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Update to miniaudio 0.10 (#1092) 

* Update to miniaudio 0.10

This replaces the old ma_pcm_converter with ma_data_converter.

At this time of this commit, miniaudio 0.10 is still in the testing
phase. To make it easier to update miniaudio.h during this period, I've
temporarily moved the @raysan5 Win32 customizations to raudio.c because
there may be quite a few updates to miniaudio.h during this time.

* Use miniaudio's built-in volume control.
pull/1108/head
David Reid 6 years ago
committed by GitHub
parent
d2bb6185f1
commit
7024628c65
No known key found for this signature in database GPG Key ID: 4AEE18F83AFDEB23
2 changed files with 11072 additions and 9865 deletions
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  1. +10838
    -9740
      src/external/miniaudio.h
  2. +234
    -125
      src/raudio.c

+ 10838
- 9740
src/external/miniaudio.h
File diff suppressed because it is too large
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+ 234
- 125
src/raudio.c View File

@ -78,6 +78,85 @@
#include "utils.h" // Required for: fopen() Android mapping
#endif
#if defined(_WIN32)
// @raysan5: To avoid conflicting windows.h symbols with raylib, so flags are defined
// WARNING: Those flags avoid inclusion of some Win32 headers that could be required
// by user at some point and won't be included...
//-------------------------------------------------------------------------------------
// If defined, the following flags inhibit definition of the indicated items.
#define NOGDICAPMASKS // CC_*, LC_*, PC_*, CP_*, TC_*, RC_
#define NOVIRTUALKEYCODES // VK_*
#define NOWINMESSAGES // WM_*, EM_*, LB_*, CB_*
#define NOWINSTYLES // WS_*, CS_*, ES_*, LBS_*, SBS_*, CBS_*
#define NOSYSMETRICS // SM_*
#define NOMENUS // MF_*
#define NOICONS // IDI_*
#define NOKEYSTATES // MK_*
#define NOSYSCOMMANDS // SC_*
#define NORASTEROPS // Binary and Tertiary raster ops
#define NOSHOWWINDOW // SW_*
#define OEMRESOURCE // OEM Resource values
#define NOATOM // Atom Manager routines
#define NOCLIPBOARD // Clipboard routines
#define NOCOLOR // Screen colors
#define NOCTLMGR // Control and Dialog routines
#define NODRAWTEXT // DrawText() and DT_*
#define NOGDI // All GDI defines and routines
#define NOKERNEL // All KERNEL defines and routines
#define NOUSER // All USER defines and routines
//#define NONLS // All NLS defines and routines
#define NOMB // MB_* and MessageBox()
#define NOMEMMGR // GMEM_*, LMEM_*, GHND, LHND, associated routines
#define NOMETAFILE // typedef METAFILEPICT
#define NOMINMAX // Macros min(a,b) and max(a,b)
#define NOMSG // typedef MSG and associated routines
#define NOOPENFILE // OpenFile(), OemToAnsi, AnsiToOem, and OF_*
#define NOSCROLL // SB_* and scrolling routines
#define NOSERVICE // All Service Controller routines, SERVICE_ equates, etc.
#define NOSOUND // Sound driver routines
#define NOTEXTMETRIC // typedef TEXTMETRIC and associated routines
#define NOWH // SetWindowsHook and WH_*
#define NOWINOFFSETS // GWL_*, GCL_*, associated routines
#define NOCOMM // COMM driver routines
#define NOKANJI // Kanji support stuff.
#define NOHELP // Help engine interface.
#define NOPROFILER // Profiler interface.
#define NODEFERWINDOWPOS // DeferWindowPos routines
#define NOMCX // Modem Configuration Extensions
// Type required before windows.h inclusion
typedef struct tagMSG *LPMSG;
#include <windows.h>
// Type required by some unused function...
typedef struct tagBITMAPINFOHEADER {
DWORD biSize;
LONG biWidth;
LONG biHeight;
WORD biPlanes;
WORD biBitCount;
DWORD biCompression;
DWORD biSizeImage;
LONG biXPelsPerMeter;
LONG biYPelsPerMeter;
DWORD biClrUsed;
DWORD biClrImportant;
} BITMAPINFOHEADER, *PBITMAPINFOHEADER;
#include <objbase.h>
#include <mmreg.h>
#include <mmsystem.h>
// @raysan5: Some required types defined for MSVC/TinyC compiler
#if defined(_MSC_VER) || defined(__TINYC__)
#include "propidl.h"
#endif
#endif
#define MA_NO_JACK
#define MINIAUDIO_IMPLEMENTATION
#include "external/miniaudio.h" // miniaudio library
@ -172,7 +251,7 @@ typedef enum {
// Audio buffer structure
struct rAudioBuffer {
ma_pcm_converter dsp; // PCM data converter
ma_data_converter converter; // Audio data converter
float volume; // Audio buffer volume
float pitch; // Audio buffer pitch
@ -185,7 +264,7 @@ struct rAudioBuffer {
bool isSubBufferProcessed[2]; // SubBuffer processed (virtual double buffer)
unsigned int sizeInFrames; // Total buffer size in frames
unsigned int frameCursorPos; // Frame cursor position
unsigned int totalFramesProcessed; // Total frames processed in this buffer (required for play timming)
unsigned int totalFramesProcessed; // Total frames processed in this buffer (required for play timing)
unsigned char *data; // Data buffer, on music stream keeps filling
@ -202,7 +281,6 @@ typedef struct AudioData {
ma_device device; // miniaudio device
ma_mutex lock; // miniaudio mutex lock
bool isReady; // Check if audio device is ready
float masterVolume; // Master volume (multiplied on output mixing)
} System;
struct {
AudioBuffer *first; // Pointer to first AudioBuffer in the list
@ -225,7 +303,6 @@ static AudioData AUDIO = { 0 }; // Global CORE context
//----------------------------------------------------------------------------------
static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message);
static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount);
static ma_uint32 OnAudioBufferDSPRead(ma_pcm_converter *pDSP, void *pFramesOut, ma_uint32 frameCount, void *pUserData);
static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume);
static void InitAudioBufferPool(void); // Initialise the multichannel buffer pool
@ -273,7 +350,6 @@ void UntrackAudioBuffer(AudioBuffer *buffer);
void InitAudioDevice(void)
{
// TODO: Load AUDIO context memory dynamically?
AUDIO.System.masterVolume = 1.0f;
// Init audio context
ma_context_config ctxConfig = ma_context_config_init();
@ -366,10 +442,7 @@ bool IsAudioDeviceReady(void)
// Set master volume (listener)
void SetMasterVolume(float volume)
{
if (volume < 0.0f) volume = 0.0f;
else if (volume > 1.0f) volume = 1.0f;
AUDIO.System.masterVolume = volume;
ma_device_set_master_volume(&AUDIO.System.device, volume);
}
//----------------------------------------------------------------------------------
@ -390,19 +463,10 @@ AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sam
audioBuffer->data = RL_CALLOC(sizeInFrames*channels*ma_get_bytes_per_sample(format), 1);
// Audio data runs through a format converter
ma_pcm_converter_config dspConfig;
memset(&dspConfig, 0, sizeof(dspConfig));
dspConfig.formatIn = format;
dspConfig.formatOut = DEVICE_FORMAT;
dspConfig.channelsIn = channels;
dspConfig.channelsOut = DEVICE_CHANNELS;
dspConfig.sampleRateIn = sampleRate;
dspConfig.sampleRateOut = DEVICE_SAMPLE_RATE;
dspConfig.onRead = OnAudioBufferDSPRead; // Callback on data reading
dspConfig.pUserData = audioBuffer; // Audio data pointer
dspConfig.allowDynamicSampleRate = true; // Required for pitch shifting
ma_result result = ma_pcm_converter_init(&dspConfig, &audioBuffer->dsp);
ma_data_converter_config converterConfig = ma_data_converter_config_init(format, DEVICE_FORMAT, channels, DEVICE_CHANNELS, sampleRate, DEVICE_SAMPLE_RATE);
converterConfig.resampling.allowDynamicSampleRate = true; // Required for pitch shifting
ma_result result = ma_data_converter_init(&converterConfig, &audioBuffer->converter);
if (result != MA_SUCCESS)
{
@ -437,6 +501,7 @@ void CloseAudioBuffer(AudioBuffer *buffer)
{
if (buffer != NULL)
{
ma_data_converter_uninit(&buffer->converter);
UntrackAudioBuffer(buffer);
RL_FREE(buffer->data);
RL_FREE(buffer);
@ -519,10 +584,10 @@ 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 newOutputSampleRate = (ma_uint32)((float)buffer->dsp.src.config.sampleRateOut/pitchMul);
buffer->pitch *= (float)buffer->dsp.src.config.sampleRateOut/newOutputSampleRate;
ma_uint32 newOutputSampleRate = (ma_uint32)((float)buffer->converter.config.sampleRateOut/pitchMul);
buffer->pitch *= (float)buffer->converter.config.sampleRateOut/newOutputSampleRate;
ma_pcm_converter_set_output_sample_rate(&buffer->dsp, newOutputSampleRate);
ma_data_converter_set_rate(&buffer->converter, buffer->converter.config.sampleRateIn, newOutputSampleRate);
}
else TRACELOG(LOG_WARNING, "SetAudioBufferPitch() : No audio buffer");
}
@ -621,13 +686,13 @@ Sound LoadSoundFromWave(Wave wave)
ma_format formatIn = ((wave.sampleSize == 8)? ma_format_u8 : ((wave.sampleSize == 16)? ma_format_s16 : ma_format_f32));
ma_uint32 frameCountIn = wave.sampleCount/wave.channels;
ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, NULL, formatIn, wave.channels, wave.sampleRate, frameCountIn);
ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, mi">0, DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, NULL, frameCountIn, formatIn, wave.channels, wave.sampleRate);
if (frameCount == 0) TRACELOG(LOG_WARNING, "LoadSoundFromWave() : Failed to get frame count for format conversion");
AudioBuffer *audioBuffer = InitAudioBuffer(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, frameCount, AUDIO_BUFFER_USAGE_STATIC);
if (audioBuffer == NULL) TRACELOG(LOG_WARNING, "LoadSoundFromWave() : Failed to create audio buffer");
frameCount = (ma_uint32)ma_convert_frames(audioBuffer->data, audioBuffer->dsp.formatConverterIn.config.formatIn, audioBuffer->dsp.formatConverterIn.config.channels, audioBuffer->dsp.src.config.sampleRateIn, wave.data, formatIn, wave.channels, wave.sampleRate, frameCountIn);
frameCount = (ma_uint32)ma_convert_frames(audioBuffer->data, frameCount, DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, wave.data, frameCountIn, formatIn, wave.channels, wave.sampleRate);
if (frameCount == 0) TRACELOG(LOG_WARNING, "LoadSoundFromWave() : Format conversion failed");
sound.sampleCount = frameCount*DEVICE_CHANNELS;
@ -666,7 +731,7 @@ void UpdateSound(Sound sound, const void *data, int samplesCount)
StopAudioBuffer(audioBuffer);
// TODO: May want to lock/unlock this since this data buffer is read at mixing time
memcpy(audioBuffer->data, data, samplesCount*audioBuffer->dsp.formatConverterIn.config.channels*ma_get_bytes_per_sample(audioBuffer->dspclass="p">.formatConverterIn.config.formatIn));
memcpy(audioBuffer->data, data, samplesCount*ma_get_bytes_per_frame(audioBuffer->converter.config.formatIn, audioBuffer->converter.config.channelsIn));
}
else TRACELOG(LOG_ERROR, "UpdateSound() : Invalid sound - no audio buffer");
}
@ -869,7 +934,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
ma_uint32 frameCountIn = wave->sampleCount; // Is wave->sampleCount actually the frame count? That terminology needs to change, if so.
ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, formatOut, channels, sampleRate, NULL, formatIn, wave->channels, wave->sampleRate, frameCountIn);
ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, mi">0, formatOut, channels, sampleRate, NULL, frameCountIn, formatIn, wave->channels, wave->sampleRate);
if (frameCount == 0)
{
TRACELOG(LOG_ERROR, "WaveFormat() : Failed to get frame count for format conversion.");
@ -878,7 +943,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
void *data = RL_MALLOC(frameCount*channels*(sampleSize/8));
frameCount = (ma_uint32)ma_convert_frames(data, formatOut, channels, sampleRate, wave->data, formatIn, wave->channels, wave->sampleRate, frameCountIn);
frameCount = (ma_uint32)ma_convert_frames(data, frameCount, formatOut, channels, sampleRate, wave->data, frameCountIn, formatIn, wave->channels, wave->sampleRate);
if (frameCount == 0)
{
TRACELOG(LOG_ERROR, "WaveFormat() : Format conversion failed.");
@ -1012,7 +1077,7 @@ Music LoadMusicStream(const char *fileName)
music.ctxType = MUSIC_AUDIO_MP3;
music.stream = InitAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
music.sampleCount = drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels;
music.sampleCount = p">(unsigned int)drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels;
music.loopCount = 0; // Infinite loop by default
musicLoaded = true;
}
@ -1487,6 +1552,142 @@ static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel,
TRACELOG(LOG_ERROR, message); // All log messages from miniaudio are errors
}
// Reads audio data from an AudioBuffer object in internal format.
static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer, void *framesOut, ma_uint32 frameCount)
{
ma_uint32 subBufferSizeInFrames = (audioBuffer->sizeInFrames > 1)? audioBuffer->sizeInFrames/2 : audioBuffer->sizeInFrames;
ma_uint32 currentSubBufferIndex = audioBuffer->frameCursorPos/subBufferSizeInFrames;
if (currentSubBufferIndex > 1)
{
TRACELOGD("Frame cursor position moved too far forward in audio stream");
return 0;
}
// Another thread can update the processed state of buffers so
// we just take a copy here to try and avoid potential synchronization problems
bool isSubBufferProcessed[2];
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);
// Fill out every frame until we find a buffer that's marked as processed. Then fill the remainder with 0
ma_uint32 framesRead = 0;
while (1)
{
// We break from this loop differently depending on the buffer's usage
// - For static buffers, we simply fill as much data as we can
// - For streaming buffers we only fill the halves of the buffer that are processed
// Unprocessed halves must keep their audio data in-tact
if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
{
if (framesRead >= frameCount) break;
}
else
{
if (isSubBufferProcessed[currentSubBufferIndex]) break;
}
ma_uint32 totalFramesRemaining = (frameCount - framesRead);
if (totalFramesRemaining == 0) break;
ma_uint32 framesRemainingInOutputBuffer;
if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
{
framesRemainingInOutputBuffer = audioBuffer->sizeInFrames - audioBuffer->frameCursorPos;
}
else
{
ma_uint32 firstFrameIndexOfThisSubBuffer = subBufferSizeInFrames*currentSubBufferIndex;
framesRemainingInOutputBuffer = subBufferSizeInFrames - (audioBuffer->frameCursorPos - firstFrameIndexOfThisSubBuffer);
}
ma_uint32 framesToRead = totalFramesRemaining;
if (framesToRead > framesRemainingInOutputBuffer) framesToRead = framesRemainingInOutputBuffer;
memcpy((unsigned char *)framesOut + (framesRead*frameSizeInBytes), audioBuffer->data + (audioBuffer->frameCursorPos*frameSizeInBytes), framesToRead*frameSizeInBytes);
audioBuffer->frameCursorPos = (audioBuffer->frameCursorPos + framesToRead)%audioBuffer->sizeInFrames;
framesRead += framesToRead;
// If we've read to the end of the buffer, mark it as processed
if (framesToRead == framesRemainingInOutputBuffer)
{
audioBuffer->isSubBufferProcessed[currentSubBufferIndex] = true;
isSubBufferProcessed[currentSubBufferIndex] = true;
currentSubBufferIndex = (currentSubBufferIndex + 1)%2;
// We need to break from this loop if we're not looping
if (!audioBuffer->looping)
{
StopAudioBuffer(audioBuffer);
break;
}
}
}
// Zero-fill excess
ma_uint32 totalFramesRemaining = (frameCount - framesRead);
if (totalFramesRemaining > 0)
{
memset((unsigned char *)framesOut + (framesRead*frameSizeInBytes), 0, totalFramesRemaining*frameSizeInBytes);
// For static buffers we can fill the remaining frames with silence for safety, but we don't want
// to report those frames as "read". The reason for this is that the caller uses the return value
// to know whether or not a non-looping sound has finished playback.
if (audioBuffer->usage != AUDIO_BUFFER_USAGE_STATIC) framesRead += totalFramesRemaining;
}
return framesRead;
}
// Reads audio data from an AudioBuffer object in device format. Returned data will be in a format appropriate for mixing.
static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, float *framesOut, ma_uint32 frameCount)
{
// What's going on here is that we're continuously converting data from the AudioBuffer's internal format to the mixing format, which
// should be defined by the output format of the data converter. We do this until frameCount frames have been output. The important
// 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];
ma_uint32 inputBufferFrameCap = sizeof(inputBuffer) / ma_get_bytes_per_frame(audioBuffer->converter.config.formatIn, audioBuffer->converter.config.channelsIn);
ma_uint32 totalOutputFramesProcessed = 0;
while (totalOutputFramesProcessed < frameCount)
{
ma_uint64 outputFramesToProcessThisIteration = frameCount - totalOutputFramesProcessed;
ma_uint64 inputFramesToProcessThisIteration = ma_data_converter_get_required_input_frame_count(&audioBuffer->converter, outputFramesToProcessThisIteration);
if (inputFramesToProcessThisIteration > inputBufferFrameCap)
{
inputFramesToProcessThisIteration = inputBufferFrameCap;
}
float *runningFramesOut = framesOut + (totalOutputFramesProcessed * audioBuffer->converter.config.channelsOut);
/* At this point we can convert the data to our mixing format. */
ma_uint64 inputFramesProcessedThisIteration = ReadAudioBufferFramesInInternalFormat(audioBuffer, inputBuffer, (ma_uint32)inputFramesToProcessThisIteration); /* Safe cast. */
ma_uint64 outputFramesProcessedThisIteration = outputFramesToProcessThisIteration;
ma_data_converter_process_pcm_frames(&audioBuffer->converter, inputBuffer, &inputFramesProcessedThisIteration, runningFramesOut, &outputFramesProcessedThisIteration);
totalOutputFramesProcessed += (ma_uint32)outputFramesProcessedThisIteration; /* Safe cast. */
if (inputFramesProcessedThisIteration < inputFramesToProcessThisIteration)
{
break; /* Ran out of input data. */
}
/* This should never be hit, but will add it here for safety. Ensures we get out of the loop when no input nor output frames are processed. */
if (inputFramesProcessedThisIteration == 0 && outputFramesProcessedThisIteration == 0)
{
break;
}
}
return totalOutputFramesProcessed;
}
// Sending audio data to device callback function
// NOTE: All the mixing takes place here
static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount)
@ -1530,7 +1731,7 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const
framesToReadRightNow = sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS;
}
ma_uint32 framesJustRead = p">(ma_uint32)ma_pcm_converter_read(&audioBuffer->dsp, tempBuffer, framesToReadRightNow);
ma_uint32 framesJustRead = n">ReadAudioBufferFramesInMixingFormat(audioBuffer, tempBuffer, framesToReadRightNow);
if (framesJustRead > 0)
{
float *framesOut = (float *)pFramesOut + (framesRead*AUDIO.System.device.playback.channels);
@ -1576,98 +1777,6 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const
ma_mutex_unlock(&AUDIO.System.lock);
}
// DSP read from audio buffer callback function
static ma_uint32 OnAudioBufferDSPRead(ma_pcm_converter *pDSP, void *pFramesOut, ma_uint32 frameCount, void *pUserData)
{
AudioBuffer *audioBuffer = (AudioBuffer *)pUserData;
ma_uint32 subBufferSizeInFrames = (audioBuffer->sizeInFrames > 1)? audioBuffer->sizeInFrames/2 : audioBuffer->sizeInFrames;
ma_uint32 currentSubBufferIndex = audioBuffer->frameCursorPos/subBufferSizeInFrames;
if (currentSubBufferIndex > 1)
{
TRACELOGD("Frame cursor position moved too far forward in audio stream");
return 0;
}
// Another thread can update the processed state of buffers so
// we just take a copy here to try and avoid potential synchronization problems
bool isSubBufferProcessed[2];
isSubBufferProcessed[0] = audioBuffer->isSubBufferProcessed[0];
isSubBufferProcessed[1] = audioBuffer->isSubBufferProcessed[1];
ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(audioBuffer->dsp.formatConverterIn.config.formatIn)*audioBuffer->dsp.formatConverterIn.config.channels;
// Fill out every frame until we find a buffer that's marked as processed. Then fill the remainder with 0
ma_uint32 framesRead = 0;
while (1)
{
// We break from this loop differently depending on the buffer's usage
// - For static buffers, we simply fill as much data as we can
// - For streaming buffers we only fill the halves of the buffer that are processed
// Unprocessed halves must keep their audio data in-tact
if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
{
if (framesRead >= frameCount) break;
}
else
{
if (isSubBufferProcessed[currentSubBufferIndex]) break;
}
ma_uint32 totalFramesRemaining = (frameCount - framesRead);
if (totalFramesRemaining == 0) break;
ma_uint32 framesRemainingInOutputBuffer;
if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
{
framesRemainingInOutputBuffer = audioBuffer->sizeInFrames - audioBuffer->frameCursorPos;
}
else
{
ma_uint32 firstFrameIndexOfThisSubBuffer = subBufferSizeInFrames*currentSubBufferIndex;
framesRemainingInOutputBuffer = subBufferSizeInFrames - (audioBuffer->frameCursorPos - firstFrameIndexOfThisSubBuffer);
}
ma_uint32 framesToRead = totalFramesRemaining;
if (framesToRead > framesRemainingInOutputBuffer) framesToRead = framesRemainingInOutputBuffer;
memcpy((unsigned char *)pFramesOut + (framesRead*frameSizeInBytes), audioBuffer->data + (audioBuffer->frameCursorPos*frameSizeInBytes), framesToRead*frameSizeInBytes);
audioBuffer->frameCursorPos = (audioBuffer->frameCursorPos + framesToRead)%audioBuffer->sizeInFrames;
framesRead += framesToRead;
// If we've read to the end of the buffer, mark it as processed
if (framesToRead == framesRemainingInOutputBuffer)
{
audioBuffer->isSubBufferProcessed[currentSubBufferIndex] = true;
isSubBufferProcessed[currentSubBufferIndex] = true;
currentSubBufferIndex = (currentSubBufferIndex + 1)%2;
// We need to break from this loop if we're not looping
if (!audioBuffer->looping)
{
StopAudioBuffer(audioBuffer);
break;
}
}
}
// Zero-fill excess
ma_uint32 totalFramesRemaining = (frameCount - framesRead);
if (totalFramesRemaining > 0)
{
memset((unsigned char *)pFramesOut + (framesRead*frameSizeInBytes), 0, totalFramesRemaining*frameSizeInBytes);
// For static buffers we can fill the remaining frames with silence for safety, but we don't want
// to report those frames as "read". The reason for this is that the caller uses the return value
// to know whether or not a non-looping sound has finished playback.
if (audioBuffer->usage != AUDIO_BUFFER_USAGE_STATIC) framesRead += totalFramesRemaining;
}
return framesRead;
}
// This is the main mixing function. Mixing is pretty simple in this project - it's just an accumulation.
// NOTE: framesOut is both an input and an output. It will be initially filled with zeros outside of this function.
static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume)
@ -1679,7 +1788,7 @@ static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 fr
float *frameOut = framesOut + (iFrame*AUDIO.System.device.playback.channels);
const float *frameIn = framesIn + (iFrame*AUDIO.System.device.playback.channels);
frameOut[iChannel] += (frameIn[iChannel]*AUDIO.System.masterVolume*localVolume);
frameOut[iChannel] += (frameIn[iChannel]*localVolume);
}
}
}

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