raudio.c

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#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(SUPPORT_MODULE_RAUDIO)
 
#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 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, ...) (void)0
    #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_OGG)
    // TODO: Remap stb_vorbis malloc()/free() calls to RL_MALLOC/RL_FREE
 
    #define STB_VORBIS_IMPLEMENTATION
    #include "external/stb_vorbis.h"    // OGG loading functions
#endif
 
#if defined(SUPPORT_FILEFORMAT_XM)
    #define JARXM_MALLOC RL_MALLOC
    #define JARXM_FREE RL_FREE
 
    #define JAR_XM_IMPLEMENTATION
    #include "external/jar_xm.h"        // XM loading functions
#endif
 
#if defined(SUPPORT_FILEFORMAT_MOD)
    #define JARMOD_MALLOC RL_MALLOC
    #define JARMOD_FREE RL_FREE
 
    #define JAR_MOD_IMPLEMENTATION
    #include "external/jar_mod.h"       // MOD loading functions
#endif
 
#if defined(SUPPORT_FILEFORMAT_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_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_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(_MSC_VER)
    #undef bool
#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
#ifndef DEFAULT_AUDIO_BUFFER_SIZE
    #define DEFAULT_AUDIO_BUFFER_SIZE       4096    // Default audio buffer size
#endif
 
 
//----------------------------------------------------------------------------------
// 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_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_MODULE_XM,        // XM module audio context
    MUSIC_MODULE_MOD        // MOD module audio context
} MusicContextType;
 
#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
 
// 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 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, 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 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
    } System;
    struct {
        AudioBuffer *first;         // Pointer to first AudioBuffer in the list
        AudioBuffer *last;          // Pointer to last AudioBuffer in the list
        int defaultSize;            // Default audio buffer size for audio streams
    } Buffer;
    struct {
        unsigned int poolCounter;                               // AudioBuffer pointers pool counter
        AudioBuffer *pool[MAX_AUDIO_BUFFER_POOL_CHANNELS];      // Multichannel AudioBuffer pointers pool
        unsigned int channels[MAX_AUDIO_BUFFER_POOL_CHANNELS];  // AudioBuffer pool channels
    } MultiChannel;
} AudioData;
 
//----------------------------------------------------------------------------------
// Global Variables Definition
//----------------------------------------------------------------------------------
static AudioData AUDIO = {          // Global AUDIO context
 
    // NOTE: Music buffer size is defined by number of samples, independent of sample size and channels number
    // After some math, considering a sampleRate of 48000, a buffer refill rate of 1/60 seconds and a
    // standard double-buffering system, a 4096 samples buffer has been chosen, it should be enough
    // In case of music-stalls, just increase this number
    .Buffer.defaultSize = 0
};
 
//----------------------------------------------------------------------------------
// Module specific Functions Declaration
//----------------------------------------------------------------------------------
static void OnLog(void *pUserData, ma_uint32 level, const char *pMessage);
static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount);
static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, AudioBuffer *buffer);
 
#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 unsigned char *LoadFileData(const char *fileName, unsigned int *bytesRead);     // Load file data as byte array (read)
static bool SaveFileData(const char *fileName, void *data, unsigned int bytesToWrite); // 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;
    }
 
    // 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;
    }
 
    // 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.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;
    }
 
    // Init dummy audio buffers pool for multichannel sound playing
    for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
    {
        // WARNING: An empty audio buffer is created (data = 0) and added to list, AudioBuffer data is filled on PlaySoundMulti()
        AUDIO.MultiChannel.pool[i] = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, 0, AUDIO_BUFFER_USAGE_STATIC);
    }
 
    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)
    {
        // Unload dummy audio buffers pool
        // WARNING: They can be pointing to already unloaded data
        for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
        {
            //UnloadAudioBuffer(AUDIO.MultiChannel.pool[i]);
            if (AUDIO.MultiChannel.pool[i] != NULL)
            {
                ma_data_converter_uninit(&AUDIO.MultiChannel.pool[i]->converter, NULL);
                UntrackAudioBuffer(AUDIO.MultiChannel.pool[i]);
                //RL_FREE(buffer->data);    // Already unloaded by UnloadSound()
                RL_FREE(AUDIO.MultiChannel.pool[i]);
            }
        }
 
        ma_mutex_uninit(&AUDIO.System.lock);
        ma_device_uninit(&AUDIO.System.device);
        ma_context_uninit(&AUDIO.System.context);
 
        AUDIO.System.isReady = false;
 
        TRACELOG(LOG_INFO, "AUDIO: Device closed successfully");
    }
    else TRACELOG(LOG_WARNING, "AUDIO: Device could not be closed, not currently initialized");
}
 
// Check if device has been initialized successfully
bool IsAudioDeviceReady(void)
{
    return AUDIO.System.isReady;
}
 
// Set master volume (listener)
void SetMasterVolume(float volume)
{
    ma_device_set_master_volume(&AUDIO.System.device, volume);
}
 
//----------------------------------------------------------------------------------
// Module Functions Definition - Audio Buffer management
//----------------------------------------------------------------------------------
 
// Initialize a new audio buffer (filled with silence)
AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage)
{
    AudioBuffer *audioBuffer = (AudioBuffer *)RL_CALLOC(1, sizeof(AudioBuffer));
 
    if (audioBuffer == NULL)
    {
        TRACELOG(LOG_WARNING, "AUDIO: Failed to allocate memory for buffer");
        return NULL;
    }
 
    if (sizeInFrames > 0) audioBuffer->data = RL_CALLOC(sizeInFrames*channels*ma_get_bytes_per_sample(format), 1);
 
    // Audio data runs through a format converter
    ma_data_converter_config converterConfig = ma_data_converter_config_init(format, AUDIO_DEVICE_FORMAT, channels, AUDIO_DEVICE_CHANNELS, sampleRate, AUDIO.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 = 0.5f;
 
    audioBuffer->callback = NULL;
    audioBuffer->processor = NULL;
 
    audioBuffer->playing = false;
    audioBuffer->paused = false;
    audioBuffer->looping = false;
 
    audioBuffer->usage = usage;
    audioBuffer->frameCursorPos = 0;
    audioBuffer->sizeInFrames = sizeInFrames;
 
    // Buffers should be marked as processed by default so that a call to
    // UpdateAudioStream() immediately after initialization works correctly
    audioBuffer->isSubBufferProcessed[0] = true;
    audioBuffer->isSubBufferProcessed[1] = true;
 
    // Track audio buffer to linked list next position
    TrackAudioBuffer(audioBuffer);
 
    return audioBuffer;
}
 
// Delete an audio buffer
void UnloadAudioBuffer(AudioBuffer *buffer)
{
    if (buffer != NULL)
    {
        ma_data_converter_uninit(&buffer->converter, NULL);
        UntrackAudioBuffer(buffer);
        RL_FREE(buffer->data);
        RL_FREE(buffer);
    }
}
 
// Check if an audio buffer is playing
bool IsAudioBufferPlaying(AudioBuffer *buffer)
{
    bool result = false;
 
    if (buffer != NULL) result = (buffer->playing && !buffer->paused);
 
    return result;
}
 
// Play an audio buffer
// NOTE: Buffer is restarted to the start.
// Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position should be maintained.
void PlayAudioBuffer(AudioBuffer *buffer)
{
    if (buffer != NULL)
    {
        buffer->playing = true;
        buffer->paused = false;
        buffer->frameCursorPos = 0;
    }
}
 
// Stop an audio buffer
void StopAudioBuffer(AudioBuffer *buffer)
{
    if (buffer != NULL)
    {
        if (IsAudioBufferPlaying(buffer))
        {
            buffer->playing = false;
            buffer->paused = false;
            buffer->frameCursorPos = 0;
            buffer->framesProcessed = 0;
            buffer->isSubBufferProcessed[0] = true;
            buffer->isSubBufferProcessed[1] = true;
        }
    }
}
 
// Pause an audio buffer
void PauseAudioBuffer(AudioBuffer *buffer)
{
    if (buffer != NULL) buffer->paused = true;
}
 
// Resume an audio buffer
void ResumeAudioBuffer(AudioBuffer *buffer)
{
    if (buffer != NULL) buffer->paused = false;
}
 
// Set volume for an audio buffer
void SetAudioBufferVolume(AudioBuffer *buffer, float volume)
{
    if (buffer != NULL) buffer->volume = volume;
}
 
// Set pitch for an audio buffer
void SetAudioBufferPitch(AudioBuffer *buffer, float pitch)
{
    if ((buffer != NULL) && (pitch > 0.0f))
    {
        // 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;
    }
}
 
// Set pan for an audio buffer
void SetAudioBufferPan(AudioBuffer *buffer, float pan)
{
    if (pan < 0.0f) pan = 0.0f;
    else if (pan > 1.0f) pan = 1.0f;
 
    if (buffer != NULL) buffer->pan = pan;
}
 
// 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
    unsigned int fileSize = 0;
    unsigned char *fileData = LoadFileData(fileName, &fileSize);
 
    // Loading wave from memory data
    if (fileData != NULL) wave = LoadWaveFromMemory(GetFileExtension(fileName), fileData, fileSize);
 
    RL_FREE(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)
    {
        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(wave.frameCount*wave.channels*sizeof(short));
 
            // NOTE: We are forcing conversion to 16bit sample size on reading
            drwav_read_pcm_frames_s16(&wav, wav.totalPCMFrameCount, 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)
    {
        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_FLAC)
    else if (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
#if defined(SUPPORT_FILEFORMAT_MP3)
    else if (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
    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;
}
 
// 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;
}
 
// 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");
}
 
// Update sound buffer with new data
void UpdateSound(Sound sound, const void *data, int sampleCount)
{
    if (sound.stream.buffer != NULL)
    {
        StopAudioBuffer(sound.stream.buffer);
 
        // TODO: May want to lock/unlock this since this data buffer is read at mixing time
        memcpy(sound.stream.buffer->data, data, sampleCount*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
    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 6 char bytes for every byte: "0x00, "
    char *txtData = (char *)RL_CALLOC(waveDataSize*6 + 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) 2018-2022 Ramon Santamaria (@raysan5)                          //\n");
    byteCount += sprintf(txtData + byteCount, "//                                                                              //\n");
    byteCount += sprintf(txtData + byteCount, "//////////////////////////////////////////////////////////////////////////////////\n\n");
 
    char fileNameLower[256] = { 0 };
    char fileNameUpper[256] = { 0 };
    for (int i = 0; fileName[i] != '.'; i++) { fileNameLower[i] = fileName[i]; }      // Get filename without extension
    for (int i = 0; fileNameLower[i] != '\0'; i++) if (fileNameLower[i] >= 'a' && fileNameLower[i] <= 'z') { fileNameUpper[i] = fileNameLower[i] - 32; }
 
    byteCount += sprintf(txtData + byteCount, "// Wave data information\n");
    byteCount += sprintf(txtData + byteCount, "#define %s_FRAME_COUNT      %u\n", fileNameUpper, wave.frameCount);
    byteCount += sprintf(txtData + byteCount, "#define %s_SAMPLE_RATE      %u\n", fileNameUpper, wave.sampleRate);
    byteCount += sprintf(txtData + byteCount, "#define %s_SAMPLE_SIZE      %u\n", fileNameUpper, wave.sampleSize);
    byteCount += sprintf(txtData + byteCount, "#define %s_CHANNELS         %u\n\n", fileNameUpper, 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 %sData[%i] = {\n", fileNameLower, 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 %sData[%i] = { ", fileNameLower, 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);
}
 
// Play a sound in the multichannel buffer pool
void PlaySoundMulti(Sound sound)
{
    int index = -1;
    unsigned int oldAge = 0;
    int oldIndex = -1;
 
    // find the first non playing pool entry
    for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
    {
        if (AUDIO.MultiChannel.channels[i] > oldAge)
        {
            oldAge = AUDIO.MultiChannel.channels[i];
            oldIndex = i;
        }
 
        if (!IsAudioBufferPlaying(AUDIO.MultiChannel.pool[i]))
        {
            index = i;
            break;
        }
    }
 
    // If no none playing pool members can be index choose the oldest
    if (index == -1)
    {
        TRACELOG(LOG_WARNING, "SOUND: Buffer pool is already full, count: %i", AUDIO.MultiChannel.poolCounter);
 
        if (oldIndex == -1)
        {
            // Shouldn't be able to get here... but just in case something odd happens!
            TRACELOG(LOG_WARNING, "SOUND: Buffer pool could not determine oldest buffer not playing sound");
            return;
        }
 
        index = oldIndex;
 
        // Just in case...
        StopAudioBuffer(AUDIO.MultiChannel.pool[index]);
    }
 
    // Experimentally mutex lock doesn't seem to be needed this makes sense
    // as pool[index] isn't playing and the only stuff we're copying
    // shouldn't be changing...
 
    AUDIO.MultiChannel.channels[index] = AUDIO.MultiChannel.poolCounter;
    AUDIO.MultiChannel.poolCounter++;
 
    SetAudioBufferVolume(AUDIO.MultiChannel.pool[index], sound.stream.buffer->volume);
    SetAudioBufferPitch(AUDIO.MultiChannel.pool[index], sound.stream.buffer->pitch);
    SetAudioBufferPan(AUDIO.MultiChannel.pool[index], sound.stream.buffer->pan);
 
    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;       // Fill dummy track with data for playing
 
    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);
}
 
// 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)
    {
        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 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->frameCount*wave->channels)))
    {
        int sampleCount = finalSample - initSample;
 
        void *data = RL_MALLOC(sampleCount*wave->sampleSize/8);
 
        memcpy(data, (unsigned char *)wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->sampleSize/8);
 
        RL_FREE(wave->data);
        wave->data = data;
    }
    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] - 127)/256.0f;
        else if (wave.sampleSize == 16) samples[i] = (float)(((short *)wave.data)[i])/32767.0f;
        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 (.OGG)
//----------------------------------------------------------------------------------
 
// 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 = RL_CALLOC(1, sizeof(drwav));
        bool success = drwav_init_file(ctxWav, fileName, NULL);
 
        music.ctxType = MUSIC_AUDIO_WAV;
        music.ctxData = ctxWav;
 
        if (success)
        {
            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;
        }
    }
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
    else if (IsFileExtension(fileName, ".ogg"))
    {
        // Open ogg audio stream
        music.ctxType = MUSIC_AUDIO_OGG;
        music.ctxData = stb_vorbis_open_filename(fileName, NULL, NULL);
 
        if (music.ctxData != NULL)
        {
            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;
        }
    }
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
    else if (IsFileExtension(fileName, ".flac"))
    {
        music.ctxType = MUSIC_AUDIO_FLAC;
        music.ctxData = drflac_open_file(fileName, NULL);
 
        if (music.ctxData != NULL)
        {
            drflac *ctxFlac = (drflac *)music.ctxData;
 
            music.stream = LoadAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
            music.frameCount = (unsigned int)ctxFlac->totalPCMFrameCount;
            music.looping = true;   // Looping enabled by default
            musicLoaded = true;
        }
    }
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
    else if (IsFileExtension(fileName, ".mp3"))
    {
        drmp3 *ctxMp3 = RL_CALLOC(1, sizeof(drmp3));
        int result = drmp3_init_file(ctxMp3, fileName, NULL);
 
        music.ctxType = MUSIC_AUDIO_MP3;
        music.ctxData = ctxMp3;
 
        if (result > 0)
        {
            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;
        }
    }
#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);
 
        music.ctxType = MUSIC_MODULE_XM;
        music.ctxData = ctxXm;
 
        if (result == 0)    // XM AUDIO.System.context created successfully
        {
            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;
        }
    }
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
    else if (IsFileExtension(fileName, ".mod"))
    {
        jar_mod_context_t *ctxMod = RL_CALLOC(1, sizeof(jar_mod_context_t));
        jar_mod_init(ctxMod);
        int result = jar_mod_load_file(ctxMod, fileName);
 
        music.ctxType = MUSIC_MODULE_MOD;
        music.ctxData = ctxMod;
 
        if (result > 0)
        {
            // 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;
        }
    }
#endif
    else TRACELOG(LOG_WARNING, "STREAM: [%s] File format not supported", fileName);
 
    if (!musicLoaded)
    {
        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_FLAC)
        else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL);
    #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
 
        music.ctxData = NULL;
        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)
    {
        drwav *ctxWav = RL_CALLOC(1, sizeof(drwav));
 
        bool success = drwav_init_memory(ctxWav, (const void *)data, dataSize, NULL);
 
        music.ctxType = MUSIC_AUDIO_WAV;
        music.ctxData = ctxWav;
 
        if (success)
        {
            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;
        }
    }
#endif
#if defined(SUPPORT_FILEFORMAT_FLAC)
    else if (strcmp(fileType, ".flac") == 0)
    {
        music.ctxType = MUSIC_AUDIO_FLAC;
        music.ctxData = drflac_open_memory((const void*)data, dataSize, NULL);
 
        if (music.ctxData != NULL)
        {
            drflac *ctxFlac = (drflac *)music.ctxData;
 
            music.stream = LoadAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
            music.frameCount = (unsigned int)ctxFlac->totalPCMFrameCount;
            music.looping = true;   // Looping enabled by default
            musicLoaded = true;
        }
    }
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
    else if (strcmp(fileType, ".mp3") == 0)
    {
        drmp3 *ctxMp3 = RL_CALLOC(1, sizeof(drmp3));
        int success = drmp3_init_memory(ctxMp3, (const void*)data, dataSize, NULL);
 
        music.ctxType = MUSIC_AUDIO_MP3;
        music.ctxData = ctxMp3;
 
        if (success)
        {
            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;
        }
    }
#endif
#if defined(SUPPORT_FILEFORMAT_OGG)
    else if (strcmp(fileType, ".ogg") == 0)
    {
        // Open ogg audio stream
        music.ctxType = MUSIC_AUDIO_OGG;
        //music.ctxData = stb_vorbis_open_filename(fileName, NULL, NULL);
        music.ctxData = stb_vorbis_open_memory((const unsigned char *)data, dataSize, NULL, NULL);
 
        if (music.ctxData != NULL)
        {
            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;
        }
    }
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
    else if (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;
            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
 
            music.ctxData = ctxXm;
            musicLoaded = true;
        }
    }
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
    else if (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;
 
            // 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;
 
            music.ctxData = ctxMod;
            musicLoaded = true;
        }
    }
#endif
    else TRACELOG(LOG_WARNING, "STREAM: Data format not supported");
 
    if (!musicLoaded)
    {
        if (false) { }
    #if defined(SUPPORT_FILEFORMAT_WAV)
        else if (music.ctxType == MUSIC_AUDIO_WAV) drwav_uninit((drwav *)music.ctxData);
    #endif
    #if defined(SUPPORT_FILEFORMAT_FLAC)
        else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL);
    #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_OGG)
        else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)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
 
        music.ctxData = NULL;
        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;
}
 
// 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_FLAC)
        else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL);
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
    else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
#endif
#if defined(SUPPORT_FILEFORMAT_XM)
        else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData);
#endif
#if defined(SUPPORT_FILEFORMAT_MOD)
        else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); }
#endif
    }
}
 
// Start music playing (open stream)
void PlayMusicStream(Music music)
{
    if (music.stream.buffer != NULL)
    {
        // For music streams, we need to make sure we maintain the frame cursor position
        // This is a hack for this section of code in UpdateMusicStream()
        // NOTE: In case window is minimized, music stream is stopped, just make sure to
        // play again on window restore: if (IsMusicStreamPlaying(music)) PlayMusicStream(music);
        ma_uint32 frameCursorPos = music.stream.buffer->frameCursorPos;
        PlayAudioStream(music.stream);  // WARNING: This resets the cursor position.
        music.stream.buffer->frameCursorPos = frameCursorPos;
    }
}
 
// Pause music playing
void PauseMusicStream(Music music)
{
    PauseAudioStream(music.stream);
}
 
// Resume music playing
void ResumeMusicStream(Music music)
{
    ResumeAudioStream(music.stream);
}
 
// Stop music playing (close stream)
void StopMusicStream(Music music)
{
    StopAudioStream(music.stream);
 
    switch (music.ctxType)
    {
#if defined(SUPPORT_FILEFORMAT_WAV)
        case MUSIC_AUDIO_WAV: drwav_seek_to_pcm_frame((drwav *)music.ctxData, 0); 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_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;
    }
}
 
// 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_FLAC)
        case MUSIC_AUDIO_FLAC: drflac_seek_to_pcm_frame((drflac *)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
        default: break;
    }
 
    music.stream.buffer->framesProcessed = positionInFrames;
}
 
// Update (re-fill) music buffers if data already processed
void UpdateMusicStream(Music music)
{
    if (music.stream.buffer == NULL) return;
 
    bool streamEnding = false;
    unsigned int subBufferSizeInFrames = music.stream.buffer->sizeInFrames/2;
 
    // NOTE: Using dynamic allocation because it could require more than 16KB
    void *pcm = RL_CALLOC(subBufferSizeInFrames*music.stream.channels*music.stream.sampleSize/8, 1);
 
    int frameCountToStream = 0;    // Total size of data in frames to be streamed
 
    // TODO: Get the framesLeft using framesProcessed... 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;
    unsigned int framesLeft = music.frameCount - music.stream.buffer->framesProcessed;
 
    while (IsAudioStreamProcessed(music.stream))
    {
        if (framesLeft >= subBufferSizeInFrames) frameCountToStream = subBufferSizeInFrames;
        else frameCountToStream = framesLeft;
 
        switch (music.ctxType)
        {
        #if defined(SUPPORT_FILEFORMAT_WAV)
            case MUSIC_AUDIO_WAV:
            {
                // NOTE: Returns the number of samples to process (not required)
                if (music.stream.sampleSize == 16) drwav_read_pcm_frames_s16((drwav *)music.ctxData, frameCountToStream, (short *)pcm);
                else if (music.stream.sampleSize == 32) drwav_read_pcm_frames_f32((drwav *)music.ctxData, frameCountToStream, (float *)pcm);
 
            } break;
        #endif
        #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, frameCountToStream*music.stream.channels);
 
            } 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, frameCountToStream*music.stream.channels, (short *)pcm);
 
            } break;
        #endif
        #if defined(SUPPORT_FILEFORMAT_MP3)
            case MUSIC_AUDIO_MP3:
            {
                drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, frameCountToStream, (float *)pcm);
 
            } 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 *)pcm, frameCountToStream);
                else if (AUDIO_DEVICE_FORMAT == ma_format_s16) jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)pcm, frameCountToStream);
                else if (AUDIO_DEVICE_FORMAT == ma_format_u8) jar_xm_generate_samples_8bit((jar_xm_context_t *)music.ctxData, (char *)pcm, frameCountToStream);
 
            } 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, frameCountToStream, 0);
            } break;
        #endif
            default: break;
        }
 
        UpdateAudioStream(music.stream, pcm, frameCountToStream);
 
        framesLeft -= frameCountToStream;
 
        if (framesLeft <= 0)
        {
            streamEnding = true;
            break;
        }
    }
 
    // Free allocated pcm data
    RL_FREE(pcm);
 
    // Reset audio stream for looping
    if (streamEnding)
    {
        StopMusicStream(music);                     // Stop music (and reset)
        if (music.looping) PlayMusicStream(music);  // Play again
    }
    else
    {
        // NOTE: In case window is minimized, music stream is stopped,
        // just make sure to play again on window restore
        if (IsMusicStreamPlaying(music)) PlayMusicStream(music);
    }
}
 
// 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(music.ctxData, NULL, NULL, NULL, &framesPlayed);
            secondsPlayed = (float)framesPlayed/music.stream.sampleRate;
        }
        else
    #endif
        {
            //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
            unsigned int framesPlayed = music.stream.buffer->framesProcessed;
            secondsPlayed = (float)framesPlayed/music.stream.sampleRate;
        }
    }
 
    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
    unsigned int subBufferSize = (AUDIO.Buffer.defaultSize == 0)? AUDIO.System.device.sampleRate/30 : 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;
}
 
// 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: unqueue -> update -> queue
// NOTE 2: To unqueue a buffer it needs to be processed: IsAudioStreamProcessed()
void UpdateAudioStream(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);
 
            // TODO: Get total frames processed on this buffer... DOES NOT WORK.
            stream.buffer->framesProcessed += 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)frameCount)
            {
                ma_uint32 framesToWrite = subBufferSizeInFrames;
 
                if (framesToWrite > (ma_uint32)frameCount) 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");
    }
}
 
// Check if any audio stream buffers requires refill
bool IsAudioStreamProcessed(AudioStream stream)
{
    if (stream.buffer == NULL) return false;
 
    return (stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1]);
}
 
// Play audio stream
void PlayAudioStream(AudioStream stream)
{
    PlayAudioBuffer(stream.buffer);
}
 
// Play audio stream
void PauseAudioStream(AudioStream stream)
{
    PauseAudioBuffer(stream.buffer);
}
 
// Resume audio stream playing
void ResumeAudioStream(AudioStream stream)
{
    ResumeAudioBuffer(stream.buffer);
}
 
// Check if audio stream is playing.
bool IsAudioStreamPlaying(AudioStream stream)
{
    return IsAudioBufferPlaying(stream.buffer);
}
 
// Stop audio stream
void StopAudioStream(AudioStream stream)
{
    StopAudioBuffer(stream.buffer);
}
 
// Set volume for audio stream (1.0 is max level)
void SetAudioStreamVolume(AudioStream stream, float volume)
{
    SetAudioBufferVolume(stream.buffer, volume);
}
 
// Set pitch for audio stream (1.0 is base level)
void SetAudioStreamPitch(AudioStream stream, float pitch)
{
    SetAudioBufferPitch(stream.buffer, pitch);
}
 
// 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) stream.buffer->callback = callback;
}
 
// 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);
}
 
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);
}
 
//----------------------------------------------------------------------------------
// Module specific 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 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] = { 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;
 
        ma_result result = ma_data_converter_get_required_input_frame_count(&audioBuffer->converter, outputFramesToProcessThisIteration, &inputFramesToProcessThisIteration);
        if (inputFramesToProcessThisIteration > inputBufferFrameCap)
        {
            inputFramesToProcessThisIteration = inputBufferFrameCap;
        }
 
        float *runningFramesOut = framesOut + (totalOutputFramesProcessed*audioBuffer->converter.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
// 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)
                        {
                            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);
}
 
// 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 left = buffer->pan;
        const float right = 1.0f - left;
 
        // 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);
            }
        }
    }
}
 
// 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;
}
 
// Load data from file into a buffer
static unsigned char *LoadFileData(const char *fileName, unsigned int *bytesRead)
{
    unsigned char *data = NULL;
    *bytesRead = 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);
                *bytesRead = 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, unsigned int bytesToWrite)
{
    if (fileName != NULL)
    {
        FILE *file = fopen(fileName, "wb");
 
        if (file != NULL)
        {
            unsigned int count = (unsigned int)fwrite(data, sizeof(unsigned char), bytesToWrite, file);
 
            if (count == 0) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to write file", fileName);
            else if (count != bytesToWrite) 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);
    }
    else TRACELOG(LOG_WARNING, "FILEIO: File name provided is not valid");
}
 
// 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 = 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);
    }
    else TRACELOG(LOG_WARNING, "FILEIO: File name provided is not valid");
}
#endif
 
#undef AudioBuffer
 
#endif      // SUPPORT_MODULE_RAUDIO