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raylib/src/raudio.c
Ray dd7a1948f1 WARNING: REDESIGN: REMOVED: utils module, functionality moved to rcore module: logging and file-system #4551 
[utils] was created long time ago, when [rcore] contained all the platforms code, the purpose of the file was exposing basic filesystem functionality across modules and also logging mechanism but many things have changed since then and there is no need to keep using this module.

 - Logging system has been move to [rcore] module and macros are exposed through `config.h` to other modules
 - File system functionality has also been centralized in [rcore] module that along the years it was already adding more and more file-system functions, now they are all in the same module
 - Android specific code has been moved to `rcore_android.c`, it had no sense to have specific platform code in `utils`, [rcore] is responsible of all platform code.
2026-01-10 12:13:07 +01:00

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/**********************************************************************************************
*
* raudio v1.1 - 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 SUPPORT_MODULE_RAUDIO
* raudio module is included in the build
*
* #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_MP3
* #define SUPPORT_FILEFORMAT_QOA
* #define SUPPORT_FILEFORMAT_FLAC
* #define SUPPORT_FILEFORMAT_XM
* #define SUPPORT_FILEFORMAT_MOD
* 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/mackron/miniaudio)
* stb_vorbis.h - Ogg audio files loading (http://www.nothings.org/stb_vorbis/)
* dr_wav.h - WAV audio files loading (http://github.com/mackron/dr_libs)
* 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) 2013-2026 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"
#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(SUPPORT_MODULE_RAUDIO) || defined(RAUDIO_STANDALONE)
#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> // Windows functionality (miniaudio)
// 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> // Component Object Model (COM) header
#include <mmreg.h> // Windows Multimedia, defines some WAVE structs
#include <mmsystem.h> // Windows Multimedia, used by Windows GDI, defines DIBINDEX macro
// 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 MA_NO_WAV
#define MA_NO_FLAC
#define MA_NO_MP3
#define MA_NO_RESOURCE_MANAGER
#define MA_NO_NODE_GRAPH
#define MA_NO_ENGINE
#define MA_NO_GENERATION
// Threading model: Default: [0] COINIT_MULTITHREADED: COM calls objects on any thread (free threading)
#define MA_COINIT_VALUE 2 // [2] COINIT_APARTMENTTHREADED: Each object has its own thread (apartment model)
#define MINIAUDIO_IMPLEMENTATION
//#define MA_DEBUG_OUTPUT
#include "external/miniaudio.h" // Audio device initialization and management
#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()
#include <string.h> // Required for: strcmp() [Used in IsFileExtension(), LoadWaveFromMemory(), LoadMusicStreamFromMemory()]
#if defined(RAUDIO_STANDALONE)
#ifndef TRACELOG
#define TRACELOG(level, ...) printf(__VA_ARGS__)
#endif
// Allow custom memory allocators
#ifndef RL_MALLOC
#define RL_MALLOC(sz) malloc(sz)
#endif
#ifndef RL_CALLOC
#define RL_CALLOC(n,sz) calloc(n,sz)
#endif
#ifndef RL_REALLOC
#define RL_REALLOC(ptr,sz) realloc(ptr,sz)
#endif
#ifndef RL_FREE
#define RL_FREE(ptr) free(ptr)
#endif
#endif
#if defined(SUPPORT_FILEFORMAT_WAV)
#define DRWAV_MALLOC RL_MALLOC
#define DRWAV_REALLOC RL_REALLOC
#define DRWAV_FREE RL_FREE
#define DR_WAV_IMPLEMENTATION
#include "external/dr_wav.h" // WAV loading functions
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
// TODO: Remap stb_vorbis malloc()/free() calls to RL_MALLOC/RL_FREE
#include "external/stb_vorbis.c" // OGG 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(SUPPORT_FILEFORMAT_QOA)
#define QOA_MALLOC RL_MALLOC
#define QOA_FREE RL_FREE
#if defined(_MSC_VER) // Disable some MSVC warning
#pragma warning(push)
#pragma warning(disable : 4018)
#pragma warning(disable : 4267)
#pragma warning(disable : 4244)
#endif
#define QOA_IMPLEMENTATION
#include "external/qoa.h" // QOA loading and saving functions
#include "external/qoaplay.c" // QOA stream playing helper functions
#if defined(_MSC_VER)
#pragma warning(pop) // Disable MSVC warning suppression
#endif
#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_XM)
#define JARXM_MALLOC RL_MALLOC
#define JARXM_FREE RL_FREE
#if defined(_MSC_VER) // Disable some MSVC warning
#pragma warning(push)
#pragma warning(disable : 4244)
#endif
#define JAR_XM_IMPLEMENTATION
#include "external/jar_xm.h" // XM loading functions
#if defined(_MSC_VER)
#pragma warning(pop) // Disable MSVC warning suppression
#endif
#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
//----------------------------------------------------------------------------------
// Defines and Macros
//----------------------------------------------------------------------------------
#ifndef AUDIO_DEVICE_FORMAT
#define AUDIO_DEVICE_FORMAT ma_format_f32 // Device output format (float-32bit)
#endif
#ifndef AUDIO_DEVICE_CHANNELS
#define AUDIO_DEVICE_CHANNELS 2 // Device output channels: stereo
#endif
#ifndef AUDIO_DEVICE_SAMPLE_RATE
#define AUDIO_DEVICE_SAMPLE_RATE 0 // Device output sample rate
#endif
#ifndef MAX_AUDIO_BUFFER_POOL_CHANNELS
#define MAX_AUDIO_BUFFER_POOL_CHANNELS 16 // Audio pool channels
#endif
//----------------------------------------------------------------------------------
// Types and Structures Definition
//----------------------------------------------------------------------------------
#if defined(RAUDIO_STANDALONE)
// Trace log level
// NOTE: Organized by priority level
typedef enum {
LOG_ALL = 0, // Display all logs
LOG_TRACE, // Trace logging, intended for internal use only
LOG_DEBUG, // Debug logging, used for internal debugging, it should be disabled on release builds
LOG_INFO, // Info logging, used for program execution info
LOG_WARNING, // Warning logging, used on recoverable failures
LOG_ERROR, // Error logging, used on unrecoverable failures
LOG_FATAL, // Fatal logging, used to abort program: exit(EXIT_FAILURE)
LOG_NONE // Disable logging
} TraceLogLevel;
#endif
// Music context type
// NOTE: Depends on data structure provided by the library
// in charge of reading the different file types
typedef enum {
MUSIC_AUDIO_NONE = 0, // No audio context loaded
MUSIC_AUDIO_WAV, // WAV audio context
MUSIC_AUDIO_OGG, // OGG audio context
MUSIC_AUDIO_FLAC, // FLAC audio context
MUSIC_AUDIO_MP3, // MP3 audio context
MUSIC_AUDIO_QOA, // QOA audio context
MUSIC_MODULE_XM, // XM module audio context
MUSIC_MODULE_MOD // MOD module audio context
} MusicContextType;
// NOTE: Different logic is used when feeding data to the playback device
// depending on whether data is streamed (Music vs Sound)
typedef enum {
AUDIO_BUFFER_USAGE_STATIC = 0,
AUDIO_BUFFER_USAGE_STREAM
} AudioBufferUsage;
// Audio buffer struct
struct rAudioBuffer {
ma_data_converter converter; // Audio data converter
AudioCallback callback; // Audio buffer callback for buffer filling on audio threads
rAudioProcessor *processor; // Audio processor
float volume; // Audio buffer volume
float pitch; // Audio buffer pitch
float pan; // Audio buffer pan (0.0f to 1.0f)
bool playing; // Audio buffer state: AUDIO_PLAYING
bool paused; // Audio buffer state: AUDIO_PAUSED
bool looping; // Audio buffer looping, default to 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 framesProcessed; // 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
};
// Audio processor struct
// NOTE: Useful to apply effects to an AudioBuffer
struct rAudioProcessor {
AudioCallback process; // Processor callback function
rAudioProcessor *next; // Next audio processor on the list
rAudioProcessor *prev; // Previous audio processor 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
size_t pcmBufferSize; // Pre-allocated buffer size
void *pcmBuffer; // Pre-allocated buffer to read audio data from file/memory
} 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;
rAudioProcessor *mixedProcessor;
} 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 = 0,
.mixedProcessor = NULL
};
//----------------------------------------------------------------------------------
// Module Internal Functions Declaration
//----------------------------------------------------------------------------------
static void OnLog(void *pUserData, ma_uint32 level, const char *pMessage);
// Reads audio data from an AudioBuffer object in internal/device formats
static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer, void *framesOut, ma_uint32 frameCount);
static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, float *framesOut, ma_uint32 frameCount);
static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount);
static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, AudioBuffer *buffer);
static bool IsAudioBufferPlayingInLockedState(AudioBuffer *buffer);
static void StopAudioBufferInLockedState(AudioBuffer *buffer);
static void UpdateAudioStreamInLockedState(AudioStream stream, const void *data, int frameCount);
#if defined(RAUDIO_STANDALONE)
static bool IsFileExtension(const char *fileName, const char *ext); // Check file extension
static const char *GetFileExtension(const char *fileName); // Get pointer to extension for a filename string (includes the dot: .png)
static const char *GetFileName(const char *filePath); // Get pointer to filename for a path string
static const char *GetFileNameWithoutExt(const char *filePath); // Get filename string without extension (uses static string)
static unsigned char *LoadFileData(const char *fileName, int *dataSize); // Load file data as byte array (read)
static bool SaveFileData(const char *fileName, void *data, int dataSize); // Save data to file from byte array (write)
static bool SaveFileText(const char *fileName, char *text); // Save text data to file (write), string must be '\0' terminated
#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 SetAudioBufferPan(AudioBuffer *buffer, float pan);
void TrackAudioBuffer(AudioBuffer *buffer);
void UntrackAudioBuffer(AudioBuffer *buffer);
//----------------------------------------------------------------------------------
// Module Functions Definition - Audio Device initialization and Closing
//----------------------------------------------------------------------------------
// Initialize audio device
void InitAudioDevice(void)
{
// Init audio context
ma_context_config ctxConfig = ma_context_config_init();
ma_log_callback_init(OnLog, NULL);
ma_result result = ma_context_init(NULL, 0, &ctxConfig, &AUDIO.System.context);
if (result != MA_SUCCESS)
{
TRACELOG(LOG_WARNING, "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_WARNING, "AUDIO: Failed to initialize playback device");
ma_context_uninit(&AUDIO.System.context);
return;
}
// Mixing happens on a separate 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.lock) != MA_SUCCESS)
{
TRACELOG(LOG_WARNING, "AUDIO: Failed to create mutex for mixing");
ma_device_uninit(&AUDIO.System.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_WARNING, "AUDIO: Failed to start playback device");
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);
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);
AUDIO.System.isReady = false;
RL_FREE(AUDIO.System.pcmBuffer);
AUDIO.System.pcmBuffer = NULL;
AUDIO.System.pcmBufferSize = 0;
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);
}
// Get master volume (listener)
float GetMasterVolume(void)
{
float volume = 0.0f;
ma_device_get_master_volume(&AUDIO.System.device, &volume);
return 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_WARNING, "AUDIO: Failed to allocate memory for buffer");
return NULL;
}
if (sizeInFrames > 0) audioBuffer->data = (unsigned char *)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.System.device.sampleRate);
converterConfig.allowDynamicSampleRate = true;
ma_result result = ma_data_converter_init(&converterConfig, NULL, &audioBuffer->converter);
if (result != MA_SUCCESS)
{
TRACELOG(LOG_WARNING, "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->pan = 0.0f; // Center
audioBuffer->callback = NULL;
audioBuffer->processor = NULL;
audioBuffer->playing = false;
audioBuffer->paused = false;
audioBuffer->looping = false;
audioBuffer->usage = usage;
audioBuffer->frameCursorPos = 0;
audioBuffer->framesProcessed = 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)
{
UntrackAudioBuffer(buffer);
ma_data_converter_uninit(&buffer->converter, NULL);
RL_FREE(buffer->data);
RL_FREE(buffer);
}
}
// Check if an audio buffer is playing from a program state without lock
bool IsAudioBufferPlaying(AudioBuffer *buffer)
{
bool result = false;
ma_mutex_lock(&AUDIO.System.lock);
result = IsAudioBufferPlayingInLockedState(buffer);
ma_mutex_unlock(&AUDIO.System.lock);
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)
{
ma_mutex_lock(&AUDIO.System.lock);
buffer->playing = true;
buffer->paused = false;
buffer->frameCursorPos = 0;
buffer->framesProcessed = 0;
buffer->isSubBufferProcessed[0] = true;
buffer->isSubBufferProcessed[1] = true;
ma_mutex_unlock(&AUDIO.System.lock);
}
}
// Stop an audio buffer from a program state without lock
void StopAudioBuffer(AudioBuffer *buffer)
{
ma_mutex_lock(&AUDIO.System.lock);
StopAudioBufferInLockedState(buffer);
ma_mutex_unlock(&AUDIO.System.lock);
}
// Pause an audio buffer
void PauseAudioBuffer(AudioBuffer *buffer)
{
if (buffer != NULL)
{
ma_mutex_lock(&AUDIO.System.lock);
buffer->paused = true;
ma_mutex_unlock(&AUDIO.System.lock);
}
}
// Resume an audio buffer
void ResumeAudioBuffer(AudioBuffer *buffer)
{
if (buffer != NULL)
{
ma_mutex_lock(&AUDIO.System.lock);
buffer->paused = false;
ma_mutex_unlock(&AUDIO.System.lock);
}
}
// Set volume for an audio buffer
void SetAudioBufferVolume(AudioBuffer *buffer, float volume)
{
if (buffer != NULL)
{
ma_mutex_lock(&AUDIO.System.lock);
buffer->volume = volume;
ma_mutex_unlock(&AUDIO.System.lock);
}
}
// Set pitch for an audio buffer
void SetAudioBufferPitch(AudioBuffer *buffer, float pitch)
{
if ((buffer != NULL) && (pitch > 0.0f))
{
ma_mutex_lock(&AUDIO.System.lock);
// 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 outputSampleRate = (ma_uint32)((float)buffer->converter.sampleRateOut/pitch);
ma_data_converter_set_rate(&buffer->converter, buffer->converter.sampleRateIn, outputSampleRate);
buffer->pitch = pitch;
ma_mutex_unlock(&AUDIO.System.lock);
}
}
// Set pan for an audio buffer
void SetAudioBufferPan(AudioBuffer *buffer, float pan)
{
if (pan < -1.0f) pan = -1.0f;
else if (pan > 1.0f) pan = 1.0f;
if (buffer != NULL)
{
ma_mutex_lock(&AUDIO.System.lock);
buffer->pan = pan;
ma_mutex_unlock(&AUDIO.System.lock);
}
}
// 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 };
// Loading file to memory
int dataSize = 0;
unsigned char *fileData = LoadFileData(fileName, &dataSize);
// Loading wave from memory data
if (fileData != NULL) wave = LoadWaveFromMemory(GetFileExtension(fileName), fileData, dataSize);
UnloadFileData(fileData);
return wave;
}
// Load wave from memory buffer, fileType refers to extension: i.e. ".wav"
// WARNING: File extension must be provided in lower-case
Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int dataSize)
{
Wave wave = { 0 };
if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV)
else if ((strcmp(fileType, ".wav") == 0) || (strcmp(fileType, ".WAV") == 0))
{
drwav wav = { 0 };
bool success = drwav_init_memory(&wav, fileData, dataSize, NULL);
if (success)
{
wave.frameCount = (unsigned int)wav.totalPCMFrameCount;
wave.sampleRate = wav.sampleRate;
wave.sampleSize = 16;
wave.channels = wav.channels;
wave.data = (short *)RL_MALLOC((size_t)wave.frameCount*wave.channels*sizeof(short));
// NOTE: We are forcing conversion to 16bit sample size on reading
drwav_read_pcm_frames_s16(&wav, wave.frameCount, (drwav_int16 *)wave.data);
}
else TRACELOG(LOG_WARNING, "WAVE: Failed to load WAV data");
drwav_uninit(&wav);
}
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
else if ((strcmp(fileType, ".ogg") == 0) || (strcmp(fileType, ".OGG") == 0))
{
stb_vorbis *oggData = stb_vorbis_open_memory((unsigned char *)fileData, dataSize, NULL, NULL);
if (oggData != NULL)
{
stb_vorbis_info info = stb_vorbis_get_info(oggData);
wave.sampleRate = info.sample_rate;
wave.sampleSize = 16; // By default, ogg data is 16 bit per sample (short)
wave.channels = info.channels;
wave.frameCount = (unsigned int)stb_vorbis_stream_length_in_samples(oggData); // NOTE: It returns frames!
wave.data = (short *)RL_MALLOC(wave.frameCount*wave.channels*sizeof(short));
// NOTE: Get the number of samples to process (be careful! we ask for number of shorts, not bytes!)
stb_vorbis_get_samples_short_interleaved(oggData, info.channels, (short *)wave.data, wave.frameCount*wave.channels);
stb_vorbis_close(oggData);
}
else TRACELOG(LOG_WARNING, "WAVE: Failed to load OGG data");
}
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
else if ((strcmp(fileType, ".mp3") == 0) || (strcmp(fileType, ".MP3") == 0))
{
drmp3_config config = { 0 };
unsigned long long int totalFrameCount = 0;
// NOTE: We are forcing conversion to 32bit float sample size on reading
wave.data = drmp3_open_memory_and_read_pcm_frames_f32(fileData, dataSize, &config, &totalFrameCount, NULL);
wave.sampleSize = 32;
if (wave.data != NULL)
{
wave.channels = config.channels;
wave.sampleRate = config.sampleRate;
wave.frameCount = (int)totalFrameCount;
}
else TRACELOG(LOG_WARNING, "WAVE: Failed to load MP3 data");
}
#endif
#if defined(SUPPORT_FILEFORMAT_QOA)
else if ((strcmp(fileType, ".qoa") == 0) || (strcmp(fileType, ".QOA") == 0))
{
qoa_desc qoa = { 0 };
// NOTE: Returned sample data is always 16 bit?
wave.data = qoa_decode(fileData, dataSize, &qoa);
wave.sampleSize = 16;
if (wave.data != NULL)
{
wave.channels = qoa.channels;
wave.sampleRate = qoa.samplerate;
wave.frameCount = qoa.samples;
}
else TRACELOG(LOG_WARNING, "WAVE: Failed to load QOA data");
}
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if ((strcmp(fileType, ".flac") == 0) || (strcmp(fileType, ".FLAC") == 0))
{
unsigned long long int totalFrameCount = 0;
// NOTE: We are forcing conversion to 16bit sample size on reading
wave.data = drflac_open_memory_and_read_pcm_frames_s16(fileData, dataSize, &wave.channels, &wave.sampleRate, &totalFrameCount, NULL);
wave.sampleSize = 16;
if (wave.data != NULL) wave.frameCount = (unsigned int)totalFrameCount;
else TRACELOG(LOG_WARNING, "WAVE: Failed to load FLAC data");
}
#endif
else TRACELOG(LOG_WARNING, "WAVE: Data format not supported");
TRACELOG(LOG_INFO, "WAVE: Data loaded successfully (%i Hz, %i bit, %i channels)", wave.sampleRate, wave.sampleSize, wave.channels);
return wave;
}
// Checks if wave data is valid (data loaded and parameters)
bool IsWaveValid(Wave wave)
{
bool result = false;
if ((wave.data != NULL) && // Validate wave data available
(wave.frameCount > 0) && // Validate frame count
(wave.sampleRate > 0) && // Validate sample rate is supported
(wave.sampleSize > 0) && // Validate sample size is supported
(wave.channels > 0)) result = true; // Validate number of channels supported
return result;
}
// 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.frameCount;
ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, 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.System.device.sampleRate, frameCount, AUDIO_BUFFER_USAGE_STATIC);
if (audioBuffer == NULL)
{
TRACELOG(LOG_WARNING, "SOUND: Failed to create buffer");
return sound; // early return to avoid dereferencing the audioBuffer null pointer
}
frameCount = (ma_uint32)ma_convert_frames(audioBuffer->data, frameCount, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, wave.data, frameCountIn, formatIn, wave.channels, wave.sampleRate);
if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed format conversion");
sound.frameCount = frameCount;
sound.stream.sampleRate = AUDIO.System.device.sampleRate;
sound.stream.sampleSize = 32;
sound.stream.channels = AUDIO_DEVICE_CHANNELS;
sound.stream.buffer = audioBuffer;
}
return sound;
}
// Clone sound from existing sound data, clone does not own wave data
// NOTE: Wave data must be unallocated manually and will be shared across all clones
Sound LoadSoundAlias(Sound source)
{
Sound sound = { 0 };
if (source.stream.buffer->data != NULL)
{
AudioBuffer *audioBuffer = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, 0, AUDIO_BUFFER_USAGE_STATIC);
if (audioBuffer == NULL)
{
TRACELOG(LOG_WARNING, "SOUND: Failed to create buffer");
return sound; // Early return to avoid dereferencing the audioBuffer null pointer
}
audioBuffer->sizeInFrames = source.stream.buffer->sizeInFrames;
audioBuffer->data = source.stream.buffer->data;
// Initalize the buffer as if it was new
audioBuffer->volume = 1.0f;
audioBuffer->pitch = 1.0f;
audioBuffer->pan = 0.0f; // Center
sound.frameCount = source.frameCount;
sound.stream.sampleRate = AUDIO.System.device.sampleRate;
sound.stream.sampleSize = 32;
sound.stream.channels = AUDIO_DEVICE_CHANNELS;
sound.stream.buffer = audioBuffer;
}
return sound;
}
// Checks if a sound is valid (data loaded and buffers initialized)
bool IsSoundValid(Sound sound)
{
bool result = false;
if ((sound.frameCount > 0) && // Validate frame count
(sound.stream.buffer != NULL) && // Validate stream buffer
(sound.stream.sampleRate > 0) && // Validate sample rate is supported
(sound.stream.sampleSize > 0) && // Validate sample size is supported
(sound.stream.channels > 0)) result = true; // Validate number of channels supported
return result;
}
// Unload wave data
void UnloadWave(Wave wave)
{
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, "SOUND: Unloaded sound data from RAM");
}
void UnloadSoundAlias(Sound alias)
{
// Untrack and unload just the sound buffer, not the sample data, it is shared with the source for the alias
if (alias.stream.buffer != NULL)
{
UntrackAudioBuffer(alias.stream.buffer);
ma_data_converter_uninit(&alias.stream.buffer->converter, NULL);
RL_FREE(alias.stream.buffer);
}
}
// Update sound buffer with new data
// PARAMS: [data], format must match sound.stream.sampleSize, default 32 bit float - stereo
// PARAMS: [frameCount] must not exceed sound.frameCount
void UpdateSound(Sound sound, const void *data, int frameCount)
{
if (sound.stream.buffer != NULL)
{
StopAudioBuffer(sound.stream.buffer);
memcpy(sound.stream.buffer->data, data, frameCount*ma_get_bytes_per_frame(sound.stream.buffer->converter.formatIn, sound.stream.buffer->converter.channelsIn));
}
}
// Export wave data to file
bool ExportWave(Wave wave, const char *fileName)
{
bool success = false;
if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV)
else if (IsFileExtension(fileName, ".wav"))
{
drwav wav = { 0 };
drwav_data_format format = { 0 };
format.container = drwav_container_riff;
if (wave.sampleSize == 32) format.format = DR_WAVE_FORMAT_IEEE_FLOAT;
else format.format = DR_WAVE_FORMAT_PCM;
format.channels = wave.channels;
format.sampleRate = wave.sampleRate;
format.bitsPerSample = wave.sampleSize;
void *fileData = NULL;
size_t fileDataSize = 0;
success = drwav_init_memory_write(&wav, &fileData, &fileDataSize, &format, NULL);
if (success) success = (int)drwav_write_pcm_frames(&wav, wave.frameCount, wave.data);
drwav_result result = drwav_uninit(&wav);
if (result == DRWAV_SUCCESS) success = SaveFileData(fileName, (unsigned char *)fileData, (unsigned int)fileDataSize);
drwav_free(fileData, NULL);
}
#endif
#if defined(SUPPORT_FILEFORMAT_QOA)
else if (IsFileExtension(fileName, ".qoa"))
{
if (wave.sampleSize == 16)
{
qoa_desc qoa = { 0 };
qoa.channels = wave.channels;
qoa.samplerate = wave.sampleRate;
qoa.samples = wave.frameCount;
int bytesWritten = qoa_write(fileName, (const short *)wave.data, &qoa);
if (bytesWritten > 0) success = true;
}
else TRACELOG(LOG_WARNING, "AUDIO: Wave data must be 16 bit per sample for QOA format export");
}
#endif
else if (IsFileExtension(fileName, ".raw"))
{
// Export raw sample data (without header)
// NOTE: It's up to the user to track wave parameters
success = SaveFileData(fileName, wave.data, wave.frameCount*wave.channels*wave.sampleSize/8);
}
if (success) TRACELOG(LOG_INFO, "FILEIO: [%s] Wave data exported successfully", fileName);
else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to export wave data", fileName);
return success;
}
// Export wave sample data to code (.h)
bool ExportWaveAsCode(Wave wave, const char *fileName)
{
bool success = false;
#ifndef TEXT_BYTES_PER_LINE
#define TEXT_BYTES_PER_LINE 20
#endif
int waveDataSize = wave.frameCount*wave.channels*wave.sampleSize/8;
// NOTE: Text data buffer size is estimated considering wave data size in bytes
// and requiring 12 char bytes for every byte; the actual size varies, but
// the longest possible char being appended is "%.4ff,\n ", which is 12 bytes
char *txtData = (char *)RL_CALLOC(waveDataSize*12 + 2000, sizeof(char));
int byteCount = 0;
byteCount += sprintf(txtData + byteCount, "\n//////////////////////////////////////////////////////////////////////////////////\n");
byteCount += sprintf(txtData + byteCount, "// //\n");
byteCount += sprintf(txtData + byteCount, "// WaveAsCode exporter v1.1 - Wave data exported as an array of bytes //\n");
byteCount += sprintf(txtData + byteCount, "// //\n");
byteCount += sprintf(txtData + byteCount, "// more info and bugs-report: github.com/raysan5/raylib //\n");
byteCount += sprintf(txtData + byteCount, "// feedback and support: ray[at]raylib.com //\n");
byteCount += sprintf(txtData + byteCount, "// //\n");
byteCount += sprintf(txtData + byteCount, "// Copyright (c) 2018-2026 Ramon Santamaria (@raysan5) //\n");
byteCount += sprintf(txtData + byteCount, "// //\n");
byteCount += sprintf(txtData + byteCount, "//////////////////////////////////////////////////////////////////////////////////\n\n");
// Get file name from path and convert variable name to uppercase
char varFileName[256] = { 0 };
strncpy(varFileName, GetFileNameWithoutExt(fileName), 256 - 1);
for (int i = 0; varFileName[i] != '\0'; i++) if (varFileName[i] >= 'a' && varFileName[i] <= 'z') { varFileName[i] = varFileName[i] - 32; }
// Add wave information
byteCount += sprintf(txtData + byteCount, "// Wave data information\n");
byteCount += sprintf(txtData + byteCount, "#define %s_FRAME_COUNT %u\n", varFileName, wave.frameCount);
byteCount += sprintf(txtData + byteCount, "#define %s_SAMPLE_RATE %u\n", varFileName, wave.sampleRate);
byteCount += sprintf(txtData + byteCount, "#define %s_SAMPLE_SIZE %u\n", varFileName, wave.sampleSize);
byteCount += sprintf(txtData + byteCount, "#define %s_CHANNELS %u\n\n", varFileName, wave.channels);
// Write wave data as an array of values
// Wave data is exported as byte array for 8/16bit and float array for 32bit float data
// NOTE: Frame data exported is channel-interlaced: frame01[sampleChannel1, sampleChannel2, ...], frame02[], frame03[]
if (wave.sampleSize == 32)
{
byteCount += sprintf(txtData + byteCount, "static float %s_DATA[%i] = {\n", varFileName, waveDataSize/4);
for (int i = 1; i < waveDataSize/4; i++) byteCount += sprintf(txtData + byteCount, ((i%TEXT_BYTES_PER_LINE == 0)? "%.4ff,\n " : "%.4ff, "), ((float *)wave.data)[i - 1]);
byteCount += sprintf(txtData + byteCount, "%.4ff };\n", ((float *)wave.data)[waveDataSize/4 - 1]);
}
else
{
byteCount += sprintf(txtData + byteCount, "static unsigned char %s_DATA[%i] = { ", varFileName, waveDataSize);
for (int i = 1; i < waveDataSize; i++) byteCount += sprintf(txtData + byteCount, ((i%TEXT_BYTES_PER_LINE == 0)? "0x%x,\n " : "0x%x, "), ((unsigned char *)wave.data)[i - 1]);
byteCount += sprintf(txtData + byteCount, "0x%x };\n", ((unsigned char *)wave.data)[waveDataSize - 1]);
}
// NOTE: Text data length exported is determined by '\0' (NULL) character
success = SaveFileText(fileName, txtData);
RL_FREE(txtData);
if (success != 0) TRACELOG(LOG_INFO, "FILEIO: [%s] Wave as code exported successfully", fileName);
else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to export wave as code", fileName);
return success;
}
// Play a sound
void PlaySound(Sound sound)
{
PlayAudioBuffer(sound.stream.buffer);
}
// 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)
{
bool result = false;
if (IsAudioBufferPlaying(sound.stream.buffer)) result = true;
return result;
}
// 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);
}
// Set pan for a sound
void SetSoundPan(Sound sound, float pan)
{
SetAudioBufferPan(sound.stream.buffer, pan);
}
// 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->frameCount;
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)
{
RL_FREE(data);
TRACELOG(LOG_WARNING, "WAVE: Failed format conversion");
return;
}
wave->frameCount = 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.frameCount*wave.channels*wave.sampleSize/8);
if (newWave.data != NULL)
{
// NOTE: Size must be provided in bytes
memcpy(newWave.data, wave.data, wave.frameCount*wave.channels*wave.sampleSize/8);
newWave.frameCount = wave.frameCount;
newWave.sampleRate = wave.sampleRate;
newWave.sampleSize = wave.sampleSize;
newWave.channels = wave.channels;
}
return newWave;
}
// Crop a wave to defined frames range
// NOTE: Security check in case of out-of-range
void WaveCrop(Wave *wave, int initFrame, int finalFrame)
{
if ((initFrame >= 0) && (initFrame < finalFrame) && ((unsigned int)finalFrame <= wave->frameCount))
{
int frameCount = finalFrame - initFrame;
void *data = RL_MALLOC(frameCount*wave->channels*wave->sampleSize/8);
memcpy(data, (unsigned char *)wave->data + (initFrame*wave->channels*wave->sampleSize/8), frameCount*wave->channels*wave->sampleSize/8);
RL_FREE(wave->data);
wave->data = data;
wave->frameCount = (unsigned int)frameCount;
}
else TRACELOG(LOG_WARNING, "WAVE: Crop range out of bounds");
}
// Load samples data from wave as a floats array
// NOTE 1: Returned sample values are normalized to range [-1..1]
// NOTE 2: Sample data allocated should be freed with UnloadWaveSamples()
float *LoadWaveSamples(Wave wave)
{
float *samples = (float *)RL_MALLOC(wave.frameCount*wave.channels*sizeof(float));
// NOTE: sampleCount is the total number of interlaced samples (including channels)
for (unsigned int i = 0; i < wave.frameCount*wave.channels; i++)
{
if (wave.sampleSize == 8) samples[i] = (float)(((unsigned char *)wave.data)[i] - 128)/128.0f;
else if (wave.sampleSize == 16) samples[i] = (float)(((short *)wave.data)[i])/32768.0f;
else if (wave.sampleSize == 32) samples[i] = ((float *)wave.data)[i];
}
return samples;
}
// Unload samples data loaded with LoadWaveSamples()
void UnloadWaveSamples(float *samples)
{
RL_FREE(samples);
}
//----------------------------------------------------------------------------------
// Module Functions Definition - Music loading and stream playing
//----------------------------------------------------------------------------------
// Load music stream from file
Music LoadMusicStream(const char *fileName)
{
Music music = { 0 };
bool musicLoaded = false;
if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV)
else if (IsFileExtension(fileName, ".wav"))
{
drwav *ctxWav = (drwav *)RL_CALLOC(1, sizeof(drwav));
bool success = drwav_init_file(ctxWav, fileName, NULL);
if (success)
{
music.ctxType = MUSIC_AUDIO_WAV;
music.ctxData = ctxWav;
int sampleSize = ctxWav->bitsPerSample;
if (ctxWav->bitsPerSample == 24) sampleSize = 16; // Forcing conversion to s16 on UpdateMusicStream()
music.stream = LoadAudioStream(ctxWav->sampleRate, sampleSize, ctxWav->channels);
music.frameCount = (unsigned int)ctxWav->totalPCMFrameCount;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
else
{
RL_FREE(ctxWav);
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
else if (IsFileExtension(fileName, ".ogg"))
{
// Open ogg audio stream
stb_vorbis *ctxOgg = stb_vorbis_open_filename(fileName, NULL, NULL);
if (ctxOgg != NULL)
{
music.ctxType = MUSIC_AUDIO_OGG;
music.ctxData = ctxOgg;
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 = LoadAudioStream(info.sample_rate, 16, info.channels);
// WARNING: It seems this function returns length in frames, not samples, so we multiply by channels
music.frameCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData);
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
else
{
stb_vorbis_close(ctxOgg);
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
else if (IsFileExtension(fileName, ".mp3"))
{
drmp3 *ctxMp3 = (drmp3 *)RL_CALLOC(1, sizeof(drmp3));
int result = drmp3_init_file(ctxMp3, fileName, NULL);
if (result > 0)
{
music.ctxType = MUSIC_AUDIO_MP3;
music.ctxData = ctxMp3;
music.stream = LoadAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
music.frameCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3);
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
else
{
RL_FREE(ctxMp3);
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_QOA)
else if (IsFileExtension(fileName, ".qoa"))
{
qoaplay_desc *ctxQoa = qoaplay_open(fileName);
if (ctxQoa != NULL)
{
music.ctxType = MUSIC_AUDIO_QOA;
music.ctxData = ctxQoa;
// NOTE: We are loading samples are 32bit float normalized data, so,
// we configure the output audio stream to also use float 32bit
music.stream = LoadAudioStream(ctxQoa->info.samplerate, 32, ctxQoa->info.channels);
music.frameCount = ctxQoa->info.samples;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
else{} //No uninit required
}
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if (IsFileExtension(fileName, ".flac"))
{
drflac *ctxFlac = drflac_open_file(fileName, NULL);
if (ctxFlac != NULL)
{
music.ctxType = MUSIC_AUDIO_FLAC;
music.ctxData = ctxFlac;
int sampleSize = ctxFlac->bitsPerSample;
if (ctxFlac->bitsPerSample == 24) sampleSize = 16; // Forcing conversion to s16 on UpdateMusicStream()
music.stream = LoadAudioStream(ctxFlac->sampleRate, sampleSize, ctxFlac->channels);
music.frameCount = (unsigned int)ctxFlac->totalPCMFrameCount;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
else
{
drflac_free(ctxFlac, NULL);
}
}
#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, AUDIO.System.device.sampleRate, fileName);
if (result == 0) // XM AUDIO.System.context created successfully
{
music.ctxType = MUSIC_MODULE_XM;
music.ctxData = ctxXm;
jar_xm_set_max_loop_count(ctxXm, 0); // Set infinite number of loops
unsigned int bits = 32;
if (AUDIO_DEVICE_FORMAT == ma_format_s16) bits = 16;
else if (AUDIO_DEVICE_FORMAT == ma_format_u8) bits = 8;
// NOTE: Only stereo is supported for XM
music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, bits, AUDIO_DEVICE_CHANNELS);
music.frameCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm); // NOTE: Always 2 channels (stereo)
music.looping = true; // Looping enabled by default
jar_xm_reset(ctxXm); // Make sure we start at the beginning of the song
musicLoaded = true;
}
else
{
jar_xm_free_context(ctxXm);
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
else if (IsFileExtension(fileName, ".mod"))
{
jar_mod_context_t *ctxMod = (jar_mod_context_t *)RL_CALLOC(1, 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;
music.ctxData = ctxMod;
// NOTE: Only stereo is supported for MOD
music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, 16, AUDIO_DEVICE_CHANNELS);
music.frameCount = (unsigned int)jar_mod_max_samples(ctxMod); // NOTE: Always 2 channels (stereo)
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
else
{
jar_mod_unload(ctxMod);
RL_FREE(ctxMod);
}
}
#endif
else TRACELOG(LOG_WARNING, "STREAM: [%s] File format not supported", fileName);
if (!musicLoaded)
{
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 loaded successfully", fileName);
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");
TRACELOG(LOG_INFO, " > Total frames: %i", music.frameCount);
}
return music;
}
// Load music stream from memory buffer, fileType refers to extension: i.e. ".wav"
// WARNING: File extension must be provided in lower-case
Music LoadMusicStreamFromMemory(const char *fileType, const unsigned char *data, int dataSize)
{
Music music = { 0 };
bool musicLoaded = false;
if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV)
else if ((strcmp(fileType, ".wav") == 0) || (strcmp(fileType, ".WAV") == 0))
{
drwav *ctxWav = (drwav *)RL_CALLOC(1, sizeof(drwav));
bool success = drwav_init_memory(ctxWav, (const void *)data, dataSize, NULL);
if (success)
{
music.ctxType = MUSIC_AUDIO_WAV;
music.ctxData = ctxWav;
int sampleSize = ctxWav->bitsPerSample;
if (ctxWav->bitsPerSample == 24) sampleSize = 16; // Forcing conversion to s16 on UpdateMusicStream()
music.stream = LoadAudioStream(ctxWav->sampleRate, sampleSize, ctxWav->channels);
music.frameCount = (unsigned int)ctxWav->totalPCMFrameCount;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
else
{
drwav_uninit(ctxWav);
RL_FREE(ctxWav);
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
else if ((strcmp(fileType, ".ogg") == 0) || (strcmp(fileType, ".OGG") == 0))
{
// Open ogg audio stream
stb_vorbis *ctxOgg = stb_vorbis_open_memory((const unsigned char *)data, dataSize, NULL, NULL);
if (ctxOgg != NULL)
{
music.ctxType = MUSIC_AUDIO_OGG;
music.ctxData = ctxOgg;
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 = LoadAudioStream(info.sample_rate, 16, info.channels);
// WARNING: It seems this function returns length in frames, not samples, so we multiply by channels
music.frameCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData);
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
else
{
stb_vorbis_close(ctxOgg);
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
else if ((strcmp(fileType, ".mp3") == 0) || (strcmp(fileType, ".MP3") == 0))
{
drmp3 *ctxMp3 = (drmp3 *)RL_CALLOC(1, sizeof(drmp3));
int success = drmp3_init_memory(ctxMp3, (const void *)data, dataSize, NULL);
if (success)
{
music.ctxType = MUSIC_AUDIO_MP3;
music.ctxData = ctxMp3;
music.stream = LoadAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
music.frameCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3);
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
else
{
drmp3_uninit(ctxMp3);
RL_FREE(ctxMp3);
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_QOA)
else if ((strcmp(fileType, ".qoa") == 0) || (strcmp(fileType, ".QOA") == 0))
{
qoaplay_desc *ctxQoa = NULL;
if ((data != NULL) && (dataSize > 0))
{
ctxQoa = qoaplay_open_memory(data, dataSize);
}
if (ctxQoa != NULL)
{
music.ctxType = MUSIC_AUDIO_QOA;
music.ctxData = ctxQoa;
// NOTE: We are loading samples are 32bit float normalized data, so,
// we configure the output audio stream to also use float 32bit
music.stream = LoadAudioStream(ctxQoa->info.samplerate, 32, ctxQoa->info.channels);
music.frameCount = ctxQoa->info.samples;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
else{} //No uninit required
}
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if ((strcmp(fileType, ".flac") == 0) || (strcmp(fileType, ".FLAC") == 0))
{
drflac *ctxFlac = drflac_open_memory((const void *)data, dataSize, NULL);
if (ctxFlac != NULL)
{
music.ctxType = MUSIC_AUDIO_FLAC;
music.ctxData = ctxFlac;
int sampleSize = ctxFlac->bitsPerSample;
if (ctxFlac->bitsPerSample == 24) sampleSize = 16; // Forcing conversion to s16 on UpdateMusicStream()
music.stream = LoadAudioStream(ctxFlac->sampleRate, sampleSize, ctxFlac->channels);
music.frameCount = (unsigned int)ctxFlac->totalPCMFrameCount;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
else
{
drflac_free(ctxFlac, NULL);
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
else if ((strcmp(fileType, ".xm") == 0) || (strcmp(fileType, ".XM") == 0))
{
jar_xm_context_t *ctxXm = NULL;
int result = jar_xm_create_context_safe(&ctxXm, (const char *)data, dataSize, AUDIO.System.device.sampleRate);
if (result == 0) // XM AUDIO.System.context created successfully
{
music.ctxType = MUSIC_MODULE_XM;
music.ctxData = ctxXm;
jar_xm_set_max_loop_count(ctxXm, 0); // Set infinite number of loops
unsigned int bits = 32;
if (AUDIO_DEVICE_FORMAT == ma_format_s16) bits = 16;
else if (AUDIO_DEVICE_FORMAT == ma_format_u8) bits = 8;
// NOTE: Only stereo is supported for XM
music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, bits, 2);
music.frameCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm); // NOTE: Always 2 channels (stereo)
music.looping = true; // Looping enabled by default
jar_xm_reset(ctxXm); // Make sure we start at the beginning of the song
musicLoaded = true;
}
else
{
jar_xm_free_context(ctxXm);
}
}
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
else if ((strcmp(fileType, ".mod") == 0) || (strcmp(fileType, ".MOD") == 0))
{
jar_mod_context_t *ctxMod = (jar_mod_context_t *)RL_MALLOC(sizeof(jar_mod_context_t));
int result = 0;
jar_mod_init(ctxMod);
// Copy data to allocated memory for default UnloadMusicStream
unsigned char *newData = (unsigned char *)RL_MALLOC(dataSize);
int it = dataSize/sizeof(unsigned char);
for (int i = 0; i < it; i++) newData[i] = data[i];
// Memory loaded version for jar_mod_load_file()
if (dataSize && (dataSize < 32*1024*1024))
{
ctxMod->modfilesize = dataSize;
ctxMod->modfile = newData;
if (jar_mod_load(ctxMod, (void *)ctxMod->modfile, dataSize)) result = dataSize;
}
if (result > 0)
{
music.ctxType = MUSIC_MODULE_MOD;
music.ctxData = ctxMod;
// NOTE: Only stereo is supported for MOD
music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, 16, 2);
music.frameCount = (unsigned int)jar_mod_max_samples(ctxMod); // NOTE: Always 2 channels (stereo)
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
else
{
jar_mod_unload(ctxMod);
RL_FREE(ctxMod);
}
}
#endif
else TRACELOG(LOG_WARNING, "STREAM: Data format not supported");
if (!musicLoaded)
{
TRACELOG(LOG_WARNING, "FILEIO: Music data could not be loaded");
}
else
{
// Show some music stream info
TRACELOG(LOG_INFO, "FILEIO: Music data loaded successfully");
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");
TRACELOG(LOG_INFO, " > Total frames: %i", music.frameCount);
}
return music;
}
// Checks if a music stream is valid (context and buffers initialized)
bool IsMusicValid(Music music)
{
return ((music.ctxData != NULL) && // Validate context loaded
(music.frameCount > 0) && // Validate audio frame count
(music.stream.sampleRate > 0) && // Validate sample rate is supported
(music.stream.sampleSize > 0) && // Validate sample size is supported
(music.stream.channels > 0)); // Validate number of channels supported
}
// Unload music stream
void UnloadMusicStream(Music music)
{
UnloadAudioStream(music.stream);
if (music.ctxData != NULL)
{
if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV)
else if (music.ctxType == MUSIC_AUDIO_WAV) drwav_uninit((drwav *)music.ctxData);
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)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_QOA)
else if (music.ctxType == MUSIC_AUDIO_QOA) qoaplay_close((qoaplay_desc *)music.ctxData);
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
else if (music.ctxType == MUSIC_AUDIO_FLAC) { drflac_close((drflac *)music.ctxData); drflac_free((drflac *)music.ctxData, NULL); }
#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) from beginning
void PlayMusicStream(Music music)
{
PlayAudioStream(music.stream);
}
// 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_WAV)
case MUSIC_AUDIO_WAV: drwav_seek_to_first_pcm_frame((drwav *)music.ctxData); break;
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
case MUSIC_AUDIO_OGG: stb_vorbis_seek_start((stb_vorbis *)music.ctxData); break;
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
case MUSIC_AUDIO_MP3: drmp3_seek_to_start_of_stream((drmp3 *)music.ctxData); break;
#endif
#if defined(SUPPORT_FILEFORMAT_QOA)
case MUSIC_AUDIO_QOA: qoaplay_rewind((qoaplay_desc *)music.ctxData); break;
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
case MUSIC_AUDIO_FLAC: drflac__seek_to_first_frame((drflac *)music.ctxData); 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;
}
}
// Seek music to a certain position (in seconds)
void SeekMusicStream(Music music, float position)
{
// Seeking is not supported in module formats
if ((music.ctxType == MUSIC_MODULE_XM) || (music.ctxType == MUSIC_MODULE_MOD)) return;
unsigned int positionInFrames = (unsigned int)(position*music.stream.sampleRate);
switch (music.ctxType)
{
#if defined(SUPPORT_FILEFORMAT_WAV)
case MUSIC_AUDIO_WAV: drwav_seek_to_pcm_frame((drwav *)music.ctxData, positionInFrames); break;
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
case MUSIC_AUDIO_OGG: stb_vorbis_seek_frame((stb_vorbis *)music.ctxData, positionInFrames); break;
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame((drmp3 *)music.ctxData, positionInFrames); break;
#endif
#if defined(SUPPORT_FILEFORMAT_QOA)
case MUSIC_AUDIO_QOA:
{
int qoaFrame = positionInFrames/QOA_FRAME_LEN;
qoaplay_seek_frame((qoaplay_desc *)music.ctxData, qoaFrame); // Seeks to QOA frame, not PCM frame
// We need to compute QOA frame number and update positionInFrames
positionInFrames = ((qoaplay_desc *)music.ctxData)->sample_position;
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
case MUSIC_AUDIO_FLAC: drflac_seek_to_pcm_frame((drflac *)music.ctxData, positionInFrames); break;
#endif
default: break;
}
ma_mutex_lock(&AUDIO.System.lock);
music.stream.buffer->framesProcessed = positionInFrames;
music.stream.buffer->isSubBufferProcessed[0] = true;
music.stream.buffer->isSubBufferProcessed[1] = true;
ma_mutex_unlock(&AUDIO.System.lock);
}
// Update (re-fill) music buffers if data already processed
void UpdateMusicStream(Music music)
{
if (music.stream.buffer == NULL) return;
if (!music.stream.buffer->playing) return;
ma_mutex_lock(&AUDIO.System.lock);
unsigned int subBufferSizeInFrames = music.stream.buffer->sizeInFrames/2;
// On first call of this function we lazily pre-allocated a temp buffer to read audio files/memory data in
int frameSize = music.stream.channels*music.stream.sampleSize/8;
unsigned int pcmSize = subBufferSizeInFrames*frameSize;
if (AUDIO.System.pcmBufferSize < pcmSize)
{
RL_FREE(AUDIO.System.pcmBuffer);
AUDIO.System.pcmBuffer = RL_CALLOC(1, pcmSize);
AUDIO.System.pcmBufferSize = pcmSize;
}
// Check both sub-buffers to check if they require refilling
for (int i = 0; i < 2; i++)
{
unsigned int framesLeft = music.frameCount - music.stream.buffer->framesProcessed; // Frames left to be processed
unsigned int framesToStream = 0; // Total frames to be streamed
if ((framesLeft >= subBufferSizeInFrames) || music.looping) framesToStream = subBufferSizeInFrames;
else framesToStream = framesLeft;
if (framesToStream == 0)
{
// Check if both buffers have been processed
if (music.stream.buffer->isSubBufferProcessed[0] && music.stream.buffer->isSubBufferProcessed[1])
{
ma_mutex_unlock(&AUDIO.System.lock);
StopMusicStream(music);
return;
}
ma_mutex_unlock(&AUDIO.System.lock);
return;
}
if (!music.stream.buffer->isSubBufferProcessed[i]) continue; // No refilling required, move to next sub-buffer
int frameCountStillNeeded = framesToStream;
int frameCountReadTotal = 0;
switch (music.ctxType)
{
#if defined(SUPPORT_FILEFORMAT_WAV)
case MUSIC_AUDIO_WAV:
{
if (music.stream.sampleSize == 16)
{
while (true)
{
int frameCountRead = (int)drwav_read_pcm_frames_s16((drwav *)music.ctxData, frameCountStillNeeded, (short *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize));
frameCountReadTotal += frameCountRead;
frameCountStillNeeded -= frameCountRead;
if (frameCountStillNeeded == 0) break;
else drwav_seek_to_first_pcm_frame((drwav *)music.ctxData);
}
}
else if (music.stream.sampleSize == 32)
{
while (true)
{
int frameCountRead = (int)drwav_read_pcm_frames_f32((drwav *)music.ctxData, frameCountStillNeeded, (float *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize));
frameCountReadTotal += frameCountRead;
frameCountStillNeeded -= frameCountRead;
if (frameCountStillNeeded == 0) break;
else drwav_seek_to_first_pcm_frame((drwav *)music.ctxData);
}
}
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
case MUSIC_AUDIO_OGG:
{
while (true)
{
int frameCountRead = stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize), frameCountStillNeeded*music.stream.channels);
frameCountReadTotal += frameCountRead;
frameCountStillNeeded -= frameCountRead;
if (frameCountStillNeeded == 0) break;
else stb_vorbis_seek_start((stb_vorbis *)music.ctxData);
}
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
case MUSIC_AUDIO_MP3:
{
while (true)
{
int frameCountRead = (int)drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, frameCountStillNeeded, (float *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize));
frameCountReadTotal += frameCountRead;
frameCountStillNeeded -= frameCountRead;
if (frameCountStillNeeded == 0) break;
else drmp3_seek_to_start_of_stream((drmp3 *)music.ctxData);
}
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_QOA)
case MUSIC_AUDIO_QOA:
{
unsigned int frameCountRead = qoaplay_decode((qoaplay_desc *)music.ctxData, (float *)AUDIO.System.pcmBuffer, framesToStream);
frameCountReadTotal += frameCountRead;
/*
while (true)
{
int frameCountRead = (int)qoaplay_decode((qoaplay_desc *)music.ctxData, (float *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize), frameCountStillNeeded);
frameCountReadTotal += frameCountRead;
frameCountStillNeeded -= frameCountRead;
if (frameCountStillNeeded == 0) break;
else qoaplay_rewind((qoaplay_desc *)music.ctxData);
}
*/
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
case MUSIC_AUDIO_FLAC:
{
while (true)
{
int frameCountRead = (int)drflac_read_pcm_frames_s16((drflac *)music.ctxData, frameCountStillNeeded, (short *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize));
frameCountReadTotal += frameCountRead;
frameCountStillNeeded -= frameCountRead;
if (frameCountStillNeeded == 0) break;
else drflac__seek_to_first_frame((drflac *)music.ctxData);
}
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
case MUSIC_MODULE_XM:
{
// NOTE: Internally we consider 2 channels generation, so sampleCount/2
if (AUDIO_DEVICE_FORMAT == ma_format_f32) jar_xm_generate_samples((jar_xm_context_t *)music.ctxData, (float *)AUDIO.System.pcmBuffer, framesToStream);
else if (AUDIO_DEVICE_FORMAT == ma_format_s16) jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)AUDIO.System.pcmBuffer, framesToStream);
else if (AUDIO_DEVICE_FORMAT == ma_format_u8) jar_xm_generate_samples_8bit((jar_xm_context_t *)music.ctxData, (char *)AUDIO.System.pcmBuffer, framesToStream);
//jar_xm_reset((jar_xm_context_t *)music.ctxData);
} 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 *)AUDIO.System.pcmBuffer, framesToStream, 0);
//jar_mod_seek_start((jar_mod_context_t *)music.ctxData);
} break;
#endif
default: break;
}
UpdateAudioStreamInLockedState(music.stream, AUDIO.System.pcmBuffer, framesToStream);
}
ma_mutex_unlock(&AUDIO.System.lock);
}
// Check if any music is playing
bool IsMusicStreamPlaying(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)
{
SetAudioBufferPitch(music.stream.buffer, pitch);
}
// Set pan for a music
void SetMusicPan(Music music, float pan)
{
SetAudioBufferPan(music.stream.buffer, pan);
}
// Get music time length (in seconds)
float GetMusicTimeLength(Music music)
{
float totalSeconds = 0.0f;
totalSeconds = (float)music.frameCount/music.stream.sampleRate;
return totalSeconds;
}
// Get current music time played (in seconds)
float GetMusicTimePlayed(Music music)
{
float secondsPlayed = 0.0f;
if (music.stream.buffer != NULL)
{
#if defined(SUPPORT_FILEFORMAT_XM)
if (music.ctxType == MUSIC_MODULE_XM)
{
uint64_t framesPlayed = 0;
jar_xm_get_position((jar_xm_context_t *)music.ctxData, NULL, NULL, NULL, &framesPlayed);
secondsPlayed = (float)framesPlayed/music.stream.sampleRate;
}
else
#endif
{
ma_mutex_lock(&AUDIO.System.lock);
//ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
int framesProcessed = (int)music.stream.buffer->framesProcessed;
int subBufferSize = (int)music.stream.buffer->sizeInFrames/2;
int framesInFirstBuffer = music.stream.buffer->isSubBufferProcessed[0]? 0 : subBufferSize;
int framesInSecondBuffer = music.stream.buffer->isSubBufferProcessed[1]? 0 : subBufferSize;
int framesInBuffers = framesInFirstBuffer + framesInSecondBuffer;
if (((unsigned int)framesInBuffers > music.frameCount) && !music.looping) framesInBuffers = music.frameCount;
int framesSentToMix = music.stream.buffer->frameCursorPos%subBufferSize;
int framesPlayed = (framesProcessed - framesInBuffers + framesSentToMix)%(int)music.frameCount;
if (framesPlayed < 0) framesPlayed += music.frameCount;
secondsPlayed = (float)framesPlayed/music.stream.sampleRate;
ma_mutex_unlock(&AUDIO.System.lock);
}
}
return secondsPlayed;
}
// Load audio stream (to stream audio pcm data)
AudioStream LoadAudioStream(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;
// If the buffer is not set, compute one that would give us a buffer good enough for a decent frame rate at the device bit size/rate
int deviceBitsPerSample = AUDIO.System.device.playback.format;
if (deviceBitsPerSample > 4) deviceBitsPerSample = 4;
deviceBitsPerSample *= AUDIO.System.device.playback.channels;
unsigned int subBufferSize = (AUDIO.Buffer.defaultSize == 0)? (AUDIO.System.device.sampleRate/30*deviceBitsPerSample) : AUDIO.Buffer.defaultSize;
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;
}
// Checks if an audio stream is valid (buffers initialized)
bool IsAudioStreamValid(AudioStream stream)
{
return ((stream.buffer != NULL) && // Validate stream buffer
(stream.sampleRate > 0) && // Validate sample rate is supported
(stream.sampleSize > 0) && // Validate sample size is supported
(stream.channels > 0)); // Validate number of channels supported
}
// Unload audio stream and free memory
void UnloadAudioStream(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: dequeue -> update -> queue
// NOTE 2: To dequeue a buffer it needs to be processed: IsAudioStreamProcessed()
void UpdateAudioStream(AudioStream stream, const void *data, int frameCount)
{
ma_mutex_lock(&AUDIO.System.lock);
UpdateAudioStreamInLockedState(stream, data, frameCount);
ma_mutex_unlock(&AUDIO.System.lock);
}
// Check if any audio stream buffers requires refill
bool IsAudioStreamProcessed(AudioStream stream)
{
if (stream.buffer == NULL) return false;
bool result = false;
ma_mutex_lock(&AUDIO.System.lock);
result = stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1];
ma_mutex_unlock(&AUDIO.System.lock);
return result;
}
// 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);
}
// Set pan for audio stream
void SetAudioStreamPan(AudioStream stream, float pan)
{
SetAudioBufferPan(stream.buffer, pan);
}
// Default size for new audio streams
void SetAudioStreamBufferSizeDefault(int size)
{
AUDIO.Buffer.defaultSize = size;
}
// Audio thread callback to request new data
void SetAudioStreamCallback(AudioStream stream, AudioCallback callback)
{
if (stream.buffer != NULL)
{
ma_mutex_lock(&AUDIO.System.lock);
stream.buffer->callback = callback;
ma_mutex_unlock(&AUDIO.System.lock);
}
}
// Add processor to audio stream. Contrary to buffers, the order of processors is important
// The new processor must be added at the end. As there aren't supposed to be a lot of processors attached to
// a given stream, we iterate through the list to find the end. That way we don't need a pointer to the last element
void AttachAudioStreamProcessor(AudioStream stream, AudioCallback process)
{
ma_mutex_lock(&AUDIO.System.lock);
rAudioProcessor *processor = (rAudioProcessor *)RL_CALLOC(1, sizeof(rAudioProcessor));
processor->process = process;
rAudioProcessor *last = stream.buffer->processor;
while (last && last->next)
{
last = last->next;
}
if (last)
{
processor->prev = last;
last->next = processor;
}
else stream.buffer->processor = processor;
ma_mutex_unlock(&AUDIO.System.lock);
}
// Remove processor from audio stream
void DetachAudioStreamProcessor(AudioStream stream, AudioCallback process)
{
ma_mutex_lock(&AUDIO.System.lock);
rAudioProcessor *processor = stream.buffer->processor;
while (processor)
{
rAudioProcessor *next = processor->next;
rAudioProcessor *prev = processor->prev;
if (processor->process == process)
{
if (stream.buffer->processor == processor) stream.buffer->processor = next;
if (prev) prev->next = next;
if (next) next->prev = prev;
RL_FREE(processor);
}
processor = next;
}
ma_mutex_unlock(&AUDIO.System.lock);
}
// Add processor to audio pipeline. Order of processors is important
// Works the same way as {Attach,Detach}AudioStreamProcessor() functions, except
// these two work on the already mixed output just before sending it to the sound hardware
void AttachAudioMixedProcessor(AudioCallback process)
{
ma_mutex_lock(&AUDIO.System.lock);
rAudioProcessor *processor = (rAudioProcessor *)RL_CALLOC(1, sizeof(rAudioProcessor));
processor->process = process;
rAudioProcessor *last = AUDIO.mixedProcessor;
while (last && last->next)
{
last = last->next;
}
if (last)
{
processor->prev = last;
last->next = processor;
}
else AUDIO.mixedProcessor = processor;
ma_mutex_unlock(&AUDIO.System.lock);
}
// Remove processor from audio pipeline
void DetachAudioMixedProcessor(AudioCallback process)
{
ma_mutex_lock(&AUDIO.System.lock);
rAudioProcessor *processor = AUDIO.mixedProcessor;
while (processor)
{
rAudioProcessor *next = processor->next;
rAudioProcessor *prev = processor->prev;
if (processor->process == process)
{
if (AUDIO.mixedProcessor == processor) AUDIO.mixedProcessor = next;
if (prev) prev->next = next;
if (next) next->prev = prev;
RL_FREE(processor);
}
processor = next;
}
ma_mutex_unlock(&AUDIO.System.lock);
}
//----------------------------------------------------------------------------------
// Module Internal Functions Definition
//----------------------------------------------------------------------------------
// Log callback function
static void OnLog(void *pUserData, ma_uint32 level, const char *pMessage)
{
TRACELOG(LOG_WARNING, "miniaudio: %s", pMessage); // 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)
{
// Using audio buffer callback
if (audioBuffer->callback)
{
audioBuffer->callback(framesOut, frameCount);
audioBuffer->framesProcessed += frameCount;
return 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] = { 0 };
isSubBufferProcessed[0] = audioBuffer->isSubBufferProcessed[0];
isSubBufferProcessed[1] = audioBuffer->isSubBufferProcessed[1];
ma_uint32 frameSizeInBytes = ma_get_bytes_per_frame(audioBuffer->converter.formatIn, audioBuffer->converter.channelsIn);
// Fill out every frame until we find a buffer that's marked as processed. Then fill the remainder with 0
ma_uint32 framesRead = 0;
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 half 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)
{
StopAudioBufferInLockedState(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 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] = { 0 };
ma_uint32 inputBufferFrameCap = sizeof(inputBuffer)/ma_get_bytes_per_frame(audioBuffer->converter.formatIn, audioBuffer->converter.channelsIn);
ma_uint32 totalOutputFramesProcessed = 0;
while (totalOutputFramesProcessed < frameCount)
{
ma_uint64 outputFramesToProcessThisIteration = frameCount - totalOutputFramesProcessed;
ma_uint64 inputFramesToProcessThisIteration = 0;
(void)ma_data_converter_get_required_input_frame_count(&audioBuffer->converter, outputFramesToProcessThisIteration, &inputFramesToProcessThisIteration);
if (inputFramesToProcessThisIteration > inputBufferFrameCap)
{
inputFramesToProcessThisIteration = inputBufferFrameCap;
}
float *runningFramesOut = framesOut + (totalOutputFramesProcessed*audioBuffer->converter.channelsOut);
// At this point we can convert the data to our mixing format
ma_uint64 inputFramesProcessedThisIteration = ReadAudioBufferFramesInInternalFormat(audioBuffer, inputBuffer, (ma_uint32)inputFramesToProcessThisIteration);
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 added 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
// This function will be called when miniaudio needs more data
// 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] = { 0 }; // 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;
// Apply processors chain if defined
rAudioProcessor *processor = audioBuffer->processor;
while (processor)
{
processor->process(framesIn, framesJustRead);
processor = processor->next;
}
MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer);
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)
{
StopAudioBufferInLockedState(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;
}
}
}
rAudioProcessor *processor = AUDIO.mixedProcessor;
while (processor)
{
processor->process(pFramesOut, frameCount);
processor = processor->next;
}
ma_mutex_unlock(&AUDIO.System.lock);
}
// Main mixing function, pretty simple in this project, just an accumulation
// NOTE: framesOut is both an input and an output, it is initially filled with zeros outside of this function
static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, AudioBuffer *buffer)
{
const float localVolume = buffer->volume;
const ma_uint32 channels = AUDIO.System.device.playback.channels;
if (channels == 2) // We consider panning
{
const float right = (buffer->pan + 1.0f)/2.0f; // Normalize: [-1..1] -> [0..1]
const float left = 1.0f - right;
// Fast sine approximation in [0..1] for pan law: y = 0.5f*x*(3 - x*x);
const float levels[2] = { localVolume*0.5f*left*(3.0f - left*left), localVolume*0.5f*right*(3.0f - right*right) };
float *frameOut = framesOut;
const float *frameIn = framesIn;
for (ma_uint32 frame = 0; frame < frameCount; frame++)
{
frameOut[0] += (frameIn[0]*levels[0]);
frameOut[1] += (frameIn[1]*levels[1]);
frameOut += 2;
frameIn += 2;
}
}
else // We do not consider panning
{
for (ma_uint32 frame = 0; frame < frameCount; frame++)
{
for (ma_uint32 c = 0; c < channels; c++)
{
float *frameOut = framesOut + (frame*channels);
const float *frameIn = framesIn + (frame*channels);
// Output accumulates input multiplied by volume to provided output (usually 0)
frameOut[c] += (frameIn[c]*localVolume);
}
}
}
}
// Check if an audio buffer is playing, assuming the audio system mutex has been locked
static bool IsAudioBufferPlayingInLockedState(AudioBuffer *buffer)
{
bool result = false;
if (buffer != NULL) result = (buffer->playing && !buffer->paused);
return result;
}
// Stop an audio buffer, assuming the audio system mutex has been locked
static void StopAudioBufferInLockedState(AudioBuffer *buffer)
{
if (buffer != NULL)
{
if (IsAudioBufferPlayingInLockedState(buffer))
{
buffer->playing = false;
buffer->paused = false;
buffer->frameCursorPos = 0;
buffer->framesProcessed = 0;
buffer->isSubBufferProcessed[0] = true;
buffer->isSubBufferProcessed[1] = true;
}
}
}
// Update audio stream, assuming the audio system mutex has been locked
static void UpdateAudioStreamInLockedState(AudioStream stream, const void *data, int frameCount)
{
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);
stream.buffer->framesProcessed += frameCount;
// 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)frameCount)
{
ma_uint32 framesToWrite = (ma_uint32)frameCount;
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");
}
}
// Some required functions for audio standalone module version
#if defined(RAUDIO_STANDALONE)
// Check file extension
static 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;
}
// Get pointer to extension for a filename string (includes the dot: .png)
static const char *GetFileExtension(const char *fileName)
{
const char *dot = strrchr(fileName, '.');
if (!dot || dot == fileName) return NULL;
return dot;
}
// String pointer reverse break: returns right-most occurrence of charset in text
static const char *strprbrk(const char *text, const char *charset)
{
const char *latestMatch = NULL;
for (; (text != NULL) && (text = strpbrk(text, charset)); latestMatch = text++) { }
return latestMatch;
}
// Get pointer to filename for a path string
static const char *GetFileName(const char *filePath)
{
const char *fileName = NULL;
if (filePath != NULL) fileName = strprbrk(filePath, "\\/");
if (!fileName) return filePath;
return fileName + 1;
}
// Get filename string without extension (uses static string)
static const char *GetFileNameWithoutExt(const char *filePath)
{
#define MAX_FILENAMEWITHOUTEXT_LENGTH 256
static char fileName[MAX_FILENAMEWITHOUTEXT_LENGTH] = { 0 };
memset(fileName, 0, MAX_FILENAMEWITHOUTEXT_LENGTH);
if (filePath != NULL) strncpy(fileName, GetFileName(filePath), MAX_FILENAMEWITHOUTEXT_LENGTH - 1); // Get filename with extension
int fileNameLength = (int)strlen(fileName); // Get size in bytes
for (int i = 0; (i < fileNameLength) && (i < MAX_FILENAMEWITHOUTEXT_LENGTH); i++)
{
if (fileName[i] == '.')
{
// NOTE: We break on first '.' found
fileName[i] = '\0';
break;
}
}
return fileName;
}
// Load data from file into a buffer
static unsigned char *LoadFileData(const char *fileName, int *dataSize)
{
unsigned char *data = NULL;
*dataSize = 0;
if (fileName != NULL)
{
FILE *file = fopen(fileName, "rb");
if (file != NULL)
{
// WARNING: On binary streams SEEK_END could not be found,
// using fseek() and ftell() could not work in some (rare) cases
fseek(file, 0, SEEK_END);
int size = ftell(file);
fseek(file, 0, SEEK_SET);
if (size > 0)
{
data = (unsigned char *)RL_MALLOC(size*sizeof(unsigned char));
// NOTE: fread() returns number of read elements instead of bytes, so we read [1 byte, size elements]
unsigned int count = (unsigned int)fread(data, sizeof(unsigned char), size, file);
*dataSize = count;
if (count != size) TRACELOG(LOG_WARNING, "FILEIO: [%s] File partially loaded", fileName);
else TRACELOG(LOG_INFO, "FILEIO: [%s] File loaded successfully", fileName);
}
else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to read file", fileName);
fclose(file);
}
else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open file", fileName);
}
else TRACELOG(LOG_WARNING, "FILEIO: File name provided is not valid");
return data;
}
// Save data to file from buffer
static bool SaveFileData(const char *fileName, void *data, int dataSize)
{
if (fileName != NULL)
{
FILE *file = fopen(fileName, "wb");
if (file != NULL)
{
unsigned int count = (unsigned int)fwrite(data, sizeof(unsigned char), dataSize, file);
if (count == 0) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to write file", fileName);
else if (count != dataSize) TRACELOG(LOG_WARNING, "FILEIO: [%s] File partially written", fileName);
else TRACELOG(LOG_INFO, "FILEIO: [%s] File saved successfully", fileName);
fclose(file);
}
else
{
TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open file", fileName);
return false;
}
}
else
{
TRACELOG(LOG_WARNING, "FILEIO: File name provided is not valid");
return false;
}
return true;
}
// Save text data to file (write), string must be '\0' terminated
static bool SaveFileText(const char *fileName, char *text)
{
if (fileName != NULL)
{
FILE *file = fopen(fileName, "wt");
if (file != NULL)
{
int count = 0;
if (text != NULL) count = fprintf(file, "%s", text);
if (count == 0) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to write text file", fileName);
else TRACELOG(LOG_INFO, "FILEIO: [%s] Text file saved successfully", fileName);
fclose(file);
}
else
{
TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open text file", fileName);
return false;
}
}
else
{
TRACELOG(LOG_WARNING, "FILEIO: File name provided is not valid");
return false;
}
return true;
}
#endif
#undef AudioBuffer
#endif // SUPPORT_MODULE_RAUDIO
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