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raylib-src/src/raudio.c

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/**********************************************************************************************
*
* raudio - A simple and easy-to-use audio library based on miniaudio
*
* FEATURES:
* - Manage audio device (init/close)
* - Manage raw audio context
* - Manage mixing channels
* - Load and unload audio files
* - Format wave data (sample rate, size, channels)
* - Play/Stop/Pause/Resume loaded audio
*
* CONFIGURATION:
*
* #define RAUDIO_STANDALONE
* Define to use the module as standalone library (independently of raylib).
* Required types and functions are defined in the same module.
*
* #define SUPPORT_FILEFORMAT_WAV
* #define SUPPORT_FILEFORMAT_OGG
* #define SUPPORT_FILEFORMAT_XM
* #define SUPPORT_FILEFORMAT_MOD
* #define SUPPORT_FILEFORMAT_FLAC
* #define SUPPORT_FILEFORMAT_MP3
* Selected desired fileformats to be supported for loading. Some of those formats are
* supported by default, to remove support, just comment unrequired #define in this module
*
* DEPENDENCIES:
* miniaudio.h - Audio device management lib (https://github.com/dr-soft/miniaudio)
* stb_vorbis.h - Ogg audio files loading (http://www.nothings.org/stb_vorbis/)
* dr_mp3.h - MP3 audio file loading (https://github.com/mackron/dr_libs)
* dr_flac.h - FLAC audio file loading (https://github.com/mackron/dr_libs)
* jar_xm.h - XM module file loading
* jar_mod.h - MOD audio file loading
*
* CONTRIBUTORS:
* David Reid (github: @mackron) (Nov. 2017):
* - Complete port to miniaudio library
*
* Joshua Reisenauer (github: @kd7tck) (2015)
* - XM audio module support (jar_xm)
* - MOD audio module support (jar_mod)
* - Mixing channels support
* - Raw audio context support
*
*
* LICENSE: zlib/libpng
*
* Copyright (c) 2013-2020 Ramon Santamaria (@raysan5)
*
* This software is provided "as-is", without any express or implied warranty. In no event
* will the authors be held liable for any damages arising from the use of this software.
*
* Permission is granted to anyone to use this software for any purpose, including commercial
* applications, and to alter it and redistribute it freely, subject to the following restrictions:
*
* 1. The origin of this software must not be misrepresented; you must not claim that you
* wrote the original software. If you use this software in a product, an acknowledgment
* in the product documentation would be appreciated but is not required.
*
* 2. Altered source versions must be plainly marked as such, and must not be misrepresented
* as being the original software.
*
* 3. This notice may not be removed or altered from any source distribution.
*
**********************************************************************************************/
#if defined(RAUDIO_STANDALONE)
#include "raudio.h"
#include <stdarg.h> // Required for: va_list, va_start(), vfprintf(), va_end()
#else
#include "raylib.h" // Declares module functions
// Check if config flags have been externally provided on compilation line
#if !defined(EXTERNAL_CONFIG_FLAGS)
#include "config.h" // Defines module configuration flags
#endif
#include "utils.h" // Required for: fopen() Android mapping
#endif
#if defined(_WIN32)
// To avoid conflicting windows.h symbols with raylib, some 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>
// Some required types defined for MSVC/TinyC compiler
#if defined(_MSC_VER) || defined(__TINYC__)
#include "propidl.h"
#endif
#endif
#define MA_MALLOC RL_MALLOC
#define MA_FREE RL_FREE
#define MA_NO_JACK
#define MINIAUDIO_IMPLEMENTATION
#include "external/miniaudio.h" // miniaudio library
#undef PlaySound // Win32 API: windows.h > mmsystem.h defines PlaySound macro
#include <stdlib.h> // Required for: malloc(), free()
#include <stdio.h> // Required for: FILE, fopen(), fclose(), fread()
#if defined(RAUDIO_STANDALONE)
#include <string.h> // Required for: strcmp() [Used in IsFileExtension()]
#if !defined(TRACELOG)
#define TRACELOG(level, ...) (void)0
#endif
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
// TODO: Remap malloc()/free() calls to RL_MALLOC/RL_FREE
#define STB_VORBIS_IMPLEMENTATION
#include "external/stb_vorbis.h" // OGG loading functions
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
#define JARXM_MALLOC RL_MALLOC
#define JARXM_FREE RL_FREE
#define JAR_XM_IMPLEMENTATION
#include "external/jar_xm.h" // XM loading functions
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
#define JARMOD_MALLOC RL_MALLOC
#define JARMOD_FREE RL_FREE
#define JAR_MOD_IMPLEMENTATION
#include "external/jar_mod.h" // MOD loading functions
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
#define DRFLAC_MALLOC RL_MALLOC
#define DRFLAC_REALLOC RL_REALLOC
#define DRFLAC_FREE RL_FREE
#define DR_FLAC_IMPLEMENTATION
#define DR_FLAC_NO_WIN32_IO
#include "external/dr_flac.h" // FLAC loading functions
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
#define DRMP3_MALLOC RL_MALLOC
#define DRMP3_REALLOC RL_REALLOC
#define DRMP3_FREE RL_FREE
#define DR_MP3_IMPLEMENTATION
#include "external/dr_mp3.h" // MP3 loading functions
#endif
#if defined(_MSC_VER)
#undef bool
#endif
//----------------------------------------------------------------------------------
// Defines and Macros
//----------------------------------------------------------------------------------
#define AUDIO_DEVICE_FORMAT ma_format_f32
#define AUDIO_DEVICE_CHANNELS 2
#define AUDIO_DEVICE_SAMPLE_RATE 44100
#define MAX_AUDIO_BUFFER_POOL_CHANNELS 16
//----------------------------------------------------------------------------------
// Types and Structures Definition
//----------------------------------------------------------------------------------
// Music context type
// NOTE: Depends on data structure provided by the library
// in charge of reading the different file types
typedef enum {
MUSIC_AUDIO_WAV = 0,
MUSIC_AUDIO_OGG,
MUSIC_AUDIO_FLAC,
MUSIC_AUDIO_MP3,
MUSIC_MODULE_XM,
MUSIC_MODULE_MOD
} MusicContextType;
#if defined(RAUDIO_STANDALONE)
typedef enum {
LOG_ALL,
LOG_TRACE,
LOG_DEBUG,
LOG_INFO,
LOG_WARNING,
LOG_ERROR,
LOG_FATAL,
LOG_NONE
} TraceLogType;
#endif
// NOTE: Different logic is used when feeding data to the playback device
// depending on whether or not data is streamed (Music vs Sound)
typedef enum {
AUDIO_BUFFER_USAGE_STATIC = 0,
AUDIO_BUFFER_USAGE_STREAM
} AudioBufferUsage;
// Audio buffer structure
struct rAudioBuffer {
ma_data_converter converter; // Audio data converter
float volume; // Audio buffer volume
float pitch; // Audio buffer pitch
bool playing; // Audio buffer state: AUDIO_PLAYING
bool paused; // Audio buffer state: AUDIO_PAUSED
bool looping; // Audio buffer looping, always true for AudioStreams
int usage; // Audio buffer usage mode: STATIC or STREAM
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 timing)
unsigned char *data; // Data buffer, on music stream keeps filling
rAudioBuffer *next; // Next audio buffer on the list
rAudioBuffer *prev; // Previous audio buffer on the list
};
#define AudioBuffer rAudioBuffer // HACK: To avoid CoreAudio (macOS) symbol collision
// Audio data context
typedef struct AudioData {
struct {
ma_context context; // miniaudio context data
ma_device device; // miniaudio device
ma_mutex lock; // miniaudio mutex lock
bool isReady; // Check if audio device is ready
} System;
struct {
AudioBuffer *first; // Pointer to first AudioBuffer in the list
AudioBuffer *last; // Pointer to last AudioBuffer in the list
int defaultSize; // Default audio buffer size for audio streams
} Buffer;
struct {
AudioBuffer *pool[MAX_AUDIO_BUFFER_POOL_CHANNELS]; // Multichannel AudioBuffer pointers pool
unsigned int poolCounter; // AudioBuffer pointers pool counter
unsigned int channels[MAX_AUDIO_BUFFER_POOL_CHANNELS]; // AudioBuffer pool channels
} MultiChannel;
} AudioData;
//----------------------------------------------------------------------------------
// Global Variables Definition
//----------------------------------------------------------------------------------
static AudioData AUDIO = { // Global AUDIO context
// NOTE: Music buffer size is defined by number of samples, independent of sample size and channels number
// After some math, considering a sampleRate of 48000, a buffer refill rate of 1/60 seconds and a
// standard double-buffering system, a 4096 samples buffer has been chosen, it should be enough
// In case of music-stalls, just increase this number
.Buffer.defaultSize = 4096
};
//----------------------------------------------------------------------------------
// Module specific Functions Declaration
//----------------------------------------------------------------------------------
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 void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume);
static void InitAudioBufferPool(void); // Initialise the multichannel buffer pool
static void CloseAudioBufferPool(void); // Close the audio buffers pool
#if defined(SUPPORT_FILEFORMAT_WAV)
static Wave LoadWAV(const char *fileName); // Load WAV file
static int SaveWAV(Wave wave, const char *fileName); // Save wave data as WAV file
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
static Wave LoadOGG(const char *fileName); // Load OGG file
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
static Wave LoadFLAC(const char *fileName); // Load FLAC file
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
static Wave LoadMP3(const char *fileName); // Load MP3 file
#endif
#if defined(RAUDIO_STANDALONE)
bool IsFileExtension(const char *fileName, const char *ext);// Check file extension
void TraceLog(int msgType, const char *text, ...); // Show trace log messages (LOG_INFO, LOG_WARNING, LOG_ERROR, LOG_DEBUG)
#endif
//----------------------------------------------------------------------------------
// AudioBuffer management functions declaration
// NOTE: Those functions are not exposed by raylib... for the moment
//----------------------------------------------------------------------------------
AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage);
void UnloadAudioBuffer(AudioBuffer *buffer);
bool IsAudioBufferPlaying(AudioBuffer *buffer);
void PlayAudioBuffer(AudioBuffer *buffer);
void StopAudioBuffer(AudioBuffer *buffer);
void PauseAudioBuffer(AudioBuffer *buffer);
void ResumeAudioBuffer(AudioBuffer *buffer);
void SetAudioBufferVolume(AudioBuffer *buffer, float volume);
void SetAudioBufferPitch(AudioBuffer *buffer, float pitch);
void TrackAudioBuffer(AudioBuffer *buffer);
void UntrackAudioBuffer(AudioBuffer *buffer);
//----------------------------------------------------------------------------------
// Module Functions Definition - Audio Device initialization and Closing
//----------------------------------------------------------------------------------
// Initialize audio device
void InitAudioDevice(void)
{
// TODO: Load AUDIO context memory dynamically?
// Init audio context
ma_context_config ctxConfig = ma_context_config_init();
ctxConfig.logCallback = OnLog;
ma_result result = ma_context_init(NULL, 0, &ctxConfig, &AUDIO.System.context);
if (result != MA_SUCCESS)
{
TRACELOG(LOG_ERROR, "AUDIO: Failed to initialize context");
return;
}
// Init audio device
// NOTE: Using the default device. Format is floating point because it simplifies mixing.
ma_device_config config = ma_device_config_init(ma_device_type_playback);
config.playback.pDeviceID = NULL; // NULL for the default playback AUDIO.System.device.
config.playback.format = AUDIO_DEVICE_FORMAT;
config.playback.channels = AUDIO_DEVICE_CHANNELS;
config.capture.pDeviceID = NULL; // NULL for the default capture AUDIO.System.device.
config.capture.format = ma_format_s16;
config.capture.channels = 1;
config.sampleRate = AUDIO_DEVICE_SAMPLE_RATE;
config.dataCallback = OnSendAudioDataToDevice;
config.pUserData = NULL;
result = ma_device_init(&AUDIO.System.context, &config, &AUDIO.System.device);
if (result != MA_SUCCESS)
{
TRACELOG(LOG_ERROR, "AUDIO: Failed to initialize playback device");
ma_context_uninit(&AUDIO.System.context);
return;
}
// Keep the device running the whole time. May want to consider doing something a bit smarter and only have the device running
// while there's at least one sound being played.
result = ma_device_start(&AUDIO.System.device);
if (result != MA_SUCCESS)
{
TRACELOG(LOG_ERROR, "AUDIO: Failed to start playback device");
ma_device_uninit(&AUDIO.System.device);
ma_context_uninit(&AUDIO.System.context);
return;
}
// Mixing happens on a seperate thread which means we need to synchronize. I'm using a mutex here to make things simple, but may
// want to look at something a bit smarter later on to keep everything real-time, if that's necessary.
if (ma_mutex_init(&AUDIO.System.context, &AUDIO.System.lock) != MA_SUCCESS)
{
TRACELOG(LOG_ERROR, "AUDIO: Failed to create mutex for mixing");
ma_device_uninit(&AUDIO.System.device);
ma_context_uninit(&AUDIO.System.context);
return;
}
TRACELOG(LOG_INFO, "AUDIO: Device initialized successfully");
TRACELOG(LOG_INFO, " > Backend: miniaudio / %s", ma_get_backend_name(AUDIO.System.context.backend));
TRACELOG(LOG_INFO, " > Format: %s -> %s", ma_get_format_name(AUDIO.System.device.playback.format), ma_get_format_name(AUDIO.System.device.playback.internalFormat));
TRACELOG(LOG_INFO, " > Channels: %d -> %d", AUDIO.System.device.playback.channels, AUDIO.System.device.playback.internalChannels);
TRACELOG(LOG_INFO, " > Sample rate: %d -> %d", AUDIO.System.device.sampleRate, AUDIO.System.device.playback.internalSampleRate);
TRACELOG(LOG_INFO, " > Periods size: %d", AUDIO.System.device.playback.internalPeriodSizeInFrames*AUDIO.System.device.playback.internalPeriods);
InitAudioBufferPool();
AUDIO.System.isReady = true;
}
// Close the audio device for all contexts
void CloseAudioDevice(void)
{
if (AUDIO.System.isReady)
{
ma_mutex_uninit(&AUDIO.System.lock);
ma_device_uninit(&AUDIO.System.device);
ma_context_uninit(&AUDIO.System.context);
CloseAudioBufferPool();
TRACELOG(LOG_INFO, "AUDIO: Device closed successfully");
}
else TRACELOG(LOG_WARNING, "AUDIO: Device could not be closed, not currently initialized");
}
// Check if device has been initialized successfully
bool IsAudioDeviceReady(void)
{
return AUDIO.System.isReady;
}
// Set master volume (listener)
void SetMasterVolume(float volume)
{
ma_device_set_master_volume(&AUDIO.System.device, volume);
}
//----------------------------------------------------------------------------------
// Module Functions Definition - Audio Buffer management
//----------------------------------------------------------------------------------
// Initialize a new audio buffer (filled with silence)
AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage)
{
AudioBuffer *audioBuffer = (AudioBuffer *)RL_CALLOC(1, sizeof(AudioBuffer));
if (audioBuffer == NULL)
{
TRACELOG(LOG_ERROR, "AUDIO: Failed to allocate memory for buffer");
return NULL;
}
audioBuffer->data = RL_CALLOC(sizeInFrames*channels*ma_get_bytes_per_sample(format), 1);
// 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_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)
{
TRACELOG(LOG_ERROR, "AUDIO: Failed to create data conversion pipeline");
RL_FREE(audioBuffer);
return NULL;
}
// Init audio buffer values
audioBuffer->volume = 1.0f;
audioBuffer->pitch = 1.0f;
audioBuffer->playing = false;
audioBuffer->paused = false;
audioBuffer->looping = false;
audioBuffer->usage = usage;
audioBuffer->frameCursorPos = 0;
audioBuffer->sizeInFrames = sizeInFrames;
// Buffers should be marked as processed by default so that a call to
// UpdateAudioStream() immediately after initialization works correctly
audioBuffer->isSubBufferProcessed[0] = true;
audioBuffer->isSubBufferProcessed[1] = true;
// Track audio buffer to linked list next position
TrackAudioBuffer(audioBuffer);
return audioBuffer;
}
// Delete an audio buffer
void UnloadAudioBuffer(AudioBuffer *buffer)
{
if (buffer != NULL)
{
ma_data_converter_uninit(&buffer->converter);
UntrackAudioBuffer(buffer);
RL_FREE(buffer->data);
RL_FREE(buffer);
}
}
// Check if an audio buffer is playing
bool IsAudioBufferPlaying(AudioBuffer *buffer)
{
bool result = false;
if (buffer != NULL) result = (buffer->playing && !buffer->paused);
return result;
}
// Play an audio buffer
// NOTE: Buffer is restarted to the start.
// Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position should be maintained.
void PlayAudioBuffer(AudioBuffer *buffer)
{
if (buffer != NULL)
{
buffer->playing = true;
buffer->paused = false;
buffer->frameCursorPos = 0;
}
}
// Stop an audio buffer
void StopAudioBuffer(AudioBuffer *buffer)
{
if (buffer != NULL)
{
if (IsAudioBufferPlaying(buffer))
{
buffer->playing = false;
buffer->paused = false;
buffer->frameCursorPos = 0;
buffer->totalFramesProcessed = 0;
buffer->isSubBufferProcessed[0] = true;
buffer->isSubBufferProcessed[1] = true;
}
}
}
// Pause an audio buffer
void PauseAudioBuffer(AudioBuffer *buffer)
{
if (buffer != NULL) buffer->paused = true;
}
// Resume an audio buffer
void ResumeAudioBuffer(AudioBuffer *buffer)
{
if (buffer != NULL) buffer->paused = false;
}
// Set volume for an audio buffer
void SetAudioBufferVolume(AudioBuffer *buffer, float volume)
{
if (buffer != NULL) buffer->volume = volume;
}
// Set pitch for an audio buffer
void SetAudioBufferPitch(AudioBuffer *buffer, float pitch)
{
if (buffer != NULL)
{
float pitchMul = pitch/buffer->pitch;
// Pitching is just an adjustment of the sample rate.
// 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->converter.config.sampleRateOut/pitchMul);
buffer->pitch *= (float)buffer->converter.config.sampleRateOut/newOutputSampleRate;
ma_data_converter_set_rate(&buffer->converter, buffer->converter.config.sampleRateIn, newOutputSampleRate);
}
}
// Track audio buffer to linked list next position
void TrackAudioBuffer(AudioBuffer *buffer)
{
ma_mutex_lock(&AUDIO.System.lock);
{
if (AUDIO.Buffer.first == NULL) AUDIO.Buffer.first = buffer;
else
{
AUDIO.Buffer.last->next = buffer;
buffer->prev = AUDIO.Buffer.last;
}
AUDIO.Buffer.last = buffer;
}
ma_mutex_unlock(&AUDIO.System.lock);
}
// Untrack audio buffer from linked list
void UntrackAudioBuffer(AudioBuffer *buffer)
{
ma_mutex_lock(&AUDIO.System.lock);
{
if (buffer->prev == NULL) AUDIO.Buffer.first = buffer->next;
else buffer->prev->next = buffer->next;
if (buffer->next == NULL) AUDIO.Buffer.last = buffer->prev;
else buffer->next->prev = buffer->prev;
buffer->prev = NULL;
buffer->next = NULL;
}
ma_mutex_unlock(&AUDIO.System.lock);
}
//----------------------------------------------------------------------------------
// Module Functions Definition - Sounds loading and playing (.WAV)
//----------------------------------------------------------------------------------
// Load wave data from file
Wave LoadWave(const char *fileName)
{
Wave wave = { 0 };
if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV)
else if (IsFileExtension(fileName, ".wav")) wave = LoadWAV(fileName);
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
else if (IsFileExtension(fileName, ".ogg")) wave = LoadOGG(fileName);
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if (IsFileExtension(fileName, ".flac")) wave = LoadFLAC(fileName);
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
else if (IsFileExtension(fileName, ".mp3")) wave = LoadMP3(fileName);
#endif
else TRACELOG(LOG_WARNING, "FILEIO: [%s] File format not supported", fileName);
return wave;
}
// Load sound from file
// NOTE: The entire file is loaded to memory to be played (no-streaming)
Sound LoadSound(const char *fileName)
{
Wave wave = LoadWave(fileName);
Sound sound = LoadSoundFromWave(wave);
UnloadWave(wave); // Sound is loaded, we can unload wave
return sound;
}
// Load sound from wave data
// NOTE: Wave data must be unallocated manually
Sound LoadSoundFromWave(Wave wave)
{
Sound sound = { 0 };
if (wave.data != NULL)
{
// When using miniaudio we need to do our own mixing.
// To simplify this we need convert the format of each sound to be consistent with
// the format used to open the playback AUDIO.System.device. We can do this two ways:
//
// 1) Convert the whole sound in one go at load time (here).
// 2) Convert the audio data in chunks at mixing time.
//
// First option has been selected, format conversion is done on the loading stage.
// The downside is that it uses more memory if the original sound is u8 or s16.
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, 0, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, NULL, frameCountIn, formatIn, wave.channels, wave.sampleRate);
if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed to get frame count for format conversion");
AudioBuffer *audioBuffer = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, frameCount, AUDIO_BUFFER_USAGE_STATIC);
if (audioBuffer == NULL) TRACELOG(LOG_WARNING, "SOUND: Failed to create buffer");
frameCount = (ma_uint32)ma_convert_frames(audioBuffer->data, frameCount, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, wave.data, frameCountIn, formatIn, wave.channels, wave.sampleRate);
if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed format conversion");
sound.sampleCount = frameCount*AUDIO_DEVICE_CHANNELS;
sound.stream.sampleRate = AUDIO_DEVICE_SAMPLE_RATE;
sound.stream.sampleSize = 32;
sound.stream.channels = AUDIO_DEVICE_CHANNELS;
sound.stream.buffer = audioBuffer;
}
return sound;
}
// Unload wave data
void UnloadWave(Wave wave)
{
if (wave.data != NULL) RL_FREE(wave.data);
TRACELOG(LOG_INFO, "WAVE: Unloaded wave data from RAM");
}
// Unload sound
void UnloadSound(Sound sound)
{
UnloadAudioBuffer(sound.stream.buffer);
TRACELOG(LOG_INFO, "WAVE: Unloaded sound data from RAM");
}
// Update sound buffer with new data
void UpdateSound(Sound sound, const void *data, int samplesCount)
{
if (sound.stream.buffer != NULL)
{
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, samplesCount*ma_get_bytes_per_frame(sound.stream.buffer->converter.config.formatIn, sound.stream.buffer->converter.config.channelsIn));
}
}
// Export wave data to file
void ExportWave(Wave wave, const char *fileName)
{
bool success = false;
if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV)
else if (IsFileExtension(fileName, ".wav")) success = SaveWAV(wave, fileName);
#endif
else if (IsFileExtension(fileName, ".raw"))
{
// Export raw sample data (without header)
// NOTE: It's up to the user to track wave parameters
SaveFileData(fileName, wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8);
success = true;
}
if (success) TRACELOG(LOG_INFO, "FILEIO: [%s] Wave data exported successfully", fileName);
else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to export wave data", fileName);
}
// Export wave sample data to code (.h)
void ExportWaveAsCode(Wave wave, const char *fileName)
{
#define BYTES_TEXT_PER_LINE 20
char varFileName[256] = { 0 };
int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8;
FILE *txtFile = fopen(fileName, "wt");
if (txtFile != NULL)
{
fprintf(txtFile, "\n//////////////////////////////////////////////////////////////////////////////////\n");
fprintf(txtFile, "// //\n");
fprintf(txtFile, "// WaveAsCode exporter v1.0 - Wave data exported as an array of bytes //\n");
fprintf(txtFile, "// //\n");
fprintf(txtFile, "// more info and bugs-report: github.com/raysan5/raylib //\n");
fprintf(txtFile, "// feedback and support: ray[at]raylib.com //\n");
fprintf(txtFile, "// //\n");
fprintf(txtFile, "// Copyright (c) 2018 Ramon Santamaria (@raysan5) //\n");
fprintf(txtFile, "// //\n");
fprintf(txtFile, "//////////////////////////////////////////////////////////////////////////////////\n\n");
#if !defined(RAUDIO_STANDALONE)
// Get file name from path and convert variable name to uppercase
strcpy(varFileName, GetFileNameWithoutExt(fileName));
for (int i = 0; varFileName[i] != '\0'; i++) if (varFileName[i] >= 'a' && varFileName[i] <= 'z') { varFileName[i] = varFileName[i] - 32; }
#else
strcpy(varFileName, fileName);
#endif
fprintf(txtFile, "// Wave data information\n");
fprintf(txtFile, "#define %s_SAMPLE_COUNT %u\n", varFileName, wave.sampleCount);
fprintf(txtFile, "#define %s_SAMPLE_RATE %u\n", varFileName, wave.sampleRate);
fprintf(txtFile, "#define %s_SAMPLE_SIZE %u\n", varFileName, wave.sampleSize);
fprintf(txtFile, "#define %s_CHANNELS %u\n\n", varFileName, wave.channels);
// Write byte data as hexadecimal text
fprintf(txtFile, "static unsigned char %s_DATA[%i] = { ", varFileName, dataSize);
for (int i = 0; i < dataSize - 1; i++) fprintf(txtFile, ((i%BYTES_TEXT_PER_LINE == 0)? "0x%x,\n" : "0x%x, "), ((unsigned char *)wave.data)[i]);
fprintf(txtFile, "0x%x };\n", ((unsigned char *)wave.data)[dataSize - 1]);
fclose(txtFile);
}
}
// Play a sound
void PlaySound(Sound sound)
{
PlayAudioBuffer(sound.stream.buffer);
}
// Play a sound in the multichannel buffer pool
void PlaySoundMulti(Sound sound)
{
int index = -1;
unsigned int oldAge = 0;
int oldIndex = -1;
// find the first non playing pool entry
for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
{
if (AUDIO.MultiChannel.channels[i] > oldAge)
{
oldAge = AUDIO.MultiChannel.channels[i];
oldIndex = i;
}
if (!IsAudioBufferPlaying(AUDIO.MultiChannel.pool[i]))
{
index = i;
break;
}
}
// If no none playing pool members can be index choose the oldest
if (index == -1)
{
TRACELOG(LOG_WARNING, "SOUND: Buffer pool is already full, count: %i", AUDIO.MultiChannel.poolCounter);
if (oldIndex == -1)
{
// Shouldn't be able to get here... but just in case something odd happens!
TRACELOG(LOG_WARNING, "SOUND: Buffer pool could not determine oldest buffer not playing sound");
return;
}
index = oldIndex;
// Just in case...
StopAudioBuffer(AUDIO.MultiChannel.pool[index]);
}
// Experimentally mutex lock doesn't seem to be needed this makes sense
// as pool[index] isn't playing and the only stuff we're copying
// shouldn't be changing...
AUDIO.MultiChannel.channels[index] = AUDIO.MultiChannel.poolCounter;
AUDIO.MultiChannel.poolCounter++;
AUDIO.MultiChannel.pool[index]->volume = sound.stream.buffer->volume;
AUDIO.MultiChannel.pool[index]->pitch = sound.stream.buffer->pitch;
AUDIO.MultiChannel.pool[index]->looping = sound.stream.buffer->looping;
AUDIO.MultiChannel.pool[index]->usage = sound.stream.buffer->usage;
AUDIO.MultiChannel.pool[index]->isSubBufferProcessed[0] = false;
AUDIO.MultiChannel.pool[index]->isSubBufferProcessed[1] = false;
AUDIO.MultiChannel.pool[index]->sizeInFrames = sound.stream.buffer->sizeInFrames;
AUDIO.MultiChannel.pool[index]->data = sound.stream.buffer->data;
PlayAudioBuffer(AUDIO.MultiChannel.pool[index]);
}
// Stop any sound played with PlaySoundMulti()
void StopSoundMulti(void)
{
for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) StopAudioBuffer(AUDIO.MultiChannel.pool[i]);
}
// Get number of sounds playing in the multichannel buffer pool
int GetSoundsPlaying(void)
{
int counter = 0;
for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
{
if (IsAudioBufferPlaying(AUDIO.MultiChannel.pool[i])) counter++;
}
return counter;
}
// Pause a sound
void PauseSound(Sound sound)
{
PauseAudioBuffer(sound.stream.buffer);
}
// Resume a paused sound
void ResumeSound(Sound sound)
{
ResumeAudioBuffer(sound.stream.buffer);
}
// Stop reproducing a sound
void StopSound(Sound sound)
{
StopAudioBuffer(sound.stream.buffer);
}
// Check if a sound is playing
bool IsSoundPlaying(Sound sound)
{
return IsAudioBufferPlaying(sound.stream.buffer);
}
// Set volume for a sound
void SetSoundVolume(Sound sound, float volume)
{
SetAudioBufferVolume(sound.stream.buffer, volume);
}
// Set pitch for a sound
void SetSoundPitch(Sound sound, float pitch)
{
SetAudioBufferPitch(sound.stream.buffer, pitch);
}
// Convert wave data to desired format
void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
{
ma_format formatIn = ((wave->sampleSize == 8)? ma_format_u8 : ((wave->sampleSize == 16)? ma_format_s16 : ma_format_f32));
ma_format formatOut = (( sampleSize == 8)? ma_format_u8 : (( sampleSize == 16)? ma_format_s16 : ma_format_f32));
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, 0, formatOut, channels, sampleRate, NULL, frameCountIn, formatIn, wave->channels, wave->sampleRate);
if (frameCount == 0)
{
TRACELOG(LOG_WARNING, "WAVE: Failed to get frame count for format conversion");
return;
}
void *data = RL_MALLOC(frameCount*channels*(sampleSize/8));
frameCount = (ma_uint32)ma_convert_frames(data, frameCount, formatOut, channels, sampleRate, wave->data, frameCountIn, formatIn, wave->channels, wave->sampleRate);
if (frameCount == 0)
{
TRACELOG(LOG_WARNING, "WAVE: Failed format conversion");
return;
}
wave->sampleCount = frameCount;
wave->sampleSize = sampleSize;
wave->sampleRate = sampleRate;
wave->channels = channels;
RL_FREE(wave->data);
wave->data = data;
}
// Copy a wave to a new wave
Wave WaveCopy(Wave wave)
{
Wave newWave = { 0 };
newWave.data = RL_MALLOC(wave.sampleCount*wave.sampleSize/8*wave.channels);
if (newWave.data != NULL)
{
// NOTE: Size must be provided in bytes
memcpy(newWave.data, wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8);
newWave.sampleCount = wave.sampleCount;
newWave.sampleRate = wave.sampleRate;
newWave.sampleSize = wave.sampleSize;
newWave.channels = wave.channels;
}
return newWave;
}
// Crop a wave to defined samples range
// NOTE: Security check in case of out-of-range
void WaveCrop(Wave *wave, int initSample, int finalSample)
{
if ((initSample >= 0) && (initSample < finalSample) &&
(finalSample > 0) && ((unsigned int)finalSample < wave->sampleCount))
{
int sampleCount = finalSample - initSample;
void *data = RL_MALLOC(sampleCount*wave->sampleSize/8*wave->channels);
memcpy(data, (unsigned char *)wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->channels*wave->sampleSize/8);
RL_FREE(wave->data);
wave->data = data;
}
else TRACELOG(LOG_WARNING, "WAVE: Crop range out of bounds");
}
// Get samples data from wave as a floats array
// NOTE: Returned sample values are normalized to range [-1..1]
float *GetWaveData(Wave wave)
{
float *samples = (float *)RL_MALLOC(wave.sampleCount*wave.channels*sizeof(float));
for (unsigned int i = 0; i < wave.sampleCount; i++)
{
for (unsigned int j = 0; j < wave.channels; j++)
{
if (wave.sampleSize == 8) samples[wave.channels*i + j] = (float)(((unsigned char *)wave.data)[wave.channels*i + j] - 127)/256.0f;
else if (wave.sampleSize == 16) samples[wave.channels*i + j] = (float)((short *)wave.data)[wave.channels*i + j]/32767.0f;
else if (wave.sampleSize == 32) samples[wave.channels*i + j] = ((float *)wave.data)[wave.channels*i + j];
}
}
return samples;
}
//----------------------------------------------------------------------------------
// Module Functions Definition - Music loading and stream playing (.OGG)
//----------------------------------------------------------------------------------
// Load music stream from file
Music LoadMusicStream(const char *fileName)
{
Music music = { 0 };
bool musicLoaded = false;
if (false) { }
#if defined(SUPPORT_FILEFORMAT_OGG)
else if (IsFileExtension(fileName, ".ogg"))
{
// Open ogg audio stream
music.ctxData = stb_vorbis_open_filename(fileName, NULL, NULL);
if (music.ctxData != NULL)
{
music.ctxType = MUSIC_AUDIO_OGG;
stb_vorbis_info info = stb_vorbis_get_info((stb_vorbis *)music.ctxData); // Get Ogg file info
// OGG bit rate defaults to 16 bit, it's enough for compressed format
music.stream = InitAudioStream(info.sample_rate, 16, info.channels);
music.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData)*info.channels;
music.loopCount = 0; // Infinite loop by default
musicLoaded = true;
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if (IsFileExtension(fileName, ".flac"))
{
music.ctxData = drflac_open_file(fileName);
if (music.ctxData != NULL)
{
music.ctxType = MUSIC_AUDIO_FLAC;
drflac *ctxFlac = (drflac *)music.ctxData;
music.stream = InitAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
music.sampleCount = (unsigned int)ctxFlac->totalSampleCount;
music.loopCount = 0; // Infinite loop by default
musicLoaded = true;
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
else if (IsFileExtension(fileName, ".mp3"))
{
drmp3 *ctxMp3 = RL_MALLOC(sizeof(drmp3));
music.ctxData = ctxMp3;
int result = drmp3_init_file(ctxMp3, fileName, NULL);
if (result > 0)
{
music.ctxType = MUSIC_AUDIO_MP3;
music.stream = InitAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
music.sampleCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels;
music.loopCount = 0; // Infinite loop by default
musicLoaded = true;
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
else if (IsFileExtension(fileName, ".xm"))
{
jar_xm_context_t *ctxXm = NULL;
int result = jar_xm_create_context_from_file(&ctxXm, 48000, fileName);
if (result == 0) // XM AUDIO.System.context created successfully
{
music.ctxType = MUSIC_MODULE_XM;
jar_xm_set_max_loop_count(ctxXm, 0); // Set infinite number of loops
// NOTE: Only stereo is supported for XM
music.stream = InitAudioStream(48000, 16, 2);
music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)*2;
music.loopCount = 0; // Infinite loop by default
jar_xm_reset(ctxXm); // make sure we start at the beginning of the song
musicLoaded = true;
music.ctxData = ctxXm;
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
else if (IsFileExtension(fileName, ".mod"))
{
jar_mod_context_t *ctxMod = RL_MALLOC(sizeof(jar_mod_context_t));
jar_mod_init(ctxMod);
int result = jar_mod_load_file(ctxMod, fileName);
if (result > 0)
{
music.ctxType = MUSIC_MODULE_MOD;
// NOTE: Only stereo is supported for MOD
music.stream = InitAudioStream(48000, 16, 2);
music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)*2;
music.loopCount = 0; // Infinite loop by default
musicLoaded = true;
music.ctxData = ctxMod;
}
}
#endif
else TRACELOG(LOG_WARNING, "STREAM: [%s] Fileformat not supported", fileName);
if (!musicLoaded)
{
if (false) { }
#if defined(SUPPORT_FILEFORMAT_OGG)
else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData);
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData);
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData);
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); }
#endif
TRACELOG(LOG_WARNING, "FILEIO: [%s] Music file could not be opened", fileName);
}
else
{
// Show some music stream info
TRACELOG(LOG_INFO, "FILEIO: [%s] Music file successfully loaded:", fileName);
TRACELOG(LOG_INFO, " > Total samples: %i", music.sampleCount);
TRACELOG(LOG_INFO, " > Sample rate: %i Hz", music.stream.sampleRate);
TRACELOG(LOG_INFO, " > Sample size: %i bits", music.stream.sampleSize);
TRACELOG(LOG_INFO, " > Channels: %i (%s)", music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi");
}
return music;
}
// Unload music stream
void UnloadMusicStream(Music music)
{
CloseAudioStream(music.stream);
if (false) { }
#if defined(SUPPORT_FILEFORMAT_OGG)
else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData);
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData);
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData);
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); }
#endif
}
// Start music playing (open stream)
void PlayMusicStream(Music music)
{
if (music.stream.buffer != NULL)
{
// For music streams, we need to make sure we maintain the frame cursor position
// This is a hack for this section of code in UpdateMusicStream()
// NOTE: In case window is minimized, music stream is stopped, just make sure to
// play again on window restore: if (IsMusicPlaying(music)) PlayMusicStream(music);
ma_uint32 frameCursorPos = music.stream.buffer->frameCursorPos;
PlayAudioStream(music.stream); // WARNING: This resets the cursor position.
music.stream.buffer->frameCursorPos = frameCursorPos;
}
}
// Pause music playing
void PauseMusicStream(Music music)
{
PauseAudioStream(music.stream);
}
// Resume music playing
void ResumeMusicStream(Music music)
{
ResumeAudioStream(music.stream);
}
// Stop music playing (close stream)
void StopMusicStream(Music music)
{
StopAudioStream(music.stream);
switch (music.ctxType)
{
#if defined(SUPPORT_FILEFORMAT_OGG)
case MUSIC_AUDIO_OGG: stb_vorbis_seek_start((stb_vorbis *)music.ctxData); break;
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
case MUSIC_AUDIO_FLAC: drflac_seek_to_pcm_frame((drflac *)music.ctxData, 0); break;
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame((drmp3 *)music.ctxData, 0); break;
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
case MUSIC_MODULE_XM: jar_xm_reset((jar_xm_context_t *)music.ctxData); break;
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
case MUSIC_MODULE_MOD: jar_mod_seek_start((jar_mod_context_t *)music.ctxData); break;
#endif
default: break;
}
}
// Update (re-fill) music buffers if data already processed
void UpdateMusicStream(Music music)
{
if (music.stream.buffer == NULL) return;
bool streamEnding = false;
unsigned int subBufferSizeInFrames = music.stream.buffer->sizeInFrames/2;
// NOTE: Using dynamic allocation because it could require more than 16KB
void *pcm = RL_CALLOC(subBufferSizeInFrames*music.stream.channels*music.stream.sampleSize/8, 1);
int samplesCount = 0; // Total size of data streamed in L+R samples for xm floats, individual L or R for ogg shorts
// TODO: Get the sampleLeft using totalFramesProcessed... but first, get total frames processed correctly...
//ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
int sampleLeft = music.sampleCount - (music.stream.buffer->totalFramesProcessed*music.stream.channels);
while (IsAudioStreamProcessed(music.stream))
{
if ((sampleLeft/music.stream.channels) >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames*music.stream.channels;
else samplesCount = sampleLeft;
switch (music.ctxType)
{
#if defined(SUPPORT_FILEFORMAT_OGG)
case MUSIC_AUDIO_OGG:
{
// NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)pcm, samplesCount);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
case MUSIC_AUDIO_FLAC:
{
// NOTE: Returns the number of samples to process (not required)
drflac_read_pcm_frames_s16((drflac *)music.ctxData, samplesCount, (short *)pcm);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
case MUSIC_AUDIO_MP3:
{
// NOTE: samplesCount, actually refers to framesCount and returns the number of frames processed
drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, samplesCount/music.stream.channels, (float *)pcm);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
case MUSIC_MODULE_XM:
{
// NOTE: Internally this function considers 2 channels generation, so samplesCount/2
jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)pcm, samplesCount/2);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
case MUSIC_MODULE_MOD:
{
// NOTE: 3rd parameter (nbsample) specify the number of stereo 16bits samples you want, so sampleCount/2
jar_mod_fillbuffer((jar_mod_context_t *)music.ctxData, (short *)pcm, samplesCount/2, 0);
} break;
#endif
default: break;
}
UpdateAudioStream(music.stream, pcm, samplesCount);
if ((music.ctxType == MUSIC_MODULE_XM) || (music.ctxType == MUSIC_MODULE_MOD))
{
if (samplesCount > 1) sampleLeft -= samplesCount/2;
else sampleLeft -= samplesCount;
}
else sampleLeft -= samplesCount;
if (sampleLeft <= 0)
{
streamEnding = true;
break;
}
}
// Free allocated pcm data
RL_FREE(pcm);
// Reset audio stream for looping
if (streamEnding)
{
StopMusicStream(music); // Stop music (and reset)
// Decrease loopCount to stop when required
if (music.loopCount > 1)
{
music.loopCount--; // Decrease loop count
PlayMusicStream(music); // Play again
}
else if (music.loopCount == 0) PlayMusicStream(music);
}
else
{
// NOTE: In case window is minimized, music stream is stopped,
// just make sure to play again on window restore
if (IsMusicPlaying(music)) PlayMusicStream(music);
}
}
// Check if any music is playing
bool IsMusicPlaying(Music music)
{
return IsAudioStreamPlaying(music.stream);
}
// Set volume for music
void SetMusicVolume(Music music, float volume)
{
SetAudioStreamVolume(music.stream, volume);
}
// Set pitch for music
void SetMusicPitch(Music music, float pitch)
{
SetAudioStreamPitch(music.stream, pitch);
}
// Set music loop count (loop repeats)
// NOTE: If set to 0, means infinite loop
void SetMusicLoopCount(Music music, int count)
{
music.loopCount = count;
}
// Get music time length (in seconds)
float GetMusicTimeLength(Music music)
{
float totalSeconds = 0.0f;
totalSeconds = (float)music.sampleCount/(music.stream.sampleRate*music.stream.channels);
return totalSeconds;
}
// Get current music time played (in seconds)
float GetMusicTimePlayed(Music music)
{
float secondsPlayed = 0.0f;
if (music.stream.buffer != NULL)
{
//ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
unsigned int samplesPlayed = music.stream.buffer->totalFramesProcessed*music.stream.channels;
secondsPlayed = (float)samplesPlayed / (music.stream.sampleRate*music.stream.channels);
}
return secondsPlayed;
}
// Init audio stream (to stream audio pcm data)
AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels)
{
AudioStream stream = { 0 };
stream.sampleRate = sampleRate;
stream.sampleSize = sampleSize;
stream.channels = channels;
ma_format formatIn = ((stream.sampleSize == 8)? ma_format_u8 : ((stream.sampleSize == 16)? ma_format_s16 : ma_format_f32));
// The size of a streaming buffer must be at least double the size of a period
unsigned int periodSize = AUDIO.System.device.playback.internalPeriodSizeInFrames;
unsigned int subBufferSize = AUDIO.Buffer.defaultSize; // Default buffer size (audio stream)
if (subBufferSize < periodSize) subBufferSize = periodSize;
// Create a double audio buffer of defined size
stream.buffer = LoadAudioBuffer(formatIn, stream.channels, stream.sampleRate, subBufferSize*2, AUDIO_BUFFER_USAGE_STREAM);
if (stream.buffer != NULL)
{
stream.buffer->looping = true; // Always loop for streaming buffers
TRACELOG(LOG_INFO, "STREAM: Initialized successfully (%i Hz, %i bit, %s)", stream.sampleRate, stream.sampleSize, (stream.channels == 1)? "Mono" : "Stereo");
}
else TRACELOG(LOG_WARNING, "STREAM: Failed to load audio buffer, stream could not be created");
return stream;
}
// Close audio stream and free memory
void CloseAudioStream(AudioStream stream)
{
UnloadAudioBuffer(stream.buffer);
TRACELOG(LOG_INFO, "STREAM: Unloaded audio stream data from RAM");
}
// Update audio stream buffers with data
// NOTE 1: Only updates one buffer of the stream source: unqueue -> update -> queue
// NOTE 2: To unqueue a buffer it needs to be processed: IsAudioStreamProcessed()
void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
{
if (stream.buffer != NULL)
{
if (stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1])
{
ma_uint32 subBufferToUpdate = 0;
if (stream.buffer->isSubBufferProcessed[0] && stream.buffer->isSubBufferProcessed[1])
{
// Both buffers are available for updating.
// Update the first one and make sure the cursor is moved back to the front.
subBufferToUpdate = 0;
stream.buffer->frameCursorPos = 0;
}
else
{
// Just update whichever sub-buffer is processed.
subBufferToUpdate = (stream.buffer->isSubBufferProcessed[0])? 0 : 1;
}
ma_uint32 subBufferSizeInFrames = stream.buffer->sizeInFrames/2;
unsigned char *subBuffer = stream.buffer->data + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate);
// TODO: Get total frames processed on this buffer... DOES NOT WORK.
stream.buffer->totalFramesProcessed += subBufferSizeInFrames;
// Does this API expect a whole buffer to be updated in one go?
// Assuming so, but if not will need to change this logic.
if (subBufferSizeInFrames >= (ma_uint32)samplesCount/stream.channels)
{
ma_uint32 framesToWrite = subBufferSizeInFrames;
if (framesToWrite > ((ma_uint32)samplesCount/stream.channels)) framesToWrite = (ma_uint32)samplesCount/stream.channels;
ma_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8);
memcpy(subBuffer, data, bytesToWrite);
// Any leftover frames should be filled with zeros.
ma_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite;
if (leftoverFrameCount > 0) memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8));
stream.buffer->isSubBufferProcessed[subBufferToUpdate] = false;
}
else TRACELOG(LOG_WARNING, "STREAM: Attempting to write too many frames to buffer");
}
else TRACELOG(LOG_WARNING, "STREAM: Buffer not available for updating");
}
}
// Check if any audio stream buffers requires refill
bool IsAudioStreamProcessed(AudioStream stream)
{
if (stream.buffer == NULL) return false;
return (stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1]);
}
// Play audio stream
void PlayAudioStream(AudioStream stream)
{
PlayAudioBuffer(stream.buffer);
}
// Play audio stream
void PauseAudioStream(AudioStream stream)
{
PauseAudioBuffer(stream.buffer);
}
// Resume audio stream playing
void ResumeAudioStream(AudioStream stream)
{
ResumeAudioBuffer(stream.buffer);
}
// Check if audio stream is playing.
bool IsAudioStreamPlaying(AudioStream stream)
{
return IsAudioBufferPlaying(stream.buffer);
}
// Stop audio stream
void StopAudioStream(AudioStream stream)
{
StopAudioBuffer(stream.buffer);
}
// Set volume for audio stream (1.0 is max level)
void SetAudioStreamVolume(AudioStream stream, float volume)
{
SetAudioBufferVolume(stream.buffer, volume);
}
// Set pitch for audio stream (1.0 is base level)
void SetAudioStreamPitch(AudioStream stream, float pitch)
{
SetAudioBufferPitch(stream.buffer, pitch);
}
// Default size for new audio streams
void SetAudioStreamBufferSizeDefault(int size)
{
AUDIO.Buffer.defaultSize = size;
}
//----------------------------------------------------------------------------------
// Module specific Functions Definition
//----------------------------------------------------------------------------------
// Log callback function
static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message)
{
(void)pContext;
(void)pDevice;
TRACELOG(LOG_ERROR, "miniaudio: %s", 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) 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)
{
(void)pDevice;
// Mixing is basically just an accumulation, we need to initialize the output buffer to 0
memset(pFramesOut, 0, frameCount*pDevice->playback.channels*ma_get_bytes_per_sample(pDevice->playback.format));
// Using a mutex here for thread-safety which makes things not real-time
// This is unlikely to be necessary for this project, but may want to consider how you might want to avoid this
ma_mutex_lock(&AUDIO.System.lock);
{
for (AudioBuffer *audioBuffer = AUDIO.Buffer.first; audioBuffer != NULL; audioBuffer = audioBuffer->next)
{
// Ignore stopped or paused sounds
if (!audioBuffer->playing || audioBuffer->paused) continue;
ma_uint32 framesRead = 0;
while (1)
{
if (framesRead >= frameCount) break;
// Just read as much data as we can from the stream
ma_uint32 framesToRead = (frameCount - framesRead);
while (framesToRead > 0)
{
float tempBuffer[1024]; // 512 frames for stereo
ma_uint32 framesToReadRightNow = framesToRead;
if (framesToReadRightNow > sizeof(tempBuffer)/sizeof(tempBuffer[0])/AUDIO_DEVICE_CHANNELS)
{
framesToReadRightNow = sizeof(tempBuffer)/sizeof(tempBuffer[0])/AUDIO_DEVICE_CHANNELS;
}
ma_uint32 framesJustRead = ReadAudioBufferFramesInMixingFormat(audioBuffer, tempBuffer, framesToReadRightNow);
if (framesJustRead > 0)
{
float *framesOut = (float *)pFramesOut + (framesRead*AUDIO.System.device.playback.channels);
float *framesIn = tempBuffer;
MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume);
framesToRead -= framesJustRead;
framesRead += framesJustRead;
}
if (!audioBuffer->playing)
{
framesRead = frameCount;
break;
}
// If we weren't able to read all the frames we requested, break
if (framesJustRead < framesToReadRightNow)
{
if (!audioBuffer->looping)
{
StopAudioBuffer(audioBuffer);
break;
}
else
{
// Should never get here, but just for safety,
// move the cursor position back to the start and continue the loop
audioBuffer->frameCursorPos = 0;
continue;
}
}
}
// If for some reason we weren't able to read every frame we'll need to break from the loop
// Not doing this could theoretically put us into an infinite loop
if (framesToRead > 0) break;
}
}
}
ma_mutex_unlock(&AUDIO.System.lock);
}
// 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)
{
for (ma_uint32 iFrame = 0; iFrame < frameCount; ++iFrame)
{
for (ma_uint32 iChannel = 0; iChannel < AUDIO.System.device.playback.channels; ++iChannel)
{
float *frameOut = framesOut + (iFrame*AUDIO.System.device.playback.channels);
const float *frameIn = framesIn + (iFrame*AUDIO.System.device.playback.channels);
frameOut[iChannel] += (frameIn[iChannel]*localVolume);
}
}
}
// Initialise the multichannel buffer pool
static void InitAudioBufferPool(void)
{
// Dummy buffers
for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
{
AUDIO.MultiChannel.pool[i] = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, 0, AUDIO_BUFFER_USAGE_STATIC);
}
// TODO: Verification required for log
TRACELOG(LOG_INFO, "AUDIO: Multichannel pool size: %i", MAX_AUDIO_BUFFER_POOL_CHANNELS);
}
// Close the audio buffers pool
static void CloseAudioBufferPool(void)
{
for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
{
RL_FREE(AUDIO.MultiChannel.pool[i]->data);
RL_FREE(AUDIO.MultiChannel.pool[i]);
}
}
#if defined(SUPPORT_FILEFORMAT_WAV)
// Load WAV file into Wave structure
static Wave LoadWAV(const char *fileName)
{
// Basic WAV headers structs
typedef struct {
char chunkID[4];
int chunkSize;
char format[4];
} WAVRiffHeader;
typedef struct {
char subChunkID[4];
int subChunkSize;
short audioFormat;
short numChannels;
int sampleRate;
int byteRate;
short blockAlign;
short bitsPerSample;
} WAVFormat;
typedef struct {
char subChunkID[4];
int subChunkSize;
} WAVData;
WAVRiffHeader wavRiffHeader = { 0 };
WAVFormat wavFormat = { 0 };
WAVData wavData = { 0 };
Wave wave = { 0 };
FILE *wavFile = NULL;
wavFile = fopen(fileName, "rb");
if (wavFile == NULL)
{
TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open WAV file", fileName);
wave.data = NULL;
}
else
{
// Read in the first chunk into the struct
fread(&wavRiffHeader, sizeof(WAVRiffHeader), 1, wavFile);
// Check for RIFF and WAVE tags
if ((wavRiffHeader.chunkID[0] != 'R') ||
(wavRiffHeader.chunkID[1] != 'I') ||
(wavRiffHeader.chunkID[2] != 'F') ||
(wavRiffHeader.chunkID[3] != 'F') ||
(wavRiffHeader.format[0] != 'W') ||
(wavRiffHeader.format[1] != 'A') ||
(wavRiffHeader.format[2] != 'V') ||
(wavRiffHeader.format[3] != 'E'))
{
TRACELOG(LOG_WARNING, "WAVE: [%s] RIFF or WAVE header are not valid", fileName);
}
else
{
// Read in the 2nd chunk for the wave info
fread(&wavFormat, sizeof(WAVFormat), 1, wavFile);
// Check for fmt tag
if ((wavFormat.subChunkID[0] != 'f') || (wavFormat.subChunkID[1] != 'm') ||
(wavFormat.subChunkID[2] != 't') || (wavFormat.subChunkID[3] != ' '))
{
TRACELOG(LOG_WARNING, "WAVE: [%s] Wave format header is not valid", fileName);
}
else
{
// Check for extra parameters;
if (wavFormat.subChunkSize > 16) fseek(wavFile, sizeof(short), SEEK_CUR);
// Read in the the last byte of data before the sound file
fread(&wavData, sizeof(WAVData), 1, wavFile);
// Check for data tag
if ((wavData.subChunkID[0] != 'd') || (wavData.subChunkID[1] != 'a') ||
(wavData.subChunkID[2] != 't') || (wavData.subChunkID[3] != 'a'))
{
TRACELOG(LOG_WARNING, "WAVE: [%s] Data header is not valid", fileName);
}
else
{
// Allocate memory for data
wave.data = RL_MALLOC(wavData.subChunkSize);
// Read in the sound data into the soundData variable
fread(wave.data, wavData.subChunkSize, 1, wavFile);
// Store wave parameters
wave.sampleRate = wavFormat.sampleRate;
wave.sampleSize = wavFormat.bitsPerSample;
wave.channels = wavFormat.numChannels;
// NOTE: Only support 8 bit, 16 bit and 32 bit sample sizes
if ((wave.sampleSize != 8) && (wave.sampleSize != 16) && (wave.sampleSize != 32))
{
TRACELOG(LOG_WARNING, "WAVE: [%s] Sample size (%ibit) not supported, converted to 16bit", fileName, wave.sampleSize);
WaveFormat(&wave, wave.sampleRate, 16, wave.channels);
}
// NOTE: Only support up to 2 channels (mono, stereo)
if (wave.channels > 2)
{
WaveFormat(&wave, wave.sampleRate, wave.sampleSize, 2);
TRACELOG(LOG_WARNING, "WAVE: [%s] Channels number (%i) not supported, converted to 2 channels", fileName, wave.channels);
}
// NOTE: subChunkSize comes in bytes, we need to translate it to number of samples
wave.sampleCount = (wavData.subChunkSize/(wave.sampleSize/8))/wave.channels;
TRACELOG(LOG_INFO, "WAVE: [%s] File loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
}
}
}
fclose(wavFile);
}
return wave;
}
// Save wave data as WAV file
static int SaveWAV(Wave wave, const char *fileName)
{
int success = 0;
int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8;
// Basic WAV headers structs
typedef struct {
char chunkID[4];
int chunkSize;
char format[4];
} RiffHeader;
typedef struct {
char subChunkID[4];
int subChunkSize;
short audioFormat;
short numChannels;
int sampleRate;
int byteRate;
short blockAlign;
short bitsPerSample;
} WaveFormat;
typedef struct {
char subChunkID[4];
int subChunkSize;
} WaveData;
FILE *wavFile = fopen(fileName, "wb");
if (wavFile == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open audio file", fileName);
else
{
RiffHeader riffHeader;
WaveFormat waveFormat;
WaveData waveData;
// Fill structs with data
riffHeader.chunkID[0] = 'R';
riffHeader.chunkID[1] = 'I';
riffHeader.chunkID[2] = 'F';
riffHeader.chunkID[3] = 'F';
riffHeader.chunkSize = 44 - 4 + wave.sampleCount*wave.sampleSize/8;
riffHeader.format[0] = 'W';
riffHeader.format[1] = 'A';
riffHeader.format[2] = 'V';
riffHeader.format[3] = 'E';
waveFormat.subChunkID[0] = 'f';
waveFormat.subChunkID[1] = 'm';
waveFormat.subChunkID[2] = 't';
waveFormat.subChunkID[3] = ' ';
waveFormat.subChunkSize = 16;
waveFormat.audioFormat = 1;
waveFormat.numChannels = wave.channels;
waveFormat.sampleRate = wave.sampleRate;
waveFormat.byteRate = wave.sampleRate*wave.sampleSize/8;
waveFormat.blockAlign = wave.sampleSize/8;
waveFormat.bitsPerSample = wave.sampleSize;
waveData.subChunkID[0] = 'd';
waveData.subChunkID[1] = 'a';
waveData.subChunkID[2] = 't';
waveData.subChunkID[3] = 'a';
waveData.subChunkSize = dataSize;
fwrite(&riffHeader, sizeof(RiffHeader), 1, wavFile);
fwrite(&waveFormat, sizeof(WaveFormat), 1, wavFile);
fwrite(&waveData, sizeof(WaveData), 1, wavFile);
success = fwrite(wave.data, dataSize, 1, wavFile);
fclose(wavFile);
}
// If all data has been written correctly to file, success = 1
return success;
}
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
// Load OGG file into Wave structure
// NOTE: Using stb_vorbis library
static Wave LoadOGG(const char *fileName)
{
Wave wave = { 0 };
stb_vorbis *oggFile = stb_vorbis_open_filename(fileName, NULL, NULL);
if (oggFile == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open OGG file", fileName);
else
{
stb_vorbis_info info = stb_vorbis_get_info(oggFile);
wave.sampleRate = info.sample_rate;
wave.sampleSize = 16; // 16 bit per sample (short)
wave.channels = info.channels;
wave.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples(oggFile)*info.channels; // Independent by channel
float totalSeconds = stb_vorbis_stream_length_in_seconds(oggFile);
if (totalSeconds > 10) TRACELOG(LOG_WARNING, "WAVE: [%s] Ogg audio length larger than 10 seconds (%f), that's a big file in memory, consider music streaming", fileName, totalSeconds);
wave.data = (short *)RL_MALLOC(wave.sampleCount*wave.channels*sizeof(short));
// NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
stb_vorbis_get_samples_short_interleaved(oggFile, info.channels, (short *)wave.data, wave.sampleCount*wave.channels);
TRACELOG(LOG_INFO, "WAVE: [%s] OGG file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
stb_vorbis_close(oggFile);
}
return wave;
}
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
// Load FLAC file into Wave structure
// NOTE: Using dr_flac library
static Wave LoadFLAC(const char *fileName)
{
Wave wave = { 0 };
// Decode an entire FLAC file in one go
unsigned long long int totalSampleCount = 0;
wave.data = drflac_open_file_and_read_pcm_frames_s16(fileName, &wave.channels, &wave.sampleRate, &totalSampleCount);
if (wave.data == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to load FLAC data", fileName);
else
{
wave.sampleCount = (unsigned int)totalSampleCount;
wave.sampleSize = 16;
// NOTE: Only support up to 2 channels (mono, stereo)
if (wave.channels > 2) TRACELOG(LOG_WARNING, "WAVE: [%s] FLAC channels number (%i) not supported", fileName, wave.channels);
TRACELOG(LOG_INFO, "WAVE: [%s] FLAC file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
}
return wave;
}
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
// Load MP3 file into Wave structure
// NOTE: Using dr_mp3 library
static Wave LoadMP3(const char *fileName)
{
Wave wave = { 0 };
// Decode an entire MP3 file in one go
unsigned long long int totalFrameCount = 0;
drmp3_config config = { 0 };
wave.data = drmp3_open_file_and_read_f32(fileName, &config, &totalFrameCount);
if (wave.data == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to load MP3 data", fileName);
else
{
wave.channels = config.outputChannels;
wave.sampleRate = config.outputSampleRate;
wave.sampleCount = (int)totalFrameCount*wave.channels;
wave.sampleSize = 32;
// NOTE: Only support up to 2 channels (mono, stereo)
if (wave.channels > 2) TRACELOG(LOG_WARNING, "WAVE: [%s] MP3 channels number (%i) not supported", fileName, wave.channels);
TRACELOG(LOG_INFO, "WAVE: [%s] MP3 file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
}
return wave;
}
#endif
// Some required functions for audio standalone module version
#if defined(RAUDIO_STANDALONE)
// Check file extension
bool IsFileExtension(const char *fileName, const char *ext)
{
bool result = false;
const char *fileExt;
if ((fileExt = strrchr(fileName, '.')) != NULL)
{
if (strcmp(fileExt, ext) == 0) result = true;
}
return result;
}
#endif
#undef AudioBuffer