init_vfs_mp3_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+
+MA_API ma_result ma_engine_start(ma_engine* pEngine)
{
-#ifdef MA_HAS_MP3
- ma_decoder_config config;
+ ma_result result;
- config = ma_decoder_config_init_copy(pConfig);
- config.encodingFormat = ma_encoding_format_mp3;
+ if (pEngine == NULL) {
+ return MA_INVALID_ARGS;
+ }
- return ma_decoder_init_vfs_w(pVFS, pFilePath, &config, pDecoder);
-#else
- (void)pVFS;
- (void)pFilePath;
- (void)pConfig;
- (void)pDecoder;
- return MA_NO_BACKEND;
-#endif
+ #if !defined(MA_NO_DEVICE_IO)
+ {
+ if (pEngine->pDevice != NULL) {
+ result = ma_device_start(pEngine->pDevice);
+ } else {
+ result = MA_INVALID_OPERATION; /* The engine is running without a device which means there's no real notion of "starting" the engine. */
+ }
+ }
+ #else
+ {
+ result = MA_INVALID_OPERATION; /* Device IO is disabled, so there's no real notion of "starting" the engine. */
+ }
+ #endif
+
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ return MA_SUCCESS;
}
-MA_API ma_result ma_decoder_init_vfs_vorbis_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API ma_result ma_engine_stop(ma_engine* pEngine)
{
-#ifdef MA_HAS_VORBIS
- ma_decoder_config config;
+ ma_result result;
- config = ma_decoder_config_init_copy(pConfig);
- config.encodingFormat = ma_encoding_format_vorbis;
+ if (pEngine == NULL) {
+ return MA_INVALID_ARGS;
+ }
- return ma_decoder_init_vfs_w(pVFS, pFilePath, &config, pDecoder);
-#else
- (void)pVFS;
- (void)pFilePath;
- (void)pConfig;
- (void)pDecoder;
- return MA_NO_BACKEND;
-#endif
-}
+ #if !defined(MA_NO_DEVICE_IO)
+ {
+ if (pEngine->pDevice != NULL) {
+ result = ma_device_stop(pEngine->pDevice);
+ } else {
+ result = MA_INVALID_OPERATION; /* The engine is running without a device which means there's no real notion of "stopping" the engine. */
+ }
+ }
+ #else
+ {
+ result = MA_INVALID_OPERATION; /* Device IO is disabled, so there's no real notion of "stopping" the engine. */
+ }
+ #endif
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+ return MA_SUCCESS;
+}
-MA_API ma_result ma_decoder_init_file(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API ma_result ma_engine_set_volume(ma_engine* pEngine, float volume)
{
- return ma_decoder_init_vfs(NULL, pFilePath, pConfig, pDecoder);
+ if (pEngine == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ return ma_node_set_output_bus_volume(ma_node_graph_get_endpoint(&pEngine->nodeGraph), 0, volume);
}
-MA_API ma_result ma_decoder_init_file_wav(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API ma_result ma_engine_set_gain_db(ma_engine* pEngine, float gainDB)
{
- return ma_decoder_init_vfs_wav(NULL, pFilePath, pConfig, pDecoder);
+ if (pEngine == NULL) {
+ return MA_INVALID_ARGS;
+ }
+
+ return ma_node_set_output_bus_volume(ma_node_graph_get_endpoint(&pEngine->nodeGraph), 0, ma_volume_db_to_linear(gainDB));
}
-MA_API ma_result ma_decoder_init_file_flac(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+
+MA_API ma_uint32 ma_engine_get_listener_count(const ma_engine* pEngine)
{
- return ma_decoder_init_vfs_flac(NULL, pFilePath, pConfig, pDecoder);
+ if (pEngine == NULL) {
+ return 0;
+ }
+
+ return pEngine->listenerCount;
}
-MA_API ma_result ma_decoder_init_file_mp3(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API ma_uint32 ma_engine_find_closest_listener(const ma_engine* pEngine, float absolutePosX, float absolutePosY, float absolutePosZ)
{
- return ma_decoder_init_vfs_mp3(NULL, pFilePath, pConfig, pDecoder);
+ ma_uint32 iListener;
+ ma_uint32 iListenerClosest;
+ float closestLen2 = MA_FLT_MAX;
+
+ if (pEngine == NULL || pEngine->listenerCount == 1) {
+ return 0;
+ }
+
+ iListenerClosest = 0;
+ for (iListener = 0; iListener < pEngine->listenerCount; iListener += 1) {
+ if (ma_engine_listener_is_enabled(pEngine, iListener)) {
+ float len2 = ma_vec3f_len2(ma_vec3f_sub(pEngine->listeners[iListener].position, ma_vec3f_init_3f(absolutePosX, absolutePosY, absolutePosZ)));
+ if (closestLen2 > len2) {
+ closestLen2 = len2;
+ iListenerClosest = iListener;
+ }
+ }
+ }
+
+ MA_ASSERT(iListenerClosest < 255);
+ return iListenerClosest;
}
-MA_API ma_result ma_decoder_init_file_vorbis(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API void ma_engine_listener_set_position(ma_engine* pEngine, ma_uint32 listenerIndex, float x, float y, float z)
{
- return ma_decoder_init_vfs_vorbis(NULL, pFilePath, pConfig, pDecoder);
+ if (pEngine == NULL || listenerIndex >= pEngine->listenerCount) {
+ return;
+ }
+
+ ma_spatializer_listener_set_position(&pEngine->listeners[listenerIndex], x, y, z);
}
+MA_API ma_vec3f ma_engine_listener_get_position(const ma_engine* pEngine, ma_uint32 listenerIndex)
+{
+ if (pEngine == NULL || listenerIndex >= pEngine->listenerCount) {
+ return ma_vec3f_init_3f(0, 0, 0);
+ }
+ return ma_spatializer_listener_get_position(&pEngine->listeners[listenerIndex]);
+}
-MA_API ma_result ma_decoder_init_file_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API void ma_engine_listener_set_direction(ma_engine* pEngine, ma_uint32 listenerIndex, float x, float y, float z)
{
- return ma_decoder_init_vfs_w(NULL, pFilePath, pConfig, pDecoder);
+ if (pEngine == NULL || listenerIndex >= pEngine->listenerCount) {
+ return;
+ }
+
+ ma_spatializer_listener_set_direction(&pEngine->listeners[listenerIndex], x, y, z);
}
-MA_API ma_result ma_decoder_init_file_wav_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API ma_vec3f ma_engine_listener_get_direction(const ma_engine* pEngine, ma_uint32 listenerIndex)
{
- return ma_decoder_init_vfs_wav_w(NULL, pFilePath, pConfig, pDecoder);
+ if (pEngine == NULL || listenerIndex >= pEngine->listenerCount) {
+ return ma_vec3f_init_3f(0, 0, -1);
+ }
+
+ return ma_spatializer_listener_get_direction(&pEngine->listeners[listenerIndex]);
}
-MA_API ma_result ma_decoder_init_file_flac_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API void ma_engine_listener_set_velocity(ma_engine* pEngine, ma_uint32 listenerIndex, float x, float y, float z)
{
- return ma_decoder_init_vfs_flac_w(NULL, pFilePath, pConfig, pDecoder);
+ if (pEngine == NULL || listenerIndex >= pEngine->listenerCount) {
+ return;
+ }
+
+ ma_spatializer_listener_set_velocity(&pEngine->listeners[listenerIndex], x, y, z);
}
-MA_API ma_result ma_decoder_init_file_mp3_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API ma_vec3f ma_engine_listener_get_velocity(const ma_engine* pEngine, ma_uint32 listenerIndex)
{
- return ma_decoder_init_vfs_mp3_w(NULL, pFilePath, pConfig, pDecoder);
+ if (pEngine == NULL || listenerIndex >= pEngine->listenerCount) {
+ return ma_vec3f_init_3f(0, 0, 0);
+ }
+
+ return ma_spatializer_listener_get_velocity(&pEngine->listeners[listenerIndex]);
}
-MA_API ma_result ma_decoder_init_file_vorbis_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
+MA_API void ma_engine_listener_set_cone(ma_engine* pEngine, ma_uint32 listenerIndex, float innerAngleInRadians, float outerAngleInRadians, float outerGain)
{
- return ma_decoder_init_vfs_vorbis_w(NULL, pFilePath, pConfig, pDecoder);
+ if (pEngine == NULL || listenerIndex >= pEngine->listenerCount) {
+ return;
+ }
+
+ ma_spatializer_listener_set_cone(&pEngine->listeners[listenerIndex], innerAngleInRadians, outerAngleInRadians, outerGain);
}
-MA_API ma_result ma_decoder_uninit(ma_decoder* pDecoder)
+MA_API void ma_engine_listener_get_cone(const ma_engine* pEngine, ma_uint32 listenerIndex, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain)
{
- if (pDecoder == NULL) {
- return MA_INVALID_ARGS;
+ if (pInnerAngleInRadians != NULL) {
+ *pInnerAngleInRadians = 0;
}
- if (pDecoder->pBackend != NULL) {
- if (pDecoder->pBackendVTable != NULL && pDecoder->pBackendVTable->onUninit != NULL) {
- pDecoder->pBackendVTable->onUninit(pDecoder->pBackendUserData, pDecoder->pBackend, &pDecoder->allocationCallbacks);
- }
+ if (pOuterAngleInRadians != NULL) {
+ *pOuterAngleInRadians = 0;
}
- /* Legacy. */
- if (pDecoder->onRead == ma_decoder__on_read_vfs) {
- ma_vfs_or_default_close(pDecoder->data.vfs.pVFS, pDecoder->data.vfs.file);
- pDecoder->data.vfs.file = NULL;
+ if (pOuterGain != NULL) {
+ *pOuterGain = 0;
}
- ma_data_converter_uninit(&pDecoder->converter);
- ma_data_source_uninit(&pDecoder->ds);
-
- return MA_SUCCESS;
+ ma_spatializer_listener_get_cone(&pEngine->listeners[listenerIndex], pInnerAngleInRadians, pOuterAngleInRadians, pOuterGain);
}
-MA_API ma_result ma_decoder_get_cursor_in_pcm_frames(ma_decoder* pDecoder, ma_uint64* pCursor)
+MA_API void ma_engine_listener_set_world_up(ma_engine* pEngine, ma_uint32 listenerIndex, float x, float y, float z)
{
- if (pCursor == NULL) {
- return MA_INVALID_ARGS;
+ if (pEngine == NULL || listenerIndex >= pEngine->listenerCount) {
+ return;
}
- *pCursor = 0;
+ ma_spatializer_listener_set_world_up(&pEngine->listeners[listenerIndex], x, y, z);
+}
- if (pDecoder == NULL) {
- return MA_INVALID_ARGS;
+MA_API ma_vec3f ma_engine_listener_get_world_up(const ma_engine* pEngine, ma_uint32 listenerIndex)
+{
+ if (pEngine == NULL || listenerIndex >= pEngine->listenerCount) {
+ return ma_vec3f_init_3f(0, 1, 0);
}
- *pCursor = pDecoder->readPointerInPCMFrames;
-
- return MA_SUCCESS;
+ return ma_spatializer_listener_get_world_up(&pEngine->listeners[listenerIndex]);
}
-MA_API ma_uint64 ma_decoder_get_length_in_pcm_frames(ma_decoder* pDecoder)
+MA_API void ma_engine_listener_set_enabled(ma_engine* pEngine, ma_uint32 listenerIndex, ma_bool32 isEnabled)
{
- if (pDecoder == NULL) {
- return 0;
+ if (pEngine == NULL || listenerIndex >= pEngine->listenerCount) {
+ return;
}
- if (pDecoder->pBackend != NULL) {
- ma_result result;
- ma_uint64 nativeLengthInPCMFrames;
- ma_uint32 internalSampleRate;
-
- ma_data_source_get_length_in_pcm_frames(pDecoder->pBackend, &nativeLengthInPCMFrames);
-
- result = ma_data_source_get_data_format(pDecoder->pBackend, NULL, NULL, &internalSampleRate);
- if (result != MA_SUCCESS) {
- return 0; /* Failed to retrieve the internal sample rate. */
- }
+ ma_spatializer_listener_set_enabled(&pEngine->listeners[listenerIndex], isEnabled);
+}
- if (internalSampleRate == pDecoder->outputSampleRate) {
- return nativeLengthInPCMFrames;
- } else {
- return ma_calculate_frame_count_after_resampling(pDecoder->outputSampleRate, internalSampleRate, nativeLengthInPCMFrames);
- }
+MA_API ma_bool32 ma_engine_listener_is_enabled(const ma_engine* pEngine, ma_uint32 listenerIndex)
+{
+ if (pEngine == NULL || listenerIndex >= pEngine->listenerCount) {
+ return MA_FALSE;
}
- return 0;
+ return ma_spatializer_listener_is_enabled(&pEngine->listeners[listenerIndex]);
}
-MA_API ma_uint64 ma_decoder_read_pcm_frames(ma_decoder* pDecoder, void* pFramesOut, ma_uint64 frameCount)
+
+#ifndef MA_NO_RESOURCE_MANAGER
+MA_API ma_result ma_engine_play_sound_ex(ma_engine* pEngine, const char* pFilePath, ma_node* pNode, ma_uint32 nodeInputBusIndex)
{
- ma_result result;
- ma_uint64 totalFramesReadOut;
- ma_uint64 totalFramesReadIn;
- void* pRunningFramesOut;
+ ma_result result = MA_SUCCESS;
+ ma_sound_inlined* pSound = NULL;
+ ma_sound_inlined* pNextSound = NULL;
- if (pDecoder == NULL) {
- return 0;
+ if (pEngine == NULL || pFilePath == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pDecoder->pBackend == NULL) {
- return 0;
+ /* Attach to the endpoint node if nothing is specicied. */
+ if (pNode == NULL) {
+ pNode = ma_node_graph_get_endpoint(&pEngine->nodeGraph);
+ nodeInputBusIndex = 0;
}
- /* Fast path. */
- if (pDecoder->converter.isPassthrough) {
- result = ma_data_source_read_pcm_frames(pDecoder->pBackend, pFramesOut, frameCount, &totalFramesReadOut, MA_FALSE);
- } else {
- /*
- Getting here means we need to do data conversion. If we're seeking forward and are _not_ doing resampling we can run this in a fast path. If we're doing resampling we
- need to run through each sample because we need to ensure it's internal cache is updated.
- */
- if (pFramesOut == NULL && pDecoder->converter.hasResampler == MA_FALSE) {
- result = ma_data_source_read_pcm_frames(pDecoder->pBackend, NULL, frameCount, &totalFramesReadOut, MA_FALSE);
- } else {
- /* Slow path. Need to run everything through the data converter. */
- ma_format internalFormat;
- ma_uint32 internalChannels;
+ /*
+ We want to check if we can recycle an already-allocated inlined sound. Since this is just a
+ helper I'm not *too* concerned about performance here and I'm happy to use a lock to keep
+ the implementation simple. Maybe this can be optimized later if there's enough demand, but
+ if this function is being used it probably means the caller doesn't really care too much.
+
+ What we do is check the atEnd flag. When this is true, we can recycle the sound. Otherwise
+ we just keep iterating. If we reach the end without finding a sound to recycle we just
+ allocate a new one. This doesn't scale well for a massive number of sounds being played
+ simultaneously as we don't ever actually free the sound objects. Some kind of garbage
+ collection routine might be valuable for this which I'll think about.
+ */
+ ma_spinlock_lock(&pEngine->inlinedSoundLock);
+ {
+ ma_uint32 soundFlags = 0;
- totalFramesReadOut = 0;
- totalFramesReadIn = 0;
- pRunningFramesOut = pFramesOut;
+ for (pNextSound = pEngine->pInlinedSoundHead; pNextSound != NULL; pNextSound = pNextSound->pNext) {
+ if (ma_sound_at_end(&pNextSound->sound)) {
+ /*
+ The sound is at the end which means it's available for recycling. All we need to do
+ is uninitialize it and reinitialize it. All we're doing is recycling memory.
+ */
+ pSound = pNextSound;
+ c89atomic_fetch_sub_32(&pEngine->inlinedSoundCount, 1);
+ break;
+ }
+ }
- result = ma_data_source_get_data_format(pDecoder->pBackend, &internalFormat, &internalChannels, NULL);
- if (result != MA_SUCCESS) {
- return 0; /* Failed to retrieve the internal format and channel count. */
+ if (pSound != NULL) {
+ /*
+ We actually want to detach the sound from the list here. The reason is because we want the sound
+ to be in a consistent state at the non-recycled case to simplify the logic below.
+ */
+ if (pEngine->pInlinedSoundHead == pSound) {
+ pEngine->pInlinedSoundHead = pSound->pNext;
}
- while (totalFramesReadOut < frameCount) {
- ma_uint8 pIntermediaryBuffer[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In internal format. */
- ma_uint64 intermediaryBufferCap = sizeof(pIntermediaryBuffer) / ma_get_bytes_per_frame(internalFormat, internalChannels);
- ma_uint64 framesToReadThisIterationIn;
- ma_uint64 framesReadThisIterationIn;
- ma_uint64 framesToReadThisIterationOut;
- ma_uint64 framesReadThisIterationOut;
- ma_uint64 requiredInputFrameCount;
+ if (pSound->pPrev != NULL) {
+ pSound->pPrev->pNext = pSound->pNext;
+ }
+ if (pSound->pNext != NULL) {
+ pSound->pNext->pPrev = pSound->pPrev;
+ }
- framesToReadThisIterationOut = (frameCount - totalFramesReadOut);
- framesToReadThisIterationIn = framesToReadThisIterationOut;
- if (framesToReadThisIterationIn > intermediaryBufferCap) {
- framesToReadThisIterationIn = intermediaryBufferCap;
- }
+ /* Now the previous sound needs to be uninitialized. */
+ ma_sound_uninit(&pNextSound->sound);
+ } else {
+ /* No sound available for recycling. Allocate one now. */
+ pSound = (ma_sound_inlined*)ma_malloc(sizeof(*pSound), &pEngine->allocationCallbacks);
+ }
- requiredInputFrameCount = ma_data_converter_get_required_input_frame_count(&pDecoder->converter, framesToReadThisIterationOut);
- if (framesToReadThisIterationIn > requiredInputFrameCount) {
- framesToReadThisIterationIn = requiredInputFrameCount;
- }
+ if (pSound != NULL) { /* Safety check for the allocation above. */
+ /*
+ At this point we should have memory allocated for the inlined sound. We just need
+ to initialize it like a normal sound now.
+ */
+ soundFlags |= MA_SOUND_FLAG_ASYNC; /* For inlined sounds we don't want to be sitting around waiting for stuff to load so force an async load. */
+ soundFlags |= MA_SOUND_FLAG_NO_DEFAULT_ATTACHMENT; /* We want specific control over where the sound is attached in the graph. We'll attach it manually just before playing the sound. */
+ soundFlags |= MA_SOUND_FLAG_NO_PITCH; /* Pitching isn't usable with inlined sounds, so disable it to save on speed. */
+ soundFlags |= MA_SOUND_FLAG_NO_SPATIALIZATION; /* Not currently doing spatialization with inlined sounds, but this might actually change later. For now disable spatialization. Will be removed if we ever add support for spatialization here. */
+
+ result = ma_sound_init_from_file(pEngine, pFilePath, soundFlags, NULL, NULL, &pSound->sound);
+ if (result == MA_SUCCESS) {
+ /* Now attach the sound to the graph. */
+ result = ma_node_attach_output_bus(pSound, 0, pNode, nodeInputBusIndex);
+ if (result == MA_SUCCESS) {
+ /* At this point the sound should be loaded and we can go ahead and add it to the list. The new item becomes the new head. */
+ pSound->pNext = pEngine->pInlinedSoundHead;
+ pSound->pPrev = NULL;
- if (requiredInputFrameCount > 0) {
- result = ma_data_source_read_pcm_frames(pDecoder->pBackend, pIntermediaryBuffer, framesToReadThisIterationIn, &framesReadThisIterationIn, MA_FALSE);
- totalFramesReadIn += framesReadThisIterationIn;
+ pEngine->pInlinedSoundHead = pSound; /* <-- This is what attaches the sound to the list. */
+ if (pSound->pNext != NULL) {
+ pSound->pNext->pPrev = pSound;
+ }
} else {
- framesReadThisIterationIn = 0;
+ ma_free(pSound, &pEngine->allocationCallbacks);
}
+ } else {
+ ma_free(pSound, &pEngine->allocationCallbacks);
+ }
+ } else {
+ result = MA_OUT_OF_MEMORY;
+ }
+ }
+ ma_spinlock_unlock(&pEngine->inlinedSoundLock);
- /*
- At this point we have our decoded data in input format and now we need to convert to output format. Note that even if we didn't read any
- input frames, we still want to try processing frames because there may some output frames generated from cached input data.
- */
- framesReadThisIterationOut = framesToReadThisIterationOut;
- result = ma_data_converter_process_pcm_frames(&pDecoder->converter, pIntermediaryBuffer, &framesReadThisIterationIn, pRunningFramesOut, &framesReadThisIterationOut);
- if (result != MA_SUCCESS) {
- break;
- }
+ if (result != MA_SUCCESS) {
+ return result;
+ }
- totalFramesReadOut += framesReadThisIterationOut;
+ /* Finally we can start playing the sound. */
+ result = ma_sound_start(&pSound->sound);
+ if (result != MA_SUCCESS) {
+ /* Failed to start the sound. We need to mark it for recycling and return an error. */
+ c89atomic_exchange_32(&pSound->sound.atEnd, MA_TRUE);
+ return result;
+ }
- if (pRunningFramesOut != NULL) {
- pRunningFramesOut = ma_offset_ptr(pRunningFramesOut, framesReadThisIterationOut * ma_get_bytes_per_frame(pDecoder->outputFormat, pDecoder->outputChannels));
- }
+ c89atomic_fetch_add_32(&pEngine->inlinedSoundCount, 1);
+ return result;
+}
- if (framesReadThisIterationIn == 0 && framesReadThisIterationOut == 0) {
- break; /* We're done. */
- }
- }
- }
+MA_API ma_result ma_engine_play_sound(ma_engine* pEngine, const char* pFilePath, ma_sound_group* pGroup)
+{
+ return ma_engine_play_sound_ex(pEngine, pFilePath, pGroup, 0);
+}
+#endif
+
+
+static ma_result ma_sound_preinit(ma_engine* pEngine, ma_sound* pSound)
+{
+ if (pSound == NULL) {
+ return MA_INVALID_ARGS;
}
- pDecoder->readPointerInPCMFrames += totalFramesReadOut;
+ MA_ZERO_OBJECT(pSound);
+ pSound->seekTarget = MA_SEEK_TARGET_NONE;
+
+ if (pEngine == NULL) {
+ return MA_INVALID_ARGS;
+ }
- return totalFramesReadOut;
+ return MA_SUCCESS;
}
-MA_API ma_result ma_decoder_seek_to_pcm_frame(ma_decoder* pDecoder, ma_uint64 frameIndex)
+static ma_result ma_sound_init_from_data_source_internal(ma_engine* pEngine, const ma_sound_config* pConfig, ma_sound* pSound)
{
- if (pDecoder == NULL) {
+ ma_result result;
+ ma_engine_node_config engineNodeConfig;
+ ma_engine_node_type type; /* Will be set to ma_engine_node_type_group if no data source is specified. */
+
+ /* Do not clear pSound to zero here - that's done at a higher level with ma_sound_preinit(). */
+ MA_ASSERT(pEngine != NULL);
+ MA_ASSERT(pSound != NULL);
+
+ if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
- if (pDecoder->pBackend != NULL) {
- ma_result result;
- ma_uint64 internalFrameIndex;
- ma_uint32 internalSampleRate;
+ pSound->pDataSource = pConfig->pDataSource;
+
+ if (pConfig->pDataSource != NULL) {
+ type = ma_engine_node_type_sound;
+ } else {
+ type = ma_engine_node_type_group;
+ }
- result = ma_data_source_get_data_format(pDecoder->pBackend, NULL, NULL, &internalSampleRate);
+ /*
+ Sounds are engine nodes. Before we can initialize this we need to determine the channel count.
+ If we can't do this we need to abort. It's up to the caller to ensure they're using a data
+ source that provides this information upfront.
+ */
+ engineNodeConfig = ma_engine_node_config_init(pEngine, type, pConfig->flags);
+ engineNodeConfig.channelsIn = pConfig->channelsIn;
+ engineNodeConfig.channelsOut = pConfig->channelsOut;
+
+ /* If we're loading from a data source the input channel count needs to be the data source's native channel count. */
+ if (pConfig->pDataSource != NULL) {
+ result = ma_data_source_get_data_format(pConfig->pDataSource, NULL, &engineNodeConfig.channelsIn, &engineNodeConfig.sampleRate, NULL, 0);
if (result != MA_SUCCESS) {
- return result; /* Failed to retrieve the internal sample rate. */
+ return result; /* Failed to retrieve the channel count. */
}
- if (internalSampleRate == pDecoder->outputSampleRate) {
- internalFrameIndex = frameIndex;
- } else {
- internalFrameIndex = ma_calculate_frame_count_after_resampling(internalSampleRate, pDecoder->outputSampleRate, frameIndex);
+ if (engineNodeConfig.channelsIn == 0) {
+ return MA_INVALID_OPERATION; /* Invalid channel count. */
}
- result = ma_data_source_seek_to_pcm_frame(pDecoder->pBackend, internalFrameIndex);
- if (result == MA_SUCCESS) {
- pDecoder->readPointerInPCMFrames = frameIndex;
+ if (engineNodeConfig.channelsOut == MA_SOUND_SOURCE_CHANNEL_COUNT) {
+ engineNodeConfig.channelsOut = engineNodeConfig.channelsIn;
}
+ }
+
+ /* Getting here means we should have a valid channel count and we can initialize the engine node. */
+ result = ma_engine_node_init(&engineNodeConfig, &pEngine->allocationCallbacks, &pSound->engineNode);
+ if (result != MA_SUCCESS) {
return result;
}
- /* Should never get here, but if we do it means onSeekToPCMFrame was not set by the backend. */
- return MA_INVALID_ARGS;
-}
-
-MA_API ma_result ma_decoder_get_available_frames(ma_decoder* pDecoder, ma_uint64* pAvailableFrames)
-{
- ma_uint64 totalFrameCount;
+ /* If no attachment is specified, attach the sound straight to the endpoint. */
+ if (pConfig->pInitialAttachment == NULL) {
+ /* No group. Attach straight to the endpoint by default, unless the caller has requested that do not. */
+ if ((pConfig->flags & MA_SOUND_FLAG_NO_DEFAULT_ATTACHMENT) == 0) {
+ result = ma_node_attach_output_bus(pSound, 0, ma_node_graph_get_endpoint(&pEngine->nodeGraph), 0);
+ }
+ } else {
+ /* An attachment is specified. Attach to it by default. The sound has only a single output bus, and the config will specify which input bus to attach to. */
+ result = ma_node_attach_output_bus(pSound, 0, pConfig->pInitialAttachment, pConfig->initialAttachmentInputBusIndex);
+ }
- if (pAvailableFrames == NULL) {
- return MA_INVALID_ARGS;
+ if (result != MA_SUCCESS) {
+ ma_engine_node_uninit(&pSound->engineNode, &pEngine->allocationCallbacks);
+ return result;
}
- *pAvailableFrames = 0;
- if (pDecoder == NULL) {
- return MA_INVALID_ARGS;
+ /* Apply initial range and looping state to the data source if applicable. */
+ if (pConfig->rangeBegInPCMFrames != 0 || pConfig->rangeEndInPCMFrames != ~((ma_uint64)0)) {
+ ma_data_source_set_range_in_pcm_frames(ma_sound_get_data_source(pSound), pConfig->rangeBegInPCMFrames, pConfig->rangeEndInPCMFrames);
}
- totalFrameCount = ma_decoder_get_length_in_pcm_frames(pDecoder);
- if (totalFrameCount == 0) {
- return MA_NOT_IMPLEMENTED;
+ if (pConfig->loopPointBegInPCMFrames != 0 || pConfig->loopPointEndInPCMFrames != ~((ma_uint64)0)) {
+ ma_data_source_set_range_in_pcm_frames(ma_sound_get_data_source(pSound), pConfig->loopPointBegInPCMFrames, pConfig->loopPointEndInPCMFrames);
}
- if (totalFrameCount <= pDecoder->readPointerInPCMFrames) {
- *pAvailableFrames = 0;
- } else {
- *pAvailableFrames = totalFrameCount - pDecoder->readPointerInPCMFrames;
- }
+ ma_sound_set_looping(pSound, pConfig->isLooping);
- return MA_SUCCESS; /* No frames available. */
+ return MA_SUCCESS;
}
-
-static ma_result ma_decoder__full_decode_and_uninit(ma_decoder* pDecoder, ma_decoder_config* pConfigOut, ma_uint64* pFrameCountOut, void** ppPCMFramesOut)
+#ifndef MA_NO_RESOURCE_MANAGER
+MA_API ma_result ma_sound_init_from_file_internal(ma_engine* pEngine, const ma_sound_config* pConfig, ma_sound* pSound)
{
- ma_uint64 totalFrameCount;
- ma_uint64 bpf;
- ma_uint64 dataCapInFrames;
- void* pPCMFramesOut;
-
- MA_ASSERT(pDecoder != NULL);
-
- totalFrameCount = 0;
- bpf = ma_get_bytes_per_frame(pDecoder->outputFormat, pDecoder->outputChannels);
-
- /* The frame count is unknown until we try reading. Thus, we just run in a loop. */
- dataCapInFrames = 0;
- pPCMFramesOut = NULL;
- for (;;) {
- ma_uint64 frameCountToTryReading;
- ma_uint64 framesJustRead;
-
- /* Make room if there's not enough. */
- if (totalFrameCount == dataCapInFrames) {
- void* pNewPCMFramesOut;
- ma_uint64 oldDataCapInFrames = dataCapInFrames;
- ma_uint64 newDataCapInFrames = dataCapInFrames*2;
- if (newDataCapInFrames == 0) {
- newDataCapInFrames = 4096;
- }
+ ma_result result = MA_SUCCESS;
+ ma_uint32 flags;
+ ma_sound_config config;
+ ma_resource_manager_pipeline_notifications notifications;
- if ((newDataCapInFrames * bpf) > MA_SIZE_MAX) {
- ma__free_from_callbacks(pPCMFramesOut, &pDecoder->allocationCallbacks);
- return MA_TOO_BIG;
- }
+ /*
+ The engine requires knowledge of the channel count of the underlying data source before it can
+ initialize the sound. Therefore, we need to make the resource manager wait until initialization
+ of the underlying data source to be initialized so we can get access to the channel count. To
+ do this, the MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT is forced.
+
+ Because we're initializing the data source before the sound, there's a chance the notification
+ will get triggered before this function returns. This is OK, so long as the caller is aware of
+ it and can avoid accessing the sound from within the notification.
+ */
+ flags = pConfig->flags | MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT;
+ pSound->pResourceManagerDataSource = (ma_resource_manager_data_source*)ma_malloc(sizeof(*pSound->pResourceManagerDataSource), &pEngine->allocationCallbacks);
+ if (pSound->pResourceManagerDataSource == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
- pNewPCMFramesOut = (void*)ma__realloc_from_callbacks(pPCMFramesOut, (size_t)(newDataCapInFrames * bpf), (size_t)(oldDataCapInFrames * bpf), &pDecoder->allocationCallbacks);
- if (pNewPCMFramesOut == NULL) {
- ma__free_from_callbacks(pPCMFramesOut, &pDecoder->allocationCallbacks);
- return MA_OUT_OF_MEMORY;
- }
+ notifications = ma_resource_manager_pipeline_notifications_init();
+ notifications.done.pFence = pConfig->pDoneFence;
- dataCapInFrames = newDataCapInFrames;
- pPCMFramesOut = pNewPCMFramesOut;
+ /*
+ We must wrap everything around the fence if one was specified. This ensures ma_fence_wait() does
+ not return prematurely before the sound has finished initializing.
+ */
+ if (notifications.done.pFence) { ma_fence_acquire(notifications.done.pFence); }
+ {
+ ma_resource_manager_data_source_config resourceManagerDataSourceConfig = ma_resource_manager_data_source_config_init();
+ resourceManagerDataSourceConfig.pFilePath = pConfig->pFilePath;
+ resourceManagerDataSourceConfig.pFilePathW = pConfig->pFilePathW;
+ resourceManagerDataSourceConfig.flags = flags;
+ resourceManagerDataSourceConfig.pNotifications = ¬ifications;
+ resourceManagerDataSourceConfig.initialSeekPointInPCMFrames = pConfig->initialSeekPointInPCMFrames;
+ resourceManagerDataSourceConfig.rangeBegInPCMFrames = pConfig->rangeBegInPCMFrames;
+ resourceManagerDataSourceConfig.rangeEndInPCMFrames = pConfig->rangeEndInPCMFrames;
+ resourceManagerDataSourceConfig.loopPointBegInPCMFrames = pConfig->loopPointBegInPCMFrames;
+ resourceManagerDataSourceConfig.loopPointEndInPCMFrames = pConfig->loopPointEndInPCMFrames;
+ resourceManagerDataSourceConfig.isLooping = pConfig->isLooping;
+
+ result = ma_resource_manager_data_source_init_ex(pEngine->pResourceManager, &resourceManagerDataSourceConfig, pSound->pResourceManagerDataSource);
+ if (result != MA_SUCCESS) {
+ goto done;
}
- frameCountToTryReading = dataCapInFrames - totalFrameCount;
- MA_ASSERT(frameCountToTryReading > 0);
+ pSound->ownsDataSource = MA_TRUE; /* <-- Important. Not setting this will result in the resource manager data source never getting uninitialized. */
- framesJustRead = ma_decoder_read_pcm_frames(pDecoder, (ma_uint8*)pPCMFramesOut + (totalFrameCount * bpf), frameCountToTryReading);
- totalFrameCount += framesJustRead;
+ /* We need to use a slightly customized version of the config so we'll need to make a copy. */
+ config = *pConfig;
+ config.pFilePath = NULL;
+ config.pFilePathW = NULL;
+ config.pDataSource = pSound->pResourceManagerDataSource;
- if (framesJustRead < frameCountToTryReading) {
- break;
+ result = ma_sound_init_from_data_source_internal(pEngine, &config, pSound);
+ if (result != MA_SUCCESS) {
+ ma_resource_manager_data_source_uninit(pSound->pResourceManagerDataSource);
+ ma_free(pSound->pResourceManagerDataSource, &pEngine->allocationCallbacks);
+ MA_ZERO_OBJECT(pSound);
+ goto done;
}
}
+done:
+ if (notifications.done.pFence) { ma_fence_release(notifications.done.pFence); }
+ return result;
+}
+MA_API ma_result ma_sound_init_from_file(ma_engine* pEngine, const char* pFilePath, ma_uint32 flags, ma_sound_group* pGroup, ma_fence* pDoneFence, ma_sound* pSound)
+{
+ ma_sound_config config = ma_sound_config_init();
+ config.pFilePath = pFilePath;
+ config.flags = flags;
+ config.pInitialAttachment = pGroup;
+ config.pDoneFence = pDoneFence;
+ return ma_sound_init_ex(pEngine, &config, pSound);
+}
- if (pConfigOut != NULL) {
- pConfigOut->format = pDecoder->outputFormat;
- pConfigOut->channels = pDecoder->outputChannels;
- pConfigOut->sampleRate = pDecoder->outputSampleRate;
- ma_channel_map_copy(pConfigOut->channelMap, pDecoder->outputChannelMap, pDecoder->outputChannels);
- }
-
- if (ppPCMFramesOut != NULL) {
- *ppPCMFramesOut = pPCMFramesOut;
- } else {
- ma__free_from_callbacks(pPCMFramesOut, &pDecoder->allocationCallbacks);
- }
-
- if (pFrameCountOut != NULL) {
- *pFrameCountOut = totalFrameCount;
- }
-
- ma_decoder_uninit(pDecoder);
- return MA_SUCCESS;
+MA_API ma_result ma_sound_init_from_file_w(ma_engine* pEngine, const wchar_t* pFilePath, ma_uint32 flags, ma_sound_group* pGroup, ma_fence* pDoneFence, ma_sound* pSound)
+{
+ ma_sound_config config = ma_sound_config_init();
+ config.pFilePathW = pFilePath;
+ config.flags = flags;
+ config.pInitialAttachment = pGroup;
+ config.pDoneFence = pDoneFence;
+ return ma_sound_init_ex(pEngine, &config, pSound);
}
-MA_API ma_result ma_decode_from_vfs(ma_vfs* pVFS, const char* pFilePath, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut)
+MA_API ma_result ma_sound_init_copy(ma_engine* pEngine, const ma_sound* pExistingSound, ma_uint32 flags, ma_sound_group* pGroup, ma_sound* pSound)
{
ma_result result;
- ma_decoder_config config;
- ma_decoder decoder;
+ ma_sound_config config;
- if (pFrameCountOut != NULL) {
- *pFrameCountOut = 0;
+ result = ma_sound_preinit(pEngine, pSound);
+ if (result != MA_SUCCESS) {
+ return result;
}
- if (ppPCMFramesOut != NULL) {
- *ppPCMFramesOut = NULL;
+
+ if (pExistingSound == NULL) {
+ return MA_INVALID_ARGS;
}
- config = ma_decoder_config_init_copy(pConfig);
+ /* Cloning only works for data buffers (not streams) that are loaded from the resource manager. */
+ if (pExistingSound->pResourceManagerDataSource == NULL) {
+ return MA_INVALID_OPERATION;
+ }
- result = ma_decoder_init_vfs(pVFS, pFilePath, &config, &decoder);
+ /*
+ We need to make a clone of the data source. If the data source is not a data buffer (i.e. a stream)
+ the this will fail.
+ */
+ pSound->pResourceManagerDataSource = (ma_resource_manager_data_source*)ma_malloc(sizeof(*pSound->pResourceManagerDataSource), &pEngine->allocationCallbacks);
+ if (pSound->pResourceManagerDataSource == NULL) {
+ return MA_OUT_OF_MEMORY;
+ }
+
+ result = ma_resource_manager_data_source_init_copy(pEngine->pResourceManager, pExistingSound->pResourceManagerDataSource, pSound->pResourceManagerDataSource);
if (result != MA_SUCCESS) {
+ ma_free(pSound->pResourceManagerDataSource, &pEngine->allocationCallbacks);
return result;
}
- result = ma_decoder__full_decode_and_uninit(&decoder, pConfig, pFrameCountOut, ppPCMFramesOut);
+ config = ma_sound_config_init();
+ config.pDataSource = pSound->pResourceManagerDataSource;
+ config.flags = flags;
+ config.pInitialAttachment = pGroup;
- return result;
+ result = ma_sound_init_from_data_source_internal(pEngine, &config, pSound);
+ if (result != MA_SUCCESS) {
+ ma_resource_manager_data_source_uninit(pSound->pResourceManagerDataSource);
+ ma_free(pSound->pResourceManagerDataSource, &pEngine->allocationCallbacks);
+ MA_ZERO_OBJECT(pSound);
+ return result;
+ }
+
+ return MA_SUCCESS;
}
+#endif
-MA_API ma_result ma_decode_file(const char* pFilePath, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut)
+MA_API ma_result ma_sound_init_from_data_source(ma_engine* pEngine, ma_data_source* pDataSource, ma_uint32 flags, ma_sound_group* pGroup, ma_sound* pSound)
{
- return ma_decode_from_vfs(NULL, pFilePath, pConfig, pFrameCountOut, ppPCMFramesOut);
+ ma_sound_config config = ma_sound_config_init();
+ config.pDataSource = pDataSource;
+ config.flags = flags;
+ config.pInitialAttachment = pGroup;
+ return ma_sound_init_ex(pEngine, &config, pSound);
}
-MA_API ma_result ma_decode_memory(const void* pData, size_t dataSize, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut)
+MA_API ma_result ma_sound_init_ex(ma_engine* pEngine, const ma_sound_config* pConfig, ma_sound* pSound)
{
- ma_decoder_config config;
- ma_decoder decoder;
ma_result result;
- if (pFrameCountOut != NULL) {
- *pFrameCountOut = 0;
- }
- if (ppPCMFramesOut != NULL) {
- *ppPCMFramesOut = NULL;
+ result = ma_sound_preinit(pEngine, pSound);
+ if (result != MA_SUCCESS) {
+ return result;
}
- if (pData == NULL || dataSize == 0) {
+ if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
- config = ma_decoder_config_init_copy(pConfig);
-
- result = ma_decoder_init_memory(pData, dataSize, &config, &decoder);
- if (result != MA_SUCCESS) {
- return result;
+ /* We need to load the sound differently depending on whether or not we're loading from a file. */
+#ifndef MA_NO_RESOURCE_MANAGER
+ if (pConfig->pFilePath != NULL || pConfig->pFilePathW != NULL) {
+ return ma_sound_init_from_file_internal(pEngine, pConfig, pSound);
+ } else
+#endif
+ {
+ /*
+ Getting here means we're not loading from a file. We may be loading from an already-initialized
+ data source, or none at all. If we aren't specifying any data source, we'll be initializing the
+ the equivalent to a group. ma_data_source_init_from_data_source_internal() will deal with this
+ for us, so no special treatment required here.
+ */
+ return ma_sound_init_from_data_source_internal(pEngine, pConfig, pSound);
}
-
- return ma_decoder__full_decode_and_uninit(&decoder, pConfig, pFrameCountOut, ppPCMFramesOut);
}
-#endif /* MA_NO_DECODING */
+MA_API void ma_sound_uninit(ma_sound* pSound)
+{
+ if (pSound == NULL) {
+ return;
+ }
+
+ /*
+ Always uninitialize the node first. This ensures it's detached from the graph and does not return until it has done
+ so which makes thread safety beyond this point trivial.
+ */
+ ma_engine_node_uninit(&pSound->engineNode, &pSound->engineNode.pEngine->allocationCallbacks);
-#ifndef MA_NO_ENCODING
+ /* Once the sound is detached from the group we can guarantee that it won't be referenced by the mixer thread which means it's safe for us to destroy the data source. */
+#ifndef MA_NO_RESOURCE_MANAGER
+ if (pSound->ownsDataSource) {
+ ma_resource_manager_data_source_uninit(pSound->pResourceManagerDataSource);
+ ma_free(pSound->pResourceManagerDataSource, &pSound->engineNode.pEngine->allocationCallbacks);
+ pSound->pDataSource = NULL;
+ }
+#else
+ MA_ASSERT(pSound->ownsDataSource == MA_FALSE);
+#endif
+}
-#if defined(MA_HAS_WAV)
-static size_t ma_encoder__internal_on_write_wav(void* pUserData, const void* pData, size_t bytesToWrite)
+MA_API ma_engine* ma_sound_get_engine(const ma_sound* pSound)
{
- ma_encoder* pEncoder = (ma_encoder*)pUserData;
- MA_ASSERT(pEncoder != NULL);
+ if (pSound == NULL) {
+ return NULL;
+ }
- return pEncoder->onWrite(pEncoder, pData, bytesToWrite);
+ return pSound->engineNode.pEngine;
}
-static drwav_bool32 ma_encoder__internal_on_seek_wav(void* pUserData, int offset, drwav_seek_origin origin)
+MA_API ma_data_source* ma_sound_get_data_source(const ma_sound* pSound)
{
- ma_encoder* pEncoder = (ma_encoder*)pUserData;
- MA_ASSERT(pEncoder != NULL);
+ if (pSound == NULL) {
+ return NULL;
+ }
- return pEncoder->onSeek(pEncoder, offset, (origin == drwav_seek_origin_start) ? ma_seek_origin_start : ma_seek_origin_current);
+ return pSound->pDataSource;
}
-static ma_result ma_encoder__on_init_wav(ma_encoder* pEncoder)
+MA_API ma_result ma_sound_start(ma_sound* pSound)
{
- drwav_data_format wavFormat;
- drwav_allocation_callbacks allocationCallbacks;
- drwav* pWav;
-
- MA_ASSERT(pEncoder != NULL);
+ if (pSound == NULL) {
+ return MA_INVALID_ARGS;
+ }
- pWav = (drwav*)ma__malloc_from_callbacks(sizeof(*pWav), &pEncoder->config.allocationCallbacks);
- if (pWav == NULL) {
- return MA_OUT_OF_MEMORY;
+ /* If the sound is already playing, do nothing. */
+ if (ma_sound_is_playing(pSound)) {
+ return MA_SUCCESS;
}
- wavFormat.container = drwav_container_riff;
- wavFormat.channels = pEncoder->config.channels;
- wavFormat.sampleRate = pEncoder->config.sampleRate;
- wavFormat.bitsPerSample = ma_get_bytes_per_sample(pEncoder->config.format) * 8;
- if (pEncoder->config.format == ma_format_f32) {
- wavFormat.format = DR_WAVE_FORMAT_IEEE_FLOAT;
- } else {
- wavFormat.format = DR_WAVE_FORMAT_PCM;
+ /* If the sound is at the end it means we want to start from the start again. */
+ if (ma_sound_at_end(pSound)) {
+ ma_result result = ma_data_source_seek_to_pcm_frame(pSound->pDataSource, 0);
+ if (result != MA_SUCCESS && result != MA_NOT_IMPLEMENTED) {
+ return result; /* Failed to seek back to the start. */
+ }
+
+ /* Make sure we clear the end indicator. */
+ c89atomic_exchange_32(&pSound->atEnd, MA_FALSE);
}
- allocationCallbacks.pUserData = pEncoder->config.allocationCallbacks.pUserData;
- allocationCallbacks.onMalloc = pEncoder->config.allocationCallbacks.onMalloc;
- allocationCallbacks.onRealloc = pEncoder->config.allocationCallbacks.onRealloc;
- allocationCallbacks.onFree = pEncoder->config.allocationCallbacks.onFree;
+ /* Make sure the sound is started. If there's a start delay, the sound won't actually start until the start time is reached. */
+ ma_node_set_state(pSound, ma_node_state_started);
- if (!drwav_init_write(pWav, &wavFormat, ma_encoder__internal_on_write_wav, ma_encoder__internal_on_seek_wav, pEncoder, &allocationCallbacks)) {
- return MA_ERROR;
+ return MA_SUCCESS;
+}
+
+MA_API ma_result ma_sound_stop(ma_sound* pSound)
+{
+ if (pSound == NULL) {
+ return MA_INVALID_ARGS;
}
- pEncoder->pInternalEncoder = pWav;
+ /* This will stop the sound immediately. Use ma_sound_set_stop_time() to stop the sound at a specific time. */
+ ma_node_set_state(pSound, ma_node_state_stopped);
return MA_SUCCESS;
}
-static void ma_encoder__on_uninit_wav(ma_encoder* pEncoder)
+MA_API void ma_sound_set_volume(ma_sound* pSound, float volume)
{
- drwav* pWav;
+ if (pSound == NULL) {
+ return;
+ }
- MA_ASSERT(pEncoder != NULL);
+ /* The volume is controlled via the output bus. */
+ ma_node_set_output_bus_volume(pSound, 0, volume);
+}
- pWav = (drwav*)pEncoder->pInternalEncoder;
- MA_ASSERT(pWav != NULL);
+MA_API float ma_sound_get_volume(const ma_sound* pSound)
+{
+ if (pSound == NULL) {
+ return 0;
+ }
- drwav_uninit(pWav);
- ma__free_from_callbacks(pWav, &pEncoder->config.allocationCallbacks);
+ return ma_node_get_output_bus_volume(pSound, 0);
}
-static ma_uint64 ma_encoder__on_write_pcm_frames_wav(ma_encoder* pEncoder, const void* pFramesIn, ma_uint64 frameCount)
+MA_API void ma_sound_set_pan(ma_sound* pSound, float pan)
{
- drwav* pWav;
+ if (pSound == NULL) {
+ return;
+ }
- MA_ASSERT(pEncoder != NULL);
+ ma_panner_set_pan(&pSound->engineNode.panner, pan);
+}
- pWav = (drwav*)pEncoder->pInternalEncoder;
- MA_ASSERT(pWav != NULL);
+MA_API float ma_sound_get_pan(const ma_sound* pSound)
+{
+ if (pSound == NULL) {
+ return 0;
+ }
- return drwav_write_pcm_frames(pWav, frameCount, pFramesIn);
+ return ma_panner_get_pan(&pSound->engineNode.panner);
}
-#endif
-MA_API ma_encoder_config ma_encoder_config_init(ma_resource_format resourceFormat, ma_format format, ma_uint32 channels, ma_uint32 sampleRate)
+MA_API void ma_sound_set_pan_mode(ma_sound* pSound, ma_pan_mode panMode)
{
- ma_encoder_config config;
-
- MA_ZERO_OBJECT(&config);
- config.resourceFormat = resourceFormat;
- config.format = format;
- config.channels = channels;
- config.sampleRate = sampleRate;
+ if (pSound == NULL) {
+ return;
+ }
- return config;
+ ma_panner_set_mode(&pSound->engineNode.panner, panMode);
}
-MA_API ma_result ma_encoder_preinit(const ma_encoder_config* pConfig, ma_encoder* pEncoder)
+MA_API ma_pan_mode ma_sound_get_pan_mode(const ma_sound* pSound)
{
- ma_result result;
-
- if (pEncoder == NULL) {
- return MA_INVALID_ARGS;
+ if (pSound == NULL) {
+ return ma_pan_mode_balance;
}
- MA_ZERO_OBJECT(pEncoder);
+ return ma_panner_get_mode(&pSound->engineNode.panner);
+}
- if (pConfig == NULL) {
- return MA_INVALID_ARGS;
+MA_API void ma_sound_set_pitch(ma_sound* pSound, float pitch)
+{
+ if (pSound == NULL) {
+ return;
}
- if (pConfig->format == ma_format_unknown || pConfig->channels == 0 || pConfig->sampleRate == 0) {
- return MA_INVALID_ARGS;
+ if (pitch <= 0) {
+ return;
}
- pEncoder->config = *pConfig;
+ c89atomic_exchange_explicit_f32(&pSound->engineNode.pitch, pitch, c89atomic_memory_order_release);
+}
- result = ma_allocation_callbacks_init_copy(&pEncoder->config.allocationCallbacks, &pConfig->allocationCallbacks);
- if (result != MA_SUCCESS) {
- return result;
+MA_API float ma_sound_get_pitch(const ma_sound* pSound)
+{
+ if (pSound == NULL) {
+ return 0;
}
- return MA_SUCCESS;
+ return c89atomic_load_f32(&pSound->engineNode.pitch); /* Naughty const-cast for this. */
}
-MA_API ma_result ma_encoder_init__internal(ma_encoder_write_proc onWrite, ma_encoder_seek_proc onSeek, void* pUserData, ma_encoder* pEncoder)
+MA_API void ma_sound_set_spatialization_enabled(ma_sound* pSound, ma_bool32 enabled)
{
- ma_result result = MA_SUCCESS;
-
- /* This assumes ma_encoder_preinit() has been called prior. */
- MA_ASSERT(pEncoder != NULL);
-
- if (onWrite == NULL || onSeek == NULL) {
- return MA_INVALID_ARGS;
+ if (pSound == NULL) {
+ return;
}
- pEncoder->onWrite = onWrite;
- pEncoder->onSeek = onSeek;
- pEncoder->pUserData = pUserData;
-
- switch (pEncoder->config.resourceFormat)
- {
- case ma_resource_format_wav:
- {
- #if defined(MA_HAS_WAV)
- pEncoder->onInit = ma_encoder__on_init_wav;
- pEncoder->onUninit = ma_encoder__on_uninit_wav;
- pEncoder->onWritePCMFrames = ma_encoder__on_write_pcm_frames_wav;
- #else
- result = MA_NO_BACKEND;
- #endif
- } break;
-
- default:
- {
- result = MA_INVALID_ARGS;
- } break;
- }
+ c89atomic_exchange_explicit_32(&pSound->engineNode.isSpatializationDisabled, !enabled, c89atomic_memory_order_release);
+}
- /* Getting here means we should have our backend callbacks set up. */
- if (result == MA_SUCCESS) {
- result = pEncoder->onInit(pEncoder);
- if (result != MA_SUCCESS) {
- return result;
- }
+MA_API ma_bool32 ma_sound_is_spatialization_enabled(const ma_sound* pSound)
+{
+ if (pSound == NULL) {
+ return MA_FALSE;
}
- return MA_SUCCESS;
+ return ma_engine_node_is_spatialization_enabled(&pSound->engineNode);
}
-MA_API size_t ma_encoder__on_write_stdio(ma_encoder* pEncoder, const void* pBufferIn, size_t bytesToWrite)
+MA_API void ma_sound_set_pinned_listener_index(ma_sound* pSound, ma_uint32 listenerIndex)
{
- return fwrite(pBufferIn, 1, bytesToWrite, (FILE*)pEncoder->pFile);
+ if (pSound == NULL || listenerIndex >= ma_engine_get_listener_count(ma_sound_get_engine(pSound))) {
+ return;
+ }
+
+ c89atomic_exchange_explicit_32(&pSound->engineNode.pinnedListenerIndex, listenerIndex, c89atomic_memory_order_release);
}
-MA_API ma_bool32 ma_encoder__on_seek_stdio(ma_encoder* pEncoder, int byteOffset, ma_seek_origin origin)
+MA_API ma_uint32 ma_sound_get_pinned_listener_index(const ma_sound* pSound)
{
- return fseek((FILE*)pEncoder->pFile, byteOffset, (origin == ma_seek_origin_current) ? SEEK_CUR : SEEK_SET) == 0;
+ if (pSound == NULL) {
+ return MA_LISTENER_INDEX_CLOSEST;
+ }
+
+ return c89atomic_load_explicit_32(&pSound->engineNode.pinnedListenerIndex, c89atomic_memory_order_acquire);
}
-MA_API ma_result ma_encoder_init_file(const char* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder)
+MA_API ma_uint32 ma_sound_get_listener_index(const ma_sound* pSound)
{
- ma_result result;
- FILE* pFile;
+ ma_uint32 listenerIndex;
- result = ma_encoder_preinit(pConfig, pEncoder);
- if (result != MA_SUCCESS) {
- return result;
+ if (pSound == NULL) {
+ return 0;
}
- /* Now open the file. If this fails we don't need to uninitialize the encoder. */
- result = ma_fopen(&pFile, pFilePath, "wb");
- if (pFile == NULL) {
- return result;
+ listenerIndex = ma_sound_get_pinned_listener_index(pSound);
+ if (listenerIndex == MA_LISTENER_INDEX_CLOSEST) {
+ ma_vec3f position = ma_sound_get_position(pSound);
+ return ma_engine_find_closest_listener(ma_sound_get_engine(pSound), position.x, position.y, position.z);
}
- pEncoder->pFile = pFile;
-
- return ma_encoder_init__internal(ma_encoder__on_write_stdio, ma_encoder__on_seek_stdio, NULL, pEncoder);
+ return listenerIndex;
}
-MA_API ma_result ma_encoder_init_file_w(const wchar_t* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder)
+MA_API ma_vec3f ma_sound_get_direction_to_listener(const ma_sound* pSound)
{
- ma_result result;
- FILE* pFile;
+ ma_vec3f relativePos;
+ ma_engine* pEngine;
- result = ma_encoder_preinit(pConfig, pEncoder);
- if (result != MA_SUCCESS) {
- return result;
+ if (pSound == NULL) {
+ return ma_vec3f_init_3f(0, 0, -1);
}
- /* Now open the file. If this fails we don't need to uninitialize the encoder. */
- result = ma_wfopen(&pFile, pFilePath, L"wb", &pEncoder->config.allocationCallbacks);
- if (pFile == NULL) {
- return result;
+ pEngine = ma_sound_get_engine(pSound);
+ if (pEngine == NULL) {
+ return ma_vec3f_init_3f(0, 0, -1);
}
- pEncoder->pFile = pFile;
+ ma_spatializer_get_relative_position_and_direction(&pSound->engineNode.spatializer, &pEngine->listeners[ma_sound_get_listener_index(pSound)], &relativePos, NULL);
- return ma_encoder_init__internal(ma_encoder__on_write_stdio, ma_encoder__on_seek_stdio, NULL, pEncoder);
+ return ma_vec3f_normalize(ma_vec3f_neg(relativePos));
}
-MA_API ma_result ma_encoder_init(ma_encoder_write_proc onWrite, ma_encoder_seek_proc onSeek, void* pUserData, const ma_encoder_config* pConfig, ma_encoder* pEncoder)
+MA_API void ma_sound_set_position(ma_sound* pSound, float x, float y, float z)
{
- ma_result result;
-
- result = ma_encoder_preinit(pConfig, pEncoder);
- if (result != MA_SUCCESS) {
- return result;
+ if (pSound == NULL) {
+ return;
}
- return ma_encoder_init__internal(onWrite, onSeek, pUserData, pEncoder);
+ ma_spatializer_set_position(&pSound->engineNode.spatializer, x, y, z);
}
-
-MA_API void ma_encoder_uninit(ma_encoder* pEncoder)
+MA_API ma_vec3f ma_sound_get_position(const ma_sound* pSound)
{
- if (pEncoder == NULL) {
- return;
- }
-
- if (pEncoder->onUninit) {
- pEncoder->onUninit(pEncoder);
+ if (pSound == NULL) {
+ return ma_vec3f_init_3f(0, 0, 0);
}
- /* If we have a file handle, close it. */
- if (pEncoder->onWrite == ma_encoder__on_write_stdio) {
- fclose((FILE*)pEncoder->pFile);
- }
+ return ma_spatializer_get_position(&pSound->engineNode.spatializer);
}
-
-MA_API ma_uint64 ma_encoder_write_pcm_frames(ma_encoder* pEncoder, const void* pFramesIn, ma_uint64 frameCount)
+MA_API void ma_sound_set_direction(ma_sound* pSound, float x, float y, float z)
{
- if (pEncoder == NULL || pFramesIn == NULL) {
- return 0;
+ if (pSound == NULL) {
+ return;
}
- return pEncoder->onWritePCMFrames(pEncoder, pFramesIn, frameCount);
+ ma_spatializer_set_direction(&pSound->engineNode.spatializer, x, y, z);
}
-#endif /* MA_NO_ENCODING */
+MA_API ma_vec3f ma_sound_get_direction(const ma_sound* pSound)
+{
+ if (pSound == NULL) {
+ return ma_vec3f_init_3f(0, 0, 0);
+ }
+ return ma_spatializer_get_direction(&pSound->engineNode.spatializer);
+}
-/**************************************************************************************************************************************************************
+MA_API void ma_sound_set_velocity(ma_sound* pSound, float x, float y, float z)
+{
+ if (pSound == NULL) {
+ return;
+ }
-Generation
+ ma_spatializer_set_velocity(&pSound->engineNode.spatializer, x, y, z);
+}
-**************************************************************************************************************************************************************/
-#ifndef MA_NO_GENERATION
-MA_API ma_waveform_config ma_waveform_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_waveform_type type, double amplitude, double frequency)
+MA_API ma_vec3f ma_sound_get_velocity(const ma_sound* pSound)
{
- ma_waveform_config config;
-
- MA_ZERO_OBJECT(&config);
- config.format = format;
- config.channels = channels;
- config.sampleRate = sampleRate;
- config.type = type;
- config.amplitude = amplitude;
- config.frequency = frequency;
+ if (pSound == NULL) {
+ return ma_vec3f_init_3f(0, 0, 0);
+ }
- return config;
+ return ma_spatializer_get_velocity(&pSound->engineNode.spatializer);
}
-static ma_result ma_waveform__data_source_on_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+MA_API void ma_sound_set_attenuation_model(ma_sound* pSound, ma_attenuation_model attenuationModel)
{
- ma_uint64 framesRead = ma_waveform_read_pcm_frames((ma_waveform*)pDataSource, pFramesOut, frameCount);
-
- if (pFramesRead != NULL) {
- *pFramesRead = framesRead;
+ if (pSound == NULL) {
+ return;
}
- if (framesRead == 0) {
- return MA_AT_END;
+ ma_spatializer_set_attenuation_model(&pSound->engineNode.spatializer, attenuationModel);
+}
+
+MA_API ma_attenuation_model ma_sound_get_attenuation_model(const ma_sound* pSound)
+{
+ if (pSound == NULL) {
+ return ma_attenuation_model_none;
}
- return MA_SUCCESS;
+ return ma_spatializer_get_attenuation_model(&pSound->engineNode.spatializer);
}
-static ma_result ma_waveform__data_source_on_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
+MA_API void ma_sound_set_positioning(ma_sound* pSound, ma_positioning positioning)
{
- return ma_waveform_seek_to_pcm_frame((ma_waveform*)pDataSource, frameIndex);
+ if (pSound == NULL) {
+ return;
+ }
+
+ ma_spatializer_set_positioning(&pSound->engineNode.spatializer, positioning);
}
-static ma_result ma_waveform__data_source_on_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate)
+MA_API ma_positioning ma_sound_get_positioning(const ma_sound* pSound)
{
- ma_waveform* pWaveform = (ma_waveform*)pDataSource;
-
- *pFormat = pWaveform->config.format;
- *pChannels = pWaveform->config.channels;
- *pSampleRate = pWaveform->config.sampleRate;
+ if (pSound == NULL) {
+ return ma_positioning_absolute;
+ }
- return MA_SUCCESS;
+ return ma_spatializer_get_positioning(&pSound->engineNode.spatializer);
}
-static ma_result ma_waveform__data_source_on_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
+MA_API void ma_sound_set_rolloff(ma_sound* pSound, float rolloff)
{
- ma_waveform* pWaveform = (ma_waveform*)pDataSource;
-
- *pCursor = (ma_uint64)(pWaveform->time / pWaveform->advance);
+ if (pSound == NULL) {
+ return;
+ }
- return MA_SUCCESS;
+ ma_spatializer_set_rolloff(&pSound->engineNode.spatializer, rolloff);
}
-static double ma_waveform__calculate_advance(ma_uint32 sampleRate, double frequency)
+MA_API float ma_sound_get_rolloff(const ma_sound* pSound)
{
- return (1.0 / (sampleRate / frequency));
+ if (pSound == NULL) {
+ return 0;
+ }
+
+ return ma_spatializer_get_rolloff(&pSound->engineNode.spatializer);
}
-static void ma_waveform__update_advance(ma_waveform* pWaveform)
+MA_API void ma_sound_set_min_gain(ma_sound* pSound, float minGain)
{
- pWaveform->advance = ma_waveform__calculate_advance(pWaveform->config.sampleRate, pWaveform->config.frequency);
+ if (pSound == NULL) {
+ return;
+ }
+
+ ma_spatializer_set_min_gain(&pSound->engineNode.spatializer, minGain);
}
-static ma_data_source_vtable g_ma_waveform_data_source_vtable =
+MA_API float ma_sound_get_min_gain(const ma_sound* pSound)
{
- ma_waveform__data_source_on_read,
- ma_waveform__data_source_on_seek,
- NULL, /* onMap */
- NULL, /* onUnmap */
- ma_waveform__data_source_on_get_data_format,
- ma_waveform__data_source_on_get_cursor,
- NULL /* onGetLength. There's no notion of a length in waveforms. */
-};
+ if (pSound == NULL) {
+ return 0;
+ }
-MA_API ma_result ma_waveform_init(const ma_waveform_config* pConfig, ma_waveform* pWaveform)
-{
- ma_result result;
- ma_data_source_config dataSourceConfig;
+ return ma_spatializer_get_min_gain(&pSound->engineNode.spatializer);
+}
- if (pWaveform == NULL) {
- return MA_INVALID_ARGS;
+MA_API void ma_sound_set_max_gain(ma_sound* pSound, float maxGain)
+{
+ if (pSound == NULL) {
+ return;
}
- MA_ZERO_OBJECT(pWaveform);
-
- dataSourceConfig = ma_data_source_config_init();
- dataSourceConfig.vtable = &g_ma_waveform_data_source_vtable;
+ ma_spatializer_set_max_gain(&pSound->engineNode.spatializer, maxGain);
+}
- result = ma_data_source_init(&dataSourceConfig, &pWaveform->ds);
- if (result != MA_SUCCESS) {
- return result;
+MA_API float ma_sound_get_max_gain(const ma_sound* pSound)
+{
+ if (pSound == NULL) {
+ return 0;
}
- pWaveform->config = *pConfig;
- pWaveform->advance = ma_waveform__calculate_advance(pWaveform->config.sampleRate, pWaveform->config.frequency);
- pWaveform->time = 0;
-
- return MA_SUCCESS;
+ return ma_spatializer_get_max_gain(&pSound->engineNode.spatializer);
}
-MA_API void ma_waveform_uninit(ma_waveform* pWaveform)
+MA_API void ma_sound_set_min_distance(ma_sound* pSound, float minDistance)
{
- if (pWaveform == NULL) {
+ if (pSound == NULL) {
return;
}
- ma_data_source_uninit(&pWaveform->ds);
+ ma_spatializer_set_min_distance(&pSound->engineNode.spatializer, minDistance);
}
-MA_API ma_result ma_waveform_set_amplitude(ma_waveform* pWaveform, double amplitude)
+MA_API float ma_sound_get_min_distance(const ma_sound* pSound)
{
- if (pWaveform == NULL) {
- return MA_INVALID_ARGS;
+ if (pSound == NULL) {
+ return 0;
}
- pWaveform->config.amplitude = amplitude;
- return MA_SUCCESS;
+ return ma_spatializer_get_min_distance(&pSound->engineNode.spatializer);
}
-MA_API ma_result ma_waveform_set_frequency(ma_waveform* pWaveform, double frequency)
+MA_API void ma_sound_set_max_distance(ma_sound* pSound, float maxDistance)
{
- if (pWaveform == NULL) {
- return MA_INVALID_ARGS;
+ if (pSound == NULL) {
+ return;
}
- pWaveform->config.frequency = frequency;
- ma_waveform__update_advance(pWaveform);
+ ma_spatializer_set_max_distance(&pSound->engineNode.spatializer, maxDistance);
+}
- return MA_SUCCESS;
+MA_API float ma_sound_get_max_distance(const ma_sound* pSound)
+{
+ if (pSound == NULL) {
+ return 0;
+ }
+
+ return ma_spatializer_get_max_distance(&pSound->engineNode.spatializer);
}
-MA_API ma_result ma_waveform_set_type(ma_waveform* pWaveform, ma_waveform_type type)
+MA_API void ma_sound_set_cone(ma_sound* pSound, float innerAngleInRadians, float outerAngleInRadians, float outerGain)
{
- if (pWaveform == NULL) {
- return MA_INVALID_ARGS;
+ if (pSound == NULL) {
+ return;
}
- pWaveform->config.type = type;
- return MA_SUCCESS;
+ ma_spatializer_set_cone(&pSound->engineNode.spatializer, innerAngleInRadians, outerAngleInRadians, outerGain);
}
-MA_API ma_result ma_waveform_set_sample_rate(ma_waveform* pWaveform, ma_uint32 sampleRate)
+MA_API void ma_sound_get_cone(const ma_sound* pSound, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain)
{
- if (pWaveform == NULL) {
- return MA_INVALID_ARGS;
+ if (pInnerAngleInRadians != NULL) {
+ *pInnerAngleInRadians = 0;
}
- pWaveform->config.sampleRate = sampleRate;
- ma_waveform__update_advance(pWaveform);
+ if (pOuterAngleInRadians != NULL) {
+ *pOuterAngleInRadians = 0;
+ }
- return MA_SUCCESS;
+ if (pOuterGain != NULL) {
+ *pOuterGain = 0;
+ }
+
+ ma_spatializer_get_cone(&pSound->engineNode.spatializer, pInnerAngleInRadians, pOuterAngleInRadians, pOuterGain);
}
-static float ma_waveform_sine_f32(double time, double amplitude)
+MA_API void ma_sound_set_doppler_factor(ma_sound* pSound, float dopplerFactor)
{
- return (float)(ma_sind(MA_TAU_D * time) * amplitude);
+ if (pSound == NULL) {
+ return;
+ }
+
+ ma_spatializer_set_doppler_factor(&pSound->engineNode.spatializer, dopplerFactor);
}
-static ma_int16 ma_waveform_sine_s16(double time, double amplitude)
+MA_API float ma_sound_get_doppler_factor(const ma_sound* pSound)
{
- return ma_pcm_sample_f32_to_s16(ma_waveform_sine_f32(time, amplitude));
+ if (pSound == NULL) {
+ return 0;
+ }
+
+ return ma_spatializer_get_doppler_factor(&pSound->engineNode.spatializer);
}
-static float ma_waveform_square_f32(double time, double amplitude)
+MA_API void ma_sound_set_directional_attenuation_factor(ma_sound* pSound, float directionalAttenuationFactor)
{
- double f = time - (ma_int64)time;
- double r;
-
- if (f < 0.5) {
- r = amplitude;
- } else {
- r = -amplitude;
+ if (pSound == NULL) {
+ return;
}
- return (float)r;
+ ma_spatializer_set_directional_attenuation_factor(&pSound->engineNode.spatializer, directionalAttenuationFactor);
}
-static ma_int16 ma_waveform_square_s16(double time, double amplitude)
+MA_API float ma_sound_get_directional_attenuation_factor(const ma_sound* pSound)
{
- return ma_pcm_sample_f32_to_s16(ma_waveform_square_f32(time, amplitude));
+ if (pSound == NULL) {
+ return 1;
+ }
+
+ return ma_spatializer_get_directional_attenuation_factor(&pSound->engineNode.spatializer);
}
-static float ma_waveform_triangle_f32(double time, double amplitude)
-{
- double f = time - (ma_int64)time;
- double r;
- r = 2 * ma_abs(2 * (f - 0.5)) - 1;
+MA_API void ma_sound_set_fade_in_pcm_frames(ma_sound* pSound, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInFrames)
+{
+ if (pSound == NULL) {
+ return;
+ }
- return (float)(r * amplitude);
+ ma_fader_set_fade(&pSound->engineNode.fader, volumeBeg, volumeEnd, fadeLengthInFrames);
}
-static ma_int16 ma_waveform_triangle_s16(double time, double amplitude)
+MA_API void ma_sound_set_fade_in_milliseconds(ma_sound* pSound, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInMilliseconds)
{
- return ma_pcm_sample_f32_to_s16(ma_waveform_triangle_f32(time, amplitude));
+ if (pSound == NULL) {
+ return;
+ }
+
+ ma_sound_set_fade_in_pcm_frames(pSound, volumeBeg, volumeEnd, (fadeLengthInMilliseconds * pSound->engineNode.fader.config.sampleRate) / 1000);
}
-static float ma_waveform_sawtooth_f32(double time, double amplitude)
+MA_API float ma_sound_get_current_fade_volume(ma_sound* pSound)
{
- double f = time - (ma_int64)time;
- double r;
-
- r = 2 * (f - 0.5);
+ if (pSound == NULL) {
+ return MA_INVALID_ARGS;
+ }
- return (float)(r * amplitude);
+ return ma_fader_get_current_volume(&pSound->engineNode.fader);
}
-static ma_int16 ma_waveform_sawtooth_s16(double time, double amplitude)
+MA_API void ma_sound_set_start_time_in_pcm_frames(ma_sound* pSound, ma_uint64 absoluteGlobalTimeInFrames)
{
- return ma_pcm_sample_f32_to_s16(ma_waveform_sawtooth_f32(time, amplitude));
+ if (pSound == NULL) {
+ return;
+ }
+
+ ma_node_set_state_time(pSound, ma_node_state_started, absoluteGlobalTimeInFrames);
}
-static void ma_waveform_read_pcm_frames__sine(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount)
+MA_API void ma_sound_set_start_time_in_milliseconds(ma_sound* pSound, ma_uint64 absoluteGlobalTimeInMilliseconds)
{
- ma_uint64 iFrame;
- ma_uint64 iChannel;
- ma_uint32 bps = ma_get_bytes_per_sample(pWaveform->config.format);
- ma_uint32 bpf = bps * pWaveform->config.channels;
-
- MA_ASSERT(pWaveform != NULL);
- MA_ASSERT(pFramesOut != NULL);
+ if (pSound == NULL) {
+ return;
+ }
- if (pWaveform->config.format == ma_format_f32) {
- float* pFramesOutF32 = (float*)pFramesOut;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- float s = ma_waveform_sine_f32(pWaveform->time, pWaveform->config.amplitude);
- pWaveform->time += pWaveform->advance;
+ ma_sound_set_start_time_in_pcm_frames(pSound, absoluteGlobalTimeInMilliseconds * ma_engine_get_sample_rate(ma_sound_get_engine(pSound)) / 1000);
+}
- for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
- pFramesOutF32[iFrame*pWaveform->config.channels + iChannel] = s;
- }
- }
- } else if (pWaveform->config.format == ma_format_s16) {
- ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- ma_int16 s = ma_waveform_sine_s16(pWaveform->time, pWaveform->config.amplitude);
- pWaveform->time += pWaveform->advance;
+MA_API void ma_sound_set_stop_time_in_pcm_frames(ma_sound* pSound, ma_uint64 absoluteGlobalTimeInFrames)
+{
+ if (pSound == NULL) {
+ return;
+ }
- for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
- pFramesOutS16[iFrame*pWaveform->config.channels + iChannel] = s;
- }
- }
- } else {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- float s = ma_waveform_sine_f32(pWaveform->time, pWaveform->config.amplitude);
- pWaveform->time += pWaveform->advance;
+ ma_node_set_state_time(pSound, ma_node_state_stopped, absoluteGlobalTimeInFrames);
+}
- for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
- ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pWaveform->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
- }
- }
+MA_API void ma_sound_set_stop_time_in_milliseconds(ma_sound* pSound, ma_uint64 absoluteGlobalTimeInMilliseconds)
+{
+ if (pSound == NULL) {
+ return;
}
+
+ ma_sound_set_stop_time_in_pcm_frames(pSound, absoluteGlobalTimeInMilliseconds * ma_engine_get_sample_rate(ma_sound_get_engine(pSound)) / 1000);
}
-static void ma_waveform_read_pcm_frames__square(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount)
+MA_API ma_bool32 ma_sound_is_playing(const ma_sound* pSound)
{
- ma_uint64 iFrame;
- ma_uint64 iChannel;
- ma_uint32 bps = ma_get_bytes_per_sample(pWaveform->config.format);
- ma_uint32 bpf = bps * pWaveform->config.channels;
+ if (pSound == NULL) {
+ return MA_FALSE;
+ }
- MA_ASSERT(pWaveform != NULL);
- MA_ASSERT(pFramesOut != NULL);
+ return ma_node_get_state_by_time(pSound, ma_engine_get_time(ma_sound_get_engine(pSound))) == ma_node_state_started;
+}
- if (pWaveform->config.format == ma_format_f32) {
- float* pFramesOutF32 = (float*)pFramesOut;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- float s = ma_waveform_square_f32(pWaveform->time, pWaveform->config.amplitude);
- pWaveform->time += pWaveform->advance;
+MA_API ma_uint64 ma_sound_get_time_in_pcm_frames(const ma_sound* pSound)
+{
+ if (pSound == NULL) {
+ return 0;
+ }
- for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
- pFramesOutF32[iFrame*pWaveform->config.channels + iChannel] = s;
- }
- }
- } else if (pWaveform->config.format == ma_format_s16) {
- ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- ma_int16 s = ma_waveform_square_s16(pWaveform->time, pWaveform->config.amplitude);
- pWaveform->time += pWaveform->advance;
+ return ma_node_get_time(pSound);
+}
- for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
- pFramesOutS16[iFrame*pWaveform->config.channels + iChannel] = s;
- }
- }
- } else {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- float s = ma_waveform_square_f32(pWaveform->time, pWaveform->config.amplitude);
- pWaveform->time += pWaveform->advance;
+MA_API void ma_sound_set_looping(ma_sound* pSound, ma_bool32 isLooping)
+{
+ if (pSound == NULL) {
+ return;
+ }
- for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
- ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pWaveform->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
- }
- }
+ /* Looping is only a valid concept if the sound is backed by a data source. */
+ if (pSound->pDataSource == NULL) {
+ return;
}
+
+ /* The looping state needs to be applied to the data source in order for any looping to actually happen. */
+ ma_data_source_set_looping(pSound->pDataSource, isLooping);
}
-static void ma_waveform_read_pcm_frames__triangle(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount)
+MA_API ma_bool32 ma_sound_is_looping(const ma_sound* pSound)
{
- ma_uint64 iFrame;
- ma_uint64 iChannel;
- ma_uint32 bps = ma_get_bytes_per_sample(pWaveform->config.format);
- ma_uint32 bpf = bps * pWaveform->config.channels;
-
- MA_ASSERT(pWaveform != NULL);
- MA_ASSERT(pFramesOut != NULL);
+ if (pSound == NULL) {
+ return MA_FALSE;
+ }
- if (pWaveform->config.format == ma_format_f32) {
- float* pFramesOutF32 = (float*)pFramesOut;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- float s = ma_waveform_triangle_f32(pWaveform->time, pWaveform->config.amplitude);
- pWaveform->time += pWaveform->advance;
+ /* There is no notion of looping for sounds that are not backed by a data source. */
+ if (pSound->pDataSource == NULL) {
+ return MA_FALSE;
+ }
- for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
- pFramesOutF32[iFrame*pWaveform->config.channels + iChannel] = s;
- }
- }
- } else if (pWaveform->config.format == ma_format_s16) {
- ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- ma_int16 s = ma_waveform_triangle_s16(pWaveform->time, pWaveform->config.amplitude);
- pWaveform->time += pWaveform->advance;
+ return ma_data_source_is_looping(pSound->pDataSource);
+}
- for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
- pFramesOutS16[iFrame*pWaveform->config.channels + iChannel] = s;
- }
- }
- } else {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- float s = ma_waveform_triangle_f32(pWaveform->time, pWaveform->config.amplitude);
- pWaveform->time += pWaveform->advance;
+MA_API ma_bool32 ma_sound_at_end(const ma_sound* pSound)
+{
+ if (pSound == NULL) {
+ return MA_FALSE;
+ }
- for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
- ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pWaveform->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
- }
- }
+ /* There is no notion of an end of a sound if it's not backed by a data source. */
+ if (pSound->pDataSource == NULL) {
+ return MA_FALSE;
}
+
+ return c89atomic_load_32(&pSound->atEnd);
}
-static void ma_waveform_read_pcm_frames__sawtooth(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount)
+MA_API ma_result ma_sound_seek_to_pcm_frame(ma_sound* pSound, ma_uint64 frameIndex)
{
- ma_uint64 iFrame;
- ma_uint64 iChannel;
- ma_uint32 bps = ma_get_bytes_per_sample(pWaveform->config.format);
- ma_uint32 bpf = bps * pWaveform->config.channels;
-
- MA_ASSERT(pWaveform != NULL);
- MA_ASSERT(pFramesOut != NULL);
-
- if (pWaveform->config.format == ma_format_f32) {
- float* pFramesOutF32 = (float*)pFramesOut;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- float s = ma_waveform_sawtooth_f32(pWaveform->time, pWaveform->config.amplitude);
- pWaveform->time += pWaveform->advance;
+ if (pSound == NULL) {
+ return MA_INVALID_ARGS;
+ }
- for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
- pFramesOutF32[iFrame*pWaveform->config.channels + iChannel] = s;
- }
- }
- } else if (pWaveform->config.format == ma_format_s16) {
- ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- ma_int16 s = ma_waveform_sawtooth_s16(pWaveform->time, pWaveform->config.amplitude);
- pWaveform->time += pWaveform->advance;
+ /* Seeking is only valid for sounds that are backed by a data source. */
+ if (pSound->pDataSource == NULL) {
+ return MA_INVALID_OPERATION;
+ }
- for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
- pFramesOutS16[iFrame*pWaveform->config.channels + iChannel] = s;
- }
+ /*
+ Resource manager data sources are thread safe which means we can just seek immediately. However, we cannot guarantee that other data sources are
+ thread safe as well so in that case we'll need to get the mixing thread to seek for us to ensure we don't try seeking at the same time as reading.
+ */
+#ifndef MA_NO_RESOURCE_MANAGER
+ if (pSound->pDataSource == pSound->pResourceManagerDataSource) {
+ ma_result result = ma_resource_manager_data_source_seek_to_pcm_frame(pSound->pResourceManagerDataSource, frameIndex);
+ if (result != MA_SUCCESS) {
+ return result;
}
- } else {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- float s = ma_waveform_sawtooth_f32(pWaveform->time, pWaveform->config.amplitude);
- pWaveform->time += pWaveform->advance;
- for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
- ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pWaveform->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
- }
- }
+ /* Time dependant effects need to have their timers updated. */
+ return ma_node_set_time(&pSound->engineNode, frameIndex);
}
+#endif
+
+ /* Getting here means the data source is not a resource manager data source so we'll need to get the mixing thread to do the seeking for us. */
+ pSound->seekTarget = frameIndex;
+
+ return MA_SUCCESS;
}
-MA_API ma_uint64 ma_waveform_read_pcm_frames(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount)
+MA_API ma_result ma_sound_get_data_format(ma_sound* pSound, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
- if (pWaveform == NULL) {
- return 0;
+ if (pSound == NULL) {
+ return MA_INVALID_ARGS;
}
- if (pFramesOut != NULL) {
- switch (pWaveform->config.type)
- {
- case ma_waveform_type_sine:
- {
- ma_waveform_read_pcm_frames__sine(pWaveform, pFramesOut, frameCount);
- } break;
+ /* The data format is retrieved directly from the data source if the sound is backed by one. Otherwise we pull it from the node. */
+ if (pSound->pDataSource == NULL) {
+ ma_uint32 channels;
- case ma_waveform_type_square:
- {
- ma_waveform_read_pcm_frames__square(pWaveform, pFramesOut, frameCount);
- } break;
+ if (pFormat != NULL) {
+ *pFormat = ma_format_f32;
+ }
- case ma_waveform_type_triangle:
- {
- ma_waveform_read_pcm_frames__triangle(pWaveform, pFramesOut, frameCount);
- } break;
+ channels = ma_node_get_input_channels(&pSound->engineNode, 0);
+ if (pChannels != NULL) {
+ *pChannels = channels;
+ }
- case ma_waveform_type_sawtooth:
- {
- ma_waveform_read_pcm_frames__sawtooth(pWaveform, pFramesOut, frameCount);
- } break;
+ if (pSampleRate != NULL) {
+ *pSampleRate = pSound->engineNode.resampler.config.sampleRateIn;
+ }
- default: return 0;
+ if (pChannelMap != NULL) {
+ ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, channels);
}
+
+ return MA_SUCCESS;
} else {
- pWaveform->time += pWaveform->advance * (ma_int64)frameCount; /* Cast to int64 required for VC6. Won't affect anything in practice. */
+ return ma_data_source_get_data_format(pSound->pDataSource, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
}
-
- return frameCount;
}
-MA_API ma_result ma_waveform_seek_to_pcm_frame(ma_waveform* pWaveform, ma_uint64 frameIndex)
+MA_API ma_result ma_sound_get_cursor_in_pcm_frames(ma_sound* pSound, ma_uint64* pCursor)
{
- if (pWaveform == NULL) {
+ if (pSound == NULL) {
return MA_INVALID_ARGS;
}
- pWaveform->time = pWaveform->advance * (ma_int64)frameIndex; /* Casting for VC6. Won't be an issue in practice. */
+ /* The notion of a cursor is only valid for sounds that are backed by a data source. */
+ if (pSound->pDataSource == NULL) {
+ return MA_INVALID_OPERATION;
+ }
- return MA_SUCCESS;
+ return ma_data_source_get_cursor_in_pcm_frames(pSound->pDataSource, pCursor);
}
-
-MA_API ma_noise_config ma_noise_config_init(ma_format format, ma_uint32 channels, ma_noise_type type, ma_int32 seed, double amplitude)
+MA_API ma_result ma_sound_get_length_in_pcm_frames(ma_sound* pSound, ma_uint64* pLength)
{
- ma_noise_config config;
- MA_ZERO_OBJECT(&config);
-
- config.format = format;
- config.channels = channels;
- config.type = type;
- config.seed = seed;
- config.amplitude = amplitude;
+ if (pSound == NULL) {
+ return MA_INVALID_ARGS;
+ }
- if (config.seed == 0) {
- config.seed = MA_DEFAULT_LCG_SEED;
+ /* The notion of a sound length is only valid for sounds that are backed by a data source. */
+ if (pSound->pDataSource == NULL) {
+ return MA_INVALID_OPERATION;
}
- return config;
+ return ma_data_source_get_length_in_pcm_frames(pSound->pDataSource, pLength);
}
-
-static ma_result ma_noise__data_source_on_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+MA_API ma_result ma_sound_get_cursor_in_seconds(ma_sound* pSound, float* pCursor)
{
- ma_uint64 framesRead = ma_noise_read_pcm_frames((ma_noise*)pDataSource, pFramesOut, frameCount);
-
- if (pFramesRead != NULL) {
- *pFramesRead = framesRead;
+ if (pSound == NULL) {
+ return MA_INVALID_ARGS;
}
- if (framesRead == 0) {
- return MA_AT_END;
+ /* The notion of a cursor is only valid for sounds that are backed by a data source. */
+ if (pSound->pDataSource == NULL) {
+ return MA_INVALID_OPERATION;
}
- return MA_SUCCESS;
-}
-
-static ma_result ma_noise__data_source_on_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
-{
- /* No-op. Just pretend to be successful. */
- (void)pDataSource;
- (void)frameIndex;
- return MA_SUCCESS;
+ return ma_data_source_get_cursor_in_seconds(pSound->pDataSource, pCursor);
}
-static ma_result ma_noise__data_source_on_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate)
+MA_API ma_result ma_sound_get_length_in_seconds(ma_sound* pSound, float* pLength)
{
- ma_noise* pNoise = (ma_noise*)pDataSource;
+ if (pSound == NULL) {
+ return MA_INVALID_ARGS;
+ }
- *pFormat = pNoise->config.format;
- *pChannels = pNoise->config.channels;
- *pSampleRate = 0; /* There is no notion of sample rate with noise generation. */
+ /* The notion of a sound length is only valid for sounds that are backed by a data source. */
+ if (pSound->pDataSource == NULL) {
+ return MA_INVALID_OPERATION;
+ }
- return MA_SUCCESS;
+ return ma_data_source_get_length_in_seconds(pSound->pDataSource, pLength);
}
-static ma_data_source_vtable g_ma_noise_data_source_vtable =
+
+MA_API ma_result ma_sound_group_init(ma_engine* pEngine, ma_uint32 flags, ma_sound_group* pParentGroup, ma_sound_group* pGroup)
{
- ma_noise__data_source_on_read,
- ma_noise__data_source_on_seek, /* No-op for noise. */
- NULL, /* onMap */
- NULL, /* onUnmap */
- ma_noise__data_source_on_get_data_format,
- NULL, /* onGetCursor. No notion of a cursor for noise. */
- NULL /* onGetLength. No notion of a length for noise. */
-};
+ ma_sound_group_config config = ma_sound_group_config_init();
+ config.flags = flags;
+ config.pInitialAttachment = pParentGroup;
+ return ma_sound_group_init_ex(pEngine, &config, pGroup);
+}
-MA_API ma_result ma_noise_init(const ma_noise_config* pConfig, ma_noise* pNoise)
+MA_API ma_result ma_sound_group_init_ex(ma_engine* pEngine, const ma_sound_group_config* pConfig, ma_sound_group* pGroup)
{
- ma_result result;
- ma_data_source_config dataSourceConfig;
+ ma_sound_config soundConfig;
- if (pNoise == NULL) {
+ if (pGroup == NULL) {
return MA_INVALID_ARGS;
}
- MA_ZERO_OBJECT(pNoise);
+ MA_ZERO_OBJECT(pGroup);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
- if (pConfig->channels < MA_MIN_CHANNELS || pConfig->channels > MA_MAX_CHANNELS) {
- return MA_INVALID_ARGS;
- }
+ /* A sound group is just a sound without a data source. */
+ soundConfig = *pConfig;
+ soundConfig.pFilePath = NULL;
+ soundConfig.pFilePathW = NULL;
+ soundConfig.pDataSource = NULL;
- dataSourceConfig = ma_data_source_config_init();
- dataSourceConfig.vtable = &g_ma_noise_data_source_vtable;
+ /*
+ Groups need to have spatialization disabled by default because I think it'll be pretty rare
+ that programs will want to spatialize groups (but not unheard of). Certainly it feels like
+ disabling this by default feels like the right option. Spatialization can be enabled with a
+ call to ma_sound_group_set_spatialization_enabled().
+ */
+ soundConfig.flags |= MA_SOUND_FLAG_NO_SPATIALIZATION;
- result = ma_data_source_init(&dataSourceConfig, &pNoise->ds);
- if (result != MA_SUCCESS) {
- return result;
- }
+ return ma_sound_init_ex(pEngine, &soundConfig, pGroup);
+}
- pNoise->config = *pConfig;
- ma_lcg_seed(&pNoise->lcg, pConfig->seed);
+MA_API void ma_sound_group_uninit(ma_sound_group* pGroup)
+{
+ ma_sound_uninit(pGroup);
+}
- if (pNoise->config.type == ma_noise_type_pink) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < pConfig->channels; iChannel += 1) {
- pNoise->state.pink.accumulation[iChannel] = 0;
- pNoise->state.pink.counter[iChannel] = 1;
- }
- }
+MA_API ma_engine* ma_sound_group_get_engine(const ma_sound_group* pGroup)
+{
+ return ma_sound_get_engine(pGroup);
+}
- if (pNoise->config.type == ma_noise_type_brownian) {
- ma_uint32 iChannel;
- for (iChannel = 0; iChannel < pConfig->channels; iChannel += 1) {
- pNoise->state.brownian.accumulation[iChannel] = 0;
- }
- }
+MA_API ma_result ma_sound_group_start(ma_sound_group* pGroup)
+{
+ return ma_sound_start(pGroup);
+}
- return MA_SUCCESS;
+MA_API ma_result ma_sound_group_stop(ma_sound_group* pGroup)
+{
+ return ma_sound_stop(pGroup);
}
-MA_API void ma_noise_uninit(ma_noise* pNoise)
+MA_API void ma_sound_group_set_volume(ma_sound_group* pGroup, float volume)
{
- if (pNoise == NULL) {
- return;
- }
+ ma_sound_set_volume(pGroup, volume);
+}
- ma_data_source_uninit(&pNoise->ds);
+MA_API float ma_sound_group_get_volume(const ma_sound_group* pGroup)
+{
+ return ma_sound_get_volume(pGroup);
}
-MA_API ma_result ma_noise_set_amplitude(ma_noise* pNoise, double amplitude)
+MA_API void ma_sound_group_set_pan(ma_sound_group* pGroup, float pan)
{
- if (pNoise == NULL) {
- return MA_INVALID_ARGS;
- }
+ ma_sound_set_pan(pGroup, pan);
+}
- pNoise->config.amplitude = amplitude;
- return MA_SUCCESS;
+MA_API float ma_sound_group_get_pan(const ma_sound_group* pGroup)
+{
+ return ma_sound_get_pan(pGroup);
}
-MA_API ma_result ma_noise_set_seed(ma_noise* pNoise, ma_int32 seed)
+MA_API void ma_sound_group_set_pan_mode(ma_sound_group* pGroup, ma_pan_mode panMode)
{
- if (pNoise == NULL) {
- return MA_INVALID_ARGS;
- }
+ ma_sound_set_pan_mode(pGroup, panMode);
+}
- pNoise->lcg.state = seed;
- return MA_SUCCESS;
+MA_API ma_pan_mode ma_sound_group_get_pan_mode(const ma_sound_group* pGroup)
+{
+ return ma_sound_get_pan_mode(pGroup);
}
+MA_API void ma_sound_group_set_pitch(ma_sound_group* pGroup, float pitch)
+{
+ ma_sound_set_pitch(pGroup, pitch);
+}
-MA_API ma_result ma_noise_set_type(ma_noise* pNoise, ma_noise_type type)
+MA_API float ma_sound_group_get_pitch(const ma_sound_group* pGroup)
{
- if (pNoise == NULL) {
- return MA_INVALID_ARGS;
- }
+ return ma_sound_get_pitch(pGroup);
+}
- pNoise->config.type = type;
- return MA_SUCCESS;
+MA_API void ma_sound_group_set_spatialization_enabled(ma_sound_group* pGroup, ma_bool32 enabled)
+{
+ ma_sound_set_spatialization_enabled(pGroup, enabled);
}
-static MA_INLINE float ma_noise_f32_white(ma_noise* pNoise)
+MA_API ma_bool32 ma_sound_group_is_spatialization_enabled(const ma_sound_group* pGroup)
{
- return (float)(ma_lcg_rand_f64(&pNoise->lcg) * pNoise->config.amplitude);
+ return ma_sound_is_spatialization_enabled(pGroup);
}
-static MA_INLINE ma_int16 ma_noise_s16_white(ma_noise* pNoise)
+MA_API void ma_sound_group_set_pinned_listener_index(ma_sound_group* pGroup, ma_uint32 listenerIndex)
{
- return ma_pcm_sample_f32_to_s16(ma_noise_f32_white(pNoise));
+ ma_sound_set_pinned_listener_index(pGroup, listenerIndex);
}
-static MA_INLINE ma_uint64 ma_noise_read_pcm_frames__white(ma_noise* pNoise, void* pFramesOut, ma_uint64 frameCount)
+MA_API ma_uint32 ma_sound_group_get_pinned_listener_index(const ma_sound_group* pGroup)
{
- ma_uint64 iFrame;
- ma_uint32 iChannel;
- const ma_uint32 channels = pNoise->config.channels;
- MA_ASSUME(channels >= MA_MIN_CHANNELS && channels <= MA_MAX_CHANNELS);
+ return ma_sound_get_pinned_listener_index(pGroup);
+}
- if (pNoise->config.format == ma_format_f32) {
- float* pFramesOutF32 = (float*)pFramesOut;
- if (pNoise->config.duplicateChannels) {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- float s = ma_noise_f32_white(pNoise);
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- pFramesOutF32[iFrame*channels + iChannel] = s;
- }
- }
- } else {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- pFramesOutF32[iFrame*channels + iChannel] = ma_noise_f32_white(pNoise);
- }
- }
- }
- } else if (pNoise->config.format == ma_format_s16) {
- ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
- if (pNoise->config.duplicateChannels) {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- ma_int16 s = ma_noise_s16_white(pNoise);
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- pFramesOutS16[iFrame*channels + iChannel] = s;
- }
- }
- } else {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- pFramesOutS16[iFrame*channels + iChannel] = ma_noise_s16_white(pNoise);
- }
- }
- }
- } else {
- const ma_uint32 bps = ma_get_bytes_per_sample(pNoise->config.format);
- const ma_uint32 bpf = bps * channels;
+MA_API ma_uint32 ma_sound_group_get_listener_index(const ma_sound_group* pGroup)
+{
+ return ma_sound_get_listener_index(pGroup);
+}
- if (pNoise->config.duplicateChannels) {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- float s = ma_noise_f32_white(pNoise);
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pNoise->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
- }
- }
- } else {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- float s = ma_noise_f32_white(pNoise);
- ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pNoise->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
- }
- }
- }
- }
+MA_API ma_vec3f ma_sound_group_get_direction_to_listener(const ma_sound_group* pGroup)
+{
+ return ma_sound_get_direction_to_listener(pGroup);
+}
- return frameCount;
+MA_API void ma_sound_group_set_position(ma_sound_group* pGroup, float x, float y, float z)
+{
+ ma_sound_set_position(pGroup, x, y, z);
}
+MA_API ma_vec3f ma_sound_group_get_position(const ma_sound_group* pGroup)
+{
+ return ma_sound_get_position(pGroup);
+}
-static MA_INLINE unsigned int ma_tzcnt32(unsigned int x)
+MA_API void ma_sound_group_set_direction(ma_sound_group* pGroup, float x, float y, float z)
{
- unsigned int n;
+ ma_sound_set_direction(pGroup, x, y, z);
+}
- /* Special case for odd numbers since they should happen about half the time. */
- if (x & 0x1) {
- return 0;
- }
+MA_API ma_vec3f ma_sound_group_get_direction(const ma_sound_group* pGroup)
+{
+ return ma_sound_get_direction(pGroup);
+}
- if (x == 0) {
- return sizeof(x) << 3;
- }
+MA_API void ma_sound_group_set_velocity(ma_sound_group* pGroup, float x, float y, float z)
+{
+ ma_sound_set_velocity(pGroup, x, y, z);
+}
- n = 1;
- if ((x & 0x0000FFFF) == 0) { x >>= 16; n += 16; }
- if ((x & 0x000000FF) == 0) { x >>= 8; n += 8; }
- if ((x & 0x0000000F) == 0) { x >>= 4; n += 4; }
- if ((x & 0x00000003) == 0) { x >>= 2; n += 2; }
- n -= x & 0x00000001;
+MA_API ma_vec3f ma_sound_group_get_velocity(const ma_sound_group* pGroup)
+{
+ return ma_sound_get_velocity(pGroup);
+}
- return n;
+MA_API void ma_sound_group_set_attenuation_model(ma_sound_group* pGroup, ma_attenuation_model attenuationModel)
+{
+ ma_sound_set_attenuation_model(pGroup, attenuationModel);
}
-/*
-Pink noise generation based on Tonic (public domain) with modifications. https://github.com/TonicAudio/Tonic/blob/master/src/Tonic/Noise.h
+MA_API ma_attenuation_model ma_sound_group_get_attenuation_model(const ma_sound_group* pGroup)
+{
+ return ma_sound_get_attenuation_model(pGroup);
+}
-This is basically _the_ reference for pink noise from what I've found: http://www.firstpr.com.au/dsp/pink-noise/
-*/
-static MA_INLINE float ma_noise_f32_pink(ma_noise* pNoise, ma_uint32 iChannel)
+MA_API void ma_sound_group_set_positioning(ma_sound_group* pGroup, ma_positioning positioning)
{
- double result;
- double binPrev;
- double binNext;
- unsigned int ibin;
+ ma_sound_set_positioning(pGroup, positioning);
+}
- ibin = ma_tzcnt32(pNoise->state.pink.counter[iChannel]) & (ma_countof(pNoise->state.pink.bin[0]) - 1);
+MA_API ma_positioning ma_sound_group_get_positioning(const ma_sound_group* pGroup)
+{
+ return ma_sound_get_positioning(pGroup);
+}
- binPrev = pNoise->state.pink.bin[iChannel][ibin];
- binNext = ma_lcg_rand_f64(&pNoise->lcg);
- pNoise->state.pink.bin[iChannel][ibin] = binNext;
+MA_API void ma_sound_group_set_rolloff(ma_sound_group* pGroup, float rolloff)
+{
+ ma_sound_set_rolloff(pGroup, rolloff);
+}
- pNoise->state.pink.accumulation[iChannel] += (binNext - binPrev);
- pNoise->state.pink.counter[iChannel] += 1;
+MA_API float ma_sound_group_get_rolloff(const ma_sound_group* pGroup)
+{
+ return ma_sound_get_rolloff(pGroup);
+}
- result = (ma_lcg_rand_f64(&pNoise->lcg) + pNoise->state.pink.accumulation[iChannel]);
- result /= 10;
+MA_API void ma_sound_group_set_min_gain(ma_sound_group* pGroup, float minGain)
+{
+ ma_sound_set_min_gain(pGroup, minGain);
+}
- return (float)(result * pNoise->config.amplitude);
+MA_API float ma_sound_group_get_min_gain(const ma_sound_group* pGroup)
+{
+ return ma_sound_get_min_gain(pGroup);
}
-static MA_INLINE ma_int16 ma_noise_s16_pink(ma_noise* pNoise, ma_uint32 iChannel)
+MA_API void ma_sound_group_set_max_gain(ma_sound_group* pGroup, float maxGain)
{
- return ma_pcm_sample_f32_to_s16(ma_noise_f32_pink(pNoise, iChannel));
+ ma_sound_set_max_gain(pGroup, maxGain);
}
-static MA_INLINE ma_uint64 ma_noise_read_pcm_frames__pink(ma_noise* pNoise, void* pFramesOut, ma_uint64 frameCount)
+MA_API float ma_sound_group_get_max_gain(const ma_sound_group* pGroup)
{
- ma_uint64 iFrame;
- ma_uint32 iChannel;
- const ma_uint32 channels = pNoise->config.channels;
- MA_ASSUME(channels >= MA_MIN_CHANNELS && channels <= MA_MAX_CHANNELS);
+ return ma_sound_get_max_gain(pGroup);
+}
- if (pNoise->config.format == ma_format_f32) {
- float* pFramesOutF32 = (float*)pFramesOut;
- if (pNoise->config.duplicateChannels) {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- float s = ma_noise_f32_pink(pNoise, 0);
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- pFramesOutF32[iFrame*channels + iChannel] = s;
- }
- }
- } else {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- pFramesOutF32[iFrame*channels + iChannel] = ma_noise_f32_pink(pNoise, iChannel);
- }
- }
- }
- } else if (pNoise->config.format == ma_format_s16) {
- ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
- if (pNoise->config.duplicateChannels) {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- ma_int16 s = ma_noise_s16_pink(pNoise, 0);
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- pFramesOutS16[iFrame*channels + iChannel] = s;
- }
- }
- } else {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- pFramesOutS16[iFrame*channels + iChannel] = ma_noise_s16_pink(pNoise, iChannel);
- }
- }
- }
- } else {
- const ma_uint32 bps = ma_get_bytes_per_sample(pNoise->config.format);
- const ma_uint32 bpf = bps * channels;
+MA_API void ma_sound_group_set_min_distance(ma_sound_group* pGroup, float minDistance)
+{
+ ma_sound_set_min_distance(pGroup, minDistance);
+}
- if (pNoise->config.duplicateChannels) {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- float s = ma_noise_f32_pink(pNoise, 0);
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pNoise->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
- }
- }
- } else {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- float s = ma_noise_f32_pink(pNoise, iChannel);
- ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pNoise->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
- }
- }
- }
- }
+MA_API float ma_sound_group_get_min_distance(const ma_sound_group* pGroup)
+{
+ return ma_sound_get_min_distance(pGroup);
+}
- return frameCount;
+MA_API void ma_sound_group_set_max_distance(ma_sound_group* pGroup, float maxDistance)
+{
+ ma_sound_set_max_distance(pGroup, maxDistance);
}
+MA_API float ma_sound_group_get_max_distance(const ma_sound_group* pGroup)
+{
+ return ma_sound_get_max_distance(pGroup);
+}
-static MA_INLINE float ma_noise_f32_brownian(ma_noise* pNoise, ma_uint32 iChannel)
+MA_API void ma_sound_group_set_cone(ma_sound_group* pGroup, float innerAngleInRadians, float outerAngleInRadians, float outerGain)
{
- double result;
+ ma_sound_set_cone(pGroup, innerAngleInRadians, outerAngleInRadians, outerGain);
+}
- result = (ma_lcg_rand_f64(&pNoise->lcg) + pNoise->state.brownian.accumulation[iChannel]);
- result /= 1.005; /* Don't escape the -1..1 range on average. */
+MA_API void ma_sound_group_get_cone(const ma_sound_group* pGroup, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain)
+{
+ ma_sound_get_cone(pGroup, pInnerAngleInRadians, pOuterAngleInRadians, pOuterGain);
+}
- pNoise->state.brownian.accumulation[iChannel] = result;
- result /= 20;
+MA_API void ma_sound_group_set_doppler_factor(ma_sound_group* pGroup, float dopplerFactor)
+{
+ ma_sound_set_doppler_factor(pGroup, dopplerFactor);
+}
- return (float)(result * pNoise->config.amplitude);
+MA_API float ma_sound_group_get_doppler_factor(const ma_sound_group* pGroup)
+{
+ return ma_sound_get_doppler_factor(pGroup);
}
-static MA_INLINE ma_int16 ma_noise_s16_brownian(ma_noise* pNoise, ma_uint32 iChannel)
+MA_API void ma_sound_group_set_directional_attenuation_factor(ma_sound_group* pGroup, float directionalAttenuationFactor)
{
- return ma_pcm_sample_f32_to_s16(ma_noise_f32_brownian(pNoise, iChannel));
+ ma_sound_set_directional_attenuation_factor(pGroup, directionalAttenuationFactor);
}
-static MA_INLINE ma_uint64 ma_noise_read_pcm_frames__brownian(ma_noise* pNoise, void* pFramesOut, ma_uint64 frameCount)
+MA_API float ma_sound_group_get_directional_attenuation_factor(const ma_sound_group* pGroup)
{
- ma_uint64 iFrame;
- ma_uint32 iChannel;
- const ma_uint32 channels = pNoise->config.channels;
- MA_ASSUME(channels >= MA_MIN_CHANNELS && channels <= MA_MAX_CHANNELS);
+ return ma_sound_get_directional_attenuation_factor(pGroup);
+}
- if (pNoise->config.format == ma_format_f32) {
- float* pFramesOutF32 = (float*)pFramesOut;
- if (pNoise->config.duplicateChannels) {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- float s = ma_noise_f32_brownian(pNoise, 0);
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- pFramesOutF32[iFrame*channels + iChannel] = s;
- }
- }
- } else {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- pFramesOutF32[iFrame*channels + iChannel] = ma_noise_f32_brownian(pNoise, iChannel);
- }
- }
- }
- } else if (pNoise->config.format == ma_format_s16) {
- ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
- if (pNoise->config.duplicateChannels) {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- ma_int16 s = ma_noise_s16_brownian(pNoise, 0);
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- pFramesOutS16[iFrame*channels + iChannel] = s;
- }
- }
- } else {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- pFramesOutS16[iFrame*channels + iChannel] = ma_noise_s16_brownian(pNoise, iChannel);
- }
- }
- }
- } else {
- const ma_uint32 bps = ma_get_bytes_per_sample(pNoise->config.format);
- const ma_uint32 bpf = bps * channels;
+MA_API void ma_sound_group_set_fade_in_pcm_frames(ma_sound_group* pGroup, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInFrames)
+{
+ ma_sound_set_fade_in_pcm_frames(pGroup, volumeBeg, volumeEnd, fadeLengthInFrames);
+}
- if (pNoise->config.duplicateChannels) {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- float s = ma_noise_f32_brownian(pNoise, 0);
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pNoise->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
- }
- }
- } else {
- for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
- for (iChannel = 0; iChannel < channels; iChannel += 1) {
- float s = ma_noise_f32_brownian(pNoise, iChannel);
- ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pNoise->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
- }
- }
- }
- }
+MA_API void ma_sound_group_set_fade_in_milliseconds(ma_sound_group* pGroup, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInMilliseconds)
+{
+ ma_sound_set_fade_in_milliseconds(pGroup, volumeBeg, volumeEnd, fadeLengthInMilliseconds);
+}
- return frameCount;
+MA_API float ma_sound_group_get_current_fade_volume(ma_sound_group* pGroup)
+{
+ return ma_sound_get_current_fade_volume(pGroup);
}
-MA_API ma_uint64 ma_noise_read_pcm_frames(ma_noise* pNoise, void* pFramesOut, ma_uint64 frameCount)
+MA_API void ma_sound_group_set_start_time_in_pcm_frames(ma_sound_group* pGroup, ma_uint64 absoluteGlobalTimeInFrames)
{
- if (pNoise == NULL) {
- return 0;
- }
+ ma_sound_set_start_time_in_pcm_frames(pGroup, absoluteGlobalTimeInFrames);
+}
- /* The output buffer is allowed to be NULL. Since we aren't tracking cursors or anything we can just do nothing and pretend to be successful. */
- if (pFramesOut == NULL) {
- return frameCount;
- }
+MA_API void ma_sound_group_set_start_time_in_milliseconds(ma_sound_group* pGroup, ma_uint64 absoluteGlobalTimeInMilliseconds)
+{
+ ma_sound_set_start_time_in_milliseconds(pGroup, absoluteGlobalTimeInMilliseconds);
+}
- if (pNoise->config.type == ma_noise_type_white) {
- return ma_noise_read_pcm_frames__white(pNoise, pFramesOut, frameCount);
- }
+MA_API void ma_sound_group_set_stop_time_in_pcm_frames(ma_sound_group* pGroup, ma_uint64 absoluteGlobalTimeInFrames)
+{
+ ma_sound_set_stop_time_in_pcm_frames(pGroup, absoluteGlobalTimeInFrames);
+}
- if (pNoise->config.type == ma_noise_type_pink) {
- return ma_noise_read_pcm_frames__pink(pNoise, pFramesOut, frameCount);
- }
+MA_API void ma_sound_group_set_stop_time_in_milliseconds(ma_sound_group* pGroup, ma_uint64 absoluteGlobalTimeInMilliseconds)
+{
+ ma_sound_set_stop_time_in_milliseconds(pGroup, absoluteGlobalTimeInMilliseconds);
+}
- if (pNoise->config.type == ma_noise_type_brownian) {
- return ma_noise_read_pcm_frames__brownian(pNoise, pFramesOut, frameCount);
- }
+MA_API ma_bool32 ma_sound_group_is_playing(const ma_sound_group* pGroup)
+{
+ return ma_sound_is_playing(pGroup);
+}
- /* Should never get here. */
- MA_ASSERT(MA_FALSE);
- return 0;
+MA_API ma_uint64 ma_sound_group_get_time_in_pcm_frames(const ma_sound_group* pGroup)
+{
+ return ma_sound_get_time_in_pcm_frames(pGroup);
}
-#endif /* MA_NO_GENERATION */
+#endif /* MA_NO_ENGINE */
@@ -52852,6 +73611,7 @@ code below please report the bug to the respective repository for the relevant p
#define drwav_min(a, b) (((a) < (b)) ? (a) : (b))
#define drwav_max(a, b) (((a) > (b)) ? (a) : (b))
#define drwav_clamp(x, lo, hi) (drwav_max((lo), drwav_min((hi), (x))))
+#define drwav_offset_ptr(p, offset) (((drwav_uint8*)(p)) + (offset))
#define DRWAV_MAX_SIMD_VECTOR_SIZE 64
#if defined(__x86_64__) || defined(_M_X64)
#define DRWAV_X64
@@ -53095,6 +73855,9 @@ static DRWAV_INLINE void drwav__bswap_samples_pcm(void* pSamples, drwav_uint64 s
{
switch (bytesPerSample)
{
+ case 1:
+ {
+ } break;
case 2:
{
drwav__bswap_samples_s16((drwav_int16*)pSamples, sampleCount);
@@ -53505,12 +74268,13 @@ DRWAV_PRIVATE size_t drwav__metadata_parser_read(drwav__metadata_parser* pParser
return pParser->onRead(pParser->pReadSeekUserData, pBufferOut, bytesToRead);
}
}
-DRWAV_PRIVATE drwav_uint64 drwav__read_smpl_to_metadata_obj(drwav__metadata_parser* pParser, drwav_metadata* pMetadata)
+DRWAV_PRIVATE drwav_uint64 drwav__read_smpl_to_metadata_obj(drwav__metadata_parser* pParser, const drwav_chunk_header* pChunkHeader, drwav_metadata* pMetadata)
{
drwav_uint8 smplHeaderData[DRWAV_SMPL_BYTES];
drwav_uint64 totalBytesRead = 0;
size_t bytesJustRead = drwav__metadata_parser_read(pParser, smplHeaderData, sizeof(smplHeaderData), &totalBytesRead);
DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read);
+ DRWAV_ASSERT(pChunkHeader != NULL);
if (bytesJustRead == sizeof(smplHeaderData)) {
drwav_uint32 iSampleLoop;
pMetadata->type = drwav_metadata_type_smpl;
@@ -53523,30 +74287,32 @@ DRWAV_PRIVATE drwav_uint64 drwav__read_smpl_to_metadata_obj(drwav__metadata_pars
pMetadata->data.smpl.smpteOffset = drwav_bytes_to_u32(smplHeaderData + 24);
pMetadata->data.smpl.sampleLoopCount = drwav_bytes_to_u32(smplHeaderData + 28);
pMetadata->data.smpl.samplerSpecificDataSizeInBytes = drwav_bytes_to_u32(smplHeaderData + 32);
- pMetadata->data.smpl.pLoops = (drwav_smpl_loop*)drwav__metadata_get_memory(pParser, sizeof(drwav_smpl_loop) * pMetadata->data.smpl.sampleLoopCount, DRWAV_METADATA_ALIGNMENT);
- for (iSampleLoop = 0; iSampleLoop < pMetadata->data.smpl.sampleLoopCount; ++iSampleLoop) {
- drwav_uint8 smplLoopData[DRWAV_SMPL_LOOP_BYTES];
- bytesJustRead = drwav__metadata_parser_read(pParser, smplLoopData, sizeof(smplLoopData), &totalBytesRead);
- if (bytesJustRead == sizeof(smplLoopData)) {
- pMetadata->data.smpl.pLoops[iSampleLoop].cuePointId = drwav_bytes_to_u32(smplLoopData + 0);
- pMetadata->data.smpl.pLoops[iSampleLoop].type = drwav_bytes_to_u32(smplLoopData + 4);
- pMetadata->data.smpl.pLoops[iSampleLoop].firstSampleByteOffset = drwav_bytes_to_u32(smplLoopData + 8);
- pMetadata->data.smpl.pLoops[iSampleLoop].lastSampleByteOffset = drwav_bytes_to_u32(smplLoopData + 12);
- pMetadata->data.smpl.pLoops[iSampleLoop].sampleFraction = drwav_bytes_to_u32(smplLoopData + 16);
- pMetadata->data.smpl.pLoops[iSampleLoop].playCount = drwav_bytes_to_u32(smplLoopData + 20);
- } else {
- break;
+ if (pMetadata->data.smpl.sampleLoopCount == (pChunkHeader->sizeInBytes - DRWAV_SMPL_BYTES) / DRWAV_SMPL_LOOP_BYTES) {
+ pMetadata->data.smpl.pLoops = (drwav_smpl_loop*)drwav__metadata_get_memory(pParser, sizeof(drwav_smpl_loop) * pMetadata->data.smpl.sampleLoopCount, DRWAV_METADATA_ALIGNMENT);
+ for (iSampleLoop = 0; iSampleLoop < pMetadata->data.smpl.sampleLoopCount; ++iSampleLoop) {
+ drwav_uint8 smplLoopData[DRWAV_SMPL_LOOP_BYTES];
+ bytesJustRead = drwav__metadata_parser_read(pParser, smplLoopData, sizeof(smplLoopData), &totalBytesRead);
+ if (bytesJustRead == sizeof(smplLoopData)) {
+ pMetadata->data.smpl.pLoops[iSampleLoop].cuePointId = drwav_bytes_to_u32(smplLoopData + 0);
+ pMetadata->data.smpl.pLoops[iSampleLoop].type = drwav_bytes_to_u32(smplLoopData + 4);
+ pMetadata->data.smpl.pLoops[iSampleLoop].firstSampleByteOffset = drwav_bytes_to_u32(smplLoopData + 8);
+ pMetadata->data.smpl.pLoops[iSampleLoop].lastSampleByteOffset = drwav_bytes_to_u32(smplLoopData + 12);
+ pMetadata->data.smpl.pLoops[iSampleLoop].sampleFraction = drwav_bytes_to_u32(smplLoopData + 16);
+ pMetadata->data.smpl.pLoops[iSampleLoop].playCount = drwav_bytes_to_u32(smplLoopData + 20);
+ } else {
+ break;
+ }
+ }
+ if (pMetadata->data.smpl.samplerSpecificDataSizeInBytes > 0) {
+ pMetadata->data.smpl.pSamplerSpecificData = drwav__metadata_get_memory(pParser, pMetadata->data.smpl.samplerSpecificDataSizeInBytes, 1);
+ DRWAV_ASSERT(pMetadata->data.smpl.pSamplerSpecificData != NULL);
+ drwav__metadata_parser_read(pParser, pMetadata->data.smpl.pSamplerSpecificData, pMetadata->data.smpl.samplerSpecificDataSizeInBytes, &totalBytesRead);
}
- }
- if (pMetadata->data.smpl.samplerSpecificDataSizeInBytes > 0) {
- pMetadata->data.smpl.pSamplerSpecificData = drwav__metadata_get_memory(pParser, pMetadata->data.smpl.samplerSpecificDataSizeInBytes, 1);
- DRWAV_ASSERT(pMetadata->data.smpl.pSamplerSpecificData != NULL);
- bytesJustRead = drwav__metadata_parser_read(pParser, pMetadata->data.smpl.pSamplerSpecificData, pMetadata->data.smpl.samplerSpecificDataSizeInBytes, &totalBytesRead);
}
}
return totalBytesRead;
}
-DRWAV_PRIVATE drwav_uint64 drwav__read_cue_to_metadata_obj(drwav__metadata_parser* pParser, drwav_metadata* pMetadata)
+DRWAV_PRIVATE drwav_uint64 drwav__read_cue_to_metadata_obj(drwav__metadata_parser* pParser, const drwav_chunk_header* pChunkHeader, drwav_metadata* pMetadata)
{
drwav_uint8 cueHeaderSectionData[DRWAV_CUE_BYTES];
drwav_uint64 totalBytesRead = 0;
@@ -53555,25 +74321,27 @@ DRWAV_PRIVATE drwav_uint64 drwav__read_cue_to_metadata_obj(drwav__metadata_parse
if (bytesJustRead == sizeof(cueHeaderSectionData)) {
pMetadata->type = drwav_metadata_type_cue;
pMetadata->data.cue.cuePointCount = drwav_bytes_to_u32(cueHeaderSectionData);
- pMetadata->data.cue.pCuePoints = (drwav_cue_point*)drwav__metadata_get_memory(pParser, sizeof(drwav_cue_point) * pMetadata->data.cue.cuePointCount, DRWAV_METADATA_ALIGNMENT);
- DRWAV_ASSERT(pMetadata->data.cue.pCuePoints != NULL);
- if (pMetadata->data.cue.cuePointCount > 0) {
- drwav_uint32 iCuePoint;
- for (iCuePoint = 0; iCuePoint < pMetadata->data.cue.cuePointCount; ++iCuePoint) {
- drwav_uint8 cuePointData[DRWAV_CUE_POINT_BYTES];
- bytesJustRead = drwav__metadata_parser_read(pParser, cuePointData, sizeof(cuePointData), &totalBytesRead);
- if (bytesJustRead == sizeof(cuePointData)) {
- pMetadata->data.cue.pCuePoints[iCuePoint].id = drwav_bytes_to_u32(cuePointData + 0);
- pMetadata->data.cue.pCuePoints[iCuePoint].playOrderPosition = drwav_bytes_to_u32(cuePointData + 4);
- pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId[0] = cuePointData[8];
- pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId[1] = cuePointData[9];
- pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId[2] = cuePointData[10];
- pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId[3] = cuePointData[11];
- pMetadata->data.cue.pCuePoints[iCuePoint].chunkStart = drwav_bytes_to_u32(cuePointData + 12);
- pMetadata->data.cue.pCuePoints[iCuePoint].blockStart = drwav_bytes_to_u32(cuePointData + 16);
- pMetadata->data.cue.pCuePoints[iCuePoint].sampleByteOffset = drwav_bytes_to_u32(cuePointData + 20);
- } else {
- break;
+ if (pMetadata->data.cue.cuePointCount == (pChunkHeader->sizeInBytes - DRWAV_CUE_BYTES) / DRWAV_CUE_POINT_BYTES) {
+ pMetadata->data.cue.pCuePoints = (drwav_cue_point*)drwav__metadata_get_memory(pParser, sizeof(drwav_cue_point) * pMetadata->data.cue.cuePointCount, DRWAV_METADATA_ALIGNMENT);
+ DRWAV_ASSERT(pMetadata->data.cue.pCuePoints != NULL);
+ if (pMetadata->data.cue.cuePointCount > 0) {
+ drwav_uint32 iCuePoint;
+ for (iCuePoint = 0; iCuePoint < pMetadata->data.cue.cuePointCount; ++iCuePoint) {
+ drwav_uint8 cuePointData[DRWAV_CUE_POINT_BYTES];
+ bytesJustRead = drwav__metadata_parser_read(pParser, cuePointData, sizeof(cuePointData), &totalBytesRead);
+ if (bytesJustRead == sizeof(cuePointData)) {
+ pMetadata->data.cue.pCuePoints[iCuePoint].id = drwav_bytes_to_u32(cuePointData + 0);
+ pMetadata->data.cue.pCuePoints[iCuePoint].playOrderPosition = drwav_bytes_to_u32(cuePointData + 4);
+ pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId[0] = cuePointData[8];
+ pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId[1] = cuePointData[9];
+ pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId[2] = cuePointData[10];
+ pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId[3] = cuePointData[11];
+ pMetadata->data.cue.pCuePoints[iCuePoint].chunkStart = drwav_bytes_to_u32(cuePointData + 12);
+ pMetadata->data.cue.pCuePoints[iCuePoint].blockStart = drwav_bytes_to_u32(cuePointData + 16);
+ pMetadata->data.cue.pCuePoints[iCuePoint].sampleByteOffset = drwav_bytes_to_u32(cuePointData + 20);
+ } else {
+ break;
+ }
}
}
}
@@ -53615,7 +74383,7 @@ DRWAV_PRIVATE drwav_uint64 drwav__read_acid_to_metadata_obj(drwav__metadata_pars
}
return bytesRead;
}
-DRWAV_PRIVATE size_t drwav__strlen_clamped(char* str, size_t maxToRead)
+DRWAV_PRIVATE size_t drwav__strlen_clamped(const char* str, size_t maxToRead)
{
size_t result = 0;
while (*str++ && result < maxToRead) {
@@ -53623,7 +74391,7 @@ DRWAV_PRIVATE size_t drwav__strlen_clamped(char* str, size_t maxToRead)
}
return result;
}
-DRWAV_PRIVATE char* drwav__metadata_copy_string(drwav__metadata_parser* pParser, char* str, size_t maxToRead)
+DRWAV_PRIVATE char* drwav__metadata_copy_string(drwav__metadata_parser* pParser, const char* str, size_t maxToRead)
{
size_t len = drwav__strlen_clamped(str, maxToRead);
if (len) {
@@ -53636,58 +74404,134 @@ DRWAV_PRIVATE char* drwav__metadata_copy_string(drwav__metadata_parser* pParser,
return NULL;
}
}
+typedef struct
+{
+ const void* pBuffer;
+ size_t sizeInBytes;
+ size_t cursor;
+} drwav_buffer_reader;
+DRWAV_PRIVATE drwav_result drwav_buffer_reader_init(const void* pBuffer, size_t sizeInBytes, drwav_buffer_reader* pReader)
+{
+ DRWAV_ASSERT(pBuffer != NULL);
+ DRWAV_ASSERT(pReader != NULL);
+ DRWAV_ZERO_OBJECT(pReader);
+ pReader->pBuffer = pBuffer;
+ pReader->sizeInBytes = sizeInBytes;
+ pReader->cursor = 0;
+ return DRWAV_SUCCESS;
+}
+DRWAV_PRIVATE const void* drwav_buffer_reader_ptr(const drwav_buffer_reader* pReader)
+{
+ DRWAV_ASSERT(pReader != NULL);
+ return drwav_offset_ptr(pReader->pBuffer, pReader->cursor);
+}
+DRWAV_PRIVATE drwav_result drwav_buffer_reader_seek(drwav_buffer_reader* pReader, size_t bytesToSeek)
+{
+ DRWAV_ASSERT(pReader != NULL);
+ if (pReader->cursor + bytesToSeek > pReader->sizeInBytes) {
+ return DRWAV_BAD_SEEK;
+ }
+ pReader->cursor += bytesToSeek;
+ return DRWAV_SUCCESS;
+}
+DRWAV_PRIVATE drwav_result drwav_buffer_reader_read(drwav_buffer_reader* pReader, void* pDst, size_t bytesToRead, size_t* pBytesRead)
+{
+ drwav_result result = DRWAV_SUCCESS;
+ size_t bytesRemaining;
+ DRWAV_ASSERT(pReader != NULL);
+ if (pBytesRead != NULL) {
+ *pBytesRead = 0;
+ }
+ bytesRemaining = (pReader->sizeInBytes - pReader->cursor);
+ if (bytesToRead > bytesRemaining) {
+ bytesToRead = bytesRemaining;
+ }
+ if (pDst == NULL) {
+ result = drwav_buffer_reader_seek(pReader, bytesToRead);
+ } else {
+ DRWAV_COPY_MEMORY(pDst, drwav_buffer_reader_ptr(pReader), bytesToRead);
+ pReader->cursor += bytesToRead;
+ }
+ DRWAV_ASSERT(pReader->cursor <= pReader->sizeInBytes);
+ if (result == DRWAV_SUCCESS) {
+ if (pBytesRead != NULL) {
+ *pBytesRead = bytesToRead;
+ }
+ }
+ return DRWAV_SUCCESS;
+}
+DRWAV_PRIVATE drwav_result drwav_buffer_reader_read_u16(drwav_buffer_reader* pReader, drwav_uint16* pDst)
+{
+ drwav_result result;
+ size_t bytesRead;
+ drwav_uint8 data[2];
+ DRWAV_ASSERT(pReader != NULL);
+ DRWAV_ASSERT(pDst != NULL);
+ *pDst = 0;
+ result = drwav_buffer_reader_read(pReader, data, sizeof(*pDst), &bytesRead);
+ if (result != DRWAV_SUCCESS || bytesRead != sizeof(*pDst)) {
+ return result;
+ }
+ *pDst = drwav_bytes_to_u16(data);
+ return DRWAV_SUCCESS;
+}
+DRWAV_PRIVATE drwav_result drwav_buffer_reader_read_u32(drwav_buffer_reader* pReader, drwav_uint32* pDst)
+{
+ drwav_result result;
+ size_t bytesRead;
+ drwav_uint8 data[4];
+ DRWAV_ASSERT(pReader != NULL);
+ DRWAV_ASSERT(pDst != NULL);
+ *pDst = 0;
+ result = drwav_buffer_reader_read(pReader, data, sizeof(*pDst), &bytesRead);
+ if (result != DRWAV_SUCCESS || bytesRead != sizeof(*pDst)) {
+ return result;
+ }
+ *pDst = drwav_bytes_to_u32(data);
+ return DRWAV_SUCCESS;
+}
DRWAV_PRIVATE drwav_uint64 drwav__read_bext_to_metadata_obj(drwav__metadata_parser* pParser, drwav_metadata* pMetadata, drwav_uint64 chunkSize)
{
drwav_uint8 bextData[DRWAV_BEXT_BYTES];
- drwav_uint64 bytesRead = drwav__metadata_parser_read(pParser, bextData, sizeof(bextData), NULL);
+ size_t bytesRead = drwav__metadata_parser_read(pParser, bextData, sizeof(bextData), NULL);
DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read);
if (bytesRead == sizeof(bextData)) {
- drwav_uint8* pReadPointer;
+ drwav_buffer_reader reader;
drwav_uint32 timeReferenceLow;
drwav_uint32 timeReferenceHigh;
size_t extraBytes;
pMetadata->type = drwav_metadata_type_bext;
- pReadPointer = bextData;
- pMetadata->data.bext.pDescription = drwav__metadata_copy_string(pParser, (char*)(pReadPointer), DRWAV_BEXT_DESCRIPTION_BYTES);
- pReadPointer += DRWAV_BEXT_DESCRIPTION_BYTES;
- pMetadata->data.bext.pOriginatorName = drwav__metadata_copy_string(pParser, (char*)(pReadPointer), DRWAV_BEXT_ORIGINATOR_NAME_BYTES);
- pReadPointer += DRWAV_BEXT_ORIGINATOR_NAME_BYTES;
- pMetadata->data.bext.pOriginatorReference = drwav__metadata_copy_string(pParser, (char*)(pReadPointer), DRWAV_BEXT_ORIGINATOR_REF_BYTES);
- pReadPointer += DRWAV_BEXT_ORIGINATOR_REF_BYTES;
- memcpy(pReadPointer, pMetadata->data.bext.pOriginationDate, sizeof(pMetadata->data.bext.pOriginationDate));
- pReadPointer += sizeof(pMetadata->data.bext.pOriginationDate);
- memcpy(pReadPointer, pMetadata->data.bext.pOriginationTime, sizeof(pMetadata->data.bext.pOriginationTime));
- pReadPointer += sizeof(pMetadata->data.bext.pOriginationTime);
- timeReferenceLow = drwav_bytes_to_u32(pReadPointer);
- pReadPointer += sizeof(drwav_uint32);
- timeReferenceHigh = drwav_bytes_to_u32(pReadPointer);
- pReadPointer += sizeof(drwav_uint32);
- pMetadata->data.bext.timeReference = ((drwav_uint64)timeReferenceHigh << 32) + timeReferenceLow;
- pMetadata->data.bext.version = drwav_bytes_to_u16(pReadPointer);
- pReadPointer += sizeof(drwav_uint16);
- pMetadata->data.bext.pUMID = drwav__metadata_get_memory(pParser, DRWAV_BEXT_UMID_BYTES, 1);
- memcpy(pMetadata->data.bext.pUMID, pReadPointer, DRWAV_BEXT_UMID_BYTES);
- pReadPointer += DRWAV_BEXT_UMID_BYTES;
- pMetadata->data.bext.loudnessValue = drwav_bytes_to_u16(pReadPointer);
- pReadPointer += sizeof(drwav_uint16);
- pMetadata->data.bext.loudnessRange = drwav_bytes_to_u16(pReadPointer);
- pReadPointer += sizeof(drwav_uint16);
- pMetadata->data.bext.maxTruePeakLevel = drwav_bytes_to_u16(pReadPointer);
- pReadPointer += sizeof(drwav_uint16);
- pMetadata->data.bext.maxMomentaryLoudness = drwav_bytes_to_u16(pReadPointer);
- pReadPointer += sizeof(drwav_uint16);
- pMetadata->data.bext.maxShortTermLoudness = drwav_bytes_to_u16(pReadPointer);
- pReadPointer += sizeof(drwav_uint16);
- DRWAV_ASSERT((pReadPointer + DRWAV_BEXT_RESERVED_BYTES) == (bextData + DRWAV_BEXT_BYTES));
- extraBytes = (size_t)(chunkSize - DRWAV_BEXT_BYTES);
- if (extraBytes > 0) {
- pMetadata->data.bext.pCodingHistory = (char*)drwav__metadata_get_memory(pParser, extraBytes + 1, 1);
- DRWAV_ASSERT(pMetadata->data.bext.pCodingHistory != NULL);
- bytesRead += drwav__metadata_parser_read(pParser, pMetadata->data.bext.pCodingHistory, extraBytes, NULL);
- pMetadata->data.bext.codingHistorySize = (drwav_uint32)strlen(pMetadata->data.bext.pCodingHistory);
- } else {
- pMetadata->data.bext.pCodingHistory = NULL;
- pMetadata->data.bext.codingHistorySize = 0;
+ if (drwav_buffer_reader_init(bextData, bytesRead, &reader) == DRWAV_SUCCESS) {
+ pMetadata->data.bext.pDescription = drwav__metadata_copy_string(pParser, (const char*)drwav_buffer_reader_ptr(&reader), DRWAV_BEXT_DESCRIPTION_BYTES);
+ drwav_buffer_reader_seek(&reader, DRWAV_BEXT_DESCRIPTION_BYTES);
+ pMetadata->data.bext.pOriginatorName = drwav__metadata_copy_string(pParser, (const char*)drwav_buffer_reader_ptr(&reader), DRWAV_BEXT_ORIGINATOR_NAME_BYTES);
+ drwav_buffer_reader_seek(&reader, DRWAV_BEXT_ORIGINATOR_NAME_BYTES);
+ pMetadata->data.bext.pOriginatorReference = drwav__metadata_copy_string(pParser, (const char*)drwav_buffer_reader_ptr(&reader), DRWAV_BEXT_ORIGINATOR_REF_BYTES);
+ drwav_buffer_reader_seek(&reader, DRWAV_BEXT_ORIGINATOR_REF_BYTES);
+ drwav_buffer_reader_read(&reader, pMetadata->data.bext.pOriginationDate, sizeof(pMetadata->data.bext.pOriginationDate), NULL);
+ drwav_buffer_reader_read(&reader, pMetadata->data.bext.pOriginationTime, sizeof(pMetadata->data.bext.pOriginationTime), NULL);
+ drwav_buffer_reader_read_u32(&reader, &timeReferenceLow);
+ drwav_buffer_reader_read_u32(&reader, &timeReferenceHigh);
+ pMetadata->data.bext.timeReference = ((drwav_uint64)timeReferenceHigh << 32) + timeReferenceLow;
+ drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.version);
+ pMetadata->data.bext.pUMID = drwav__metadata_get_memory(pParser, DRWAV_BEXT_UMID_BYTES, 1);
+ drwav_buffer_reader_read(&reader, pMetadata->data.bext.pUMID, DRWAV_BEXT_UMID_BYTES, NULL);
+ drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.loudnessValue);
+ drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.loudnessRange);
+ drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.maxTruePeakLevel);
+ drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.maxMomentaryLoudness);
+ drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.maxShortTermLoudness);
+ DRWAV_ASSERT((drwav_offset_ptr(drwav_buffer_reader_ptr(&reader), DRWAV_BEXT_RESERVED_BYTES)) == (bextData + DRWAV_BEXT_BYTES));
+ extraBytes = (size_t)(chunkSize - DRWAV_BEXT_BYTES);
+ if (extraBytes > 0) {
+ pMetadata->data.bext.pCodingHistory = (char*)drwav__metadata_get_memory(pParser, extraBytes + 1, 1);
+ DRWAV_ASSERT(pMetadata->data.bext.pCodingHistory != NULL);
+ bytesRead += drwav__metadata_parser_read(pParser, pMetadata->data.bext.pCodingHistory, extraBytes, NULL);
+ pMetadata->data.bext.codingHistorySize = (drwav_uint32)strlen(pMetadata->data.bext.pCodingHistory);
+ } else {
+ pMetadata->data.bext.pCodingHistory = NULL;
+ pMetadata->data.bext.codingHistorySize = 0;
+ }
}
}
return bytesRead;
@@ -53707,7 +74551,7 @@ DRWAV_PRIVATE drwav_uint64 drwav__read_list_label_or_note_to_metadata_obj(drwav_
pMetadata->data.labelOrNote.stringLength = sizeIncludingNullTerminator - 1;
pMetadata->data.labelOrNote.pString = (char*)drwav__metadata_get_memory(pParser, sizeIncludingNullTerminator, 1);
DRWAV_ASSERT(pMetadata->data.labelOrNote.pString != NULL);
- bytesJustRead = drwav__metadata_parser_read(pParser, pMetadata->data.labelOrNote.pString, sizeIncludingNullTerminator, &totalBytesRead);
+ drwav__metadata_parser_read(pParser, pMetadata->data.labelOrNote.pString, sizeIncludingNullTerminator, &totalBytesRead);
} else {
pMetadata->data.labelOrNote.stringLength = 0;
pMetadata->data.labelOrNote.pString = NULL;
@@ -53739,7 +74583,7 @@ DRWAV_PRIVATE drwav_uint64 drwav__read_list_labelled_cue_region_to_metadata_obj(
pMetadata->data.labelledCueRegion.stringLength = sizeIncludingNullTerminator - 1;
pMetadata->data.labelledCueRegion.pString = (char*)drwav__metadata_get_memory(pParser, sizeIncludingNullTerminator, 1);
DRWAV_ASSERT(pMetadata->data.labelledCueRegion.pString != NULL);
- bytesJustRead = drwav__metadata_parser_read(pParser, pMetadata->data.labelledCueRegion.pString, sizeIncludingNullTerminator, &totalBytesRead);
+ drwav__metadata_parser_read(pParser, pMetadata->data.labelledCueRegion.pString, sizeIncludingNullTerminator, &totalBytesRead);
} else {
pMetadata->data.labelledCueRegion.stringLength = 0;
pMetadata->data.labelledCueRegion.pString = NULL;
@@ -53805,11 +74649,11 @@ DRWAV_PRIVATE drwav_uint64 drwav__metadata_process_unknown_chunk(drwav__metadata
}
return bytesRead;
}
-DRWAV_PRIVATE drwav_bool32 drwav__chunk_matches(drwav_uint64 allowedMetadataTypes, const drwav_uint8* pChunkID, drwav_metadata_type type, const char* pID)
+DRWAV_PRIVATE drwav_bool32 drwav__chunk_matches(drwav_metadata_type allowedMetadataTypes, const drwav_uint8* pChunkID, drwav_metadata_type type, const char* pID)
{
return (allowedMetadataTypes & type) && drwav_fourcc_equal(pChunkID, pID);
}
-DRWAV_PRIVATE drwav_uint64 drwav__metadata_process_chunk(drwav__metadata_parser* pParser, const drwav_chunk_header* pChunkHeader, drwav_uint64 allowedMetadataTypes)
+DRWAV_PRIVATE drwav_uint64 drwav__metadata_process_chunk(drwav__metadata_parser* pParser, const drwav_chunk_header* pChunkHeader, drwav_metadata_type allowedMetadataTypes)
{
const drwav_uint8 *pChunkID = pChunkHeader->id.fourcc;
drwav_uint64 bytesRead = 0;
@@ -53825,16 +74669,21 @@ DRWAV_PRIVATE drwav_uint64 drwav__metadata_process_chunk(drwav__metadata_parser*
bytesJustRead = drwav__metadata_parser_read(pParser, buffer, sizeof(buffer), &bytesRead);
if (bytesJustRead == sizeof(buffer)) {
drwav_uint32 loopCount = drwav_bytes_to_u32(buffer);
- bytesJustRead = drwav__metadata_parser_read(pParser, buffer, sizeof(buffer), &bytesRead);
- if (bytesJustRead == sizeof(buffer)) {
- drwav_uint32 samplerSpecificDataSizeInBytes = drwav_bytes_to_u32(buffer);
- pParser->metadataCount += 1;
- drwav__metadata_request_extra_memory_for_stage_2(pParser, sizeof(drwav_smpl_loop) * loopCount, DRWAV_METADATA_ALIGNMENT);
- drwav__metadata_request_extra_memory_for_stage_2(pParser, samplerSpecificDataSizeInBytes, 1);
+ drwav_uint64 calculatedLoopCount;
+ calculatedLoopCount = (pChunkHeader->sizeInBytes - DRWAV_SMPL_BYTES) / DRWAV_SMPL_LOOP_BYTES;
+ if (calculatedLoopCount == loopCount) {
+ bytesJustRead = drwav__metadata_parser_read(pParser, buffer, sizeof(buffer), &bytesRead);
+ if (bytesJustRead == sizeof(buffer)) {
+ drwav_uint32 samplerSpecificDataSizeInBytes = drwav_bytes_to_u32(buffer);
+ pParser->metadataCount += 1;
+ drwav__metadata_request_extra_memory_for_stage_2(pParser, sizeof(drwav_smpl_loop) * loopCount, DRWAV_METADATA_ALIGNMENT);
+ drwav__metadata_request_extra_memory_for_stage_2(pParser, samplerSpecificDataSizeInBytes, 1);
+ }
+ } else {
}
}
} else {
- bytesRead = drwav__read_smpl_to_metadata_obj(pParser, &pParser->pMetadata[pParser->metadataCursor]);
+ bytesRead = drwav__read_smpl_to_metadata_obj(pParser, pChunkHeader, &pParser->pMetadata[pParser->metadataCursor]);
if (bytesRead == pChunkHeader->sizeInBytes) {
pParser->metadataCursor += 1;
} else {
@@ -53876,7 +74725,7 @@ DRWAV_PRIVATE drwav_uint64 drwav__metadata_process_chunk(drwav__metadata_parser*
cueCount = (size_t)(pChunkHeader->sizeInBytes - DRWAV_CUE_BYTES) / DRWAV_CUE_POINT_BYTES;
drwav__metadata_request_extra_memory_for_stage_2(pParser, sizeof(drwav_cue_point) * cueCount, DRWAV_METADATA_ALIGNMENT);
} else {
- bytesRead = drwav__read_cue_to_metadata_obj(pParser, &pParser->pMetadata[pParser->metadataCursor]);
+ bytesRead = drwav__read_cue_to_metadata_obj(pParser, pChunkHeader, &pParser->pMetadata[pParser->metadataCursor]);
if (bytesRead == pChunkHeader->sizeInBytes) {
pParser->metadataCursor += 1;
} else {
@@ -53991,7 +74840,7 @@ DRWAV_PRIVATE drwav_uint64 drwav__metadata_process_chunk(drwav__metadata_parser*
subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_album);
} else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_tracknumber, "ITRK")) {
subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_tracknumber);
- } else if (allowedMetadataTypes & drwav_metadata_type_unknown) {
+ } else if ((allowedMetadataTypes & drwav_metadata_type_unknown) != 0) {
subchunkBytesRead = drwav__metadata_process_unknown_chunk(pParser, subchunkId, subchunkDataSize, listType);
}
bytesRead += subchunkBytesRead;
@@ -54010,18 +74859,25 @@ DRWAV_PRIVATE drwav_uint64 drwav__metadata_process_chunk(drwav__metadata_parser*
bytesRead += 1;
}
}
- } else if (allowedMetadataTypes & drwav_metadata_type_unknown) {
+ } else if ((allowedMetadataTypes & drwav_metadata_type_unknown) != 0) {
bytesRead = drwav__metadata_process_unknown_chunk(pParser, pChunkID, pChunkHeader->sizeInBytes, drwav_metadata_location_top_level);
}
return bytesRead;
}
DRWAV_PRIVATE drwav_uint32 drwav_get_bytes_per_pcm_frame(drwav* pWav)
{
+ drwav_uint32 bytesPerFrame;
if ((pWav->bitsPerSample & 0x7) == 0) {
- return (pWav->bitsPerSample * pWav->fmt.channels) >> 3;
+ bytesPerFrame = (pWav->bitsPerSample * pWav->fmt.channels) >> 3;
} else {
- return pWav->fmt.blockAlign;
+ bytesPerFrame = pWav->fmt.blockAlign;
}
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW || pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) {
+ if (bytesPerFrame != pWav->fmt.channels) {
+ return 0;
+ }
+ }
+ return bytesPerFrame;
}
DRWAV_API drwav_uint16 drwav_fmt_get_format(const drwav_fmt* pFMT)
{
@@ -54302,7 +75158,11 @@ DRWAV_PRIVATE drwav_bool32 drwav_init__internal(drwav* pWav, drwav_chunk_proc on
if (sampleCountFromFactChunk != 0) {
pWav->totalPCMFrameCount = sampleCountFromFactChunk;
} else {
- pWav->totalPCMFrameCount = dataChunkSize / drwav_get_bytes_per_pcm_frame(pWav);
+ drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ return DRWAV_FALSE;
+ }
+ pWav->totalPCMFrameCount = dataChunkSize / bytesPerFrame;
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) {
drwav_uint64 totalBlockHeaderSizeInBytes;
drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign;
@@ -54328,6 +75188,9 @@ DRWAV_PRIVATE drwav_bool32 drwav_init__internal(drwav* pWav, drwav_chunk_proc on
return DRWAV_FALSE;
}
}
+ if (drwav_get_bytes_per_pcm_frame(pWav) == 0) {
+ return DRWAV_FALSE;
+ }
#ifdef DR_WAV_LIBSNDFILE_COMPAT
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) {
drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign;
@@ -55829,6 +76692,7 @@ DRWAV_API drwav_result drwav_uninit(drwav* pWav)
DRWAV_API size_t drwav_read_raw(drwav* pWav, size_t bytesToRead, void* pBufferOut)
{
size_t bytesRead;
+ drwav_uint32 bytesPerFrame;
if (pWav == NULL || bytesToRead == 0) {
return 0;
}
@@ -55838,6 +76702,10 @@ DRWAV_API size_t drwav_read_raw(drwav* pWav, size_t bytesToRead, void* pBufferOu
if (bytesToRead == 0) {
return 0;
}
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ return 0;
+ }
if (pBufferOut != NULL) {
bytesRead = pWav->onRead(pWav->pUserData, pBufferOut, bytesToRead);
} else {
@@ -55866,7 +76734,7 @@ DRWAV_API size_t drwav_read_raw(drwav* pWav, size_t bytesToRead, void* pBufferOu
}
}
}
- pWav->readCursorInPCMFrames += bytesRead / drwav_get_bytes_per_pcm_frame(pWav);
+ pWav->readCursorInPCMFrames += bytesRead / bytesPerFrame;
pWav->bytesRemaining -= bytesRead;
return bytesRead;
}
@@ -55897,7 +76765,11 @@ DRWAV_API drwav_uint64 drwav_read_pcm_frames_be(drwav* pWav, drwav_uint64 frames
{
drwav_uint64 framesRead = drwav_read_pcm_frames_le(pWav, framesToRead, pBufferOut);
if (pBufferOut != NULL) {
- drwav__bswap_samples(pBufferOut, framesRead*pWav->channels, drwav_get_bytes_per_pcm_frame(pWav)/pWav->channels, pWav->translatedFormatTag);
+ drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ return 0;
+ }
+ drwav__bswap_samples(pBufferOut, framesRead*pWav->channels, bytesPerFrame/pWav->channels, pWav->translatedFormatTag);
}
return framesRead;
}
@@ -55941,8 +76813,8 @@ DRWAV_API drwav_bool32 drwav_seek_to_pcm_frame(drwav* pWav, drwav_uint64 targetF
if (pWav->totalPCMFrameCount == 0) {
return DRWAV_TRUE;
}
- if (targetFrameIndex >= pWav->totalPCMFrameCount) {
- targetFrameIndex = pWav->totalPCMFrameCount - 1;
+ if (targetFrameIndex > pWav->totalPCMFrameCount) {
+ targetFrameIndex = pWav->totalPCMFrameCount;
}
if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) {
if (targetFrameIndex < pWav->readCursorInPCMFrames) {
@@ -55977,10 +76849,15 @@ DRWAV_API drwav_bool32 drwav_seek_to_pcm_frame(drwav* pWav, drwav_uint64 targetF
drwav_uint64 currentBytePos;
drwav_uint64 targetBytePos;
drwav_uint64 offset;
- totalSizeInBytes = pWav->totalPCMFrameCount * drwav_get_bytes_per_pcm_frame(pWav);
+ drwav_uint32 bytesPerFrame;
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ if (bytesPerFrame == 0) {
+ return DRWAV_FALSE;
+ }
+ totalSizeInBytes = pWav->totalPCMFrameCount * bytesPerFrame;
DRWAV_ASSERT(totalSizeInBytes >= pWav->bytesRemaining);
currentBytePos = totalSizeInBytes - pWav->bytesRemaining;
- targetBytePos = targetFrameIndex * drwav_get_bytes_per_pcm_frame(pWav);
+ targetBytePos = targetFrameIndex * bytesPerFrame;
if (currentBytePos < targetBytePos) {
offset = (targetBytePos - currentBytePos);
} else {
@@ -55994,7 +76871,7 @@ DRWAV_API drwav_bool32 drwav_seek_to_pcm_frame(drwav* pWav, drwav_uint64 targetF
if (!pWav->onSeek(pWav->pUserData, offset32, drwav_seek_origin_current)) {
return DRWAV_FALSE;
}
- pWav->readCursorInPCMFrames += offset32 / drwav_get_bytes_per_pcm_frame(pWav);
+ pWav->readCursorInPCMFrames += offset32 / bytesPerFrame;
pWav->bytesRemaining -= offset32;
offset -= offset32;
}
@@ -56080,6 +76957,9 @@ DRWAV_API drwav_uint64 drwav_write_pcm_frames_be(drwav* pWav, drwav_uint64 frame
bytesWritten = 0;
pRunningData = (const drwav_uint8*)pData;
bytesPerSample = drwav_get_bytes_per_pcm_frame(pWav) / pWav->channels;
+ if (bytesPerSample == 0) {
+ return 0;
+ }
while (bytesToWrite > 0) {
drwav_uint8 temp[4096];
drwav_uint32 sampleCount;
@@ -56278,7 +77158,7 @@ DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__ima(drwav* pWav, drwav_uin
return totalFramesRead;
}
pWav->ima.predictor[0] = drwav_bytes_to_s16(header + 0);
- pWav->ima.stepIndex[0] = header[2];
+ pWav->ima.stepIndex[0] = drwav_clamp(header[2], 0, (drwav_int32)drwav_countof(stepTable)-1);
pWav->ima.cachedFrames[drwav_countof(pWav->ima.cachedFrames) - 1] = pWav->ima.predictor[0];
pWav->ima.cachedFrameCount = 1;
} else {
@@ -56293,9 +77173,9 @@ DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__ima(drwav* pWav, drwav_uin
return totalFramesRead;
}
pWav->ima.predictor[0] = drwav_bytes_to_s16(header + 0);
- pWav->ima.stepIndex[0] = header[2];
+ pWav->ima.stepIndex[0] = drwav_clamp(header[2], 0, (drwav_int32)drwav_countof(stepTable)-1);
pWav->ima.predictor[1] = drwav_bytes_to_s16(header + 4);
- pWav->ima.stepIndex[1] = header[6];
+ pWav->ima.stepIndex[1] = drwav_clamp(header[6], 0, (drwav_int32)drwav_countof(stepTable)-1);
pWav->ima.cachedFrames[drwav_countof(pWav->ima.cachedFrames) - 2] = pWav->ima.predictor[0];
pWav->ima.cachedFrames[drwav_countof(pWav->ima.cachedFrames) - 1] = pWav->ima.predictor[1];
pWav->ima.cachedFrameCount = 1;
@@ -56409,7 +77289,7 @@ static DRWAV_INLINE drwav_int16 drwav__mulaw_to_s16(drwav_uint8 sampleIn)
}
DRWAV_PRIVATE void drwav__pcm_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample)
{
- unsigned int i;
+ size_t i;
if (bytesPerSample == 1) {
drwav_u8_to_s16(pOut, pIn, totalSampleCount);
return;
@@ -56461,8 +77341,10 @@ DRWAV_PRIVATE void drwav__ieee_to_s16(drwav_int16* pOut, const drwav_uint8* pIn,
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__pcm(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
{
drwav_uint64 totalFramesRead;
- drwav_uint8 sampleData[4096];
+ drwav_uint8 sampleData[4096] = {0};
drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
if ((pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM && pWav->bitsPerSample == 16) || pBufferOut == NULL) {
return drwav_read_pcm_frames(pWav, framesToRead, pBufferOut);
}
@@ -56470,14 +77352,25 @@ DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__pcm(drwav* pWav, drwav_uin
if (bytesPerFrame == 0) {
return 0;
}
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0;
+ }
totalFramesRead = 0;
while (framesToRead > 0) {
- drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData);
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
- drwav__pcm_to_s16(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels), bytesPerFrame/pWav->channels);
- pBufferOut += framesRead*pWav->channels;
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE);
+ break;
+ }
+ drwav__pcm_to_s16(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample);
+ pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
@@ -56486,8 +77379,10 @@ DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__pcm(drwav* pWav, drwav_uin
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__ieee(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
{
drwav_uint64 totalFramesRead;
- drwav_uint8 sampleData[4096];
+ drwav_uint8 sampleData[4096] = {0};
drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
if (pBufferOut == NULL) {
return drwav_read_pcm_frames(pWav, framesToRead, NULL);
}
@@ -56495,14 +77390,25 @@ DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__ieee(drwav* pWav, drwav_ui
if (bytesPerFrame == 0) {
return 0;
}
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0;
+ }
totalFramesRead = 0;
while (framesToRead > 0) {
- drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData);
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
- drwav__ieee_to_s16(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels), bytesPerFrame/pWav->channels);
- pBufferOut += framesRead*pWav->channels;
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE);
+ break;
+ }
+ drwav__ieee_to_s16(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample);
+ pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
@@ -56511,8 +77417,10 @@ DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__ieee(drwav* pWav, drwav_ui
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__alaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
{
drwav_uint64 totalFramesRead;
- drwav_uint8 sampleData[4096];
+ drwav_uint8 sampleData[4096] = {0};
drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
if (pBufferOut == NULL) {
return drwav_read_pcm_frames(pWav, framesToRead, NULL);
}
@@ -56520,14 +77428,25 @@ DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__alaw(drwav* pWav, drwav_ui
if (bytesPerFrame == 0) {
return 0;
}
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0;
+ }
totalFramesRead = 0;
while (framesToRead > 0) {
- drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData);
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
- drwav_alaw_to_s16(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels));
- pBufferOut += framesRead*pWav->channels;
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE);
+ break;
+ }
+ drwav_alaw_to_s16(pBufferOut, sampleData, (size_t)samplesRead);
+ pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
@@ -56536,8 +77455,10 @@ DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__alaw(drwav* pWav, drwav_ui
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__mulaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
{
drwav_uint64 totalFramesRead;
- drwav_uint8 sampleData[4096];
+ drwav_uint8 sampleData[4096] = {0};
drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
if (pBufferOut == NULL) {
return drwav_read_pcm_frames(pWav, framesToRead, NULL);
}
@@ -56545,14 +77466,25 @@ DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__mulaw(drwav* pWav, drwav_u
if (bytesPerFrame == 0) {
return 0;
}
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0;
+ }
totalFramesRead = 0;
while (framesToRead > 0) {
- drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData);
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
- drwav_mulaw_to_s16(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels));
- pBufferOut += framesRead*pWav->channels;
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE);
+ break;
+ }
+ drwav_mulaw_to_s16(pBufferOut, sampleData, (size_t)samplesRead);
+ pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
@@ -56733,49 +77665,50 @@ DRWAV_PRIVATE void drwav__ieee_to_f32(float* pOut, const drwav_uint8* pIn, size_
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__pcm(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
{
drwav_uint64 totalFramesRead;
- drwav_uint8 sampleData[4096];
- drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ drwav_uint8 sampleData[4096] = {0};
+ drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0;
+ }
totalFramesRead = 0;
while (framesToRead > 0) {
- drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData);
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
- drwav__pcm_to_f32(pBufferOut, sampleData, (size_t)framesRead*pWav->channels, bytesPerFrame/pWav->channels);
- pBufferOut += framesRead*pWav->channels;
- framesToRead -= framesRead;
- totalFramesRead += framesRead;
- }
- return totalFramesRead;
-}
-DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__msadpcm(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
-{
- drwav_uint64 totalFramesRead = 0;
- drwav_int16 samples16[2048];
- while (framesToRead > 0) {
- drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, drwav_min(framesToRead, drwav_countof(samples16)/pWav->channels), samples16);
- if (framesRead == 0) {
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE);
break;
}
- drwav_s16_to_f32(pBufferOut, samples16, (size_t)(framesRead*pWav->channels));
- pBufferOut += framesRead*pWav->channels;
+ drwav__pcm_to_f32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample);
+ pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
return totalFramesRead;
}
-DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__ima(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
+DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__msadpcm_ima(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
{
- drwav_uint64 totalFramesRead = 0;
+ drwav_uint64 totalFramesRead;
drwav_int16 samples16[2048];
+ totalFramesRead = 0;
while (framesToRead > 0) {
- drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, drwav_min(framesToRead, drwav_countof(samples16)/pWav->channels), samples16);
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, drwav_countof(samples16)/pWav->channels);
+ drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, framesToReadThisIteration, samples16);
if (framesRead == 0) {
break;
}
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
drwav_s16_to_f32(pBufferOut, samples16, (size_t)(framesRead*pWav->channels));
pBufferOut += framesRead*pWav->channels;
framesToRead -= framesRead;
@@ -56786,8 +77719,10 @@ DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__ima(drwav* pWav, drwav_uin
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__ieee(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
{
drwav_uint64 totalFramesRead;
- drwav_uint8 sampleData[4096];
+ drwav_uint8 sampleData[4096] = {0};
drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT && pWav->bitsPerSample == 32) {
return drwav_read_pcm_frames(pWav, framesToRead, pBufferOut);
}
@@ -56795,14 +77730,25 @@ DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__ieee(drwav* pWav, drwav_ui
if (bytesPerFrame == 0) {
return 0;
}
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0;
+ }
totalFramesRead = 0;
while (framesToRead > 0) {
- drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData);
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
- drwav__ieee_to_f32(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels), bytesPerFrame/pWav->channels);
- pBufferOut += framesRead*pWav->channels;
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE);
+ break;
+ }
+ drwav__ieee_to_f32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample);
+ pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
@@ -56811,19 +77757,33 @@ DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__ieee(drwav* pWav, drwav_ui
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__alaw(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
{
drwav_uint64 totalFramesRead;
- drwav_uint8 sampleData[4096];
- drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ drwav_uint8 sampleData[4096] = {0};
+ drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0;
+ }
totalFramesRead = 0;
while (framesToRead > 0) {
- drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData);
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
- drwav_alaw_to_f32(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels));
- pBufferOut += framesRead*pWav->channels;
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE);
+ break;
+ }
+ drwav_alaw_to_f32(pBufferOut, sampleData, (size_t)samplesRead);
+ pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
@@ -56832,19 +77792,33 @@ DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__alaw(drwav* pWav, drwav_ui
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__mulaw(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
{
drwav_uint64 totalFramesRead;
- drwav_uint8 sampleData[4096];
- drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ drwav_uint8 sampleData[4096] = {0};
+ drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0;
+ }
totalFramesRead = 0;
while (framesToRead > 0) {
- drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData);
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
- drwav_mulaw_to_f32(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels));
- pBufferOut += framesRead*pWav->channels;
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE);
+ break;
+ }
+ drwav_mulaw_to_f32(pBufferOut, sampleData, (size_t)samplesRead);
+ pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
@@ -56864,8 +77838,8 @@ DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32(drwav* pWav, drwav_uint64 frame
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) {
return drwav_read_pcm_frames_f32__pcm(pWav, framesToRead, pBufferOut);
}
- if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) {
- return drwav_read_pcm_frames_f32__msadpcm(pWav, framesToRead, pBufferOut);
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM || pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) {
+ return drwav_read_pcm_frames_f32__msadpcm_ima(pWav, framesToRead, pBufferOut);
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) {
return drwav_read_pcm_frames_f32__ieee(pWav, framesToRead, pBufferOut);
@@ -56876,9 +77850,6 @@ DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32(drwav* pWav, drwav_uint64 frame
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) {
return drwav_read_pcm_frames_f32__mulaw(pWav, framesToRead, pBufferOut);
}
- if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) {
- return drwav_read_pcm_frames_f32__ima(pWav, framesToRead, pBufferOut);
- }
return 0;
}
DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32le(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
@@ -57035,8 +78006,10 @@ DRWAV_PRIVATE void drwav__ieee_to_s32(drwav_int32* pOut, const drwav_uint8* pIn,
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__pcm(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
{
drwav_uint64 totalFramesRead;
- drwav_uint8 sampleData[4096];
+ drwav_uint8 sampleData[4096] = {0};
drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM && pWav->bitsPerSample == 32) {
return drwav_read_pcm_frames(pWav, framesToRead, pBufferOut);
}
@@ -57044,44 +78017,41 @@ DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__pcm(drwav* pWav, drwav_uin
if (bytesPerFrame == 0) {
return 0;
}
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0;
+ }
totalFramesRead = 0;
while (framesToRead > 0) {
- drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData);
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
- drwav__pcm_to_s32(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels), bytesPerFrame/pWav->channels);
- pBufferOut += framesRead*pWav->channels;
- framesToRead -= framesRead;
- totalFramesRead += framesRead;
- }
- return totalFramesRead;
-}
-DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__msadpcm(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
-{
- drwav_uint64 totalFramesRead = 0;
- drwav_int16 samples16[2048];
- while (framesToRead > 0) {
- drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, drwav_min(framesToRead, drwav_countof(samples16)/pWav->channels), samples16);
- if (framesRead == 0) {
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE);
break;
}
- drwav_s16_to_s32(pBufferOut, samples16, (size_t)(framesRead*pWav->channels));
- pBufferOut += framesRead*pWav->channels;
+ drwav__pcm_to_s32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample);
+ pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
return totalFramesRead;
}
-DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__ima(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
+DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__msadpcm_ima(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
{
drwav_uint64 totalFramesRead = 0;
drwav_int16 samples16[2048];
while (framesToRead > 0) {
- drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, drwav_min(framesToRead, drwav_countof(samples16)/pWav->channels), samples16);
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, drwav_countof(samples16)/pWav->channels);
+ drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, framesToReadThisIteration, samples16);
if (framesRead == 0) {
break;
}
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
drwav_s16_to_s32(pBufferOut, samples16, (size_t)(framesRead*pWav->channels));
pBufferOut += framesRead*pWav->channels;
framesToRead -= framesRead;
@@ -57092,19 +78062,33 @@ DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__ima(drwav* pWav, drwav_uin
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__ieee(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
{
drwav_uint64 totalFramesRead;
- drwav_uint8 sampleData[4096];
- drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ drwav_uint8 sampleData[4096] = {0};
+ drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0;
+ }
totalFramesRead = 0;
while (framesToRead > 0) {
- drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData);
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
- drwav__ieee_to_s32(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels), bytesPerFrame/pWav->channels);
- pBufferOut += framesRead*pWav->channels;
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE);
+ break;
+ }
+ drwav__ieee_to_s32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample);
+ pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
@@ -57113,19 +78097,33 @@ DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__ieee(drwav* pWav, drwav_ui
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__alaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
{
drwav_uint64 totalFramesRead;
- drwav_uint8 sampleData[4096];
- drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ drwav_uint8 sampleData[4096] = {0};
+ drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0;
+ }
totalFramesRead = 0;
while (framesToRead > 0) {
- drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData);
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
- drwav_alaw_to_s32(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels));
- pBufferOut += framesRead*pWav->channels;
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE);
+ break;
+ }
+ drwav_alaw_to_s32(pBufferOut, sampleData, (size_t)samplesRead);
+ pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
@@ -57134,19 +78132,33 @@ DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__alaw(drwav* pWav, drwav_ui
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__mulaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
{
drwav_uint64 totalFramesRead;
- drwav_uint8 sampleData[4096];
- drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
+ drwav_uint8 sampleData[4096] = {0};
+ drwav_uint32 bytesPerFrame;
+ drwav_uint32 bytesPerSample;
+ drwav_uint64 samplesRead;
+ bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
+ bytesPerSample = bytesPerFrame / pWav->channels;
+ if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
+ return 0;
+ }
totalFramesRead = 0;
while (framesToRead > 0) {
- drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame), sampleData);
+ drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
+ drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
- drwav_mulaw_to_s32(pBufferOut, sampleData, (size_t)(framesRead*pWav->channels));
- pBufferOut += framesRead*pWav->channels;
+ DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
+ samplesRead = framesRead * pWav->channels;
+ if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
+ DRWAV_ASSERT(DRWAV_FALSE);
+ break;
+ }
+ drwav_mulaw_to_s32(pBufferOut, sampleData, (size_t)samplesRead);
+ pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
@@ -57166,8 +78178,8 @@ DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32(drwav* pWav, drwav_uint64 frame
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) {
return drwav_read_pcm_frames_s32__pcm(pWav, framesToRead, pBufferOut);
}
- if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) {
- return drwav_read_pcm_frames_s32__msadpcm(pWav, framesToRead, pBufferOut);
+ if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM || pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) {
+ return drwav_read_pcm_frames_s32__msadpcm_ima(pWav, framesToRead, pBufferOut);
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) {
return drwav_read_pcm_frames_s32__ieee(pWav, framesToRead, pBufferOut);
@@ -57178,9 +78190,6 @@ DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32(drwav* pWav, drwav_uint64 frame
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) {
return drwav_read_pcm_frames_s32__mulaw(pWav, framesToRead, pBufferOut);
}
- if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) {
- return drwav_read_pcm_frames_s32__ima(pWav, framesToRead, pBufferOut);
- }
return 0;
}
DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32le(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
@@ -57690,7 +78699,7 @@ DRWAV_API drwav_bool32 drwav_fourcc_equal(const drwav_uint8* a, const char* b)
#define DRFLAC_X64
#elif defined(__i386) || defined(_M_IX86)
#define DRFLAC_X86
-#elif defined(__arm__) || defined(_M_ARM) || defined(_M_ARM64)
+#elif defined(__arm__) || defined(_M_ARM) || defined(__arm64) || defined(__arm64__) || defined(__aarch64__) || defined(_M_ARM64)
#define DRFLAC_ARM
#endif
#if !defined(DR_FLAC_NO_SIMD)
@@ -57727,13 +78736,6 @@ DRWAV_API drwav_bool32 drwav_fourcc_equal(const drwav_uint8* a, const char* b)
#if defined(DRFLAC_ARM)
#if !defined(DRFLAC_NO_NEON) && (defined(__ARM_NEON) || defined(__aarch64__) || defined(_M_ARM64))
#define DRFLAC_SUPPORT_NEON
- #endif
- #if !defined(__GNUC__) && !defined(__clang__) && defined(__has_include)
- #if !defined(DRFLAC_SUPPORT_NEON) && !defined(DRFLAC_NO_NEON) && __has_include()
- #define DRFLAC_SUPPORT_NEON
- #endif
- #endif
- #if defined(DRFLAC_SUPPORT_NEON)
#include
#endif
#endif
@@ -58136,6 +79138,11 @@ static DRFLAC_INLINE drflac_uint32 drflac__be2host_32(drflac_uint32 n)
}
return n;
}
+static DRFLAC_INLINE drflac_uint32 drflac__be2host_32_ptr_unaligned(const void* pData)
+{
+ const drflac_uint8* pNum = (drflac_uint8*)pData;
+ return *(pNum) << 24 | *(pNum+1) << 16 | *(pNum+2) << 8 | *(pNum+3);
+}
static DRFLAC_INLINE drflac_uint64 drflac__be2host_64(drflac_uint64 n)
{
if (drflac__is_little_endian()) {
@@ -58150,6 +79157,11 @@ static DRFLAC_INLINE drflac_uint32 drflac__le2host_32(drflac_uint32 n)
}
return n;
}
+static DRFLAC_INLINE drflac_uint32 drflac__le2host_32_ptr_unaligned(const void* pData)
+{
+ const drflac_uint8* pNum = (drflac_uint8*)pData;
+ return *pNum | *(pNum+1) << 8 | *(pNum+2) << 16 | *(pNum+3) << 24;
+}
static DRFLAC_INLINE drflac_uint32 drflac__unsynchsafe_32(drflac_uint32 n)
{
drflac_uint32 result = 0;
@@ -58535,6 +79547,9 @@ static DRFLAC_INLINE drflac_bool32 drflac__read_uint32(drflac_bs* bs, unsigned i
if (!drflac__reload_cache(bs)) {
return DRFLAC_FALSE;
}
+ if (bitCountLo > DRFLAC_CACHE_L1_BITS_REMAINING(bs)) {
+ return DRFLAC_FALSE;
+ }
*pResultOut = (resultHi << bitCountLo) | (drflac_uint32)DRFLAC_CACHE_L1_SELECT_AND_SHIFT(bs, bitCountLo);
bs->consumedBits += bitCountLo;
bs->cache <<= bitCountLo;
@@ -58876,8 +79891,18 @@ static DRFLAC_INLINE drflac_bool32 drflac__seek_past_next_set_bit(drflac_bs* bs,
return DRFLAC_FALSE;
}
}
+ if (bs->cache == 1) {
+ *pOffsetOut = zeroCounter + (drflac_uint32)DRFLAC_CACHE_L1_BITS_REMAINING(bs) - 1;
+ if (!drflac__reload_cache(bs)) {
+ return DRFLAC_FALSE;
+ }
+ return DRFLAC_TRUE;
+ }
setBitOffsetPlus1 = drflac__clz(bs->cache);
setBitOffsetPlus1 += 1;
+ if (setBitOffsetPlus1 > DRFLAC_CACHE_L1_BITS_REMAINING(bs)) {
+ return DRFLAC_FALSE;
+ }
bs->consumedBits += setBitOffsetPlus1;
bs->cache <<= setBitOffsetPlus1;
*pOffsetOut = zeroCounter + setBitOffsetPlus1 - 1;
@@ -58963,6 +79988,24 @@ static drflac_result drflac__read_utf8_coded_number(drflac_bs* bs, drflac_uint64
*pCRCOut = crc;
return DRFLAC_SUCCESS;
}
+static DRFLAC_INLINE drflac_uint32 drflac__ilog2_u32(drflac_uint32 x)
+{
+#if 1
+ drflac_uint32 result = 0;
+ while (x > 0) {
+ result += 1;
+ x >>= 1;
+ }
+ return result;
+#endif
+}
+static DRFLAC_INLINE drflac_bool32 drflac__use_64_bit_prediction(drflac_uint32 bitsPerSample, drflac_uint32 order, drflac_uint32 precision)
+{
+ return bitsPerSample + precision + drflac__ilog2_u32(order) > 32;
+}
+#if defined(__clang__)
+__attribute__((no_sanitize("signed-integer-overflow")))
+#endif
static DRFLAC_INLINE drflac_int32 drflac__calculate_prediction_32(drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pDecodedSamples)
{
drflac_int32 prediction = 0;
@@ -59173,7 +80216,7 @@ static DRFLAC_INLINE drflac_int32 drflac__calculate_prediction_64(drflac_uint32
return (drflac_int32)(prediction >> shift);
}
#if 0
-static drflac_bool32 drflac__decode_samples_with_residual__rice__reference(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pSamplesOut)
+static drflac_bool32 drflac__decode_samples_with_residual__rice__reference(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pSamplesOut)
{
drflac_uint32 i;
DRFLAC_ASSERT(bs != NULL);
@@ -59205,10 +80248,10 @@ static drflac_bool32 drflac__decode_samples_with_residual__rice__reference(drfla
} else {
decodedRice = (decodedRice >> 1);
}
- if (bitsPerSample+shift >= 32) {
- pSamplesOut[i] = decodedRice + drflac__calculate_prediction_64(order, shift, coefficients, pSamplesOut + i);
+ if (drflac__use_64_bit_prediction(bitsPerSample, lpcOrder, lpcPrecision)) {
+ pSamplesOut[i] = decodedRice + drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + i);
} else {
- pSamplesOut[i] = decodedRice + drflac__calculate_prediction_32(order, shift, coefficients, pSamplesOut + i);
+ pSamplesOut[i] = decodedRice + drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + i);
}
}
return DRFLAC_TRUE;
@@ -59287,6 +80330,9 @@ static DRFLAC_INLINE drflac_bool32 drflac__read_rice_parts(drflac_bs* bs, drflac
if (!drflac__reload_cache(bs)) {
return DRFLAC_FALSE;
}
+ if (bitCountLo > DRFLAC_CACHE_L1_BITS_REMAINING(bs)) {
+ return DRFLAC_FALSE;
+ }
}
riceParamPart = (drflac_uint32)(resultHi | DRFLAC_CACHE_L1_SELECT_AND_SHIFT_SAFE(bs, bitCountLo));
bs->consumedBits += bitCountLo;
@@ -59334,6 +80380,9 @@ static DRFLAC_INLINE drflac_bool32 drflac__read_rice_parts_x1(drflac_bs* bs, drf
if (!drflac__reload_cache(bs)) {
return DRFLAC_FALSE;
}
+ if (riceParamPartLoBitCount > DRFLAC_CACHE_L1_BITS_REMAINING(bs)) {
+ return DRFLAC_FALSE;
+ }
bs_cache = bs->cache;
bs_consumedBits = bs->consumedBits + riceParamPartLoBitCount;
}
@@ -59403,6 +80452,9 @@ static DRFLAC_INLINE drflac_bool32 drflac__seek_rice_parts(drflac_bs* bs, drflac
if (!drflac__reload_cache(bs)) {
return DRFLAC_FALSE;
}
+ if (riceParamPartLoBitCount > DRFLAC_CACHE_L1_BITS_REMAINING(bs)) {
+ return DRFLAC_FALSE;
+ }
bs_cache = bs->cache;
bs_consumedBits = bs->consumedBits + riceParamPartLoBitCount;
}
@@ -59464,7 +80516,7 @@ static drflac_bool32 drflac__decode_samples_with_residual__rice__scalar_zeroorde
}
return DRFLAC_TRUE;
}
-static drflac_bool32 drflac__decode_samples_with_residual__rice__scalar(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pSamplesOut)
+static drflac_bool32 drflac__decode_samples_with_residual__rice__scalar(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pSamplesOut)
{
drflac_uint32 t[2] = {0x00000000, 0xFFFFFFFF};
drflac_uint32 zeroCountPart0 = 0;
@@ -59480,12 +80532,12 @@ static drflac_bool32 drflac__decode_samples_with_residual__rice__scalar(drflac_b
drflac_uint32 i;
DRFLAC_ASSERT(bs != NULL);
DRFLAC_ASSERT(pSamplesOut != NULL);
- if (order == 0) {
- return drflac__decode_samples_with_residual__rice__scalar_zeroorder(bs, bitsPerSample, count, riceParam, order, shift, coefficients, pSamplesOut);
+ if (lpcOrder == 0) {
+ return drflac__decode_samples_with_residual__rice__scalar_zeroorder(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, coefficients, pSamplesOut);
}
riceParamMask = (drflac_uint32)~((~0UL) << riceParam);
pSamplesOutEnd = pSamplesOut + (count & ~3);
- if (bitsPerSample+shift > 32) {
+ if (drflac__use_64_bit_prediction(bitsPerSample, lpcOrder, lpcPrecision)) {
while (pSamplesOut < pSamplesOutEnd) {
if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart0, &riceParamPart0) ||
!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart1, &riceParamPart1) ||
@@ -59505,10 +80557,10 @@ static drflac_bool32 drflac__decode_samples_with_residual__rice__scalar(drflac_b
riceParamPart1 = (riceParamPart1 >> 1) ^ t[riceParamPart1 & 0x01];
riceParamPart2 = (riceParamPart2 >> 1) ^ t[riceParamPart2 & 0x01];
riceParamPart3 = (riceParamPart3 >> 1) ^ t[riceParamPart3 & 0x01];
- pSamplesOut[0] = riceParamPart0 + drflac__calculate_prediction_64(order, shift, coefficients, pSamplesOut + 0);
- pSamplesOut[1] = riceParamPart1 + drflac__calculate_prediction_64(order, shift, coefficients, pSamplesOut + 1);
- pSamplesOut[2] = riceParamPart2 + drflac__calculate_prediction_64(order, shift, coefficients, pSamplesOut + 2);
- pSamplesOut[3] = riceParamPart3 + drflac__calculate_prediction_64(order, shift, coefficients, pSamplesOut + 3);
+ pSamplesOut[0] = riceParamPart0 + drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + 0);
+ pSamplesOut[1] = riceParamPart1 + drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + 1);
+ pSamplesOut[2] = riceParamPart2 + drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + 2);
+ pSamplesOut[3] = riceParamPart3 + drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + 3);
pSamplesOut += 4;
}
} else {
@@ -59531,10 +80583,10 @@ static drflac_bool32 drflac__decode_samples_with_residual__rice__scalar(drflac_b
riceParamPart1 = (riceParamPart1 >> 1) ^ t[riceParamPart1 & 0x01];
riceParamPart2 = (riceParamPart2 >> 1) ^ t[riceParamPart2 & 0x01];
riceParamPart3 = (riceParamPart3 >> 1) ^ t[riceParamPart3 & 0x01];
- pSamplesOut[0] = riceParamPart0 + drflac__calculate_prediction_32(order, shift, coefficients, pSamplesOut + 0);
- pSamplesOut[1] = riceParamPart1 + drflac__calculate_prediction_32(order, shift, coefficients, pSamplesOut + 1);
- pSamplesOut[2] = riceParamPart2 + drflac__calculate_prediction_32(order, shift, coefficients, pSamplesOut + 2);
- pSamplesOut[3] = riceParamPart3 + drflac__calculate_prediction_32(order, shift, coefficients, pSamplesOut + 3);
+ pSamplesOut[0] = riceParamPart0 + drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + 0);
+ pSamplesOut[1] = riceParamPart1 + drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + 1);
+ pSamplesOut[2] = riceParamPart2 + drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + 2);
+ pSamplesOut[3] = riceParamPart3 + drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + 3);
pSamplesOut += 4;
}
}
@@ -59546,10 +80598,10 @@ static drflac_bool32 drflac__decode_samples_with_residual__rice__scalar(drflac_b
riceParamPart0 &= riceParamMask;
riceParamPart0 |= (zeroCountPart0 << riceParam);
riceParamPart0 = (riceParamPart0 >> 1) ^ t[riceParamPart0 & 0x01];
- if (bitsPerSample+shift > 32) {
- pSamplesOut[0] = riceParamPart0 + drflac__calculate_prediction_64(order, shift, coefficients, pSamplesOut + 0);
+ if (drflac__use_64_bit_prediction(bitsPerSample, lpcOrder, lpcPrecision)) {
+ pSamplesOut[0] = riceParamPart0 + drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + 0);
} else {
- pSamplesOut[0] = riceParamPart0 + drflac__calculate_prediction_32(order, shift, coefficients, pSamplesOut + 0);
+ pSamplesOut[0] = riceParamPart0 + drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + 0);
}
i += 1;
pSamplesOut += 1;
@@ -59894,18 +80946,18 @@ static drflac_bool32 drflac__decode_samples_with_residual__rice__sse41_64(drflac
}
return DRFLAC_TRUE;
}
-static drflac_bool32 drflac__decode_samples_with_residual__rice__sse41(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pSamplesOut)
+static drflac_bool32 drflac__decode_samples_with_residual__rice__sse41(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pSamplesOut)
{
DRFLAC_ASSERT(bs != NULL);
DRFLAC_ASSERT(pSamplesOut != NULL);
- if (order > 0 && order <= 12) {
- if (bitsPerSample+shift > 32) {
- return drflac__decode_samples_with_residual__rice__sse41_64(bs, count, riceParam, order, shift, coefficients, pSamplesOut);
+ if (lpcOrder > 0 && lpcOrder <= 12) {
+ if (drflac__use_64_bit_prediction(bitsPerSample, lpcOrder, lpcPrecision)) {
+ return drflac__decode_samples_with_residual__rice__sse41_64(bs, count, riceParam, lpcOrder, lpcShift, coefficients, pSamplesOut);
} else {
- return drflac__decode_samples_with_residual__rice__sse41_32(bs, count, riceParam, order, shift, coefficients, pSamplesOut);
+ return drflac__decode_samples_with_residual__rice__sse41_32(bs, count, riceParam, lpcOrder, lpcShift, coefficients, pSamplesOut);
}
} else {
- return drflac__decode_samples_with_residual__rice__scalar(bs, bitsPerSample, count, riceParam, order, shift, coefficients, pSamplesOut);
+ return drflac__decode_samples_with_residual__rice__scalar(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pSamplesOut);
}
}
#endif
@@ -60244,37 +81296,37 @@ static drflac_bool32 drflac__decode_samples_with_residual__rice__neon_64(drflac_
}
return DRFLAC_TRUE;
}
-static drflac_bool32 drflac__decode_samples_with_residual__rice__neon(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pSamplesOut)
+static drflac_bool32 drflac__decode_samples_with_residual__rice__neon(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pSamplesOut)
{
DRFLAC_ASSERT(bs != NULL);
DRFLAC_ASSERT(pSamplesOut != NULL);
- if (order > 0 && order <= 12) {
- if (bitsPerSample+shift > 32) {
- return drflac__decode_samples_with_residual__rice__neon_64(bs, count, riceParam, order, shift, coefficients, pSamplesOut);
+ if (lpcOrder > 0 && lpcOrder <= 12) {
+ if (drflac__use_64_bit_prediction(bitsPerSample, lpcOrder, lpcPrecision)) {
+ return drflac__decode_samples_with_residual__rice__neon_64(bs, count, riceParam, lpcOrder, lpcShift, coefficients, pSamplesOut);
} else {
- return drflac__decode_samples_with_residual__rice__neon_32(bs, count, riceParam, order, shift, coefficients, pSamplesOut);
+ return drflac__decode_samples_with_residual__rice__neon_32(bs, count, riceParam, lpcOrder, lpcShift, coefficients, pSamplesOut);
}
} else {
- return drflac__decode_samples_with_residual__rice__scalar(bs, bitsPerSample, count, riceParam, order, shift, coefficients, pSamplesOut);
+ return drflac__decode_samples_with_residual__rice__scalar(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pSamplesOut);
}
}
#endif
-static drflac_bool32 drflac__decode_samples_with_residual__rice(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pSamplesOut)
+static drflac_bool32 drflac__decode_samples_with_residual__rice(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pSamplesOut)
{
#if defined(DRFLAC_SUPPORT_SSE41)
if (drflac__gIsSSE41Supported) {
- return drflac__decode_samples_with_residual__rice__sse41(bs, bitsPerSample, count, riceParam, order, shift, coefficients, pSamplesOut);
+ return drflac__decode_samples_with_residual__rice__sse41(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pSamplesOut);
} else
#elif defined(DRFLAC_SUPPORT_NEON)
if (drflac__gIsNEONSupported) {
- return drflac__decode_samples_with_residual__rice__neon(bs, bitsPerSample, count, riceParam, order, shift, coefficients, pSamplesOut);
+ return drflac__decode_samples_with_residual__rice__neon(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pSamplesOut);
} else
#endif
{
#if 0
- return drflac__decode_samples_with_residual__rice__reference(bs, bitsPerSample, count, riceParam, order, shift, coefficients, pSamplesOut);
+ return drflac__decode_samples_with_residual__rice__reference(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pSamplesOut);
#else
- return drflac__decode_samples_with_residual__rice__scalar(bs, bitsPerSample, count, riceParam, order, shift, coefficients, pSamplesOut);
+ return drflac__decode_samples_with_residual__rice__scalar(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pSamplesOut);
#endif
}
}
@@ -60289,7 +81341,10 @@ static drflac_bool32 drflac__read_and_seek_residual__rice(drflac_bs* bs, drflac_
}
return DRFLAC_TRUE;
}
-static drflac_bool32 drflac__decode_samples_with_residual__unencoded(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 unencodedBitsPerSample, drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pSamplesOut)
+#if defined(__clang__)
+__attribute__((no_sanitize("signed-integer-overflow")))
+#endif
+static drflac_bool32 drflac__decode_samples_with_residual__unencoded(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 unencodedBitsPerSample, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pSamplesOut)
{
drflac_uint32 i;
DRFLAC_ASSERT(bs != NULL);
@@ -60303,15 +81358,15 @@ static drflac_bool32 drflac__decode_samples_with_residual__unencoded(drflac_bs*
} else {
pSamplesOut[i] = 0;
}
- if (bitsPerSample >= 24) {
- pSamplesOut[i] += drflac__calculate_prediction_64(order, shift, coefficients, pSamplesOut + i);
+ if (drflac__use_64_bit_prediction(bitsPerSample, lpcOrder, lpcPrecision)) {
+ pSamplesOut[i] += drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + i);
} else {
- pSamplesOut[i] += drflac__calculate_prediction_32(order, shift, coefficients, pSamplesOut + i);
+ pSamplesOut[i] += drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + i);
}
}
return DRFLAC_TRUE;
}
-static drflac_bool32 drflac__decode_samples_with_residual(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 blockSize, drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pDecodedSamples)
+static drflac_bool32 drflac__decode_samples_with_residual(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 blockSize, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pDecodedSamples)
{
drflac_uint8 residualMethod;
drflac_uint8 partitionOrder;
@@ -60326,17 +81381,17 @@ static drflac_bool32 drflac__decode_samples_with_residual(drflac_bs* bs, drflac_
if (residualMethod != DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE && residualMethod != DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE2) {
return DRFLAC_FALSE;
}
- pDecodedSamples += order;
+ pDecodedSamples += lpcOrder;
if (!drflac__read_uint8(bs, 4, &partitionOrder)) {
return DRFLAC_FALSE;
}
if (partitionOrder > 8) {
return DRFLAC_FALSE;
}
- if ((blockSize / (1 << partitionOrder)) < order) {
+ if ((blockSize / (1 << partitionOrder)) < lpcOrder) {
return DRFLAC_FALSE;
}
- samplesInPartition = (blockSize / (1 << partitionOrder)) - order;
+ samplesInPartition = (blockSize / (1 << partitionOrder)) - lpcOrder;
partitionsRemaining = (1 << partitionOrder);
for (;;) {
drflac_uint8 riceParam = 0;
@@ -60356,7 +81411,7 @@ static drflac_bool32 drflac__decode_samples_with_residual(drflac_bs* bs, drflac_
}
}
if (riceParam != 0xFF) {
- if (!drflac__decode_samples_with_residual__rice(bs, bitsPerSample, samplesInPartition, riceParam, order, shift, coefficients, pDecodedSamples)) {
+ if (!drflac__decode_samples_with_residual__rice(bs, bitsPerSample, samplesInPartition, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pDecodedSamples)) {
return DRFLAC_FALSE;
}
} else {
@@ -60364,7 +81419,7 @@ static drflac_bool32 drflac__decode_samples_with_residual(drflac_bs* bs, drflac_
if (!drflac__read_uint8(bs, 5, &unencodedBitsPerSample)) {
return DRFLAC_FALSE;
}
- if (!drflac__decode_samples_with_residual__unencoded(bs, bitsPerSample, samplesInPartition, unencodedBitsPerSample, order, shift, coefficients, pDecodedSamples)) {
+ if (!drflac__decode_samples_with_residual__unencoded(bs, bitsPerSample, samplesInPartition, unencodedBitsPerSample, lpcOrder, lpcShift, lpcPrecision, coefficients, pDecodedSamples)) {
return DRFLAC_FALSE;
}
}
@@ -60484,7 +81539,7 @@ static drflac_bool32 drflac__decode_samples__fixed(drflac_bs* bs, drflac_uint32
}
pDecodedSamples[i] = sample;
}
- if (!drflac__decode_samples_with_residual(bs, subframeBitsPerSample, blockSize, lpcOrder, 0, lpcCoefficientsTable[lpcOrder], pDecodedSamples)) {
+ if (!drflac__decode_samples_with_residual(bs, subframeBitsPerSample, blockSize, lpcOrder, 0, 4, lpcCoefficientsTable[lpcOrder], pDecodedSamples)) {
return DRFLAC_FALSE;
}
return DRFLAC_TRUE;
@@ -60521,7 +81576,7 @@ static drflac_bool32 drflac__decode_samples__lpc(drflac_bs* bs, drflac_uint32 bl
return DRFLAC_FALSE;
}
}
- if (!drflac__decode_samples_with_residual(bs, bitsPerSample, blockSize, lpcOrder, lpcShift, coefficients, pDecodedSamples)) {
+ if (!drflac__decode_samples_with_residual(bs, bitsPerSample, blockSize, lpcOrder, lpcShift, lpcPrecision, coefficients, pDecodedSamples)) {
return DRFLAC_FALSE;
}
return DRFLAC_TRUE;
@@ -60630,6 +81685,9 @@ static drflac_bool32 drflac__read_next_flac_frame_header(drflac_bs* bs, drflac_u
return DRFLAC_FALSE;
}
crc8 = drflac_crc8(crc8, header->blockSizeInPCMFrames, 16);
+ if (header->blockSizeInPCMFrames == 0xFFFF) {
+ return DRFLAC_FALSE;
+ }
header->blockSizeInPCMFrames += 1;
} else {
DRFLAC_ASSERT(blockSize >= 8);
@@ -60662,6 +81720,9 @@ static drflac_bool32 drflac__read_next_flac_frame_header(drflac_bs* bs, drflac_u
if (header->bitsPerSample == 0) {
header->bitsPerSample = streaminfoBitsPerSample;
}
+ if (header->bitsPerSample != streaminfoBitsPerSample) {
+ return DRFLAC_FALSE;
+ }
if (!drflac__read_uint8(bs, 8, &header->crc8)) {
return DRFLAC_FALSE;
}
@@ -60732,6 +81793,9 @@ static drflac_bool32 drflac__decode_subframe(drflac_bs* bs, drflac_frame* frame,
} else if (frame->header.channelAssignment == DRFLAC_CHANNEL_ASSIGNMENT_RIGHT_SIDE && subframeIndex == 0) {
subframeBitsPerSample += 1;
}
+ if (subframeBitsPerSample > 32) {
+ return DRFLAC_FALSE;
+ }
if (pSubframe->wastedBitsPerSample >= subframeBitsPerSample) {
return DRFLAC_FALSE;
}
@@ -61215,6 +82279,9 @@ static drflac_bool32 drflac__seek_to_pcm_frame__seek_table(drflac* pFlac, drflac
if (pFlac->pSeekpoints == NULL || pFlac->seekpointCount == 0) {
return DRFLAC_FALSE;
}
+ if (pFlac->pSeekpoints[0].firstPCMFrame > pcmFrameIndex) {
+ return DRFLAC_FALSE;
+ }
for (iSeekpoint = 0; iSeekpoint < pFlac->seekpointCount; ++iSeekpoint) {
if (pFlac->pSeekpoints[iSeekpoint].firstPCMFrame >= pcmFrameIndex) {
break;
@@ -61558,13 +82625,13 @@ static drflac_bool32 drflac__read_and_decode_metadata(drflac_read_proc onRead, d
metadata.rawDataSize = blockSize;
pRunningData = (const char*)pRawData;
pRunningDataEnd = (const char*)pRawData + blockSize;
- metadata.data.vorbis_comment.vendorLength = drflac__le2host_32(*(const drflac_uint32*)pRunningData); pRunningData += 4;
+ metadata.data.vorbis_comment.vendorLength = drflac__le2host_32_ptr_unaligned(pRunningData); pRunningData += 4;
if ((pRunningDataEnd - pRunningData) - 4 < (drflac_int64)metadata.data.vorbis_comment.vendorLength) {
drflac__free_from_callbacks(pRawData, pAllocationCallbacks);
return DRFLAC_FALSE;
}
metadata.data.vorbis_comment.vendor = pRunningData; pRunningData += metadata.data.vorbis_comment.vendorLength;
- metadata.data.vorbis_comment.commentCount = drflac__le2host_32(*(const drflac_uint32*)pRunningData); pRunningData += 4;
+ metadata.data.vorbis_comment.commentCount = drflac__le2host_32_ptr_unaligned(pRunningData); pRunningData += 4;
if ((pRunningDataEnd - pRunningData) / sizeof(drflac_uint32) < metadata.data.vorbis_comment.commentCount) {
drflac__free_from_callbacks(pRawData, pAllocationCallbacks);
return DRFLAC_FALSE;
@@ -61576,7 +82643,7 @@ static drflac_bool32 drflac__read_and_decode_metadata(drflac_read_proc onRead, d
drflac__free_from_callbacks(pRawData, pAllocationCallbacks);
return DRFLAC_FALSE;
}
- commentLength = drflac__le2host_32(*(const drflac_uint32*)pRunningData); pRunningData += 4;
+ commentLength = drflac__le2host_32_ptr_unaligned(pRunningData); pRunningData += 4;
if (pRunningDataEnd - pRunningData < (drflac_int64)commentLength) {
drflac__free_from_callbacks(pRawData, pAllocationCallbacks);
return DRFLAC_FALSE;
@@ -61660,24 +82727,24 @@ static drflac_bool32 drflac__read_and_decode_metadata(drflac_read_proc onRead, d
metadata.rawDataSize = blockSize;
pRunningData = (const char*)pRawData;
pRunningDataEnd = (const char*)pRawData + blockSize;
- metadata.data.picture.type = drflac__be2host_32(*(const drflac_uint32*)pRunningData); pRunningData += 4;
- metadata.data.picture.mimeLength = drflac__be2host_32(*(const drflac_uint32*)pRunningData); pRunningData += 4;
+ metadata.data.picture.type = drflac__be2host_32_ptr_unaligned(pRunningData); pRunningData += 4;
+ metadata.data.picture.mimeLength = drflac__be2host_32_ptr_unaligned(pRunningData); pRunningData += 4;
if ((pRunningDataEnd - pRunningData) - 24 < (drflac_int64)metadata.data.picture.mimeLength) {
drflac__free_from_callbacks(pRawData, pAllocationCallbacks);
return DRFLAC_FALSE;
}
metadata.data.picture.mime = pRunningData; pRunningData += metadata.data.picture.mimeLength;
- metadata.data.picture.descriptionLength = drflac__be2host_32(*(const drflac_uint32*)pRunningData); pRunningData += 4;
+ metadata.data.picture.descriptionLength = drflac__be2host_32_ptr_unaligned(pRunningData); pRunningData += 4;
if ((pRunningDataEnd - pRunningData) - 20 < (drflac_int64)metadata.data.picture.descriptionLength) {
drflac__free_from_callbacks(pRawData, pAllocationCallbacks);
return DRFLAC_FALSE;
}
metadata.data.picture.description = pRunningData; pRunningData += metadata.data.picture.descriptionLength;
- metadata.data.picture.width = drflac__be2host_32(*(const drflac_uint32*)pRunningData); pRunningData += 4;
- metadata.data.picture.height = drflac__be2host_32(*(const drflac_uint32*)pRunningData); pRunningData += 4;
- metadata.data.picture.colorDepth = drflac__be2host_32(*(const drflac_uint32*)pRunningData); pRunningData += 4;
- metadata.data.picture.indexColorCount = drflac__be2host_32(*(const drflac_uint32*)pRunningData); pRunningData += 4;
- metadata.data.picture.pictureDataSize = drflac__be2host_32(*(const drflac_uint32*)pRunningData); pRunningData += 4;
+ metadata.data.picture.width = drflac__be2host_32_ptr_unaligned(pRunningData); pRunningData += 4;
+ metadata.data.picture.height = drflac__be2host_32_ptr_unaligned(pRunningData); pRunningData += 4;
+ metadata.data.picture.colorDepth = drflac__be2host_32_ptr_unaligned(pRunningData); pRunningData += 4;
+ metadata.data.picture.indexColorCount = drflac__be2host_32_ptr_unaligned(pRunningData); pRunningData += 4;
+ metadata.data.picture.pictureDataSize = drflac__be2host_32_ptr_unaligned(pRunningData); pRunningData += 4;
metadata.data.picture.pPictureData = (const drflac_uint8*)pRunningData;
if (pRunningDataEnd - pRunningData < (drflac_int64)metadata.data.picture.pictureDataSize) {
drflac__free_from_callbacks(pRawData, pAllocationCallbacks);
@@ -65798,7 +86865,7 @@ DRFLAC_API const char* drflac_next_vorbis_comment(drflac_vorbis_comment_iterator
if (pIter == NULL || pIter->countRemaining == 0 || pIter->pRunningData == NULL) {
return NULL;
}
- length = drflac__le2host_32(*(const drflac_uint32*)pIter->pRunningData);
+ length = drflac__le2host_32_ptr_unaligned(pIter->pRunningData);
pIter->pRunningData += 4;
pComment = pIter->pRunningData;
pIter->pRunningData += length;
@@ -69134,1093 +90201,6 @@ DRMP3_API void drmp3_free(void* p, const drmp3_allocation_callbacks* pAllocation
#endif /* miniaudio_c */
#endif /* MINIAUDIO_IMPLEMENTATION */
-/*
-RELEASE NOTES - VERSION 0.10.x
-==============================
-Version 0.10 includes major API changes and refactoring, mostly concerned with the data conversion system. Data conversion is performed internally to convert
-audio data between the format requested when initializing the `ma_device` object and the format of the internal device used by the backend. The same applies
-to the `ma_decoder` object. The previous design has several design flaws and missing features which necessitated a complete redesign.
-
-
-Changes to Data Conversion
---------------------------
-The previous data conversion system used callbacks to deliver input data for conversion. This design works well in some specific situations, but in other
-situations it has some major readability and maintenance issues. The decision was made to replace this with a more iterative approach where you just pass in a
-pointer to the input data directly rather than dealing with a callback.
-
-The following are the data conversion APIs that have been removed and their replacements:
-
- - ma_format_converter -> ma_convert_pcm_frames_format()
- - ma_channel_router -> ma_channel_converter
- - ma_src -> ma_resampler
- - ma_pcm_converter -> ma_data_converter
-
-The previous conversion APIs accepted a callback in their configs. There are no longer any callbacks to deal with. Instead you just pass the data into the
-`*_process_pcm_frames()` function as a pointer to a buffer.
-
-The simplest aspect of data conversion is sample format conversion. To convert between two formats, just call `ma_convert_pcm_frames_format()`. Channel
-conversion is also simple which you can do with `ma_channel_converter` via `ma_channel_converter_process_pcm_frames()`.
-
-Resampling is more complicated because the number of output frames that are processed is different to the number of input frames that are consumed. When you
-call `ma_resampler_process_pcm_frames()` you need to pass in the number of input frames available for processing and the number of output frames you want to
-output. Upon returning they will receive the number of input frames that were consumed and the number of output frames that were generated.
-
-The `ma_data_converter` API is a wrapper around format, channel and sample rate conversion and handles all of the data conversion you'll need which probably
-makes it the best option if you need to do data conversion.
-
-In addition to changes to the API design, a few other changes have been made to the data conversion pipeline:
-
- - The sinc resampler has been removed. This was completely broken and never actually worked properly.
- - The linear resampler now uses low-pass filtering to remove aliasing. The quality of the low-pass filter can be controlled via the resampler config with the
- `lpfOrder` option, which has a maximum value of MA_MAX_FILTER_ORDER.
- - Data conversion now supports s16 natively which runs through a fixed point pipeline. Previously everything needed to be converted to floating point before
- processing, whereas now both s16 and f32 are natively supported. Other formats still require conversion to either s16 or f32 prior to processing, however
- `ma_data_converter` will handle this for you.
-
-
-Custom Memory Allocators
-------------------------
-miniaudio has always supported macro level customization for memory allocation via MA_MALLOC, MA_REALLOC and MA_FREE, however some scenarios require more
-flexibility by allowing a user data pointer to be passed to the custom allocation routines. Support for this has been added to version 0.10 via the
-`ma_allocation_callbacks` structure. Anything making use of heap allocations has been updated to accept this new structure.
-
-The `ma_context_config` structure has been updated with a new member called `allocationCallbacks`. Leaving this set to it's defaults returned by
-`ma_context_config_init()` will cause it to use MA_MALLOC, MA_REALLOC and MA_FREE. Likewise, The `ma_decoder_config` structure has been updated in the same
-way, and leaving everything as-is after `ma_decoder_config_init()` will cause it to use the same defaults.
-
-The following APIs have been updated to take a pointer to a `ma_allocation_callbacks` object. Setting this parameter to NULL will cause it to use defaults.
-Otherwise they will use the relevant callback in the structure.
-
- - ma_malloc()
- - ma_realloc()
- - ma_free()
- - ma_aligned_malloc()
- - ma_aligned_free()
- - ma_rb_init() / ma_rb_init_ex()
- - ma_pcm_rb_init() / ma_pcm_rb_init_ex()
-
-Note that you can continue to use MA_MALLOC, MA_REALLOC and MA_FREE as per normal. These will continue to be used by default if you do not specify custom
-allocation callbacks.
-
-
-Buffer and Period Configuration Changes
----------------------------------------
-The way in which the size of the internal buffer and periods are specified in the device configuration have changed. In previous versions, the config variables
-`bufferSizeInFrames` and `bufferSizeInMilliseconds` defined the size of the entire buffer, with the size of a period being the size of this variable divided by
-the period count. This became confusing because people would expect the value of `bufferSizeInFrames` or `bufferSizeInMilliseconds` to independantly determine
-latency, when in fact it was that value divided by the period count that determined it. These variables have been removed and replaced with new ones called
-`periodSizeInFrames` and `periodSizeInMilliseconds`.
-
-These new configuration variables work in the same way as their predecessors in that if one is set to 0, the other will be used, but the main difference is
-that you now set these to you desired latency rather than the size of the entire buffer. The benefit of this is that it's much easier and less confusing to
-configure latency.
-
-The following unused APIs have been removed:
-
- ma_get_default_buffer_size_in_milliseconds()
- ma_get_default_buffer_size_in_frames()
-
-The following macros have been removed:
-
- MA_BASE_BUFFER_SIZE_IN_MILLISECONDS_LOW_LATENCY
- MA_BASE_BUFFER_SIZE_IN_MILLISECONDS_CONSERVATIVE
-
-
-Other API Changes
------------------
-Other less major API changes have also been made in version 0.10.
-
-`ma_device_set_stop_callback()` has been removed. If you require a stop callback, you must now set it via the device config just like the data callback.
-
-The `ma_sine_wave` API has been replaced with a more general API called `ma_waveform`. This supports generation of different types of waveforms, including
-sine, square, triangle and sawtooth. Use `ma_waveform_init()` in place of `ma_sine_wave_init()` to initialize the waveform object. This takes a configuration
-object called `ma_waveform_config` which defines the properties of the waveform. Use `ma_waveform_config_init()` to initialize a `ma_waveform_config` object.
-Use `ma_waveform_read_pcm_frames()` in place of `ma_sine_wave_read_f32()` and `ma_sine_wave_read_f32_ex()`.
-
-`ma_convert_frames()` and `ma_convert_frames_ex()` have been changed. Both of these functions now take a new parameter called `frameCountOut` which specifies
-the size of the output buffer in PCM frames. This has been added for safety. In addition to this, the parameters for `ma_convert_frames_ex()` have changed to
-take a pointer to a `ma_data_converter_config` object to specify the input and output formats to convert between. This was done to make it more flexible, to
-prevent the parameter list getting too long, and to prevent API breakage whenever a new conversion property is added.
-
-`ma_calculate_frame_count_after_src()` has been renamed to `ma_calculate_frame_count_after_resampling()` for consistency with the new `ma_resampler` API.
-
-
-Filters
--------
-The following filters have been added:
-
- |-------------|-------------------------------------------------------------------|
- | API | Description |
- |-------------|-------------------------------------------------------------------|
- | ma_biquad | Biquad filter (transposed direct form 2) |
- | ma_lpf1 | First order low-pass filter |
- | ma_lpf2 | Second order low-pass filter |
- | ma_lpf | High order low-pass filter (Butterworth) |
- | ma_hpf1 | First order high-pass filter |
- | ma_hpf2 | Second order high-pass filter |
- | ma_hpf | High order high-pass filter (Butterworth) |
- | ma_bpf2 | Second order band-pass filter |
- | ma_bpf | High order band-pass filter |
- | ma_peak2 | Second order peaking filter |
- | ma_notch2 | Second order notching filter |
- | ma_loshelf2 | Second order low shelf filter |
- | ma_hishelf2 | Second order high shelf filter |
- |-------------|-------------------------------------------------------------------|
-
-These filters all support 32-bit floating point and 16-bit signed integer formats natively. Other formats need to be converted beforehand.
-
-
-Sine, Square, Triangle and Sawtooth Waveforms
----------------------------------------------
-Previously miniaudio supported only sine wave generation. This has now been generalized to support sine, square, triangle and sawtooth waveforms. The old
-`ma_sine_wave` API has been removed and replaced with the `ma_waveform` API. Use `ma_waveform_config_init()` to initialize a config object, and then pass it
-into `ma_waveform_init()`. Then use `ma_waveform_read_pcm_frames()` to read PCM data.
-
-
-Noise Generation
-----------------
-A noise generation API has been added. This is used via the `ma_noise` API. Currently white, pink and Brownian noise is supported. The `ma_noise` API is
-similar to the waveform API. Use `ma_noise_config_init()` to initialize a config object, and then pass it into `ma_noise_init()` to initialize a `ma_noise`
-object. Then use `ma_noise_read_pcm_frames()` to read PCM data.
-
-
-Miscellaneous Changes
----------------------
-The MA_NO_STDIO option has been removed. This would disable file I/O APIs, however this has proven to be too hard to maintain for it's perceived value and was
-therefore removed.
-
-Internal functions have all been made static where possible. If you get warnings about unused functions, please submit a bug report.
-
-The `ma_device` structure is no longer defined as being aligned to MA_SIMD_ALIGNMENT. This resulted in a possible crash when allocating a `ma_device` object on
-the heap, but not aligning it to MA_SIMD_ALIGNMENT. This crash would happen due to the compiler seeing the alignment specified on the structure and assuming it
-was always aligned as such and thinking it was safe to emit alignment-dependant SIMD instructions. Since miniaudio's philosophy is for things to just work,
-this has been removed from all structures.
-
-Results codes have been overhauled. Unnecessary result codes have been removed, and some have been renumbered for organisation purposes. If you are are binding
-maintainer you will need to update your result codes. Support has also been added for retrieving a human readable description of a given result code via the
-`ma_result_description()` API.
-
-ALSA: The automatic format conversion, channel conversion and resampling performed by ALSA is now disabled by default as they were causing some compatibility
-issues with certain devices and configurations. These can be individually enabled via the device config:
-
- ```c
- deviceConfig.alsa.noAutoFormat = MA_TRUE;
- deviceConfig.alsa.noAutoChannels = MA_TRUE;
- deviceConfig.alsa.noAutoResample = MA_TRUE;
- ```
-*/
-
-/*
-RELEASE NOTES - VERSION 0.9.x
-=============================
-Version 0.9 includes major API changes, centered mostly around full-duplex and the rebrand to "miniaudio". Before I go into detail about the major changes I
-would like to apologize. I know it's annoying dealing with breaking API changes, but I think it's best to get these changes out of the way now while the
-library is still relatively young and unknown.
-
-There's been a lot of refactoring with this release so there's a good chance a few bugs have been introduced. I apologize in advance for this. You may want to
-hold off on upgrading for the short term if you're worried. If mini_al v0.8.14 works for you, and you don't need full-duplex support, you can avoid upgrading
-(though you won't be getting future bug fixes).
-
-
-Rebranding to "miniaudio"
--------------------------
-The decision was made to rename mini_al to miniaudio. Don't worry, it's the same project. The reason for this is simple:
-
-1) Having the word "audio" in the title makes it immediately clear that the library is related to audio; and
-2) I don't like the look of the underscore.
-
-This rebrand has necessitated a change in namespace from "mal" to "ma". I know this is annoying, and I apologize, but it's better to get this out of the road
-now rather than later. Also, since there are necessary API changes for full-duplex support I think it's better to just get the namespace change over and done
-with at the same time as the full-duplex changes. I'm hoping this will be the last of the major API changes. Fingers crossed!
-
-The implementation define is now "#define MINIAUDIO_IMPLEMENTATION". You can also use "#define MA_IMPLEMENTATION" if that's your preference.
-
-
-Full-Duplex Support
--------------------
-The major feature added to version 0.9 is full-duplex. This has necessitated a few API changes.
-
-1) The data callback has now changed. Previously there was one type of callback for playback and another for capture. I wanted to avoid a third callback just
- for full-duplex so the decision was made to break this API and unify the callbacks. Now, there is just one callback which is the same for all three modes
- (playback, capture, duplex). The new callback looks like the following:
-
- void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount);
-
- This callback allows you to move data straight out of the input buffer and into the output buffer in full-duplex mode. In playback-only mode, pInput will be
- null. Likewise, pOutput will be null in capture-only mode. The sample count is no longer returned from the callback since it's not necessary for miniaudio
- anymore.
-
-2) The device config needed to change in order to support full-duplex. Full-duplex requires the ability to allow the client to choose a different PCM format
- for the playback and capture sides. The old ma_device_config object simply did not allow this and needed to change. With these changes you now specify the
- device ID, format, channels, channel map and share mode on a per-playback and per-capture basis (see example below). The sample rate must be the same for
- playback and capture.
-
- Since the device config API has changed I have also decided to take the opportunity to simplify device initialization. Now, the device ID, device type and
- callback user data are set in the config. ma_device_init() is now simplified down to taking just the context, device config and a pointer to the device
- object being initialized. The rationale for this change is that it just makes more sense to me that these are set as part of the config like everything
- else.
-
- Example device initialization:
-
- ma_device_config config = ma_device_config_init(ma_device_type_duplex); // Or ma_device_type_playback or ma_device_type_capture.
- config.playback.pDeviceID = &myPlaybackDeviceID; // Or NULL for the default playback device.
- config.playback.format = ma_format_f32;
- config.playback.channels = 2;
- config.capture.pDeviceID = &myCaptureDeviceID; // Or NULL for the default capture device.
- config.capture.format = ma_format_s16;
- config.capture.channels = 1;
- config.sampleRate = 44100;
- config.dataCallback = data_callback;
- config.pUserData = &myUserData;
-
- result = ma_device_init(&myContext, &config, &device);
- if (result != MA_SUCCESS) {
- ... handle error ...
- }
-
- Note that the "onDataCallback" member of ma_device_config has been renamed to "dataCallback". Also, "onStopCallback" has been renamed to "stopCallback".
-
-This is the first pass for full-duplex and there is a known bug. You will hear crackling on the following backends when sample rate conversion is required for
-the playback device:
- - Core Audio
- - JACK
- - AAudio
- - OpenSL
- - WebAudio
-
-In addition to the above, not all platforms have been absolutely thoroughly tested simply because I lack the hardware for such thorough testing. If you
-experience a bug, an issue report on GitHub or an email would be greatly appreciated (and a sample program that reproduces the issue if possible).
-
-
-Other API Changes
------------------
-In addition to the above, the following API changes have been made:
-
-- The log callback is no longer passed to ma_context_config_init(). Instead you need to set it manually after initialization.
-- The onLogCallback member of ma_context_config has been renamed to "logCallback".
-- The log callback now takes a logLevel parameter. The new callback looks like: void log_callback(ma_context* pContext, ma_device* pDevice, ma_uint32 logLevel, const char* message)
- - You can use ma_log_level_to_string() to convert the logLevel to human readable text if you want to log it.
-- Some APIs have been renamed:
- - mal_decoder_read() -> ma_decoder_read_pcm_frames()
- - mal_decoder_seek_to_frame() -> ma_decoder_seek_to_pcm_frame()
- - mal_sine_wave_read() -> ma_sine_wave_read_f32()
- - mal_sine_wave_read_ex() -> ma_sine_wave_read_f32_ex()
-- Some APIs have been removed:
- - mal_device_get_buffer_size_in_bytes()
- - mal_device_set_recv_callback()
- - mal_device_set_send_callback()
- - mal_src_set_input_sample_rate()
- - mal_src_set_output_sample_rate()
-- Error codes have been rearranged. If you're a binding maintainer you will need to update.
-- The ma_backend enums have been rearranged to priority order. The rationale for this is to simplify automatic backend selection and to make it easier to see
- the priority. If you're a binding maintainer you will need to update.
-- ma_dsp has been renamed to ma_pcm_converter. The rationale for this change is that I'm expecting "ma_dsp" to conflict with some future planned high-level
- APIs.
-- For functions that take a pointer/count combo, such as ma_decoder_read_pcm_frames(), the parameter order has changed so that the pointer comes before the
- count. The rationale for this is to keep it consistent with things like memcpy().
-
-
-Miscellaneous Changes
----------------------
-The following miscellaneous changes have also been made.
-
-- The AAudio backend has been added for Android 8 and above. This is Android's new "High-Performance Audio" API. (For the record, this is one of the nicest
- audio APIs out there, just behind the BSD audio APIs).
-- The WebAudio backend has been added. This is based on ScriptProcessorNode. This removes the need for SDL.
-- The SDL and OpenAL backends have been removed. These were originally implemented to add support for platforms for which miniaudio was not explicitly
- supported. These are no longer needed and have therefore been removed.
-- Device initialization now fails if the requested share mode is not supported. If you ask for exclusive mode, you either get an exclusive mode device, or an
- error. The rationale for this change is to give the client more control over how to handle cases when the desired shared mode is unavailable.
-- A lock-free ring buffer API has been added. There are two varients of this. "ma_rb" operates on bytes, whereas "ma_pcm_rb" operates on PCM frames.
-- The library is now licensed as a choice of Public Domain (Unlicense) _or_ MIT-0 (No Attribution) which is the same as MIT, but removes the attribution
- requirement. The rationale for this is to support countries that don't recognize public domain.
-*/
-
-/*
-REVISION HISTORY
-================
-v0.10.42 - 2021-08-22
- - Fix a possible deadlock when stopping devices.
-
-v0.10.41 - 2021-08-15
- - Core Audio: Fix some deadlock errors.
-
-v0.10.40 - 2021-07-23
- - Fix a bug when converting from stereo to mono.
- - PulseAudio: Fix a glitch when pausing and resuming a device.
-
-v0.10.39 - 2021-07-20
- - Core Audio: Fix a deadlock when the default device is changed.
- - Core Audio: Fix compilation errors on macOS and iOS.
- - PulseAudio: Fix a bug where the stop callback is not fired when a device is unplugged.
- - PulseAudio: Fix a null pointer dereference.
-
-v0.10.38 - 2021-07-14
- - Fix a linking error when MA_DEBUG_OUTPUT is not enabled.
- - Fix an error where ma_log_postv() does not return a value.
- - OpenSL: Fix a bug with setting of stream types and recording presets.
-
-0.10.37 - 2021-07-06
- - Fix a bug with log message formatting.
- - Fix build when compiling with MA_NO_THREADING.
- - Minor updates to channel mapping.
-
-0.10.36 - 2021-07-03
- - Add support for custom decoding backends.
- - Fix some bugs with the Vorbis decoder.
- - PulseAudio: Fix a bug with channel mapping.
- - PulseAudio: Fix a bug where miniaudio does not fall back to a supported format when PulseAudio
- defaults to a format not known to miniaudio.
- - OpenSL: Fix a crash when initializing a capture device when a recording preset other than the
- default is specified.
- - Silence some warnings when compiling with MA_DEBUG_OUTPUT
- - Improvements to logging. See the `ma_log` API for details. The logCallback variable used by
- ma_context has been deprecated and will be replaced with the new system in version 0.11.
- - Initialize an `ma_log` object with `ma_log_init()`.
- - Register a callback with `ma_log_register_callback()`.
- - In the context config, set `pLog` to your `ma_log` object and stop using `logCallback`.
- - Prep work for some upcoming changes to data sources. These changes are still compatible with
- existing code, however code will need to be updated in preparation for version 0.11 which will
- be breaking. You should make these changes now for any custom data sources:
- - Change your base data source object from `ma_data_source_callbacks` to `ma_data_source_base`.
- - Call `ma_data_source_init()` for your base object in your custom data source's initialization
- routine. This takes a config object which includes a pointer to a vtable which is now where
- your custom callbacks are defined.
- - Call `ma_data_source_uninit()` in your custom data source's uninitialization routine. This
- doesn't currently do anything, but it placeholder in case some future uninitialization code
- is required to be added at a later date.
-
-v0.10.35 - 2021-04-27
- - Fix the C++ build.
-
-v0.10.34 - 2021-04-26
- - WASAPI: Fix a bug where a result code is not getting checked at initialization time.
- - WASAPI: Bug fixes for loopback mode.
- - ALSA: Fix a possible deadlock when stopping devices.
- - Mark devices as default on the null backend.
-
-v0.10.33 - 2021-04-04
- - Core Audio: Fix a memory leak.
- - Core Audio: Fix a bug where the performance profile is not being used by playback devices.
- - JACK: Fix loading of 64-bit JACK on Windows.
- - Fix a calculation error and add a safety check to the following APIs to prevent a division by zero:
- - ma_calculate_buffer_size_in_milliseconds_from_frames()
- - ma_calculate_buffer_size_in_frames_from_milliseconds()
- - Fix compilation errors relating to c89atomic.
- - Update FLAC decoder.
-
-v0.10.32 - 2021-02-23
- - WASAPI: Fix a deadlock in exclusive mode.
- - WASAPI: No longer return an error from ma_context_get_device_info() when an exclusive mode format
- cannot be retrieved.
- - WASAPI: Attempt to fix some bugs with device uninitialization.
- - PulseAudio: Yet another refactor, this time to remove the dependency on `pa_threaded_mainloop`.
- - Web Audio: Fix a bug on Chrome and any other browser using the same engine.
- - Web Audio: Automatically start the device on some user input if the device has been started. This
- is to work around Google's policy of not starting audio if no user input has yet been performed.
- - Fix a bug where thread handles are not being freed.
- - Fix some static analysis warnings in FLAC, WAV and MP3 decoders.
- - Fix a warning due to referencing _MSC_VER when it is undefined.
- - Update to latest version of c89atomic.
- - Internal refactoring to migrate over to the new backend callback system for the following backends:
- - PulseAudio
- - ALSA
- - Core Audio
- - AAudio
- - OpenSL|ES
- - OSS
- - audio(4)
- - sndio
-
-v0.10.31 - 2021-01-17
- - Make some functions const correct.
- - Update ma_data_source_read_pcm_frames() to initialize pFramesRead to 0 for safety.
- - Add the MA_ATOMIC annotation for use with variables that should be used atomically and remove unnecessary volatile qualifiers.
- - Add support for enabling only specific backends at compile time. This is the reverse of the pre-existing system. With the new
- system, all backends are first disabled with `MA_ENABLE_ONLY_SPECIFIC_BACKENDS`, which is then followed with `MA_ENABLE_*`. The
- old system where you disable backends with `MA_NO_*` still exists and is still the default.
-
-v0.10.30 - 2021-01-10
- - Fix a crash in ma_audio_buffer_read_pcm_frames().
- - Update spinlock APIs to take a volatile parameter as input.
- - Silence some unused parameter warnings.
- - Fix a warning on GCC when compiling as C++.
-
-v0.10.29 - 2020-12-26
- - Fix some subtle multi-threading bugs on non-x86 platforms.
- - Fix a bug resulting in superfluous memory allocations when enumerating devices.
- - Core Audio: Fix a compilation error when compiling for iOS.
-
-v0.10.28 - 2020-12-16
- - Fix a crash when initializing a POSIX thread.
- - OpenSL|ES: Respect the MA_NO_RUNTIME_LINKING option.
-
-v0.10.27 - 2020-12-04
- - Add support for dynamically configuring some properties of `ma_noise` objects post-initialization.
- - Add support for configuring the channel mixing mode in the device config.
- - Fix a bug with simple channel mixing mode (drop or silence excess channels).
- - Fix some bugs with trying to access uninitialized variables.
- - Fix some errors with stopping devices for synchronous backends where the backend's stop callback would get fired twice.
- - Fix a bug in the decoder due to using an uninitialized variable.
- - Fix some data race errors.
-
-v0.10.26 - 2020-11-24
- - WASAPI: Fix a bug where the exclusive mode format may not be retrieved correctly due to accessing freed memory.
- - Fix a bug with ma_waveform where glitching occurs after changing frequency.
- - Fix compilation with OpenWatcom.
- - Fix compilation with TCC.
- - Fix compilation with Digital Mars.
- - Fix compilation warnings.
- - Remove bitfields from public structures to aid in binding maintenance.
-
-v0.10.25 - 2020-11-15
- - PulseAudio: Fix a bug where the stop callback isn't fired.
- - WebAudio: Fix an error that occurs when Emscripten increases the size of it's heap.
- - Custom Backends: Change the onContextInit and onDeviceInit callbacks to take a parameter which is a pointer to the config that was
- passed into ma_context_init() and ma_device_init(). This replaces the deviceType parameter of onDeviceInit.
- - Fix compilation warnings on older versions of GCC.
-
-v0.10.24 - 2020-11-10
- - Fix a bug where initialization of a backend can fail due to some bad state being set from a prior failed attempt at initializing a
- lower priority backend.
-
-v0.10.23 - 2020-11-09
- - AAudio: Add support for configuring a playback stream's usage.
- - Fix a compilation error when all built-in asynchronous backends are disabled at compile time.
- - Fix compilation errors when compiling as C++.
-
-v0.10.22 - 2020-11-08
- - Add support for custom backends.
- - Add support for detecting default devices during device enumeration and with `ma_context_get_device_info()`.
- - Refactor to the PulseAudio backend. This simplifies the implementation and fixes a capture bug.
- - ALSA: Fix a bug in `ma_context_get_device_info()` where the PCM handle is left open in the event of an error.
- - Core Audio: Further improvements to sample rate selection.
- - Core Audio: Fix some bugs with capture mode.
- - OpenSL: Add support for configuring stream types and recording presets.
- - AAudio: Add support for configuring content types and input presets.
- - Fix bugs in `ma_decoder_init_file*()` where the file handle is not closed after a decoding error.
- - Fix some compilation warnings on GCC and Clang relating to the Speex resampler.
- - Fix a compilation error for the Linux build when the ALSA and JACK backends are both disabled.
- - Fix a compilation error for the BSD build.
- - Fix some compilation errors on older versions of GCC.
- - Add documentation for `MA_NO_RUNTIME_LINKING`.
-
-v0.10.21 - 2020-10-30
- - Add ma_is_backend_enabled() and ma_get_enabled_backends() for retrieving enabled backends at run-time.
- - WASAPI: Fix a copy and paste bug relating to loopback mode.
- - Core Audio: Fix a bug when using multiple contexts.
- - Core Audio: Fix a compilation warning.
- - Core Audio: Improvements to sample rate selection.
- - Core Audio: Improvements to format/channels/rate selection when requesting defaults.
- - Core Audio: Add notes regarding the Apple notarization process.
- - Fix some bugs due to null pointer dereferences.
-
-v0.10.20 - 2020-10-06
- - Fix build errors with UWP.
- - Minor documentation updates.
-
-v0.10.19 - 2020-09-22
- - WASAPI: Return an error when exclusive mode is requested, but the native format is not supported by miniaudio.
- - Fix a bug where ma_decoder_seek_to_pcm_frames() never returns MA_SUCCESS even though it was successful.
- - Store the sample rate in the `ma_lpf` and `ma_hpf` structures.
-
-v0.10.18 - 2020-08-30
- - Fix build errors with VC6.
- - Fix a bug in channel converter for s32 format.
- - Change channel converter configs to use the default channel map instead of a blank channel map when no channel map is specified when initializing the
- config. This fixes an issue where the optimized mono expansion path would never get used.
- - Use a more appropriate default format for FLAC decoders. This will now use ma_format_s16 when the FLAC is encoded as 16-bit.
- - Update FLAC decoder.
- - Update links to point to the new repository location (https://github.com/mackron/miniaudio).
-
-v0.10.17 - 2020-08-28
- - Fix an error where the WAV codec is incorrectly excluded from the build depending on which compile time options are set.
- - Fix a bug in ma_audio_buffer_read_pcm_frames() where it isn't returning the correct number of frames processed.
- - Fix compilation error on Android.
- - Core Audio: Fix a bug with full-duplex mode.
- - Add ma_decoder_get_cursor_in_pcm_frames().
- - Update WAV codec.
-
-v0.10.16 - 2020-08-14
- - WASAPI: Fix a potential crash due to using an uninitialized variable.
- - OpenSL: Enable runtime linking.
- - OpenSL: Fix a multithreading bug when initializing and uninitializing multiple contexts at the same time.
- - iOS: Improvements to device enumeration.
- - Fix a crash in ma_data_source_read_pcm_frames() when the output frame count parameter is NULL.
- - Fix a bug in ma_data_source_read_pcm_frames() where looping doesn't work.
- - Fix some compilation warnings on Windows when both DirectSound and WinMM are disabled.
- - Fix some compilation warnings when no decoders are enabled.
- - Add ma_audio_buffer_get_available_frames().
- - Add ma_decoder_get_available_frames().
- - Add sample rate to ma_data_source_get_data_format().
- - Change volume APIs to take 64-bit frame counts.
- - Updates to documentation.
-
-v0.10.15 - 2020-07-15
- - Fix a bug when converting bit-masked channel maps to miniaudio channel maps. This affects the WASAPI and OpenSL backends.
-
-v0.10.14 - 2020-07-14
- - Fix compilation errors on Android.
- - Fix compilation errors with -march=armv6.
- - Updates to the documentation.
-
-v0.10.13 - 2020-07-11
- - Fix some potential buffer overflow errors with channel maps when channel counts are greater than MA_MAX_CHANNELS.
- - Fix compilation error on Emscripten.
- - Silence some unused function warnings.
- - Increase the default buffer size on the Web Audio backend. This fixes glitching issues on some browsers.
- - Bring FLAC decoder up-to-date with dr_flac.
- - Bring MP3 decoder up-to-date with dr_mp3.
-
-v0.10.12 - 2020-07-04
- - Fix compilation errors on the iOS build.
-
-v0.10.11 - 2020-06-28
- - Fix some bugs with device tracking on Core Audio.
- - Updates to documentation.
-
-v0.10.10 - 2020-06-26
- - Add include guard for the implementation section.
- - Mark ma_device_sink_info_callback() as static.
- - Fix compilation errors with MA_NO_DECODING and MA_NO_ENCODING.
- - Fix compilation errors with MA_NO_DEVICE_IO
-
-v0.10.9 - 2020-06-24
- - Amalgamation of dr_wav, dr_flac and dr_mp3. With this change, including the header section of these libraries before the implementation of miniaudio is no
- longer required. Decoding of WAV, FLAC and MP3 should be supported seamlessly without any additional libraries. Decoders can be excluded from the build
- with the following options:
- - MA_NO_WAV
- - MA_NO_FLAC
- - MA_NO_MP3
- If you get errors about multiple definitions you need to either enable the options above, move the implementation of dr_wav, dr_flac and/or dr_mp3 to before
- the implementation of miniaudio, or update dr_wav, dr_flac and/or dr_mp3.
- - Changes to the internal atomics library. This has been replaced with c89atomic.h which is embedded within this file.
- - Fix a bug when a decoding backend reports configurations outside the limits of miniaudio's decoder abstraction.
- - Fix the UWP build.
- - Fix the Core Audio build.
- - Fix the -std=c89 build on GCC.
-
-v0.10.8 - 2020-06-22
- - Remove dependency on ma_context from mutexes.
- - Change ma_data_source_read_pcm_frames() to return a result code and output the frames read as an output parameter.
- - Change ma_data_source_seek_pcm_frames() to return a result code and output the frames seeked as an output parameter.
- - Change ma_audio_buffer_unmap() to return MA_AT_END when the end has been reached. This should be considered successful.
- - Change playback.pDeviceID and capture.pDeviceID to constant pointers in ma_device_config.
- - Add support for initializing decoders from a virtual file system object. This is achieved via the ma_vfs API and allows the application to customize file
- IO for the loading and reading of raw audio data. Passing in NULL for the VFS will use defaults. New APIs:
- - ma_decoder_init_vfs()
- - ma_decoder_init_vfs_wav()
- - ma_decoder_init_vfs_flac()
- - ma_decoder_init_vfs_mp3()
- - ma_decoder_init_vfs_vorbis()
- - ma_decoder_init_vfs_w()
- - ma_decoder_init_vfs_wav_w()
- - ma_decoder_init_vfs_flac_w()
- - ma_decoder_init_vfs_mp3_w()
- - ma_decoder_init_vfs_vorbis_w()
- - Add support for memory mapping to ma_data_source.
- - ma_data_source_map()
- - ma_data_source_unmap()
- - Add ma_offset_pcm_frames_ptr() and ma_offset_pcm_frames_const_ptr() which can be used for offsetting a pointer by a specified number of PCM frames.
- - Add initial implementation of ma_yield() which is useful for spin locks which will be used in some upcoming work.
- - Add documentation for log levels.
- - The ma_event API has been made public in preparation for some uncoming work.
- - Fix a bug in ma_decoder_seek_to_pcm_frame() where the internal sample rate is not being taken into account for determining the seek location.
- - Fix some bugs with the linear resampler when dynamically changing the sample rate.
- - Fix compilation errors with MA_NO_DEVICE_IO.
- - Fix some warnings with GCC and -std=c89.
- - Fix some formatting warnings with GCC and -Wall and -Wpedantic.
- - Fix some warnings with VC6.
- - Minor optimization to ma_copy_pcm_frames(). This is now a no-op when the input and output buffers are the same.
-
-v0.10.7 - 2020-05-25
- - Fix a compilation error in the C++ build.
- - Silence a warning.
-
-v0.10.6 - 2020-05-24
- - Change ma_clip_samples_f32() and ma_clip_pcm_frames_f32() to take a 64-bit sample/frame count.
- - Change ma_zero_pcm_frames() to clear to 128 for ma_format_u8.
- - Add ma_silence_pcm_frames() which replaces ma_zero_pcm_frames(). ma_zero_pcm_frames() will be removed in version 0.11.
- - Add support for u8, s24 and s32 formats to ma_channel_converter.
- - Add compile-time and run-time version querying.
- - MA_VERSION_MINOR
- - MA_VERSION_MAJOR
- - MA_VERSION_REVISION
- - MA_VERSION_STRING
- - ma_version()
- - ma_version_string()
- - Add ma_audio_buffer for reading raw audio data directly from memory.
- - Fix a bug in shuffle mode in ma_channel_converter.
- - Fix compilation errors in certain configurations for ALSA and PulseAudio.
- - The data callback now initializes the output buffer to 128 when the playback sample format is ma_format_u8.
-
-v0.10.5 - 2020-05-05
- - Change ma_zero_pcm_frames() to take a 64-bit frame count.
- - Add ma_copy_pcm_frames().
- - Add MA_NO_GENERATION build option to exclude the `ma_waveform` and `ma_noise` APIs from the build.
- - Add support for formatted logging to the VC6 build.
- - Fix a crash in the linear resampler when LPF order is 0.
- - Fix compilation errors and warnings with older versions of Visual Studio.
- - Minor documentation updates.
-
-v0.10.4 - 2020-04-12
- - Fix a data conversion bug when converting from the client format to the native device format.
-
-v0.10.3 - 2020-04-07
- - Bring up to date with breaking changes to dr_mp3.
- - Remove MA_NO_STDIO. This was causing compilation errors and the maintenance cost versus practical benefit is no longer worthwhile.
- - Fix a bug with data conversion where it was unnecessarily converting to s16 or f32 and then straight back to the original format.
- - Fix compilation errors and warnings with Visual Studio 2005.
- - ALSA: Disable ALSA's automatic data conversion by default and add configuration options to the device config:
- - alsa.noAutoFormat
- - alsa.noAutoChannels
- - alsa.noAutoResample
- - WASAPI: Add some overrun recovery for ma_device_type_capture devices.
-
-v0.10.2 - 2020-03-22
- - Decorate some APIs with MA_API which were missed in the previous version.
- - Fix a bug in ma_linear_resampler_set_rate() and ma_linear_resampler_set_rate_ratio().
-
-v0.10.1 - 2020-03-17
- - Add MA_API decoration. This can be customized by defining it before including miniaudio.h.
- - Fix a bug where opening a file would return a success code when in fact it failed.
- - Fix compilation errors with Visual Studio 6 and 2003.
- - Fix warnings on macOS.
-
-v0.10.0 - 2020-03-07
- - API CHANGE: Refactor data conversion APIs
- - ma_format_converter has been removed. Use ma_convert_pcm_frames_format() instead.
- - ma_channel_router has been replaced with ma_channel_converter.
- - ma_src has been replaced with ma_resampler
- - ma_pcm_converter has been replaced with ma_data_converter
- - API CHANGE: Add support for custom memory allocation callbacks. The following APIs have been updated to take an extra parameter for the allocation
- callbacks:
- - ma_malloc()
- - ma_realloc()
- - ma_free()
- - ma_aligned_malloc()
- - ma_aligned_free()
- - ma_rb_init() / ma_rb_init_ex()
- - ma_pcm_rb_init() / ma_pcm_rb_init_ex()
- - API CHANGE: Simplify latency specification in device configurations. The bufferSizeInFrames and bufferSizeInMilliseconds parameters have been replaced with
- periodSizeInFrames and periodSizeInMilliseconds respectively. The previous variables defined the size of the entire buffer, whereas the new ones define the
- size of a period. The following APIs have been removed since they are no longer relevant:
- - ma_get_default_buffer_size_in_milliseconds()
- - ma_get_default_buffer_size_in_frames()
- - API CHANGE: ma_device_set_stop_callback() has been removed. If you require a stop callback, you must now set it via the device config just like the data
- callback.
- - API CHANGE: The ma_sine_wave API has been replaced with ma_waveform. The following APIs have been removed:
- - ma_sine_wave_init()
- - ma_sine_wave_read_f32()
- - ma_sine_wave_read_f32_ex()
- - API CHANGE: ma_convert_frames() has been updated to take an extra parameter which is the size of the output buffer in PCM frames. Parameters have also been
- reordered.
- - API CHANGE: ma_convert_frames_ex() has been changed to take a pointer to a ma_data_converter_config object to specify the input and output formats to
- convert between.
- - API CHANGE: ma_calculate_frame_count_after_src() has been renamed to ma_calculate_frame_count_after_resampling().
- - Add support for the following filters:
- - Biquad (ma_biquad)
- - First order low-pass (ma_lpf1)
- - Second order low-pass (ma_lpf2)
- - Low-pass with configurable order (ma_lpf)
- - First order high-pass (ma_hpf1)
- - Second order high-pass (ma_hpf2)
- - High-pass with configurable order (ma_hpf)
- - Second order band-pass (ma_bpf2)
- - Band-pass with configurable order (ma_bpf)
- - Second order peaking EQ (ma_peak2)
- - Second order notching (ma_notch2)
- - Second order low shelf (ma_loshelf2)
- - Second order high shelf (ma_hishelf2)
- - Add waveform generation API (ma_waveform) with support for the following:
- - Sine
- - Square
- - Triangle
- - Sawtooth
- - Add noise generation API (ma_noise) with support for the following:
- - White
- - Pink
- - Brownian
- - Add encoding API (ma_encoder). This only supports outputting to WAV files via dr_wav.
- - Add ma_result_description() which is used to retrieve a human readable description of a given result code.
- - Result codes have been changed. Binding maintainers will need to update their result code constants.
- - More meaningful result codes are now returned when a file fails to open.
- - Internal functions have all been made static where possible.
- - Fix potential crash when ma_device object's are not aligned to MA_SIMD_ALIGNMENT.
- - Fix a bug in ma_decoder_get_length_in_pcm_frames() where it was returning the length based on the internal sample rate rather than the output sample rate.
- - Fix bugs in some backends where the device is not drained properly in ma_device_stop().
- - Improvements to documentation.
-
-v0.9.10 - 2020-01-15
- - Fix compilation errors due to #if/#endif mismatches.
- - WASAPI: Fix a bug where automatic stream routing is being performed for devices that are initialized with an explicit device ID.
- - iOS: Fix a crash on device uninitialization.
-
-v0.9.9 - 2020-01-09
- - Fix compilation errors with MinGW.
- - Fix compilation errors when compiling on Apple platforms.
- - WASAPI: Add support for disabling hardware offloading.
- - WASAPI: Add support for disabling automatic stream routing.
- - Core Audio: Fix bugs in the case where the internal device uses deinterleaved buffers.
- - Core Audio: Add support for controlling the session category (AVAudioSessionCategory) and options (AVAudioSessionCategoryOptions).
- - JACK: Fix bug where incorrect ports are connected.
-
-v0.9.8 - 2019-10-07
- - WASAPI: Fix a potential deadlock when starting a full-duplex device.
- - WASAPI: Enable automatic resampling by default. Disable with config.wasapi.noAutoConvertSRC.
- - Core Audio: Fix bugs with automatic stream routing.
- - Add support for controlling whether or not the content of the output buffer passed in to the data callback is pre-initialized
- to zero. By default it will be initialized to zero, but this can be changed by setting noPreZeroedOutputBuffer in the device
- config. Setting noPreZeroedOutputBuffer to true will leave the contents undefined.
- - Add support for clipping samples after the data callback has returned. This only applies when the playback sample format is
- configured as ma_format_f32. If you are doing clipping yourself, you can disable this overhead by setting noClip to true in
- the device config.
- - Add support for master volume control for devices.
- - Use ma_device_set_master_volume() to set the volume to a factor between 0 and 1, where 0 is silence and 1 is full volume.
- - Use ma_device_set_master_gain_db() to set the volume in decibels where 0 is full volume and < 0 reduces the volume.
- - Fix warnings emitted by GCC when `__inline__` is undefined or defined as nothing.
-
-v0.9.7 - 2019-08-28
- - Add support for loopback mode (WASAPI only).
- - To use this, set the device type to ma_device_type_loopback, and then fill out the capture section of the device config.
- - If you need to capture from a specific output device, set the capture device ID to that of a playback device.
- - Fix a crash when an error is posted in ma_device_init().
- - Fix a compilation error when compiling for ARM architectures.
- - Fix a bug with the audio(4) backend where the device is incorrectly being opened in non-blocking mode.
- - Fix memory leaks in the Core Audio backend.
- - Minor refactoring to the WinMM, ALSA, PulseAudio, OSS, audio(4), sndio and null backends.
-
-v0.9.6 - 2019-08-04
- - Add support for loading decoders using a wchar_t string for file paths.
- - Don't trigger an assert when ma_device_start() is called on a device that is already started. This will now log a warning
- and return MA_INVALID_OPERATION. The same applies for ma_device_stop().
- - Try fixing an issue with PulseAudio taking a long time to start playback.
- - Fix a bug in ma_convert_frames() and ma_convert_frames_ex().
- - Fix memory leaks in the WASAPI backend.
- - Fix a compilation error with Visual Studio 2010.
-
-v0.9.5 - 2019-05-21
- - Add logging to ma_dlopen() and ma_dlsym().
- - Add ma_decoder_get_length_in_pcm_frames().
- - Fix a bug with capture on the OpenSL|ES backend.
- - Fix a bug with the ALSA backend where a device would not restart after being stopped.
-
-v0.9.4 - 2019-05-06
- - Add support for C89. With this change, miniaudio should compile clean with GCC/Clang with "-std=c89 -ansi -pedantic" and
- Microsoft compilers back to VC6. Other compilers should also work, but have not been tested.
-
-v0.9.3 - 2019-04-19
- - Fix compiler errors on GCC when compiling with -std=c99.
-
-v0.9.2 - 2019-04-08
- - Add support for per-context user data.
- - Fix a potential bug with context configs.
- - Fix some bugs with PulseAudio.
-
-v0.9.1 - 2019-03-17
- - Fix a bug where the output buffer is not getting zeroed out before calling the data callback. This happens when
- the device is running in passthrough mode (not doing any data conversion).
- - Fix an issue where the data callback is getting called too frequently on the WASAPI and DirectSound backends.
- - Fix error on the UWP build.
- - Fix a build error on Apple platforms.
-
-v0.9 - 2019-03-06
- - Rebranded to "miniaudio". All namespaces have been renamed from "mal" to "ma".
- - API CHANGE: ma_device_init() and ma_device_config_init() have changed significantly:
- - The device type, device ID and user data pointer have moved from ma_device_init() to the config.
- - All variations of ma_device_config_init_*() have been removed in favor of just ma_device_config_init().
- - ma_device_config_init() now takes only one parameter which is the device type. All other properties need
- to be set on the returned object directly.
- - The onDataCallback and onStopCallback members of ma_device_config have been renamed to "dataCallback"
- and "stopCallback".
- - The ID of the physical device is now split into two: one for the playback device and the other for the
- capture device. This is required for full-duplex. These are named "pPlaybackDeviceID" and "pCaptureDeviceID".
- - API CHANGE: The data callback has changed. It now uses a unified callback for all device types rather than
- being separate for each. It now takes two pointers - one containing input data and the other output data. This
- design in required for full-duplex. The return value is now void instead of the number of frames written. The
- new callback looks like the following:
- void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount);
- - API CHANGE: Remove the log callback parameter from ma_context_config_init(). With this change,
- ma_context_config_init() now takes no parameters and the log callback is set via the structure directly. The
- new policy for config initialization is that only mandatory settings are passed in to *_config_init(). The
- "onLog" member of ma_context_config has been renamed to "logCallback".
- - API CHANGE: Remove ma_device_get_buffer_size_in_bytes().
- - API CHANGE: Rename decoding APIs to "pcm_frames" convention.
- - mal_decoder_read() -> ma_decoder_read_pcm_frames()
- - mal_decoder_seek_to_frame() -> ma_decoder_seek_to_pcm_frame()
- - API CHANGE: Rename sine wave reading APIs to f32 convention.
- - mal_sine_wave_read() -> ma_sine_wave_read_f32()
- - mal_sine_wave_read_ex() -> ma_sine_wave_read_f32_ex()
- - API CHANGE: Remove some deprecated APIs
- - mal_device_set_recv_callback()
- - mal_device_set_send_callback()
- - mal_src_set_input_sample_rate()
- - mal_src_set_output_sample_rate()
- - API CHANGE: Add log level to the log callback. New signature:
- - void on_log(ma_context* pContext, ma_device* pDevice, ma_uint32 logLevel, const char* message)
- - API CHANGE: Changes to result codes. Constants have changed and unused codes have been removed. If you're
- a binding mainainer you will need to update your result code constants.
- - API CHANGE: Change the order of the ma_backend enums to priority order. If you are a binding maintainer, you
- will need to update.
- - API CHANGE: Rename mal_dsp to ma_pcm_converter. All functions have been renamed from mal_dsp_*() to
- ma_pcm_converter_*(). All structures have been renamed from mal_dsp* to ma_pcm_converter*.
- - API CHANGE: Reorder parameters of ma_decoder_read_pcm_frames() to be consistent with the new parameter order scheme.
- - The resampling algorithm has been changed from sinc to linear. The rationale for this is that the sinc implementation
- is too inefficient right now. This will hopefully be improved at a later date.
- - Device initialization will no longer fall back to shared mode if exclusive mode is requested but is unusable.
- With this change, if you request an device in exclusive mode, but exclusive mode is not supported, it will not
- automatically fall back to shared mode. The client will need to reinitialize the device in shared mode if that's
- what they want.
- - Add ring buffer API. This is ma_rb and ma_pcm_rb, the difference being that ma_rb operates on bytes and
- ma_pcm_rb operates on PCM frames.
- - Add Web Audio backend. This is used when compiling with Emscripten. The SDL backend, which was previously
- used for web support, will be removed in a future version.
- - Add AAudio backend (Android Audio). This is the new priority backend for Android. Support for AAudio starts
- with Android 8. OpenSL|ES is used as a fallback for older versions of Android.
- - Remove OpenAL and SDL backends.
- - Fix a possible deadlock when rapidly stopping the device after it has started.
- - Update documentation.
- - Change licensing to a choice of public domain _or_ MIT-0 (No Attribution).
-
-v0.8.14 - 2018-12-16
- - Core Audio: Fix a bug where the device state is not set correctly after stopping.
- - Add support for custom weights to the channel router.
- - Update decoders to use updated APIs in dr_flac, dr_mp3 and dr_wav.
-
-v0.8.13 - 2018-12-04
- - Core Audio: Fix a bug with channel mapping.
- - Fix a bug with channel routing where the back/left and back/right channels have the wrong weight.
-
-v0.8.12 - 2018-11-27
- - Drop support for SDL 1.2. The Emscripten build now requires "-s USE_SDL=2".
- - Fix a linking error with ALSA.
- - Fix a bug on iOS where the device name is not set correctly.
-
-v0.8.11 - 2018-11-21
- - iOS bug fixes.
- - Minor tweaks to PulseAudio.
-
-v0.8.10 - 2018-10-21
- - Core Audio: Fix a hang when uninitializing a device.
- - Fix a bug where an incorrect value is returned from mal_device_stop().
-
-v0.8.9 - 2018-09-28
- - Fix a bug with the SDL backend where device initialization fails.
-
-v0.8.8 - 2018-09-14
- - Fix Linux build with the ALSA backend.
- - Minor documentation fix.
-
-v0.8.7 - 2018-09-12
- - Fix a bug with UWP detection.
-
-v0.8.6 - 2018-08-26
- - Automatically switch the internal device when the default device is unplugged. Note that this is still in the
- early stages and not all backends handle this the same way. As of this version, this will not detect a default
- device switch when changed from the operating system's audio preferences (unless the backend itself handles
- this automatically). This is not supported in exclusive mode.
- - WASAPI and Core Audio: Add support for stream routing. When the application is using a default device and the
- user switches the default device via the operating system's audio preferences, miniaudio will automatically switch
- the internal device to the new default. This is not supported in exclusive mode.
- - WASAPI: Add support for hardware offloading via IAudioClient2. Only supported on Windows 8 and newer.
- - WASAPI: Add support for low-latency shared mode via IAudioClient3. Only supported on Windows 10 and newer.
- - Add support for compiling the UWP build as C.
- - mal_device_set_recv_callback() and mal_device_set_send_callback() have been deprecated. You must now set this
- when the device is initialized with mal_device_init*(). These will be removed in version 0.9.0.
-
-v0.8.5 - 2018-08-12
- - Add support for specifying the size of a device's buffer in milliseconds. You can still set the buffer size in
- frames if that suits you. When bufferSizeInFrames is 0, bufferSizeInMilliseconds will be used. If both are non-0
- then bufferSizeInFrames will take priority. If both are set to 0 the default buffer size is used.
- - Add support for the audio(4) backend to OpenBSD.
- - Fix a bug with the ALSA backend that was causing problems on Raspberry Pi. This significantly improves the
- Raspberry Pi experience.
- - Fix a bug where an incorrect number of samples is returned from sinc resampling.
- - Add support for setting the value to be passed to internal calls to CoInitializeEx().
- - WASAPI and WinMM: Stop the device when it is unplugged.
-
-v0.8.4 - 2018-08-06
- - Add sndio backend for OpenBSD.
- - Add audio(4) backend for NetBSD.
- - Drop support for the OSS backend on everything except FreeBSD and DragonFly BSD.
- - Formats are now native-endian (were previously little-endian).
- - Mark some APIs as deprecated:
- - mal_src_set_input_sample_rate() and mal_src_set_output_sample_rate() are replaced with mal_src_set_sample_rate().
- - mal_dsp_set_input_sample_rate() and mal_dsp_set_output_sample_rate() are replaced with mal_dsp_set_sample_rate().
- - Fix a bug when capturing using the WASAPI backend.
- - Fix some aliasing issues with resampling, specifically when increasing the sample rate.
- - Fix warnings.
-
-v0.8.3 - 2018-07-15
- - Fix a crackling bug when resampling in capture mode.
- - Core Audio: Fix a bug where capture does not work.
- - ALSA: Fix a bug where the worker thread can get stuck in an infinite loop.
- - PulseAudio: Fix a bug where mal_context_init() succeeds when PulseAudio is unusable.
- - JACK: Fix a bug where mal_context_init() succeeds when JACK is unusable.
-
-v0.8.2 - 2018-07-07
- - Fix a bug on macOS with Core Audio where the internal callback is not called.
-
-v0.8.1 - 2018-07-06
- - Fix compilation errors and warnings.
-
-v0.8 - 2018-07-05
- - Changed MAL_IMPLEMENTATION to MINI_AL_IMPLEMENTATION for consistency with other libraries. The old
- way is still supported for now, but you should update as it may be removed in the future.
- - API CHANGE: Replace device enumeration APIs. mal_enumerate_devices() has been replaced with
- mal_context_get_devices(). An additional low-level device enumration API has been introduced called
- mal_context_enumerate_devices() which uses a callback to report devices.
- - API CHANGE: Rename mal_get_sample_size_in_bytes() to mal_get_bytes_per_sample() and add
- mal_get_bytes_per_frame().
- - API CHANGE: Replace mal_device_config.preferExclusiveMode with mal_device_config.shareMode.
- - This new config can be set to mal_share_mode_shared (default) or mal_share_mode_exclusive.
- - API CHANGE: Remove excludeNullDevice from mal_context_config.alsa.
- - API CHANGE: Rename MAL_MAX_SAMPLE_SIZE_IN_BYTES to MAL_MAX_PCM_SAMPLE_SIZE_IN_BYTES.
- - API CHANGE: Change the default channel mapping to the standard Microsoft mapping.
- - API CHANGE: Remove backend-specific result codes.
- - API CHANGE: Changes to the format conversion APIs (mal_pcm_f32_to_s16(), etc.)
- - Add support for Core Audio (Apple).
- - Add support for PulseAudio.
- - This is the highest priority backend on Linux (higher priority than ALSA) since it is commonly
- installed by default on many of the popular distros and offer's more seamless integration on
- platforms where PulseAudio is used. In addition, if PulseAudio is installed and running (which
- is extremely common), it's better to just use PulseAudio directly rather than going through the
- "pulse" ALSA plugin (which is what the "default" ALSA device is likely set to).
- - Add support for JACK.
- - Remove dependency on asound.h for the ALSA backend. This means the ALSA development packages are no
- longer required to build miniaudio.
- - Remove dependency on dsound.h for the DirectSound backend. This fixes build issues with some
- distributions of MinGW.
- - Remove dependency on audioclient.h for the WASAPI backend. This fixes build issues with some
- distributions of MinGW.
- - Add support for dithering to format conversion.
- - Add support for configuring the priority of the worker thread.
- - Add a sine wave generator.
- - Improve efficiency of sample rate conversion.
- - Introduce the notion of standard channel maps. Use mal_get_standard_channel_map().
- - Introduce the notion of default device configurations. A default config uses the same configuration
- as the backend's internal device, and as such results in a pass-through data transmission pipeline.
- - Add support for passing in NULL for the device config in mal_device_init(), which uses a default
- config. This requires manually calling mal_device_set_send/recv_callback().
- - Add support for decoding from raw PCM data (mal_decoder_init_raw(), etc.)
- - Make mal_device_init_ex() more robust.
- - Make some APIs more const-correct.
- - Fix errors with SDL detection on Apple platforms.
- - Fix errors with OpenAL detection.
- - Fix some memory leaks.
- - Fix a bug with opening decoders from memory.
- - Early work on SSE2, AVX2 and NEON optimizations.
- - Miscellaneous bug fixes.
- - Documentation updates.
-
-v0.7 - 2018-02-25
- - API CHANGE: Change mal_src_read_frames() and mal_dsp_read_frames() to use 64-bit sample counts.
- - Add decoder APIs for loading WAV, FLAC, Vorbis and MP3 files.
- - Allow opening of devices without a context.
- - In this case the context is created and managed internally by the device.
- - Change the default channel mapping to the same as that used by FLAC.
- - Fix build errors with macOS.
-
-v0.6c - 2018-02-12
- - Fix build errors with BSD/OSS.
-
-v0.6b - 2018-02-03
- - Fix some warnings when compiling with Visual C++.
-
-v0.6a - 2018-01-26
- - Fix errors with channel mixing when increasing the channel count.
- - Improvements to the build system for the OpenAL backend.
- - Documentation fixes.
-
-v0.6 - 2017-12-08
- - API CHANGE: Expose and improve mutex APIs. If you were using the mutex APIs before this version you'll
- need to update.
- - API CHANGE: SRC and DSP callbacks now take a pointer to a mal_src and mal_dsp object respectively.
- - API CHANGE: Improvements to event and thread APIs. These changes make these APIs more consistent.
- - Add support for SDL and Emscripten.
- - Simplify the build system further for when development packages for various backends are not installed.
- With this change, when the compiler supports __has_include, backends without the relevant development
- packages installed will be ignored. This fixes the build for old versions of MinGW.
- - Fixes to the Android build.
- - Add mal_convert_frames(). This is a high-level helper API for performing a one-time, bulk conversion of
- audio data to a different format.
- - Improvements to f32 -> u8/s16/s24/s32 conversion routines.
- - Fix a bug where the wrong value is returned from mal_device_start() for the OpenSL backend.
- - Fixes and improvements for Raspberry Pi.
- - Warning fixes.
-
-v0.5 - 2017-11-11
- - API CHANGE: The mal_context_init() function now takes a pointer to a mal_context_config object for
- configuring the context. The works in the same kind of way as the device config. The rationale for this
- change is to give applications better control over context-level properties, add support for backend-
- specific configurations, and support extensibility without breaking the API.
- - API CHANGE: The alsa.preferPlugHW device config variable has been removed since it's not really useful for
- anything anymore.
- - ALSA: By default, device enumeration will now only enumerate over unique card/device pairs. Applications
- can enable verbose device enumeration by setting the alsa.useVerboseDeviceEnumeration context config
- variable.
- - ALSA: When opening a device in shared mode (the default), the dmix/dsnoop plugin will be prioritized. If
- this fails it will fall back to the hw plugin. With this change the preferExclusiveMode config is now
- honored. Note that this does not happen when alsa.useVerboseDeviceEnumeration is set to true (see above)
- which is by design.
- - ALSA: Add support for excluding the "null" device using the alsa.excludeNullDevice context config variable.
- - ALSA: Fix a bug with channel mapping which causes an assertion to fail.
- - Fix errors with enumeration when pInfo is set to NULL.
- - OSS: Fix a bug when starting a device when the client sends 0 samples for the initial buffer fill.
-
-v0.4 - 2017-11-05
- - API CHANGE: The log callback is now per-context rather than per-device and as is thus now passed to
- mal_context_init(). The rationale for this change is that it allows applications to capture diagnostic
- messages at the context level. Previously this was only available at the device level.
- - API CHANGE: The device config passed to mal_device_init() is now const.
- - Added support for OSS which enables support on BSD platforms.
- - Added support for WinMM (waveOut/waveIn).
- - Added support for UWP (Universal Windows Platform) applications. Currently C++ only.
- - Added support for exclusive mode for selected backends. Currently supported on WASAPI.
- - POSIX builds no longer require explicit linking to libpthread (-lpthread).
- - ALSA: Explicit linking to libasound (-lasound) is no longer required.
- - ALSA: Latency improvements.
- - ALSA: Use MMAP mode where available. This can be disabled with the alsa.noMMap config.
- - ALSA: Use "hw" devices instead of "plughw" devices by default. This can be disabled with the
- alsa.preferPlugHW config.
- - WASAPI is now the highest priority backend on Windows platforms.
- - Fixed an error with sample rate conversion which was causing crackling when capturing.
- - Improved error handling.
- - Improved compiler support.
- - Miscellaneous bug fixes.
-
-v0.3 - 2017-06-19
- - API CHANGE: Introduced the notion of a context. The context is the highest level object and is required for
- enumerating and creating devices. Now, applications must first create a context, and then use that to
- enumerate and create devices. The reason for this change is to ensure device enumeration and creation is
- tied to the same backend. In addition, some backends are better suited to this design.
- - API CHANGE: Removed the rewinding APIs because they're too inconsistent across the different backends, hard
- to test and maintain, and just generally unreliable.
- - Added helper APIs for initializing mal_device_config objects.
- - Null Backend: Fixed a crash when recording.
- - Fixed build for UWP.
- - Added support for f32 formats to the OpenSL|ES backend.
- - Added initial implementation of the WASAPI backend.
- - Added initial implementation of the OpenAL backend.
- - Added support for low quality linear sample rate conversion.
- - Added early support for basic channel mapping.
-
-v0.2 - 2016-10-28
- - API CHANGE: Add user data pointer as the last parameter to mal_device_init(). The rationale for this
- change is to ensure the logging callback has access to the user data during initialization.
- - API CHANGE: Have device configuration properties be passed to mal_device_init() via a structure. Rationale:
- 1) The number of parameters is just getting too much.
- 2) It makes it a bit easier to add new configuration properties in the future. In particular, there's a
- chance there will be support added for backend-specific properties.
- - Dropped support for f64, A-law and Mu-law formats since they just aren't common enough to justify the
- added maintenance cost.
- - DirectSound: Increased the default buffer size for capture devices.
- - Added initial implementation of the OpenSL|ES backend.
-
-v0.1 - 2016-10-21
- - Initial versioned release.
-*/
-
/*
This software is available as a choice of the following licenses. Choose
diff --git a/src/raudio.c b/src/raudio.c
index ccd156c1e..8397abbbb 100644
--- a/src/raudio.c
+++ b/src/raudio.c
@@ -372,7 +372,7 @@ static AudioData AUDIO = { // Global AUDIO context
//----------------------------------------------------------------------------------
// Module specific Functions Declaration
//----------------------------------------------------------------------------------
-static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message);
+static void OnLog(void *pUserData, ma_uint32 level, const char *pMessage);
static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount);
static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, AudioBuffer *buffer);
@@ -411,7 +411,7 @@ void InitAudioDevice(void)
{
// Init audio context
ma_context_config ctxConfig = ma_context_config_init();
- ctxConfig.logCallback = OnLog;
+ ma_log_callback_init(OnLog, NULL);
ma_result result = ma_context_init(NULL, 0, &ctxConfig, &AUDIO.System.context);
if (result != MA_SUCCESS)
@@ -492,7 +492,7 @@ void CloseAudioDevice(void)
//UnloadAudioBuffer(AUDIO.MultiChannel.pool[i]);
if (AUDIO.MultiChannel.pool[i] != NULL)
{
- ma_data_converter_uninit(&AUDIO.MultiChannel.pool[i]->converter);
+ ma_data_converter_uninit(&AUDIO.MultiChannel.pool[i]->converter, NULL);
UntrackAudioBuffer(AUDIO.MultiChannel.pool[i]);
//RL_FREE(buffer->data); // Already unloaded by UnloadSound()
RL_FREE(AUDIO.MultiChannel.pool[i]);
@@ -541,9 +541,9 @@ AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sam
// Audio data runs through a format converter
ma_data_converter_config converterConfig = ma_data_converter_config_init(format, AUDIO_DEVICE_FORMAT, channels, AUDIO_DEVICE_CHANNELS, sampleRate, AUDIO.System.device.sampleRate);
- converterConfig.resampling.allowDynamicSampleRate = true; // Pitch shifting
+ converterConfig.allowDynamicSampleRate = true;
- ma_result result = ma_data_converter_init(&converterConfig, &audioBuffer->converter);
+ ma_result result = ma_data_converter_init(&converterConfig, NULL, &audioBuffer->converter);
if (result != MA_SUCCESS)
{
@@ -580,7 +580,7 @@ void UnloadAudioBuffer(AudioBuffer *buffer)
{
if (buffer != NULL)
{
- ma_data_converter_uninit(&buffer->converter);
+ ma_data_converter_uninit(&buffer->converter, NULL);
UntrackAudioBuffer(buffer);
RL_FREE(buffer->data);
RL_FREE(buffer);
@@ -654,8 +654,8 @@ void SetAudioBufferPitch(AudioBuffer *buffer, float pitch)
// Note that this changes the duration of the sound:
// - higher pitches will make the sound faster
// - lower pitches make it slower
- ma_uint32 outputSampleRate = (ma_uint32)((float)buffer->converter.config.sampleRateOut/pitch);
- ma_data_converter_set_rate(&buffer->converter, buffer->converter.config.sampleRateIn, outputSampleRate);
+ ma_uint32 outputSampleRate = (ma_uint32)((float)buffer->converter.sampleRateOut/pitch);
+ ma_data_converter_set_rate(&buffer->converter, buffer->converter.sampleRateIn, outputSampleRate);
buffer->pitch = pitch;
}
@@ -894,7 +894,7 @@ void UpdateSound(Sound sound, const void *data, int sampleCount)
StopAudioBuffer(sound.stream.buffer);
// TODO: May want to lock/unlock this since this data buffer is read at mixing time
- memcpy(sound.stream.buffer->data, data, sampleCount*ma_get_bytes_per_frame(sound.stream.buffer->converter.config.formatIn, sound.stream.buffer->converter.config.channelsIn));
+ memcpy(sound.stream.buffer->data, data, sampleCount*ma_get_bytes_per_frame(sound.stream.buffer->converter.formatIn, sound.stream.buffer->converter.channelsIn));
}
}
@@ -2033,12 +2033,9 @@ void SetAudioStreamBufferSizeDefault(int size)
//----------------------------------------------------------------------------------
// Log callback function
-static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message)
+static void OnLog(void *pUserData, ma_uint32 level, const char *pMessage)
{
- (void)pContext;
- (void)pDevice;
-
- TRACELOG(LOG_WARNING, "miniaudio: %s", message); // All log messages from miniaudio are errors
+ TRACELOG(LOG_WARNING, "miniaudio: %s", pMessage); // All log messages from miniaudio are errors
}
// Reads audio data from an AudioBuffer object in internal format.
@@ -2055,7 +2052,7 @@ static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer,
isSubBufferProcessed[0] = audioBuffer->isSubBufferProcessed[0];
isSubBufferProcessed[1] = audioBuffer->isSubBufferProcessed[1];
- ma_uint32 frameSizeInBytes = ma_get_bytes_per_frame(audioBuffer->converter.config.formatIn, audioBuffer->converter.config.channelsIn);
+ ma_uint32 frameSizeInBytes = ma_get_bytes_per_frame(audioBuffer->converter.formatIn, audioBuffer->converter.channelsIn);
// Fill out every frame until we find a buffer that's marked as processed. Then fill the remainder with 0
ma_uint32 framesRead = 0;
@@ -2135,20 +2132,21 @@ static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, f
// detail to remember here is that we never, ever attempt to read more input data than is required for the specified number of output
// frames. This can be achieved with ma_data_converter_get_required_input_frame_count().
ma_uint8 inputBuffer[4096] = { 0 };
- ma_uint32 inputBufferFrameCap = sizeof(inputBuffer)/ma_get_bytes_per_frame(audioBuffer->converter.config.formatIn, audioBuffer->converter.config.channelsIn);
+ ma_uint32 inputBufferFrameCap = sizeof(inputBuffer)/ma_get_bytes_per_frame(audioBuffer->converter.formatIn, audioBuffer->converter.channelsIn);
ma_uint32 totalOutputFramesProcessed = 0;
while (totalOutputFramesProcessed < frameCount)
{
ma_uint64 outputFramesToProcessThisIteration = frameCount - totalOutputFramesProcessed;
+ ma_uint64 inputFramesToProcessThisIteration = 0;
- ma_uint64 inputFramesToProcessThisIteration = ma_data_converter_get_required_input_frame_count(&audioBuffer->converter, outputFramesToProcessThisIteration);
+ ma_result result = ma_data_converter_get_required_input_frame_count(&audioBuffer->converter, outputFramesToProcessThisIteration, &inputFramesToProcessThisIteration);
if (inputFramesToProcessThisIteration > inputBufferFrameCap)
{
inputFramesToProcessThisIteration = inputBufferFrameCap;
}
- float *runningFramesOut = framesOut + (totalOutputFramesProcessed*audioBuffer->converter.config.channelsOut);
+ float *runningFramesOut = framesOut + (totalOutputFramesProcessed*audioBuffer->converter.channelsOut);
/* At this point we can convert the data to our mixing format. */
ma_uint64 inputFramesProcessedThisIteration = ReadAudioBufferFramesInInternalFormat(audioBuffer, inputBuffer, (ma_uint32)inputFramesToProcessThisIteration); /* Safe cast. */
@@ -2282,7 +2280,7 @@ static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 fr
frameIn += 2;
}
}
- else // pan is kinda meaningless
+ else // pan is kinda meaningless
{
for (ma_uint32 iFrame = 0; iFrame < frameCount; ++iFrame)
{