raylib/src/raudio.c

/**********************************************************************************************
*
*   raudio - A simple and easy-to-use audio library based on miniaudio
*
*   FEATURES:
*       - Manage audio device (init/close)
*       - Manage raw audio context
*       - Manage mixing channels
*       - Load and unload audio files
*       - Format wave data (sample rate, size, channels)
*       - Play/Stop/Pause/Resume loaded audio
*
*   CONFIGURATION:
*
*   #define RAUDIO_STANDALONE
*       Define to use the module as standalone library (independently of raylib).
*       Required types and functions are defined in the same module.
*
*   #define SUPPORT_FILEFORMAT_WAV
*   #define SUPPORT_FILEFORMAT_OGG
*   #define SUPPORT_FILEFORMAT_XM
*   #define SUPPORT_FILEFORMAT_MOD
*   #define SUPPORT_FILEFORMAT_FLAC
*   #define SUPPORT_FILEFORMAT_MP3
*       Selected desired fileformats to be supported for loading. Some of those formats are
*       supported by default, to remove support, just comment unrequired #define in this module
*
*   DEPENDENCIES:
*       miniaudio.h  - Audio device management lib (https://github.com/dr-soft/miniaudio)
*       stb_vorbis.h - Ogg audio files loading (http://www.nothings.org/stb_vorbis/)
*       dr_mp3.h     - MP3 audio file loading (https://github.com/mackron/dr_libs)
*       dr_flac.h    - FLAC audio file loading (https://github.com/mackron/dr_libs)
*       jar_xm.h     - XM module file loading
*       jar_mod.h    - MOD audio file loading
*
*   CONTRIBUTORS:
*       David Reid (github: @mackron) (Nov. 2017):
*           - Complete port to miniaudio library
*
*       Joshua Reisenauer (github: @kd7tck) (2015)
*           - XM audio module support (jar_xm)
*           - MOD audio module support (jar_mod)
*           - Mixing channels support
*           - Raw audio context support
*
*
*   LICENSE: zlib/libpng
*
*   Copyright (c) 2013-2020 Ramon Santamaria (@raysan5)
*
*   This software is provided "as-is", without any express or implied warranty. In no event
*   will the authors be held liable for any damages arising from the use of this software.
*
*   Permission is granted to anyone to use this software for any purpose, including commercial
*   applications, and to alter it and redistribute it freely, subject to the following restrictions:
*
*     1. The origin of this software must not be misrepresented; you must not claim that you
*     wrote the original software. If you use this software in a product, an acknowledgment
*     in the product documentation would be appreciated but is not required.
*
*     2. Altered source versions must be plainly marked as such, and must not be misrepresented
*     as being the original software.
*
*     3. This notice may not be removed or altered from any source distribution.
*
**********************************************************************************************/

#if defined(RAUDIO_STANDALONE)
    #include "raudio.h"
    #include <stdarg.h>         // Required for: va_list, va_start(), vfprintf(), va_end()
#else
    #include "raylib.h"         // Declares module functions

// Check if config flags have been externally provided on compilation line
#if !defined(EXTERNAL_CONFIG_FLAGS)
    #include "config.h"         // Defines module configuration flags
#endif
    #include "utils.h"          // Required for: fopen() Android mapping
#endif

#if defined(_WIN32)
// @raysan5: To avoid conflicting windows.h symbols with raylib, so flags are defined
// WARNING: Those flags avoid inclusion of some Win32 headers that could be required 
// by user at some point and won't be included...
//-------------------------------------------------------------------------------------

// If defined, the following flags inhibit definition of the indicated items.
#define NOGDICAPMASKS     // CC_*, LC_*, PC_*, CP_*, TC_*, RC_
#define NOVIRTUALKEYCODES // VK_*
#define NOWINMESSAGES     // WM_*, EM_*, LB_*, CB_*
#define NOWINSTYLES       // WS_*, CS_*, ES_*, LBS_*, SBS_*, CBS_*
#define NOSYSMETRICS      // SM_*
#define NOMENUS           // MF_*
#define NOICONS           // IDI_*
#define NOKEYSTATES       // MK_*
#define NOSYSCOMMANDS     // SC_*
#define NORASTEROPS       // Binary and Tertiary raster ops
#define NOSHOWWINDOW      // SW_*
#define OEMRESOURCE       // OEM Resource values
#define NOATOM            // Atom Manager routines
#define NOCLIPBOARD       // Clipboard routines
#define NOCOLOR           // Screen colors
#define NOCTLMGR          // Control and Dialog routines
#define NODRAWTEXT        // DrawText() and DT_*
#define NOGDI             // All GDI defines and routines
#define NOKERNEL          // All KERNEL defines and routines
#define NOUSER            // All USER defines and routines
//#define NONLS             // All NLS defines and routines
#define NOMB              // MB_* and MessageBox()
#define NOMEMMGR          // GMEM_*, LMEM_*, GHND, LHND, associated routines
#define NOMETAFILE        // typedef METAFILEPICT
#define NOMINMAX          // Macros min(a,b) and max(a,b)
#define NOMSG             // typedef MSG and associated routines
#define NOOPENFILE        // OpenFile(), OemToAnsi, AnsiToOem, and OF_*
#define NOSCROLL          // SB_* and scrolling routines
#define NOSERVICE         // All Service Controller routines, SERVICE_ equates, etc.
#define NOSOUND           // Sound driver routines
#define NOTEXTMETRIC      // typedef TEXTMETRIC and associated routines
#define NOWH              // SetWindowsHook and WH_*
#define NOWINOFFSETS      // GWL_*, GCL_*, associated routines
#define NOCOMM            // COMM driver routines
#define NOKANJI           // Kanji support stuff.
#define NOHELP            // Help engine interface.
#define NOPROFILER        // Profiler interface.
#define NODEFERWINDOWPOS  // DeferWindowPos routines
#define NOMCX             // Modem Configuration Extensions

// Type required before windows.h inclusion
typedef struct tagMSG *LPMSG;

#include <windows.h>

// Type required by some unused function...
typedef struct tagBITMAPINFOHEADER {
  DWORD biSize;
  LONG  biWidth;
  LONG  biHeight;
  WORD  biPlanes;
  WORD  biBitCount;
  DWORD biCompression;
  DWORD biSizeImage;
  LONG  biXPelsPerMeter;
  LONG  biYPelsPerMeter;
  DWORD biClrUsed;
  DWORD biClrImportant;
} BITMAPINFOHEADER, *PBITMAPINFOHEADER;

#include <objbase.h>
#include <mmreg.h>
#include <mmsystem.h>

// @raysan5: Some required types defined for MSVC/TinyC compiler
#if defined(_MSC_VER) || defined(__TINYC__)
    #include "propidl.h"
#endif
#endif

#define MA_NO_JACK
#define MINIAUDIO_IMPLEMENTATION
#include "external/miniaudio.h" // miniaudio library
#undef PlaySound                // Win32 API: windows.h > mmsystem.h defines PlaySound macro

#include <stdlib.h>             // Required for: malloc(), free()
#include <stdio.h>              // Required for: FILE, fopen(), fclose(), fread()

#if defined(RAUDIO_STANDALONE)
    #include <string.h>         // Required for: strcmp() [Used in IsFileExtension()]
#endif

#if defined(SUPPORT_FILEFORMAT_OGG)
    #define STB_VORBIS_IMPLEMENTATION
    #include "external/stb_vorbis.h"    // OGG loading functions
#endif

#if defined(SUPPORT_FILEFORMAT_XM)
    #define JAR_XM_IMPLEMENTATION
    #include "external/jar_xm.h"        // XM loading functions
#endif

#if defined(SUPPORT_FILEFORMAT_MOD)
    #define JAR_MOD_IMPLEMENTATION
    #include "external/jar_mod.h"       // MOD loading functions
#endif

#if defined(SUPPORT_FILEFORMAT_FLAC)
    #define DR_FLAC_IMPLEMENTATION
    #define DR_FLAC_NO_WIN32_IO
    #include "external/dr_flac.h"       // FLAC loading functions
#endif

#if defined(SUPPORT_FILEFORMAT_MP3)
    #define DR_MP3_IMPLEMENTATION
    #include "external/dr_mp3.h"        // MP3 loading functions
#endif

#if defined(_MSC_VER)
    #undef bool
#endif

//----------------------------------------------------------------------------------
// Defines and Macros
//----------------------------------------------------------------------------------
// 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
#if !defined(AUDIO_BUFFER_SIZE)
    #define AUDIO_BUFFER_SIZE 4096      // PCM data samples (i.e. 16bit, Mono: 8Kb)
#endif

#define DEVICE_FORMAT       ma_format_f32
#define DEVICE_CHANNELS     2
#define DEVICE_SAMPLE_RATE  44100

#define MAX_AUDIO_BUFFER_POOL_CHANNELS 16

//----------------------------------------------------------------------------------
// Types and Structures Definition
//----------------------------------------------------------------------------------

// Music context type
// NOTE: Depends on data structure provided by the library
// in charge of reading the different file types
typedef enum {
    MUSIC_AUDIO_WAV = 0,
    MUSIC_AUDIO_OGG,
    MUSIC_AUDIO_FLAC,
    MUSIC_AUDIO_MP3,
    MUSIC_MODULE_XM,
    MUSIC_MODULE_MOD
} MusicContextType;

#if defined(RAUDIO_STANDALONE)
typedef enum {
    LOG_ALL,
    LOG_TRACE,
    LOG_DEBUG,
    LOG_INFO,
    LOG_WARNING,
    LOG_ERROR,
    LOG_FATAL,
    LOG_NONE
} TraceLogType;
#endif

// NOTE: Different logic is used when feeding data to the playback device
// depending on whether or not data is streamed (Music vs Sound)
typedef enum {
    AUDIO_BUFFER_USAGE_STATIC = 0,
    AUDIO_BUFFER_USAGE_STREAM
} AudioBufferUsage;

// Audio buffer structure
struct rAudioBuffer {
    ma_data_converter converter;    // Audio data converter

    float volume;                   // Audio buffer volume
    float pitch;                    // Audio buffer pitch

    bool playing;                   // Audio buffer state: AUDIO_PLAYING
    bool paused;                    // Audio buffer state: AUDIO_PAUSED
    bool looping;                   // Audio buffer looping, always true for AudioStreams
    int usage;                      // Audio buffer usage mode: STATIC or STREAM

    bool isSubBufferProcessed[2];   // SubBuffer processed (virtual double buffer)
    unsigned int sizeInFrames;      // Total buffer size in frames
    unsigned int frameCursorPos;    // Frame cursor position
    unsigned int totalFramesProcessed;  // Total frames processed in this buffer (required for play timing)

    unsigned char *data;            // Data buffer, on music stream keeps filling

    rAudioBuffer *next;             // Next audio buffer on the list
    rAudioBuffer *prev;             // Previous audio buffer on the list
};

#define AudioBuffer rAudioBuffer    // HACK: To avoid CoreAudio (macOS) symbol collision

// Audio data context
typedef struct AudioData {
    struct {
        ma_context context;         // miniaudio context data
        ma_device device;           // miniaudio device
        ma_mutex lock;              // miniaudio mutex lock
        bool isReady;               // Check if audio device is ready
    } System;
    struct {
        AudioBuffer *first;         // Pointer to first AudioBuffer in the list
        AudioBuffer *last;          // Pointer to last AudioBuffer in the list
    } Buffer;
    struct {
        AudioBuffer *pool[MAX_AUDIO_BUFFER_POOL_CHANNELS];      // Multichannel AudioBuffer pointers pool
        unsigned int poolCounter;                               // AudioBuffer pointers pool counter
        unsigned int channels[MAX_AUDIO_BUFFER_POOL_CHANNELS];  // AudioBuffer pool channels
    } MultiChannel;
} AudioData;

//----------------------------------------------------------------------------------
// Global Variables Definition
//----------------------------------------------------------------------------------
static AudioData AUDIO = { 0 };     // Global CORE context

//----------------------------------------------------------------------------------
// Module specific Functions Declaration
//----------------------------------------------------------------------------------
static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message);
static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount);
static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume);

static void InitAudioBufferPool(void);                  // Initialise the multichannel buffer pool
static void CloseAudioBufferPool(void);                 // Close the audio buffers pool

#if defined(SUPPORT_FILEFORMAT_WAV)
static Wave LoadWAV(const char *fileName);              // Load WAV file
static int SaveWAV(Wave wave, const char *fileName);    // Save wave data as WAV file
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
static Wave LoadOGG(const char *fileName);              // Load OGG file
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
static Wave LoadFLAC(const char *fileName);             // Load FLAC file
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
static Wave LoadMP3(const char *fileName);              // Load MP3 file
#endif

#if defined(RAUDIO_STANDALONE)
bool IsFileExtension(const char *fileName, const char *ext);// Check file extension
void TRACELOG(int msgType, const char *text, ...);      // Show trace log messages (LOG_INFO, LOG_WARNING, LOG_ERROR, LOG_DEBUG)
#endif

//----------------------------------------------------------------------------------
// AudioBuffer management functions declaration
// NOTE: Those functions are not exposed by raylib... for the moment
//----------------------------------------------------------------------------------
AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage);
void CloseAudioBuffer(AudioBuffer *buffer);
bool IsAudioBufferPlaying(AudioBuffer *buffer);
void PlayAudioBuffer(AudioBuffer *buffer);
void StopAudioBuffer(AudioBuffer *buffer);
void PauseAudioBuffer(AudioBuffer *buffer);
void ResumeAudioBuffer(AudioBuffer *buffer);
void SetAudioBufferVolume(AudioBuffer *buffer, float volume);
void SetAudioBufferPitch(AudioBuffer *buffer, float pitch);
void TrackAudioBuffer(AudioBuffer *buffer);
void UntrackAudioBuffer(AudioBuffer *buffer);

//----------------------------------------------------------------------------------
// Module Functions Definition - Audio Device initialization and Closing
//----------------------------------------------------------------------------------
// Initialize audio device
void InitAudioDevice(void)
{
    // TODO: Load AUDIO context memory dynamically?

    // Init audio context
    ma_context_config ctxConfig = ma_context_config_init();
    ctxConfig.logCallback = OnLog;

    ma_result result = ma_context_init(NULL, 0, &ctxConfig, &AUDIO.System.context);
    if (result != MA_SUCCESS)
    {
        TRACELOG(LOG_ERROR, "Failed to initialize audio 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    = DEVICE_FORMAT;
    config.playback.channels  = 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         = DEVICE_SAMPLE_RATE;
    config.dataCallback       = OnSendAudioDataToDevice;
    config.pUserData          = NULL;

    result = ma_device_init(&AUDIO.System.context, &config, &AUDIO.System.device);
    if (result != MA_SUCCESS)
    {
        TRACELOG(LOG_ERROR, "Failed to initialize audio playback 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_ERROR, "Failed to start audio playback AUDIO.System.device");
        ma_device_uninit(&AUDIO.System.device);
        ma_context_uninit(&AUDIO.System.context);
        return;
    }

    // Mixing happens on a seperate thread which means we need to synchronize. I'm using a mutex here to make things simple, but may
    // want to look at something a bit smarter later on to keep everything real-time, if that's necessary.
    if (ma_mutex_init(&AUDIO.System.context, &AUDIO.System.lock) != MA_SUCCESS)
    {
        TRACELOG(LOG_ERROR, "Failed to create mutex for audio mixing");
        ma_device_uninit(&AUDIO.System.device);
        ma_context_uninit(&AUDIO.System.context);
        return;
    }

    TRACELOG(LOG_INFO, "Audio device initialized successfully");
    TRACELOG(LOG_INFO, "Audio backend: miniaudio / %s", ma_get_backend_name(AUDIO.System.context.backend));
    TRACELOG(LOG_INFO, "Audio format: %s -> %s", ma_get_format_name(AUDIO.System.device.playback.format), ma_get_format_name(AUDIO.System.device.playback.internalFormat));
    TRACELOG(LOG_INFO, "Audio channels: %d -> %d", AUDIO.System.device.playback.channels, AUDIO.System.device.playback.internalChannels);
    TRACELOG(LOG_INFO, "Audio sample rate: %d -> %d", AUDIO.System.device.sampleRate, AUDIO.System.device.playback.internalSampleRate);
    TRACELOG(LOG_INFO, "Audio buffer size: %d", AUDIO.System.device.playback.internalPeriodSizeInFrames);

    InitAudioBufferPool();
    TRACELOG(LOG_INFO, "Audio multichannel pool size: %i", MAX_AUDIO_BUFFER_POOL_CHANNELS);

    AUDIO.System.isReady = true;
}

// Close the audio device for all contexts
void CloseAudioDevice(void)
{
    if (AUDIO.System.isReady)
    {
        ma_mutex_uninit(&AUDIO.System.lock);
        ma_device_uninit(&AUDIO.System.device);
        ma_context_uninit(&AUDIO.System.context);

        CloseAudioBufferPool();

        TRACELOG(LOG_INFO, "Audio AUDIO.System.device closed successfully");
    }
    else TRACELOG(LOG_WARNING, "Could not close audio AUDIO.System.device because it is not currently initialized");
}

// Check if device has been initialized successfully
bool IsAudioDeviceReady(void)
{
    return AUDIO.System.isReady;
}

// Set master volume (listener)
void SetMasterVolume(float volume)
{
    ma_device_set_master_volume(&AUDIO.System.device, volume);
}

//----------------------------------------------------------------------------------
// Module Functions Definition - Audio Buffer management
//----------------------------------------------------------------------------------

// Initialize a new audio buffer (filled with silence)
AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage)
{
    AudioBuffer *audioBuffer = (AudioBuffer *)RL_CALLOC(1, sizeof(AudioBuffer));

    if (audioBuffer == NULL)
    {
        TRACELOG(LOG_ERROR, "InitAudioBuffer() : Failed to allocate memory for audio buffer");
        return NULL;
    }

    audioBuffer->data = RL_CALLOC(sizeInFrames*channels*ma_get_bytes_per_sample(format), 1);

    // Audio data runs through a format converter
    ma_data_converter_config converterConfig = ma_data_converter_config_init(format, DEVICE_FORMAT, channels, DEVICE_CHANNELS, sampleRate, DEVICE_SAMPLE_RATE);
    converterConfig.resampling.allowDynamicSampleRate = true;        // Required for pitch shifting

    ma_result result = ma_data_converter_init(&converterConfig, &audioBuffer->converter);

    if (result != MA_SUCCESS)
    {
        TRACELOG(LOG_ERROR, "InitAudioBuffer() : Failed to create data conversion pipeline");
        RL_FREE(audioBuffer);
        return NULL;
    }

    // Init audio buffer values
    audioBuffer->volume = 1.0f;
    audioBuffer->pitch = 1.0f;
    audioBuffer->playing = false;
    audioBuffer->paused = false;
    audioBuffer->looping = false;
    audioBuffer->usage = usage;
    audioBuffer->frameCursorPos = 0;
    audioBuffer->sizeInFrames = sizeInFrames;

    // Buffers should be marked as processed by default so that a call to
    // UpdateAudioStream() immediately after initialization works correctly
    audioBuffer->isSubBufferProcessed[0] = true;
    audioBuffer->isSubBufferProcessed[1] = true;

    // Track audio buffer to linked list next position
    TrackAudioBuffer(audioBuffer);

    return audioBuffer;
}

// Delete an audio buffer
void CloseAudioBuffer(AudioBuffer *buffer)
{
    if (buffer != NULL)
    {
        ma_data_converter_uninit(&buffer->converter);
        UntrackAudioBuffer(buffer);
        RL_FREE(buffer->data);
        RL_FREE(buffer);
    }
    else TRACELOG(LOG_ERROR, "CloseAudioBuffer() : No audio buffer");
}

// Check if an audio buffer is playing
bool IsAudioBufferPlaying(AudioBuffer *buffer)
{
    bool result = false;

    if (buffer != NULL) result = (buffer->playing && !buffer->paused);
    else TRACELOG(LOG_WARNING, "IsAudioBufferPlaying() : No audio buffer");

    return result;
}

// Play an audio buffer
// NOTE: Buffer is restarted to the start.
// Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position should be maintained.
void PlayAudioBuffer(AudioBuffer *buffer)
{
    if (buffer != NULL)
    {
        buffer->playing = true;
        buffer->paused = false;
        buffer->frameCursorPos = 0;
    }
    else TRACELOG(LOG_ERROR, "PlayAudioBuffer() : No audio buffer");
}

// Stop an audio buffer
void StopAudioBuffer(AudioBuffer *buffer)
{
    if (buffer != NULL)
    {
        if (IsAudioBufferPlaying(buffer))
        {
            buffer->playing = false;
            buffer->paused = false;
            buffer->frameCursorPos = 0;
            buffer->totalFramesProcessed = 0;
            buffer->isSubBufferProcessed[0] = true;
            buffer->isSubBufferProcessed[1] = true;
        }
    }
    else TRACELOG(LOG_ERROR, "StopAudioBuffer() : No audio buffer");
}

// Pause an audio buffer
void PauseAudioBuffer(AudioBuffer *buffer)
{
    if (buffer != NULL) buffer->paused = true;
    else TRACELOG(LOG_ERROR, "PauseAudioBuffer() : No audio buffer");
}

// Resume an audio buffer
void ResumeAudioBuffer(AudioBuffer *buffer)
{
    if (buffer != NULL) buffer->paused = false;
    else TRACELOG(LOG_ERROR, "ResumeAudioBuffer() : No audio buffer");
}

// Set volume for an audio buffer
void SetAudioBufferVolume(AudioBuffer *buffer, float volume)
{
    if (buffer != NULL) buffer->volume = volume;
    else TRACELOG(LOG_WARNING, "SetAudioBufferVolume() : No audio buffer");
}

// Set pitch for an audio buffer
void SetAudioBufferPitch(AudioBuffer *buffer, float pitch)
{
    if (buffer != NULL)
    {
        float pitchMul = pitch/buffer->pitch;

        // Pitching is just an adjustment of the sample rate.
        // Note that this changes the duration of the sound:
        //  - higher pitches will make the sound faster
        //  - lower pitches make it slower
        ma_uint32 newOutputSampleRate = (ma_uint32)((float)buffer->converter.config.sampleRateOut/pitchMul);
        buffer->pitch *= (float)buffer->converter.config.sampleRateOut/newOutputSampleRate;

        ma_data_converter_set_rate(&buffer->converter, buffer->converter.config.sampleRateIn, newOutputSampleRate);
    }
    else TRACELOG(LOG_WARNING, "SetAudioBufferPitch() : No audio buffer");
}

// Track audio buffer to linked list next position
void TrackAudioBuffer(AudioBuffer *buffer)
{
    ma_mutex_lock(&AUDIO.System.lock);
    {
        if (AUDIO.Buffer.first == NULL) AUDIO.Buffer.first = buffer;
        else
        {
            AUDIO.Buffer.last->next = buffer;
            buffer->prev = AUDIO.Buffer.last;
        }

        AUDIO.Buffer.last = buffer;
    }
    ma_mutex_unlock(&AUDIO.System.lock);
}

// Untrack audio buffer from linked list
void UntrackAudioBuffer(AudioBuffer *buffer)
{
    ma_mutex_lock(&AUDIO.System.lock);
    {
        if (buffer->prev == NULL) AUDIO.Buffer.first = buffer->next;
        else buffer->prev->next = buffer->next;

        if (buffer->next == NULL) AUDIO.Buffer.last = buffer->prev;
        else buffer->next->prev = buffer->prev;

        buffer->prev = NULL;
        buffer->next = NULL;
    }
    ma_mutex_unlock(&AUDIO.System.lock);
}

//----------------------------------------------------------------------------------
// Module Functions Definition - Sounds loading and playing (.WAV)
//----------------------------------------------------------------------------------

// Load wave data from file
Wave LoadWave(const char *fileName)
{
    Wave wave = { 0 };

    if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV)
    else if (IsFileExtension(fileName, ".wav")) wave = LoadWAV(fileName);
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
    else if (IsFileExtension(fileName, ".ogg")) wave = LoadOGG(fileName);
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
    else if (IsFileExtension(fileName, ".flac")) wave = LoadFLAC(fileName);
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
    else if (IsFileExtension(fileName, ".mp3")) wave = LoadMP3(fileName);
#endif
    else TRACELOG(LOG_WARNING, "[%s] Audio fileformat not supported, it can't be loaded", fileName);

    return wave;
}

// Load sound from file
// NOTE: The entire file is loaded to memory to be played (no-streaming)
Sound LoadSound(const char *fileName)
{
    Wave wave = LoadWave(fileName);

    Sound sound = LoadSoundFromWave(wave);

    UnloadWave(wave);       // Sound is loaded, we can unload wave

    return sound;
}

// Load sound from wave data
// NOTE: Wave data must be unallocated manually
Sound LoadSoundFromWave(Wave wave)
{
    Sound sound = { 0 };

    if (wave.data != NULL)
    {
        // When using miniaudio we need to do our own mixing.
        // To simplify this we need convert the format of each sound to be consistent with
        // the format used to open the playback AUDIO.System.device. We can do this two ways:
        //
        //   1) Convert the whole sound in one go at load time (here).
        //   2) Convert the audio data in chunks at mixing time.
        //
        // First option has been selected, format conversion is done on the loading stage.
        // The downside is that it uses more memory if the original sound is u8 or s16.
        ma_format formatIn  = ((wave.sampleSize == 8)? ma_format_u8 : ((wave.sampleSize == 16)? ma_format_s16 : ma_format_f32));
        ma_uint32 frameCountIn = wave.sampleCount/wave.channels;

        ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, NULL, frameCountIn, formatIn, wave.channels, wave.sampleRate);
        if (frameCount == 0) TRACELOG(LOG_WARNING, "LoadSoundFromWave() : Failed to get frame count for format conversion");

        AudioBuffer *audioBuffer = InitAudioBuffer(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, frameCount, AUDIO_BUFFER_USAGE_STATIC);
        if (audioBuffer == NULL) TRACELOG(LOG_WARNING, "LoadSoundFromWave() : Failed to create audio buffer");

        frameCount = (ma_uint32)ma_convert_frames(audioBuffer->data, frameCount, DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, wave.data, frameCountIn, formatIn, wave.channels, wave.sampleRate);
        if (frameCount == 0) TRACELOG(LOG_WARNING, "LoadSoundFromWave() : Format conversion failed");

        sound.sampleCount = frameCount*DEVICE_CHANNELS;
        sound.stream.sampleRate = DEVICE_SAMPLE_RATE;
        sound.stream.sampleSize = 32;
        sound.stream.channels = DEVICE_CHANNELS;
        sound.stream.buffer = audioBuffer;
    }

    return sound;
}

// Unload wave data
void UnloadWave(Wave wave)
{
    if (wave.data != NULL) RL_FREE(wave.data);

    TRACELOG(LOG_INFO, "Unloaded wave data from RAM");
}

// Unload sound
void UnloadSound(Sound sound)
{
    CloseAudioBuffer(sound.stream.buffer);

    TRACELOG(LOG_INFO, "Unloaded sound data from RAM");
}

// Update sound buffer with new data
void UpdateSound(Sound sound, const void *data, int samplesCount)
{
    AudioBuffer *audioBuffer = sound.stream.buffer;

    if (audioBuffer != NULL)
    {
        StopAudioBuffer(audioBuffer);

        // TODO: May want to lock/unlock this since this data buffer is read at mixing time
        memcpy(audioBuffer->data, data, samplesCount*ma_get_bytes_per_frame(audioBuffer->converter.config.formatIn, audioBuffer->converter.config.channelsIn));
    }
    else TRACELOG(LOG_ERROR, "UpdateSound() : Invalid sound - no audio buffer");
}

// Export wave data to file
void ExportWave(Wave wave, const char *fileName)
{
    bool success = false;

    if (false) { }
#if defined(SUPPORT_FILEFORMAT_WAV)
    else if (IsFileExtension(fileName, ".wav")) success = SaveWAV(wave, fileName);
#endif
    else if (IsFileExtension(fileName, ".raw"))
    {
        // Export raw sample data (without header)
        // NOTE: It's up to the user to track wave parameters
        FILE *rawFile = fopen(fileName, "wb");
        success = fwrite(wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8, 1, rawFile);
        fclose(rawFile);
    }

    if (success) TRACELOG(LOG_INFO, "Wave exported successfully: %s", fileName);
    else TRACELOG(LOG_WARNING, "Wave could not be exported.");
}

// Export wave sample data to code (.h)
void ExportWaveAsCode(Wave wave, const char *fileName)
{
    #define BYTES_TEXT_PER_LINE     20

    char varFileName[256] = { 0 };
    int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8;

    FILE *txtFile = fopen(fileName, "wt");

    if (txtFile != NULL)
    {
        fprintf(txtFile, "\n//////////////////////////////////////////////////////////////////////////////////\n");
        fprintf(txtFile, "//                                                                              //\n");
        fprintf(txtFile, "// WaveAsCode exporter v1.0 - Wave data exported as an array of bytes           //\n");
        fprintf(txtFile, "//                                                                              //\n");
        fprintf(txtFile, "// more info and bugs-report:  github.com/raysan5/raylib                        //\n");
        fprintf(txtFile, "// feedback and support:       ray[at]raylib.com                                //\n");
        fprintf(txtFile, "//                                                                              //\n");
        fprintf(txtFile, "// Copyright (c) 2018 Ramon Santamaria (@raysan5)                               //\n");
        fprintf(txtFile, "//                                                                              //\n");
        fprintf(txtFile, "//////////////////////////////////////////////////////////////////////////////////\n\n");

#if !defined(RAUDIO_STANDALONE)
        // Get file name from path and convert variable name to uppercase
        strcpy(varFileName, GetFileNameWithoutExt(fileName));
        for (int i = 0; varFileName[i] != '\0'; i++) if (varFileName[i] >= 'a' && varFileName[i] <= 'z') { varFileName[i] = varFileName[i] - 32; }
#else
        strcpy(varFileName, fileName);
#endif

        fprintf(txtFile, "// Wave data information\n");
        fprintf(txtFile, "#define %s_SAMPLE_COUNT     %i\n", varFileName, wave.sampleCount);
        fprintf(txtFile, "#define %s_SAMPLE_RATE      %i\n", varFileName, wave.sampleRate);
        fprintf(txtFile, "#define %s_SAMPLE_SIZE      %i\n", varFileName, wave.sampleSize);
        fprintf(txtFile, "#define %s_CHANNELS         %i\n\n", varFileName, wave.channels);

        // Write byte data as hexadecimal text
        fprintf(txtFile, "static unsigned char %s_DATA[%i] = { ", varFileName, dataSize);
        for (int i = 0; i < dataSize - 1; i++) fprintf(txtFile, ((i%BYTES_TEXT_PER_LINE == 0)? "0x%x,\n" : "0x%x, "), ((unsigned char *)wave.data)[i]);
        fprintf(txtFile, "0x%x };\n", ((unsigned char *)wave.data)[dataSize - 1]);

        fclose(txtFile);
    }
}

// Play a sound
void PlaySound(Sound sound)
{
    PlayAudioBuffer(sound.stream.buffer);
}

// Play a sound in the multichannel buffer pool
void PlaySoundMulti(Sound sound)
{
    int index = -1;
    unsigned int oldAge = 0;
    int oldIndex = -1;

    // find the first non playing pool entry
    for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
    {
        if (AUDIO.MultiChannel.channels[i] > oldAge)
        {
            oldAge = AUDIO.MultiChannel.channels[i];
            oldIndex = i;
        }

        if (!IsAudioBufferPlaying(AUDIO.MultiChannel.pool[i]))
        {
            index = i;
            break;
        }
    }

    // If no none playing pool members can be index choose the oldest
    if (index == -1)
    {
        TRACELOG(LOG_WARNING,"pool age %i ended a sound early no room in buffer pool", AUDIO.MultiChannel.poolCounter);

        if (oldIndex == -1)
        {
            // Shouldn't be able to get here... but just in case something odd happens!
            TRACELOG(LOG_ERROR,"sound buffer pool couldn't determine oldest buffer not playing sound");

            return;
        }

        index = oldIndex;

        // Just in case...
        StopAudioBuffer(AUDIO.MultiChannel.pool[index]);
    }

    // Experimentally mutex lock doesn't seem to be needed this makes sense
    // as AUDIO.MultiChannel.pool[index] isn't playing and the only stuff we're copying
    // shouldn't be changing...

    AUDIO.MultiChannel.channels[index] = AUDIO.MultiChannel.poolCounter;
    AUDIO.MultiChannel.poolCounter++;

    AUDIO.MultiChannel.pool[index]->volume = sound.stream.buffer->volume;
    AUDIO.MultiChannel.pool[index]->pitch = sound.stream.buffer->pitch;
    AUDIO.MultiChannel.pool[index]->looping = sound.stream.buffer->looping;
    AUDIO.MultiChannel.pool[index]->usage = sound.stream.buffer->usage;
    AUDIO.MultiChannel.pool[index]->isSubBufferProcessed[0] = false;
    AUDIO.MultiChannel.pool[index]->isSubBufferProcessed[1] = false;
    AUDIO.MultiChannel.pool[index]->sizeInFrames = sound.stream.buffer->sizeInFrames;
    AUDIO.MultiChannel.pool[index]->data = sound.stream.buffer->data;

    PlayAudioBuffer(AUDIO.MultiChannel.pool[index]);
}

// Stop any sound played with PlaySoundMulti()
void StopSoundMulti(void)
{
    for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) StopAudioBuffer(AUDIO.MultiChannel.pool[i]);
}

// Get number of sounds playing in the multichannel buffer pool
int GetSoundsPlaying(void)
{
    int counter = 0;

    for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
    {
        if (IsAudioBufferPlaying(AUDIO.MultiChannel.pool[i])) counter++;
    }

    return counter;
}

// Pause a sound
void PauseSound(Sound sound)
{
    PauseAudioBuffer(sound.stream.buffer);
}

// Resume a paused sound
void ResumeSound(Sound sound)
{
    ResumeAudioBuffer(sound.stream.buffer);
}

// Stop reproducing a sound
void StopSound(Sound sound)
{
    StopAudioBuffer(sound.stream.buffer);
}

// Check if a sound is playing
bool IsSoundPlaying(Sound sound)
{
    return IsAudioBufferPlaying(sound.stream.buffer);
}

// Set volume for a sound
void SetSoundVolume(Sound sound, float volume)
{
    SetAudioBufferVolume(sound.stream.buffer, volume);
}

// Set pitch for a sound
void SetSoundPitch(Sound sound, float pitch)
{
    SetAudioBufferPitch(sound.stream.buffer, pitch);
}

// Convert wave data to desired format
void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
{
    ma_format formatIn  = ((wave->sampleSize == 8)? ma_format_u8 : ((wave->sampleSize == 16)? ma_format_s16 : ma_format_f32));
    ma_format formatOut = ((      sampleSize == 8)? ma_format_u8 : ((      sampleSize == 16)? ma_format_s16 : ma_format_f32));

    ma_uint32 frameCountIn = wave->sampleCount;  // Is wave->sampleCount actually the frame count? That terminology needs to change, if so.

    ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, formatOut, channels, sampleRate, NULL, frameCountIn, formatIn, wave->channels, wave->sampleRate);
    if (frameCount == 0)
    {
        TRACELOG(LOG_ERROR, "WaveFormat() : Failed to get frame count for format conversion.");
        return;
    }

    void *data = RL_MALLOC(frameCount*channels*(sampleSize/8));

    frameCount = (ma_uint32)ma_convert_frames(data, frameCount, formatOut, channels, sampleRate, wave->data, frameCountIn, formatIn, wave->channels, wave->sampleRate);
    if (frameCount == 0)
    {
        TRACELOG(LOG_ERROR, "WaveFormat() : Format conversion failed.");
        return;
    }

    wave->sampleCount = frameCount;
    wave->sampleSize = sampleSize;
    wave->sampleRate = sampleRate;
    wave->channels = channels;
    RL_FREE(wave->data);
    wave->data = data;
}

// Copy a wave to a new wave
Wave WaveCopy(Wave wave)
{
    Wave newWave = { 0 };

    newWave.data = RL_MALLOC(wave.sampleCount*wave.sampleSize/8*wave.channels);

    if (newWave.data != NULL)
    {
        // NOTE: Size must be provided in bytes
        memcpy(newWave.data, wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8);

        newWave.sampleCount = wave.sampleCount;
        newWave.sampleRate = wave.sampleRate;
        newWave.sampleSize = wave.sampleSize;
        newWave.channels = wave.channels;
    }

    return newWave;
}

// Crop a wave to defined samples range
// NOTE: Security check in case of out-of-range
void WaveCrop(Wave *wave, int initSample, int finalSample)
{
    if ((initSample >= 0) && (initSample < finalSample) &&
        (finalSample > 0) && ((unsigned int)finalSample < wave->sampleCount))
    {
        int sampleCount = finalSample - initSample;

        void *data = RL_MALLOC(sampleCount*wave->sampleSize/8*wave->channels);

        memcpy(data, (unsigned char *)wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->channels*wave->sampleSize/8);

        RL_FREE(wave->data);
        wave->data = data;
    }
    else TRACELOG(LOG_WARNING, "Wave crop range out of bounds");
}

// Get samples data from wave as a floats array
// NOTE: Returned sample values are normalized to range [-1..1]
float *GetWaveData(Wave wave)
{
    float *samples = (float *)RL_MALLOC(wave.sampleCount*wave.channels*sizeof(float));

    for (unsigned int i = 0; i < wave.sampleCount; i++)
    {
        for (unsigned int j = 0; j < wave.channels; j++)
        {
            if (wave.sampleSize == 8) samples[wave.channels*i + j] = (float)(((unsigned char *)wave.data)[wave.channels*i + j] - 127)/256.0f;
            else if (wave.sampleSize == 16) samples[wave.channels*i + j] = (float)((short *)wave.data)[wave.channels*i + j]/32767.0f;
            else if (wave.sampleSize == 32) samples[wave.channels*i + j] = ((float *)wave.data)[wave.channels*i + j];
        }
    }

    return samples;
}

//----------------------------------------------------------------------------------
// Module Functions Definition - Music loading and stream playing (.OGG)
//----------------------------------------------------------------------------------

// Load music stream from file
Music LoadMusicStream(const char *fileName)
{
    Music music = { 0 };
    bool musicLoaded = false;

    if (false) { }
#if defined(SUPPORT_FILEFORMAT_OGG)
    else if (IsFileExtension(fileName, ".ogg"))
    {
        // Open ogg audio stream
        music.ctxData = stb_vorbis_open_filename(fileName, NULL, NULL);

        if (music.ctxData != NULL)
        {
            music.ctxType = MUSIC_AUDIO_OGG;
            stb_vorbis_info info = stb_vorbis_get_info((stb_vorbis *)music.ctxData);  // Get Ogg file info

            // OGG bit rate defaults to 16 bit, it's enough for compressed format
            music.stream = InitAudioStream(info.sample_rate, 16, info.channels);
            music.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData)*info.channels;
            music.loopCount = 0;   // Infinite loop by default
            musicLoaded = true;
        }
    }
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
    else if (IsFileExtension(fileName, ".flac"))
    {
        music.ctxData = drflac_open_file(fileName);

        if (music.ctxData != NULL)
        {
            music.ctxType = MUSIC_AUDIO_FLAC;
            drflac *ctxFlac = (drflac *)music.ctxData;

            music.stream = InitAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
            music.sampleCount = (unsigned int)ctxFlac->totalSampleCount;
            music.loopCount = 0;   // Infinite loop by default
            musicLoaded = true;
        }
    }
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
    else if (IsFileExtension(fileName, ".mp3"))
    {
        drmp3 *ctxMp3 = RL_MALLOC(sizeof(drmp3));
        music.ctxData = ctxMp3;

        int result = drmp3_init_file(ctxMp3, fileName, NULL);

        if (result > 0)
        {
            music.ctxType = MUSIC_AUDIO_MP3;

            music.stream = InitAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
            music.sampleCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels;
            music.loopCount = 0;   // Infinite loop by default
            musicLoaded = true;
        }
    }
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
    else if (IsFileExtension(fileName, ".xm"))
    {
        jar_xm_context_t *ctxXm = NULL;

        int result = jar_xm_create_context_from_file(&ctxXm, 48000, fileName);

        if (result == 0)    // XM AUDIO.System.context created successfully
        {
            music.ctxType = MUSIC_MODULE_XM;
            jar_xm_set_max_loop_count(ctxXm, 0);    // Set infinite number of loops

            // NOTE: Only stereo is supported for XM
            music.stream = InitAudioStream(48000, 16, 2);
            music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)*2;
            music.loopCount = 0;   // Infinite loop by default
            jar_xm_reset(ctxXm);   // make sure we start at the beginning of the song
            musicLoaded = true;

            music.ctxData = ctxXm;
        }
    }
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
    else if (IsFileExtension(fileName, ".mod"))
    {
        jar_mod_context_t *ctxMod = RL_MALLOC(sizeof(jar_mod_context_t));
        music.ctxData = ctxMod;

        jar_mod_init(ctxMod);
        int result = jar_mod_load_file(ctxMod, fileName);

        if (result > 0)
        {
            music.ctxType = MUSIC_MODULE_MOD;

            // NOTE: Only stereo is supported for MOD
            music.stream = InitAudioStream(48000, 16, 2);
            music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)*2;
            music.loopCount = 0;   // Infinite loop by default
            musicLoaded = true;
        }
    }
#endif

    if (!musicLoaded)
    {
        if (false) { }
    #if defined(SUPPORT_FILEFORMAT_OGG)
        else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData);
    #endif
    #if defined(SUPPORT_FILEFORMAT_FLAC)
        else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData);
    #endif
    #if defined(SUPPORT_FILEFORMAT_MP3)
        else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
    #endif
    #if defined(SUPPORT_FILEFORMAT_XM)
        else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData);
    #endif
    #if defined(SUPPORT_FILEFORMAT_MOD)
        else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); }
    #endif

        TRACELOG(LOG_WARNING, "[%s] Music file could not be opened", fileName);
    }
    else
    {
        // Show some music stream info
        TRACELOG(LOG_INFO, "[%s] Music file successfully loaded:", fileName);
        TRACELOG(LOG_INFO, "   Total samples: %i", music.sampleCount);
        TRACELOG(LOG_INFO, "   Sample rate: %i Hz", music.stream.sampleRate);
        TRACELOG(LOG_INFO, "   Sample size: %i bits", music.stream.sampleSize);
        TRACELOG(LOG_INFO, "   Channels: %i (%s)", music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi");
    }

    return music;
}

// Unload music stream
void UnloadMusicStream(Music music)
{
    CloseAudioStream(music.stream);

    if (false) { }
#if defined(SUPPORT_FILEFORMAT_OGG)
    else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData);
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
    else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData);
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
    else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
    else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData);
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
    else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); }
#endif
}

// Start music playing (open stream)
void PlayMusicStream(Music music)
{
    AudioBuffer *audioBuffer = music.stream.buffer;

    if (audioBuffer != NULL)
    {
        // For music streams, we need to make sure we maintain the frame cursor position
        // This is a hack for this section of code in UpdateMusicStream()
        // NOTE: In case window is minimized, music stream is stopped, just make sure to
        // play again on window restore: if (IsMusicPlaying(music)) PlayMusicStream(music);
        ma_uint32 frameCursorPos = audioBuffer->frameCursorPos;
        PlayAudioStream(music.stream);  // WARNING: This resets the cursor position.
        audioBuffer->frameCursorPos = frameCursorPos;
    }
    else TRACELOG(LOG_ERROR, "PlayMusicStream() : No audio buffer");

}

// Pause music playing
void PauseMusicStream(Music music)
{
    PauseAudioStream(music.stream);
}

// Resume music playing
void ResumeMusicStream(Music music)
{
    ResumeAudioStream(music.stream);
}

// Stop music playing (close stream)
void StopMusicStream(Music music)
{
    StopAudioStream(music.stream);

    switch (music.ctxType)
    {
#if defined(SUPPORT_FILEFORMAT_OGG)
        case MUSIC_AUDIO_OGG: stb_vorbis_seek_start((stb_vorbis *)music.ctxData); break;
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
        case MUSIC_AUDIO_FLAC: drflac_seek_to_pcm_frame((drflac *)music.ctxData, 0); break;
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
        case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame((drmp3 *)music.ctxData, 0); break;
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
        case MUSIC_MODULE_XM: jar_xm_reset((jar_xm_context_t *)music.ctxData); break;
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
        case MUSIC_MODULE_MOD: jar_mod_seek_start((jar_mod_context_t *)music.ctxData); break;
#endif
        default: break;
    }
}

// Update (re-fill) music buffers if data already processed
void UpdateMusicStream(Music music)
{
    bool streamEnding = false;

    unsigned int subBufferSizeInFrames = music.stream.buffer->sizeInFrames/2;

    // NOTE: Using dynamic allocation because it could require more than 16KB
    void *pcm = RL_CALLOC(subBufferSizeInFrames*music.stream.channels*music.stream.sampleSize/8, 1);

    int samplesCount = 0;    // Total size of data streamed in L+R samples for xm floats, individual L or R for ogg shorts

    // TODO: Get the sampleLeft using totalFramesProcessed... but first, get total frames processed correctly...
    //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
    int sampleLeft = music.sampleCount - (music.stream.buffer->totalFramesProcessed*music.stream.channels);

    while (IsAudioStreamProcessed(music.stream))
    {
        if ((sampleLeft/music.stream.channels) >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames*music.stream.channels;
        else samplesCount = sampleLeft;

        switch (music.ctxType)
        {
        #if defined(SUPPORT_FILEFORMAT_OGG)
            case MUSIC_AUDIO_OGG:
            {
                // NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
                stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)pcm, samplesCount);

            } break;
        #endif
        #if defined(SUPPORT_FILEFORMAT_FLAC)
            case MUSIC_AUDIO_FLAC:
            {
                // NOTE: Returns the number of samples to process (not required)
                drflac_read_pcm_frames_s16((drflac *)music.ctxData, samplesCount, (short *)pcm);

            } break;
        #endif
        #if defined(SUPPORT_FILEFORMAT_MP3)
            case MUSIC_AUDIO_MP3:
            {
                // NOTE: samplesCount, actually refers to framesCount and returns the number of frames processed
                drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, samplesCount/music.stream.channels, (float *)pcm);

            } break;
        #endif
        #if defined(SUPPORT_FILEFORMAT_XM)
            case MUSIC_MODULE_XM:
            {
                // NOTE: Internally this function considers 2 channels generation, so samplesCount/2
                jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)pcm, samplesCount/2);
            } break;
        #endif
        #if defined(SUPPORT_FILEFORMAT_MOD)
            case MUSIC_MODULE_MOD:
            {
                // NOTE: 3rd parameter (nbsample) specify the number of stereo 16bits samples you want, so sampleCount/2
                jar_mod_fillbuffer((jar_mod_context_t *)music.ctxData, (short *)pcm, samplesCount/2, 0);
            } break;
        #endif
            default: break;
        }

        UpdateAudioStream(music.stream, pcm, samplesCount);

        if ((music.ctxType == MUSIC_MODULE_XM) || (music.ctxType == MUSIC_MODULE_MOD))
        {
            if (samplesCount > 1) sampleLeft -= samplesCount/2;
            else sampleLeft -= samplesCount;
        }
        else sampleLeft -= samplesCount;

        if (sampleLeft <= 0)
        {
            streamEnding = true;
            break;
        }
    }

    // Free allocated pcm data
    RL_FREE(pcm);

    // Reset audio stream for looping
    if (streamEnding)
    {
        StopMusicStream(music);        // Stop music (and reset)

        // Decrease loopCount to stop when required
        if (music.loopCount > 1)
        {
            music.loopCount--;         // Decrease loop count
            PlayMusicStream(music);    // Play again
        }
        else if (music.loopCount == 0) PlayMusicStream(music);
    }
    else
    {
        // NOTE: In case window is minimized, music stream is stopped,
        // just make sure to play again on window restore
        if (IsMusicPlaying(music)) PlayMusicStream(music);
    }
}

// Check if any music is playing
bool IsMusicPlaying(Music music)
{
    return IsAudioStreamPlaying(music.stream);
}

// Set volume for music
void SetMusicVolume(Music music, float volume)
{
    SetAudioStreamVolume(music.stream, volume);
}

// Set pitch for music
void SetMusicPitch(Music music, float pitch)
{
    SetAudioStreamPitch(music.stream, pitch);
}

// Set music loop count (loop repeats)
// NOTE: If set to 0, means infinite loop
void SetMusicLoopCount(Music music, int count)
{
    music.loopCount = count;
}

// Get music time length (in seconds)
float GetMusicTimeLength(Music music)
{
    float totalSeconds = 0.0f;

    totalSeconds = (float)music.sampleCount/(music.stream.sampleRate*music.stream.channels);

    return totalSeconds;
}

// Get current music time played (in seconds)
float GetMusicTimePlayed(Music music)
{
    float secondsPlayed = 0.0f;

    //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
    unsigned int samplesPlayed = music.stream.buffer->totalFramesProcessed*music.stream.channels;
    secondsPlayed = (float)samplesPlayed/(music.stream.sampleRate*music.stream.channels);

    return secondsPlayed;
}

// Init audio stream (to stream audio pcm data)
AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels)
{
    AudioStream stream = { 0 };

    stream.sampleRate = sampleRate;
    stream.sampleSize = sampleSize;
    stream.channels = channels;

    ma_format formatIn = ((stream.sampleSize == 8)? ma_format_u8 : ((stream.sampleSize == 16)? ma_format_s16 : ma_format_f32));

    // The size of a streaming buffer must be at least double the size of a period
    unsigned int periodSize = AUDIO.System.device.playback.internalPeriodSizeInFrames/AUDIO.System.device.playback.internalPeriods;
    unsigned int subBufferSize = AUDIO_BUFFER_SIZE;

    if (subBufferSize < periodSize) subBufferSize = periodSize;

    stream.buffer = InitAudioBuffer(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, "Audio stream loaded successfully (%i Hz, %i bit, %s)", stream.sampleRate, stream.sampleSize, (stream.channels == 1)? "Mono" : "Stereo");
    }
    else TRACELOG(LOG_ERROR, "InitAudioStream() : Failed to create audio buffer");

    return stream;
}

// Close audio stream and free memory
void CloseAudioStream(AudioStream stream)
{
    CloseAudioBuffer(stream.buffer);

    TRACELOG(LOG_INFO, "Unloaded audio stream data");
}

// Update audio stream buffers with data
// NOTE 1: Only updates one buffer of the stream source: unqueue -> update -> queue
// NOTE 2: To unqueue a buffer it needs to be processed: IsAudioStreamProcessed()
void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
{
    AudioBuffer *audioBuffer = stream.buffer;

    if (audioBuffer != NULL)
    {
        if (audioBuffer->isSubBufferProcessed[0] || audioBuffer->isSubBufferProcessed[1])
        {
            ma_uint32 subBufferToUpdate = 0;

            if (audioBuffer->isSubBufferProcessed[0] && audioBuffer->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;
                audioBuffer->frameCursorPos = 0;
            }
            else
            {
                // Just update whichever sub-buffer is processed.
                subBufferToUpdate = (audioBuffer->isSubBufferProcessed[0])? 0 : 1;
            }

            ma_uint32 subBufferSizeInFrames = audioBuffer->sizeInFrames/2;
            unsigned char *subBuffer = audioBuffer->data + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate);

            // TODO: Get total frames processed on this buffer... DOES NOT WORK.
            audioBuffer->totalFramesProcessed += subBufferSizeInFrames;

            // Does this API expect a whole buffer to be updated in one go?
            // Assuming so, but if not will need to change this logic.
            if (subBufferSizeInFrames >= (ma_uint32)samplesCount/stream.channels)
            {
                ma_uint32 framesToWrite = subBufferSizeInFrames;

                if (framesToWrite > ((ma_uint32)samplesCount/stream.channels)) framesToWrite = (ma_uint32)samplesCount/stream.channels;

                ma_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8);
                memcpy(subBuffer, data, bytesToWrite);

                // Any leftover frames should be filled with zeros.
                ma_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite;

                if (leftoverFrameCount > 0) memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8));

                audioBuffer->isSubBufferProcessed[subBufferToUpdate] = false;
            }
            else TRACELOG(LOG_ERROR, "UpdateAudioStream() : Attempting to write too many frames to buffer");
        }
        else TRACELOG(LOG_ERROR, "UpdateAudioStream() : Audio buffer not available for updating");
    }
    else TRACELOG(LOG_ERROR, "UpdateAudioStream() : No audio buffer");
}

// Check if any audio stream buffers requires refill
bool IsAudioStreamProcessed(AudioStream stream)
{
    if (stream.buffer == NULL)
    {
        TRACELOG(LOG_ERROR, "IsAudioStreamProcessed() : No audio buffer");
        return false;
    }

    return (stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1]);
}

// Play audio stream
void PlayAudioStream(AudioStream stream)
{
    PlayAudioBuffer(stream.buffer);
}

// Play audio stream
void PauseAudioStream(AudioStream stream)
{
    PauseAudioBuffer(stream.buffer);
}

// Resume audio stream playing
void ResumeAudioStream(AudioStream stream)
{
    ResumeAudioBuffer(stream.buffer);
}

// Check if audio stream is playing.
bool IsAudioStreamPlaying(AudioStream stream)
{
    return IsAudioBufferPlaying(stream.buffer);
}

// Stop audio stream
void StopAudioStream(AudioStream stream)
{
    StopAudioBuffer(stream.buffer);
}

void SetAudioStreamVolume(AudioStream stream, float volume)
{
    SetAudioBufferVolume(stream.buffer, volume);
}

void SetAudioStreamPitch(AudioStream stream, float pitch)
{
    SetAudioBufferPitch(stream.buffer, pitch);
}

//----------------------------------------------------------------------------------
// Module specific Functions Definition
//----------------------------------------------------------------------------------

// Log callback function
static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message)
{
    (void)pContext;
    (void)pDevice;

    TRACELOG(LOG_ERROR, message);   // All log messages from miniaudio are errors
}

// Reads audio data from an AudioBuffer object in internal format.
static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer, void *framesOut, ma_uint32 frameCount)
{
    ma_uint32 subBufferSizeInFrames = (audioBuffer->sizeInFrames > 1)? audioBuffer->sizeInFrames/2 : audioBuffer->sizeInFrames;
    ma_uint32 currentSubBufferIndex = audioBuffer->frameCursorPos/subBufferSizeInFrames;

    if (currentSubBufferIndex > 1)
    {
        TRACELOGD("Frame cursor position moved too far forward in audio stream");
        return 0;
    }

    // Another thread can update the processed state of buffers so
    // we just take a copy here to try and avoid potential synchronization problems
    bool isSubBufferProcessed[2];
    isSubBufferProcessed[0] = audioBuffer->isSubBufferProcessed[0];
    isSubBufferProcessed[1] = audioBuffer->isSubBufferProcessed[1];

    ma_uint32 frameSizeInBytes = ma_get_bytes_per_frame(audioBuffer->converter.config.formatIn, audioBuffer->converter.config.channelsIn);

    // Fill out every frame until we find a buffer that's marked as processed. Then fill the remainder with 0
    ma_uint32 framesRead = 0;
    while (1)
    {
        // We break from this loop differently depending on the buffer's usage
        //  - For static buffers, we simply fill as much data as we can
        //  - For streaming buffers we only fill the halves of the buffer that are processed
        //    Unprocessed halves must keep their audio data in-tact
        if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
        {
            if (framesRead >= frameCount) break;
        }
        else
        {
            if (isSubBufferProcessed[currentSubBufferIndex]) break;
        }

        ma_uint32 totalFramesRemaining = (frameCount - framesRead);
        if (totalFramesRemaining == 0) break;

        ma_uint32 framesRemainingInOutputBuffer;
        if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
        {
            framesRemainingInOutputBuffer = audioBuffer->sizeInFrames - audioBuffer->frameCursorPos;
        }
        else
        {
            ma_uint32 firstFrameIndexOfThisSubBuffer = subBufferSizeInFrames*currentSubBufferIndex;
            framesRemainingInOutputBuffer = subBufferSizeInFrames - (audioBuffer->frameCursorPos - firstFrameIndexOfThisSubBuffer);
        }

        ma_uint32 framesToRead = totalFramesRemaining;
        if (framesToRead > framesRemainingInOutputBuffer) framesToRead = framesRemainingInOutputBuffer;

        memcpy((unsigned char *)framesOut + (framesRead*frameSizeInBytes), audioBuffer->data + (audioBuffer->frameCursorPos*frameSizeInBytes), framesToRead*frameSizeInBytes);
        audioBuffer->frameCursorPos = (audioBuffer->frameCursorPos + framesToRead)%audioBuffer->sizeInFrames;
        framesRead += framesToRead;

        // If we've read to the end of the buffer, mark it as processed
        if (framesToRead == framesRemainingInOutputBuffer)
        {
            audioBuffer->isSubBufferProcessed[currentSubBufferIndex] = true;
            isSubBufferProcessed[currentSubBufferIndex] = true;

            currentSubBufferIndex = (currentSubBufferIndex + 1)%2;

            // We need to break from this loop if we're not looping
            if (!audioBuffer->looping)
            {
                StopAudioBuffer(audioBuffer);
                break;
            }
        }
    }

    // Zero-fill excess
    ma_uint32 totalFramesRemaining = (frameCount - framesRead);
    if (totalFramesRemaining > 0)
    {
        memset((unsigned char *)framesOut + (framesRead*frameSizeInBytes), 0, totalFramesRemaining*frameSizeInBytes);

        // For static buffers we can fill the remaining frames with silence for safety, but we don't want
        // to report those frames as "read". The reason for this is that the caller uses the return value
        // to know whether or not a non-looping sound has finished playback.
        if (audioBuffer->usage != AUDIO_BUFFER_USAGE_STATIC) framesRead += totalFramesRemaining;
    }

    return framesRead;
}

// Reads audio data from an AudioBuffer object in device format. Returned data will be in a format appropriate for mixing.
static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, float *framesOut, ma_uint32 frameCount)
{
    // What's going on here is that we're continuously converting data from the AudioBuffer's internal format to the mixing format, which 
    // should be defined by the output format of the data converter. We do this until frameCount frames have been output. The important
    // detail to remember here is that we never, ever attempt to read more input data than is required for the specified number of output
    // frames. This can be achieved with ma_data_converter_get_required_input_frame_count().
    ma_uint8 inputBuffer[4096];
    ma_uint32 inputBufferFrameCap = sizeof(inputBuffer) / ma_get_bytes_per_frame(audioBuffer->converter.config.formatIn, audioBuffer->converter.config.channelsIn);

    ma_uint32 totalOutputFramesProcessed = 0;
    while (totalOutputFramesProcessed < frameCount)
    {
        ma_uint64 outputFramesToProcessThisIteration = frameCount - totalOutputFramesProcessed;

        ma_uint64 inputFramesToProcessThisIteration = ma_data_converter_get_required_input_frame_count(&audioBuffer->converter, outputFramesToProcessThisIteration);
        if (inputFramesToProcessThisIteration > inputBufferFrameCap)
        {
            inputFramesToProcessThisIteration = inputBufferFrameCap;
        }

        float *runningFramesOut = framesOut + (totalOutputFramesProcessed * audioBuffer->converter.config.channelsOut);

        /* At this point we can convert the data to our mixing format. */
        ma_uint64 inputFramesProcessedThisIteration = ReadAudioBufferFramesInInternalFormat(audioBuffer, inputBuffer, (ma_uint32)inputFramesToProcessThisIteration);    /* Safe cast. */
        ma_uint64 outputFramesProcessedThisIteration = outputFramesToProcessThisIteration;
        ma_data_converter_process_pcm_frames(&audioBuffer->converter, inputBuffer, &inputFramesProcessedThisIteration, runningFramesOut, &outputFramesProcessedThisIteration);
        
        totalOutputFramesProcessed += (ma_uint32)outputFramesProcessedThisIteration; /* Safe cast. */

        if (inputFramesProcessedThisIteration < inputFramesToProcessThisIteration)
        {
            break;  /* Ran out of input data. */
        }

        /* This should never be hit, but will add it here for safety. Ensures we get out of the loop when no input nor output frames are processed. */
        if (inputFramesProcessedThisIteration == 0 && outputFramesProcessedThisIteration == 0)
        {
            break;
        }
    }

    return totalOutputFramesProcessed;
}


// Sending audio data to device callback function
// NOTE: All the mixing takes place here
static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount)
{
    (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)
                {
                    TRACELOGD("Mixed too many frames from audio buffer");
                    break;
                }

                if (framesRead == frameCount) break;

                // Just read as much data as we can from the stream
                ma_uint32 framesToRead = (frameCount - framesRead);

                while (framesToRead > 0)
                {
                    float tempBuffer[1024]; // 512 frames for stereo

                    ma_uint32 framesToReadRightNow = framesToRead;
                    if (framesToReadRightNow > sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS)
                    {
                        framesToReadRightNow = sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS;
                    }

                    ma_uint32 framesJustRead = ReadAudioBufferFramesInMixingFormat(audioBuffer, tempBuffer, framesToReadRightNow);
                    if (framesJustRead > 0)
                    {
                        float *framesOut = (float *)pFramesOut + (framesRead*AUDIO.System.device.playback.channels);
                        float *framesIn  = tempBuffer;

                        MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume);

                        framesToRead -= framesJustRead;
                        framesRead += framesJustRead;
                    }

                    if (!audioBuffer->playing)
                    {
                        framesRead = frameCount;
                        break;
                    }

                    // If we weren't able to read all the frames we requested, break
                    if (framesJustRead < framesToReadRightNow)
                    {
                        if (!audioBuffer->looping)
                        {
                            StopAudioBuffer(audioBuffer);
                            break;
                        }
                        else
                        {
                            // Should never get here, but just for safety,
                            // move the cursor position back to the start and continue the loop
                            audioBuffer->frameCursorPos = 0;
                            continue;
                        }
                    }
                }

                // If for some reason we weren't able to read every frame we'll need to break from the loop
                // Not doing this could theoretically put us into an infinite loop
                if (framesToRead > 0) break;
            }
        }
    }

    ma_mutex_unlock(&AUDIO.System.lock);
}

// This is the main mixing function. Mixing is pretty simple in this project - it's just an accumulation.
// NOTE: framesOut is both an input and an output. It will be initially filled with zeros outside of this function.
static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume)
{
    for (ma_uint32 iFrame = 0; iFrame < frameCount; ++iFrame)
    {
        for (ma_uint32 iChannel = 0; iChannel < AUDIO.System.device.playback.channels; ++iChannel)
        {
            float *frameOut = framesOut + (iFrame*AUDIO.System.device.playback.channels);
            const float *frameIn  = framesIn  + (iFrame*AUDIO.System.device.playback.channels);

            frameOut[iChannel] += (frameIn[iChannel]*localVolume);
        }
    }
}

// Initialise the multichannel buffer pool
static void InitAudioBufferPool(void)
{
    // Dummy buffers
    for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
    {
        AUDIO.MultiChannel.pool[i] = InitAudioBuffer(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, 0, AUDIO_BUFFER_USAGE_STATIC);
    }
}

// Close the audio buffers pool
static void CloseAudioBufferPool(void)
{
    for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
    {
        RL_FREE(AUDIO.MultiChannel.pool[i]->data);
        RL_FREE(AUDIO.MultiChannel.pool[i]);
    }
}

#if defined(SUPPORT_FILEFORMAT_WAV)
// Load WAV file into Wave structure
static Wave LoadWAV(const char *fileName)
{
    // Basic WAV headers structs
    typedef struct {
        char chunkID[4];
        int chunkSize;
        char format[4];
    } WAVRiffHeader;

    typedef struct {
        char subChunkID[4];
        int subChunkSize;
        short audioFormat;
        short numChannels;
        int sampleRate;
        int byteRate;
        short blockAlign;
        short bitsPerSample;
    } WAVFormat;

    typedef struct {
        char subChunkID[4];
        int subChunkSize;
    } WAVData;

    WAVRiffHeader wavRiffHeader = { 0 };
    WAVFormat wavFormat = { 0 };
    WAVData wavData = { 0 };

    Wave wave = { 0 };
    FILE *wavFile = NULL;

    wavFile = fopen(fileName, "rb");

    if (wavFile == NULL)
    {
        TRACELOG(LOG_WARNING, "[%s] WAV file could not be opened", fileName);
        wave.data = NULL;
    }
    else
    {
        // Read in the first chunk into the struct
        fread(&wavRiffHeader, sizeof(WAVRiffHeader), 1, wavFile);

        // Check for RIFF and WAVE tags
        if ((wavRiffHeader.chunkID[0] != 'R') ||
            (wavRiffHeader.chunkID[1] != 'I') ||
            (wavRiffHeader.chunkID[2] != 'F') ||
            (wavRiffHeader.chunkID[3] != 'F') ||
            (wavRiffHeader.format[0] != 'W') ||
            (wavRiffHeader.format[1] != 'A') ||
            (wavRiffHeader.format[2] != 'V') ||
            (wavRiffHeader.format[3] != 'E'))
        {
            TRACELOG(LOG_WARNING, "[%s] Invalid RIFF or WAVE Header", fileName);
        }
        else
        {
            // Read in the 2nd chunk for the wave info
            fread(&wavFormat, sizeof(WAVFormat), 1, wavFile);

            // Check for fmt tag
            if ((wavFormat.subChunkID[0] != 'f') || (wavFormat.subChunkID[1] != 'm') ||
                (wavFormat.subChunkID[2] != 't') || (wavFormat.subChunkID[3] != ' '))
            {
                TRACELOG(LOG_WARNING, "[%s] Invalid Wave format", fileName);
            }
            else
            {
                // Check for extra parameters;
                if (wavFormat.subChunkSize > 16) fseek(wavFile, sizeof(short), SEEK_CUR);

                // Read in the the last byte of data before the sound file
                fread(&wavData, sizeof(WAVData), 1, wavFile);

                // Check for data tag
                if ((wavData.subChunkID[0] != 'd') || (wavData.subChunkID[1] != 'a') ||
                    (wavData.subChunkID[2] != 't') || (wavData.subChunkID[3] != 'a'))
                {
                    TRACELOG(LOG_WARNING, "[%s] Invalid data header", fileName);
                }
                else
                {
                    // Allocate memory for data
                    wave.data = RL_MALLOC(wavData.subChunkSize);

                    // Read in the sound data into the soundData variable
                    fread(wave.data, wavData.subChunkSize, 1, wavFile);

                    // Store wave parameters
                    wave.sampleRate = wavFormat.sampleRate;
                    wave.sampleSize = wavFormat.bitsPerSample;
                    wave.channels = wavFormat.numChannels;

                    // NOTE: Only support 8 bit, 16 bit and 32 bit sample sizes
                    if ((wave.sampleSize != 8) && (wave.sampleSize != 16) && (wave.sampleSize != 32))
                    {
                        TRACELOG(LOG_WARNING, "[%s] WAV sample size (%ibit) not supported, converted to 16bit", fileName, wave.sampleSize);
                        WaveFormat(&wave, wave.sampleRate, 16, wave.channels);
                    }

                    // NOTE: Only support up to 2 channels (mono, stereo)
                    if (wave.channels > 2)
                    {
                        WaveFormat(&wave, wave.sampleRate, wave.sampleSize, 2);
                        TRACELOG(LOG_WARNING, "[%s] WAV channels number (%i) not supported, converted to 2 channels", fileName, wave.channels);
                    }

                    // NOTE: subChunkSize comes in bytes, we need to translate it to number of samples
                    wave.sampleCount = (wavData.subChunkSize/(wave.sampleSize/8))/wave.channels;

                    TRACELOG(LOG_INFO, "[%s] WAV file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
                }
            }
        }

        fclose(wavFile);
    }

    return wave;
}

// Save wave data as WAV file
static int SaveWAV(Wave wave, const char *fileName)
{
    int success = 0;
    int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8;

    // Basic WAV headers structs
    typedef struct {
        char chunkID[4];
        int chunkSize;
        char format[4];
    } RiffHeader;

    typedef struct {
        char subChunkID[4];
        int subChunkSize;
        short audioFormat;
        short numChannels;
        int sampleRate;
        int byteRate;
        short blockAlign;
        short bitsPerSample;
    } WaveFormat;

    typedef struct {
        char subChunkID[4];
        int subChunkSize;
    } WaveData;

    FILE *wavFile = fopen(fileName, "wb");

    if (wavFile == NULL) TRACELOG(LOG_WARNING, "[%s] WAV audio file could not be created", fileName);
    else
    {
        RiffHeader riffHeader;
        WaveFormat waveFormat;
        WaveData waveData;

        // Fill structs with data
        riffHeader.chunkID[0] = 'R';
        riffHeader.chunkID[1] = 'I';
        riffHeader.chunkID[2] = 'F';
        riffHeader.chunkID[3] = 'F';
        riffHeader.chunkSize = 44 - 4 + wave.sampleCount*wave.sampleSize/8;
        riffHeader.format[0] = 'W';
        riffHeader.format[1] = 'A';
        riffHeader.format[2] = 'V';
        riffHeader.format[3] = 'E';

        waveFormat.subChunkID[0] = 'f';
        waveFormat.subChunkID[1] = 'm';
        waveFormat.subChunkID[2] = 't';
        waveFormat.subChunkID[3] = ' ';
        waveFormat.subChunkSize = 16;
        waveFormat.audioFormat = 1;
        waveFormat.numChannels = wave.channels;
        waveFormat.sampleRate = wave.sampleRate;
        waveFormat.byteRate = wave.sampleRate*wave.sampleSize/8;
        waveFormat.blockAlign = wave.sampleSize/8;
        waveFormat.bitsPerSample = wave.sampleSize;

        waveData.subChunkID[0] = 'd';
        waveData.subChunkID[1] = 'a';
        waveData.subChunkID[2] = 't';
        waveData.subChunkID[3] = 'a';
        waveData.subChunkSize = dataSize;

        fwrite(&riffHeader, sizeof(RiffHeader), 1, wavFile);
        fwrite(&waveFormat, sizeof(WaveFormat), 1, wavFile);
        fwrite(&waveData, sizeof(WaveData), 1, wavFile);

        success = fwrite(wave.data, dataSize, 1, wavFile);

        fclose(wavFile);
    }

    // If all data has been written correctly to file, success = 1
    return success;
}
#endif

#if defined(SUPPORT_FILEFORMAT_OGG)
// Load OGG file into Wave structure
// NOTE: Using stb_vorbis library
static Wave LoadOGG(const char *fileName)
{
    Wave wave = { 0 };

    stb_vorbis *oggFile = stb_vorbis_open_filename(fileName, NULL, NULL);

    if (oggFile == NULL) TRACELOG(LOG_WARNING, "[%s] OGG file could not be opened", fileName);
    else
    {
        stb_vorbis_info info = stb_vorbis_get_info(oggFile);

        wave.sampleRate = info.sample_rate;
        wave.sampleSize = 16;                   // 16 bit per sample (short)
        wave.channels = info.channels;
        wave.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples(oggFile)*info.channels;  // Independent by channel

        float totalSeconds = stb_vorbis_stream_length_in_seconds(oggFile);
        if (totalSeconds > 10) TRACELOG(LOG_WARNING, "[%s] Ogg audio length is larger than 10 seconds (%f), that's a big file in memory, consider music streaming", fileName, totalSeconds);

        wave.data = (short *)RL_MALLOC(wave.sampleCount*wave.channels*sizeof(short));

        // NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
        //int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(oggFile, info.channels, (short *)wave.data, wave.sampleCount*wave.channels);
        //TRACELOGD("[%s] Samples obtained: %i", fileName, numSamplesOgg);

        TRACELOG(LOG_INFO, "[%s] OGG file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");

        stb_vorbis_close(oggFile);
    }

    return wave;
}
#endif

#if defined(SUPPORT_FILEFORMAT_FLAC)
// Load FLAC file into Wave structure
// NOTE: Using dr_flac library
static Wave LoadFLAC(const char *fileName)
{
    Wave wave;

    // Decode an entire FLAC file in one go
    unsigned long long int totalSampleCount;
    wave.data = drflac_open_file_and_read_pcm_frames_s16(fileName, &wave.channels, &wave.sampleRate, &totalSampleCount);

    wave.sampleCount = (unsigned int)totalSampleCount;
    wave.sampleSize = 16;

    // NOTE: Only support up to 2 channels (mono, stereo)
    if (wave.channels > 2) TRACELOG(LOG_WARNING, "[%s] FLAC channels number (%i) not supported", fileName, wave.channels);

    if (wave.data == NULL) TRACELOG(LOG_WARNING, "[%s] FLAC data could not be loaded", fileName);
    else TRACELOG(LOG_INFO, "[%s] FLAC file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");

    return wave;
}
#endif

#if defined(SUPPORT_FILEFORMAT_MP3)
// Load MP3 file into Wave structure
// NOTE: Using dr_mp3 library
static Wave LoadMP3(const char *fileName)
{
    Wave wave = { 0 };

    // Decode an entire MP3 file in one go
    unsigned long long int totalFrameCount = 0;
    drmp3_config config = { 0 };
    wave.data = drmp3_open_file_and_read_f32(fileName, &config, &totalFrameCount);

    wave.channels = config.outputChannels;
    wave.sampleRate = config.outputSampleRate;
    wave.sampleCount = (int)totalFrameCount*wave.channels;
    wave.sampleSize = 32;

    // NOTE: Only support up to 2 channels (mono, stereo)
    if (wave.channels > 2) TRACELOG(LOG_WARNING, "[%s] MP3 channels number (%i) not supported", fileName, wave.channels);

    if (wave.data == NULL) TRACELOG(LOG_WARNING, "[%s] MP3 data could not be loaded", fileName);
    else TRACELOG(LOG_INFO, "[%s] MP3 file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");

    return wave;
}
#endif

// Some required functions for audio standalone module version
#if defined(RAUDIO_STANDALONE)
// Check file extension
bool IsFileExtension(const char *fileName, const char *ext)
{
    bool result = false;
    const char *fileExt;

    if ((fileExt = strrchr(fileName, '.')) != NULL)
    {
        if (strcmp(fileExt, ext) == 0) result = true;
    }

    return result;
}

// Show trace log messages (LOG_INFO, LOG_WARNING, LOG_ERROR, LOG_DEBUG)
void TRACELOG(int msgType, const char *text, ...)
{
    va_list args;
    va_start(args, text);

    switch (msgType)
    {
        case LOG_INFO: fprintf(stdout, "INFO: "); break;
        case LOG_ERROR: fprintf(stdout, "ERROR: "); break;
        case LOG_WARNING: fprintf(stdout, "WARNING: "); break;
        case LOG_DEBUG: fprintf(stdout, "DEBUG: "); break;
        default: break;
    }

    vfprintf(stdout, text, args);
    fprintf(stdout, "\n");

    va_end(args);

    if (msgType == LOG_ERROR) exit(1);
}
#endif

#undef AudioBuffer