Changed the template to now download raylib instead of having it in the repo.

1 files changed, 0 insertions, 2138 deletions

diff --git a/libs/raylib/src/raudio.c b/libs/raylib/src/raudio.c
deleted file mode 100644
index 6313b16..0000000
--- a/libs/raylib/src/raudio.c
+++ /dev/null
@@ -1,2138 +0,0 @@
-/**********************************************************************************************
-*
-*   raudio - A simple and easy-to-use audio library based on miniaudio
-*
-*   FEATURES:
-*       - Manage audio device (init/close)
-*       - Manage raw audio context
-*       - Manage mixing channels
-*       - Load and unload audio files
-*       - Format wave data (sample rate, size, channels)
-*       - Play/Stop/Pause/Resume loaded audio
-*
-*   CONFIGURATION:
-*
-*   #define RAUDIO_STANDALONE
-*       Define to use the module as standalone library (independently of raylib).
-*       Required types and functions are defined in the same module.
-*
-*   #define SUPPORT_FILEFORMAT_WAV
-*   #define SUPPORT_FILEFORMAT_OGG
-*   #define SUPPORT_FILEFORMAT_XM
-*   #define SUPPORT_FILEFORMAT_MOD
-*   #define SUPPORT_FILEFORMAT_FLAC
-*   #define SUPPORT_FILEFORMAT_MP3
-*       Selected desired fileformats to be supported for loading. Some of those formats are
-*       supported by default, to remove support, just comment unrequired #define in this module
-*
-*   DEPENDENCIES:
-*       miniaudio.h  - Audio device management lib (https://github.com/dr-soft/miniaudio)
-*       stb_vorbis.h - Ogg audio files loading (http://www.nothings.org/stb_vorbis/)
-*       dr_mp3.h     - MP3 audio file loading (https://github.com/mackron/dr_libs)
-*       dr_flac.h    - FLAC audio file loading (https://github.com/mackron/dr_libs)
-*       jar_xm.h     - XM module file loading
-*       jar_mod.h    - MOD audio file loading
-*
-*   CONTRIBUTORS:
-*       David Reid (github: @mackron) (Nov. 2017):
-*           - Complete port to miniaudio library
-*
-*       Joshua Reisenauer (github: @kd7tck) (2015)
-*           - XM audio module support (jar_xm)
-*           - MOD audio module support (jar_mod)
-*           - Mixing channels support
-*           - Raw audio context support
-*
-*
-*   LICENSE: zlib/libpng
-*
-*   Copyright (c) 2013-2020 Ramon Santamaria (@raysan5)
-*
-*   This software is provided "as-is", without any express or implied warranty. In no event
-*   will the authors be held liable for any damages arising from the use of this software.
-*
-*   Permission is granted to anyone to use this software for any purpose, including commercial
-*   applications, and to alter it and redistribute it freely, subject to the following restrictions:
-*
-*     1. The origin of this software must not be misrepresented; you must not claim that you
-*     wrote the original software. If you use this software in a product, an acknowledgment
-*     in the product documentation would be appreciated but is not required.
-*
-*     2. Altered source versions must be plainly marked as such, and must not be misrepresented
-*     as being the original software.
-*
-*     3. This notice may not be removed or altered from any source distribution.
-*
-**********************************************************************************************/
-
-#if defined(RAUDIO_STANDALONE)
-    #include "raudio.h"
-    #include <stdarg.h>         // Required for: va_list, va_start(), vfprintf(), va_end()
-#else
-    #include "raylib.h"         // Declares module functions
-
-// Check if config flags have been externally provided on compilation line
-#if !defined(EXTERNAL_CONFIG_FLAGS)
-    #include "config.h"         // Defines module configuration flags
-#endif
-    #include "utils.h"          // Required for: fopen() Android mapping
-#endif
-
-#if defined(_WIN32)
-// To avoid conflicting windows.h symbols with raylib, some flags are defined
-// WARNING: Those flags avoid inclusion of some Win32 headers that could be required
-// by user at some point and won't be included...
-//-------------------------------------------------------------------------------------
-
-// If defined, the following flags inhibit definition of the indicated items.
-#define NOGDICAPMASKS     // CC_*, LC_*, PC_*, CP_*, TC_*, RC_
-#define NOVIRTUALKEYCODES // VK_*
-#define NOWINMESSAGES     // WM_*, EM_*, LB_*, CB_*
-#define NOWINSTYLES       // WS_*, CS_*, ES_*, LBS_*, SBS_*, CBS_*
-#define NOSYSMETRICS      // SM_*
-#define NOMENUS           // MF_*
-#define NOICONS           // IDI_*
-#define NOKEYSTATES       // MK_*
-#define NOSYSCOMMANDS     // SC_*
-#define NORASTEROPS       // Binary and Tertiary raster ops
-#define NOSHOWWINDOW      // SW_*
-#define OEMRESOURCE       // OEM Resource values
-#define NOATOM            // Atom Manager routines
-#define NOCLIPBOARD       // Clipboard routines
-#define NOCOLOR           // Screen colors
-#define NOCTLMGR          // Control and Dialog routines
-#define NODRAWTEXT        // DrawText() and DT_*
-#define NOGDI             // All GDI defines and routines
-#define NOKERNEL          // All KERNEL defines and routines
-#define NOUSER            // All USER defines and routines
-//#define NONLS             // All NLS defines and routines
-#define NOMB              // MB_* and MessageBox()
-#define NOMEMMGR          // GMEM_*, LMEM_*, GHND, LHND, associated routines
-#define NOMETAFILE        // typedef METAFILEPICT
-#define NOMINMAX          // Macros min(a,b) and max(a,b)
-#define NOMSG             // typedef MSG and associated routines
-#define NOOPENFILE        // OpenFile(), OemToAnsi, AnsiToOem, and OF_*
-#define NOSCROLL          // SB_* and scrolling routines
-#define NOSERVICE         // All Service Controller routines, SERVICE_ equates, etc.
-#define NOSOUND           // Sound driver routines
-#define NOTEXTMETRIC      // typedef TEXTMETRIC and associated routines
-#define NOWH              // SetWindowsHook and WH_*
-#define NOWINOFFSETS      // GWL_*, GCL_*, associated routines
-#define NOCOMM            // COMM driver routines
-#define NOKANJI           // Kanji support stuff.
-#define NOHELP            // Help engine interface.
-#define NOPROFILER        // Profiler interface.
-#define NODEFERWINDOWPOS  // DeferWindowPos routines
-#define NOMCX             // Modem Configuration Extensions
-
-// Type required before windows.h inclusion
-typedef struct tagMSG *LPMSG;
-
-#include <windows.h>
-
-// Type required by some unused function...
-typedef struct tagBITMAPINFOHEADER {
-  DWORD biSize;
-  LONG  biWidth;
-  LONG  biHeight;
-  WORD  biPlanes;
-  WORD  biBitCount;
-  DWORD biCompression;
-  DWORD biSizeImage;
-  LONG  biXPelsPerMeter;
-  LONG  biYPelsPerMeter;
-  DWORD biClrUsed;
-  DWORD biClrImportant;
-} BITMAPINFOHEADER, *PBITMAPINFOHEADER;
-
-#include <objbase.h>
-#include <mmreg.h>
-#include <mmsystem.h>
-
-// Some required types defined for MSVC/TinyC compiler
-#if defined(_MSC_VER) || defined(__TINYC__)
-    #include "propidl.h"
-#endif
-#endif
-
-#define MA_MALLOC RL_MALLOC
-#define MA_FREE RL_FREE
-
-#define MA_NO_JACK
-#define MINIAUDIO_IMPLEMENTATION
-#include "external/miniaudio.h"         // miniaudio library
-#undef PlaySound                        // Win32 API: windows.h > mmsystem.h defines PlaySound macro
-
-#include <stdlib.h>                     // Required for: malloc(), free()
-#include <stdio.h>                      // Required for: FILE, fopen(), fclose(), fread()
-
-#if defined(RAUDIO_STANDALONE)
-    #include <string.h>                 // Required for: strcmp() [Used in IsFileExtension()]
-
-    #if !defined(TRACELOG)
-        #define TRACELOG(level, ...) (void)0
-    #endif
-#endif
-
-#if defined(SUPPORT_FILEFORMAT_OGG)
-    // TODO: Remap malloc()/free() calls to RL_MALLOC/RL_FREE
-
-    #define STB_VORBIS_IMPLEMENTATION
-    #include "external/stb_vorbis.h"    // OGG loading functions
-#endif
-
-#if defined(SUPPORT_FILEFORMAT_XM)
-    #define JARXM_MALLOC RL_MALLOC
-    #define JARXM_FREE RL_FREE
-
-    #define JAR_XM_IMPLEMENTATION
-    #include "external/jar_xm.h"        // XM loading functions
-#endif
-
-#if defined(SUPPORT_FILEFORMAT_MOD)
-    #define JARMOD_MALLOC RL_MALLOC
-    #define JARMOD_FREE RL_FREE
-
-    #define JAR_MOD_IMPLEMENTATION
-    #include "external/jar_mod.h"       // MOD loading functions
-#endif
-
-#if defined(SUPPORT_FILEFORMAT_FLAC)
-    #define DRFLAC_MALLOC RL_MALLOC
-    #define DRFLAC_REALLOC RL_REALLOC
-    #define DRFLAC_FREE RL_FREE
-
-    #define DR_FLAC_IMPLEMENTATION
-    #define DR_FLAC_NO_WIN32_IO
-    #include "external/dr_flac.h"       // FLAC loading functions
-#endif
-
-#if defined(SUPPORT_FILEFORMAT_MP3)
-    #define DRMP3_MALLOC RL_MALLOC
-    #define DRMP3_REALLOC RL_REALLOC
-    #define DRMP3_FREE RL_FREE
-
-    #define DR_MP3_IMPLEMENTATION
-    #include "external/dr_mp3.h"        // MP3 loading functions
-#endif
-
-#if defined(_MSC_VER)
-    #undef bool
-#endif
-
-//----------------------------------------------------------------------------------
-// Defines and Macros
-//----------------------------------------------------------------------------------
-#define AUDIO_DEVICE_FORMAT         ma_format_f32
-#define AUDIO_DEVICE_CHANNELS       2
-#define AUDIO_DEVICE_SAMPLE_RATE    44100
-
-#define MAX_AUDIO_BUFFER_POOL_CHANNELS 16
-
-//----------------------------------------------------------------------------------
-// Types and Structures Definition
-//----------------------------------------------------------------------------------
-
-// Music context type
-// NOTE: Depends on data structure provided by the library
-// in charge of reading the different file types
-typedef enum {
-    MUSIC_AUDIO_WAV = 0,
-    MUSIC_AUDIO_OGG,
-    MUSIC_AUDIO_FLAC,
-    MUSIC_AUDIO_MP3,
-    MUSIC_MODULE_XM,
-    MUSIC_MODULE_MOD
-} MusicContextType;
-
-#if defined(RAUDIO_STANDALONE)
-typedef enum {
-    LOG_ALL,
-    LOG_TRACE,
-    LOG_DEBUG,
-    LOG_INFO,
-    LOG_WARNING,
-    LOG_ERROR,
-    LOG_FATAL,
-    LOG_NONE
-} TraceLogType;
-#endif
-
-// NOTE: Different logic is used when feeding data to the playback device
-// depending on whether or not data is streamed (Music vs Sound)
-typedef enum {
-    AUDIO_BUFFER_USAGE_STATIC = 0,
-    AUDIO_BUFFER_USAGE_STREAM
-} AudioBufferUsage;
-
-// Audio buffer structure
-struct rAudioBuffer {
-    ma_data_converter converter;    // Audio data converter
-
-    float volume;                   // Audio buffer volume
-    float pitch;                    // Audio buffer pitch
-
-    bool playing;                   // Audio buffer state: AUDIO_PLAYING
-    bool paused;                    // Audio buffer state: AUDIO_PAUSED
-    bool looping;                   // Audio buffer looping, always true for AudioStreams
-    int usage;                      // Audio buffer usage mode: STATIC or STREAM
-
-    bool isSubBufferProcessed[2];   // SubBuffer processed (virtual double buffer)
-    unsigned int sizeInFrames;      // Total buffer size in frames
-    unsigned int frameCursorPos;    // Frame cursor position
-    unsigned int totalFramesProcessed;  // Total frames processed in this buffer (required for play timing)
-
-    unsigned char *data;            // Data buffer, on music stream keeps filling
-
-    rAudioBuffer *next;             // Next audio buffer on the list
-    rAudioBuffer *prev;             // Previous audio buffer on the list
-};
-
-#define AudioBuffer rAudioBuffer    // HACK: To avoid CoreAudio (macOS) symbol collision
-
-// Audio data context
-typedef struct AudioData {
-    struct {
-        ma_context context;         // miniaudio context data
-        ma_device device;           // miniaudio device
-        ma_mutex lock;              // miniaudio mutex lock
-        bool isReady;               // Check if audio device is ready
-    } System;
-    struct {
-        AudioBuffer *first;         // Pointer to first AudioBuffer in the list
-        AudioBuffer *last;          // Pointer to last AudioBuffer in the list
-        int defaultSize;            // Default audio buffer size for audio streams
-    } Buffer;
-    struct {
-        AudioBuffer *pool[MAX_AUDIO_BUFFER_POOL_CHANNELS];      // Multichannel AudioBuffer pointers pool
-        unsigned int poolCounter;                               // AudioBuffer pointers pool counter
-        unsigned int channels[MAX_AUDIO_BUFFER_POOL_CHANNELS];  // AudioBuffer pool channels
-    } MultiChannel;
-} AudioData;
-
-//----------------------------------------------------------------------------------
-// Global Variables Definition
-//----------------------------------------------------------------------------------
-static AudioData AUDIO = {          // Global AUDIO context
-
-    // NOTE: Music buffer size is defined by number of samples, independent of sample size and channels number
-    // After some math, considering a sampleRate of 48000, a buffer refill rate of 1/60 seconds and a
-    // standard double-buffering system, a 4096 samples buffer has been chosen, it should be enough
-    // In case of music-stalls, just increase this number
-    .Buffer.defaultSize = 4096
-};
-
-//----------------------------------------------------------------------------------
-// Module specific Functions Declaration
-//----------------------------------------------------------------------------------
-static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message);
-static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount);
-static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume);
-
-static void InitAudioBufferPool(void);                  // Initialise the multichannel buffer pool
-static void CloseAudioBufferPool(void);                 // Close the audio buffers pool
-
-#if defined(SUPPORT_FILEFORMAT_WAV)
-static Wave LoadWAV(const char *fileName);              // Load WAV file
-static int SaveWAV(Wave wave, const char *fileName);    // Save wave data as WAV file
-#endif
-#if defined(SUPPORT_FILEFORMAT_OGG)
-static Wave LoadOGG(const char *fileName);              // Load OGG file
-#endif
-#if defined(SUPPORT_FILEFORMAT_FLAC)
-static Wave LoadFLAC(const char *fileName);             // Load FLAC file
-#endif
-#if defined(SUPPORT_FILEFORMAT_MP3)
-static Wave LoadMP3(const char *fileName);              // Load MP3 file
-#endif
-
-#if defined(RAUDIO_STANDALONE)
-bool IsFileExtension(const char *fileName, const char *ext);// Check file extension
-void TraceLog(int msgType, const char *text, ...);      // Show trace log messages (LOG_INFO, LOG_WARNING, LOG_ERROR, LOG_DEBUG)
-#endif
-
-//----------------------------------------------------------------------------------
-// AudioBuffer management functions declaration
-// NOTE: Those functions are not exposed by raylib... for the moment
-//----------------------------------------------------------------------------------
-AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage);
-void UnloadAudioBuffer(AudioBuffer *buffer);
-
-bool IsAudioBufferPlaying(AudioBuffer *buffer);
-void PlayAudioBuffer(AudioBuffer *buffer);
-void StopAudioBuffer(AudioBuffer *buffer);
-void PauseAudioBuffer(AudioBuffer *buffer);
-void ResumeAudioBuffer(AudioBuffer *buffer);
-void SetAudioBufferVolume(AudioBuffer *buffer, float volume);
-void SetAudioBufferPitch(AudioBuffer *buffer, float pitch);
-void TrackAudioBuffer(AudioBuffer *buffer);
-void UntrackAudioBuffer(AudioBuffer *buffer);
-
-//----------------------------------------------------------------------------------
-// Module Functions Definition - Audio Device initialization and Closing
-//----------------------------------------------------------------------------------
-// Initialize audio device
-void InitAudioDevice(void)
-{
-    // TODO: Load AUDIO context memory dynamically?
-
-    // Init audio context
-    ma_context_config ctxConfig = ma_context_config_init();
-    ctxConfig.logCallback = OnLog;
-
-    ma_result result = ma_context_init(NULL, 0, &ctxConfig, &AUDIO.System.context);
-    if (result != MA_SUCCESS)
-    {
-        TRACELOG(LOG_ERROR, "AUDIO: Failed to initialize context");
-        return;
-    }
-
-    // Init audio device
-    // NOTE: Using the default device. Format is floating point because it simplifies mixing.
-    ma_device_config config = ma_device_config_init(ma_device_type_playback);
-    config.playback.pDeviceID = NULL;  // NULL for the default playback AUDIO.System.device.
-    config.playback.format    = AUDIO_DEVICE_FORMAT;
-    config.playback.channels  = AUDIO_DEVICE_CHANNELS;
-    config.capture.pDeviceID  = NULL;  // NULL for the default capture AUDIO.System.device.
-    config.capture.format     = ma_format_s16;
-    config.capture.channels   = 1;
-    config.sampleRate         = AUDIO_DEVICE_SAMPLE_RATE;
-    config.dataCallback       = OnSendAudioDataToDevice;
-    config.pUserData          = NULL;
-
-    result = ma_device_init(&AUDIO.System.context, &config, &AUDIO.System.device);
-    if (result != MA_SUCCESS)
-    {
-        TRACELOG(LOG_ERROR, "AUDIO: Failed to initialize playback device");
-        ma_context_uninit(&AUDIO.System.context);
-        return;
-    }
-
-    // Keep the device running the whole time. May want to consider doing something a bit smarter and only have the device running
-    // while there's at least one sound being played.
-    result = ma_device_start(&AUDIO.System.device);
-    if (result != MA_SUCCESS)
-    {
-        TRACELOG(LOG_ERROR, "AUDIO: Failed to start playback device");
-        ma_device_uninit(&AUDIO.System.device);
-        ma_context_uninit(&AUDIO.System.context);
-        return;
-    }
-
-    // Mixing happens on a seperate thread which means we need to synchronize. I'm using a mutex here to make things simple, but may
-    // want to look at something a bit smarter later on to keep everything real-time, if that's necessary.
-    if (ma_mutex_init(&AUDIO.System.context, &AUDIO.System.lock) != MA_SUCCESS)
-    {
-        TRACELOG(LOG_ERROR, "AUDIO: Failed to create mutex for mixing");
-        ma_device_uninit(&AUDIO.System.device);
-        ma_context_uninit(&AUDIO.System.context);
-        return;
-    }
-
-    TRACELOG(LOG_INFO, "AUDIO: Device initialized successfully");
-    TRACELOG(LOG_INFO, "    > Backend:      miniaudio / %s", ma_get_backend_name(AUDIO.System.context.backend));
-    TRACELOG(LOG_INFO, "    > Format:       %s -> %s", ma_get_format_name(AUDIO.System.device.playback.format), ma_get_format_name(AUDIO.System.device.playback.internalFormat));
-    TRACELOG(LOG_INFO, "    > Channels:     %d -> %d", AUDIO.System.device.playback.channels, AUDIO.System.device.playback.internalChannels);
-    TRACELOG(LOG_INFO, "    > Sample rate:  %d -> %d", AUDIO.System.device.sampleRate, AUDIO.System.device.playback.internalSampleRate);
-    TRACELOG(LOG_INFO, "    > Periods size: %d", AUDIO.System.device.playback.internalPeriodSizeInFrames*AUDIO.System.device.playback.internalPeriods);
-
-    InitAudioBufferPool();
-
-    AUDIO.System.isReady = true;
-}
-
-// Close the audio device for all contexts
-void CloseAudioDevice(void)
-{
-    if (AUDIO.System.isReady)
-    {
-        ma_mutex_uninit(&AUDIO.System.lock);
-        ma_device_uninit(&AUDIO.System.device);
-        ma_context_uninit(&AUDIO.System.context);
-
-        CloseAudioBufferPool();
-
-        TRACELOG(LOG_INFO, "AUDIO: Device closed successfully");
-    }
-    else TRACELOG(LOG_WARNING, "AUDIO: Device could not be closed, not currently initialized");
-}
-
-// Check if device has been initialized successfully
-bool IsAudioDeviceReady(void)
-{
-    return AUDIO.System.isReady;
-}
-
-// Set master volume (listener)
-void SetMasterVolume(float volume)
-{
-    ma_device_set_master_volume(&AUDIO.System.device, volume);
-}
-
-//----------------------------------------------------------------------------------
-// Module Functions Definition - Audio Buffer management
-//----------------------------------------------------------------------------------
-
-// Initialize a new audio buffer (filled with silence)
-AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage)
-{
-    AudioBuffer *audioBuffer = (AudioBuffer *)RL_CALLOC(1, sizeof(AudioBuffer));
-
-    if (audioBuffer == NULL)
-    {
-        TRACELOG(LOG_ERROR, "AUDIO: Failed to allocate memory for buffer");
-        return NULL;
-    }
-
-    audioBuffer->data = RL_CALLOC(sizeInFrames*channels*ma_get_bytes_per_sample(format), 1);
-
-    // Audio data runs through a format converter
-    ma_data_converter_config converterConfig = ma_data_converter_config_init(format, AUDIO_DEVICE_FORMAT, channels, AUDIO_DEVICE_CHANNELS, sampleRate, AUDIO_DEVICE_SAMPLE_RATE);
-    converterConfig.resampling.allowDynamicSampleRate = true;        // Required for pitch shifting
-
-    ma_result result = ma_data_converter_init(&converterConfig, &audioBuffer->converter);
-
-    if (result != MA_SUCCESS)
-    {
-        TRACELOG(LOG_ERROR, "AUDIO: Failed to create data conversion pipeline");
-        RL_FREE(audioBuffer);
-        return NULL;
-    }
-
-    // Init audio buffer values
-    audioBuffer->volume = 1.0f;
-    audioBuffer->pitch = 1.0f;
-    audioBuffer->playing = false;
-    audioBuffer->paused = false;
-    audioBuffer->looping = false;
-    audioBuffer->usage = usage;
-    audioBuffer->frameCursorPos = 0;
-    audioBuffer->sizeInFrames = sizeInFrames;
-
-    // Buffers should be marked as processed by default so that a call to
-    // UpdateAudioStream() immediately after initialization works correctly
-    audioBuffer->isSubBufferProcessed[0] = true;
-    audioBuffer->isSubBufferProcessed[1] = true;
-
-    // Track audio buffer to linked list next position
-    TrackAudioBuffer(audioBuffer);
-
-    return audioBuffer;
-}
-
-// Delete an audio buffer
-void UnloadAudioBuffer(AudioBuffer *buffer)
-{
-    if (buffer != NULL)
-    {
-        ma_data_converter_uninit(&buffer->converter);
-        UntrackAudioBuffer(buffer);
-        RL_FREE(buffer->data);
-        RL_FREE(buffer);
-    }
-}
-
-// Check if an audio buffer is playing
-bool IsAudioBufferPlaying(AudioBuffer *buffer)
-{
-    bool result = false;
-
-    if (buffer != NULL) result = (buffer->playing && !buffer->paused);
-
-    return result;
-}
-
-// Play an audio buffer
-// NOTE: Buffer is restarted to the start.
-// Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position should be maintained.
-void PlayAudioBuffer(AudioBuffer *buffer)
-{
-    if (buffer != NULL)
-    {
-        buffer->playing = true;
-        buffer->paused = false;
-        buffer->frameCursorPos = 0;
-    }
-}
-
-// Stop an audio buffer
-void StopAudioBuffer(AudioBuffer *buffer)
-{
-    if (buffer != NULL)
-    {
-        if (IsAudioBufferPlaying(buffer))
-        {
-            buffer->playing = false;
-            buffer->paused = false;
-            buffer->frameCursorPos = 0;
-            buffer->totalFramesProcessed = 0;
-            buffer->isSubBufferProcessed[0] = true;
-            buffer->isSubBufferProcessed[1] = true;
-        }
-    }
-}
-
-// Pause an audio buffer
-void PauseAudioBuffer(AudioBuffer *buffer)
-{
-    if (buffer != NULL) buffer->paused = true;
-}
-
-// Resume an audio buffer
-void ResumeAudioBuffer(AudioBuffer *buffer)
-{
-    if (buffer != NULL) buffer->paused = false;
-}
-
-// Set volume for an audio buffer
-void SetAudioBufferVolume(AudioBuffer *buffer, float volume)
-{
-    if (buffer != NULL) buffer->volume = volume;
-}
-
-// Set pitch for an audio buffer
-void SetAudioBufferPitch(AudioBuffer *buffer, float pitch)
-{
-    if (buffer != NULL)
-    {
-        float pitchMul = pitch/buffer->pitch;
-
-        // Pitching is just an adjustment of the sample rate.
-        // Note that this changes the duration of the sound:
-        //  - higher pitches will make the sound faster
-        //  - lower pitches make it slower
-        ma_uint32 newOutputSampleRate = (ma_uint32)((float)buffer->converter.config.sampleRateOut/pitchMul);
-        buffer->pitch *= (float)buffer->converter.config.sampleRateOut/newOutputSampleRate;
-
-        ma_data_converter_set_rate(&buffer->converter, buffer->converter.config.sampleRateIn, newOutputSampleRate);
-    }
-}
-
-// Track audio buffer to linked list next position
-void TrackAudioBuffer(AudioBuffer *buffer)
-{
-    ma_mutex_lock(&AUDIO.System.lock);
-    {
-        if (AUDIO.Buffer.first == NULL) AUDIO.Buffer.first = buffer;
-        else
-        {
-            AUDIO.Buffer.last->next = buffer;
-            buffer->prev = AUDIO.Buffer.last;
-        }
-
-        AUDIO.Buffer.last = buffer;
-    }
-    ma_mutex_unlock(&AUDIO.System.lock);
-}
-
-// Untrack audio buffer from linked list
-void UntrackAudioBuffer(AudioBuffer *buffer)
-{
-    ma_mutex_lock(&AUDIO.System.lock);
-    {
-        if (buffer->prev == NULL) AUDIO.Buffer.first = buffer->next;
-        else buffer->prev->next = buffer->next;
-
-        if (buffer->next == NULL) AUDIO.Buffer.last = buffer->prev;
-        else buffer->next->prev = buffer->prev;
-
-        buffer->prev = NULL;
-        buffer->next = NULL;
-    }
-    ma_mutex_unlock(&AUDIO.System.lock);
-}
-
-//----------------------------------------------------------------------------------
-// Module Functions Definition - Sounds loading and playing (.WAV)
-//----------------------------------------------------------------------------------
-
-// Load wave data from file
-Wave LoadWave(const char *fileName)
-{
-    Wave wave = { 0 };
-
-    if (false) { }
-#if defined(SUPPORT_FILEFORMAT_WAV)
-    else if (IsFileExtension(fileName, ".wav")) wave = LoadWAV(fileName);
-#endif
-#if defined(SUPPORT_FILEFORMAT_OGG)
-    else if (IsFileExtension(fileName, ".ogg")) wave = LoadOGG(fileName);
-#endif
-#if defined(SUPPORT_FILEFORMAT_FLAC)
-    else if (IsFileExtension(fileName, ".flac")) wave = LoadFLAC(fileName);
-#endif
-#if defined(SUPPORT_FILEFORMAT_MP3)
-    else if (IsFileExtension(fileName, ".mp3")) wave = LoadMP3(fileName);
-#endif
-    else TRACELOG(LOG_WARNING, "FILEIO: [%s] File format not supported", fileName);
-
-    return wave;
-}
-
-// Load sound from file
-// NOTE: The entire file is loaded to memory to be played (no-streaming)
-Sound LoadSound(const char *fileName)
-{
-    Wave wave = LoadWave(fileName);
-
-    Sound sound = LoadSoundFromWave(wave);
-
-    UnloadWave(wave);       // Sound is loaded, we can unload wave
-
-    return sound;
-}
-
-// Load sound from wave data
-// NOTE: Wave data must be unallocated manually
-Sound LoadSoundFromWave(Wave wave)
-{
-    Sound sound = { 0 };
-
-    if (wave.data != NULL)
-    {
-        // When using miniaudio we need to do our own mixing.
-        // To simplify this we need convert the format of each sound to be consistent with
-        // the format used to open the playback AUDIO.System.device. We can do this two ways:
-        //
-        //   1) Convert the whole sound in one go at load time (here).
-        //   2) Convert the audio data in chunks at mixing time.
-        //
-        // First option has been selected, format conversion is done on the loading stage.
-        // The downside is that it uses more memory if the original sound is u8 or s16.
-        ma_format formatIn  = ((wave.sampleSize == 8)? ma_format_u8 : ((wave.sampleSize == 16)? ma_format_s16 : ma_format_f32));
-        ma_uint32 frameCountIn = wave.sampleCount/wave.channels;
-
-        ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, NULL, frameCountIn, formatIn, wave.channels, wave.sampleRate);
-        if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed to get frame count for format conversion");
-
-        AudioBuffer *audioBuffer = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, frameCount, AUDIO_BUFFER_USAGE_STATIC);
-        if (audioBuffer == NULL) TRACELOG(LOG_WARNING, "SOUND: Failed to create buffer");
-
-        frameCount = (ma_uint32)ma_convert_frames(audioBuffer->data, frameCount, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, wave.data, frameCountIn, formatIn, wave.channels, wave.sampleRate);
-        if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed format conversion");
-
-        sound.sampleCount = frameCount*AUDIO_DEVICE_CHANNELS;
-        sound.stream.sampleRate = AUDIO_DEVICE_SAMPLE_RATE;
-        sound.stream.sampleSize = 32;
-        sound.stream.channels = AUDIO_DEVICE_CHANNELS;
-        sound.stream.buffer = audioBuffer;
-    }
-
-    return sound;
-}
-
-// Unload wave data
-void UnloadWave(Wave wave)
-{
-    if (wave.data != NULL) RL_FREE(wave.data);
-
-    TRACELOG(LOG_INFO, "WAVE: Unloaded wave data from RAM");
-}
-
-// Unload sound
-void UnloadSound(Sound sound)
-{
-    UnloadAudioBuffer(sound.stream.buffer);
-
-    TRACELOG(LOG_INFO, "WAVE: Unloaded sound data from RAM");
-}
-
-// Update sound buffer with new data
-void UpdateSound(Sound sound, const void *data, int samplesCount)
-{
-    if (sound.stream.buffer != NULL)
-    {
-        StopAudioBuffer(sound.stream.buffer);
-
-        // TODO: May want to lock/unlock this since this data buffer is read at mixing time
-        memcpy(sound.stream.buffer->data, data, samplesCount*ma_get_bytes_per_frame(sound.stream.buffer->converter.config.formatIn, sound.stream.buffer->converter.config.channelsIn));
-    }
-}
-
-// Export wave data to file
-void ExportWave(Wave wave, const char *fileName)
-{
-    bool success = false;
-
-    if (false) { }
-#if defined(SUPPORT_FILEFORMAT_WAV)
-    else if (IsFileExtension(fileName, ".wav")) success = SaveWAV(wave, fileName);
-#endif
-    else if (IsFileExtension(fileName, ".raw"))
-    {
-        // Export raw sample data (without header)
-        // NOTE: It's up to the user to track wave parameters
-        SaveFileData(fileName, wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8);
-        success = true;
-    }
-
-    if (success) TRACELOG(LOG_INFO, "FILEIO: [%s] Wave data exported successfully", fileName);
-    else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to export wave data", fileName);
-}
-
-// Export wave sample data to code (.h)
-void ExportWaveAsCode(Wave wave, const char *fileName)
-{
-    #define BYTES_TEXT_PER_LINE     20
-
-    char varFileName[256] = { 0 };
-    int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8;
-
-    FILE *txtFile = fopen(fileName, "wt");
-
-    if (txtFile != NULL)
-    {
-        fprintf(txtFile, "\n//////////////////////////////////////////////////////////////////////////////////\n");
-        fprintf(txtFile, "//                                                                              //\n");
-        fprintf(txtFile, "// WaveAsCode exporter v1.0 - Wave data exported as an array of bytes           //\n");
-        fprintf(txtFile, "//                                                                              //\n");
-        fprintf(txtFile, "// more info and bugs-report:  github.com/raysan5/raylib                        //\n");
-        fprintf(txtFile, "// feedback and support:       ray[at]raylib.com                                //\n");
-        fprintf(txtFile, "//                                                                              //\n");
-        fprintf(txtFile, "// Copyright (c) 2018 Ramon Santamaria (@raysan5)                               //\n");
-        fprintf(txtFile, "//                                                                              //\n");
-        fprintf(txtFile, "//////////////////////////////////////////////////////////////////////////////////\n\n");
-
-#if !defined(RAUDIO_STANDALONE)
-        // Get file name from path and convert variable name to uppercase
-        strcpy(varFileName, GetFileNameWithoutExt(fileName));
-        for (int i = 0; varFileName[i] != '\0'; i++) if (varFileName[i] >= 'a' && varFileName[i] <= 'z') { varFileName[i] = varFileName[i] - 32; }
-#else
-        strcpy(varFileName, fileName);
-#endif
-
-        fprintf(txtFile, "// Wave data information\n");
-        fprintf(txtFile, "#define %s_SAMPLE_COUNT     %u\n", varFileName, wave.sampleCount);
-        fprintf(txtFile, "#define %s_SAMPLE_RATE      %u\n", varFileName, wave.sampleRate);
-        fprintf(txtFile, "#define %s_SAMPLE_SIZE      %u\n", varFileName, wave.sampleSize);
-        fprintf(txtFile, "#define %s_CHANNELS         %u\n\n", varFileName, wave.channels);
-
-        // Write byte data as hexadecimal text
-        fprintf(txtFile, "static unsigned char %s_DATA[%i] = { ", varFileName, dataSize);
-        for (int i = 0; i < dataSize - 1; i++) fprintf(txtFile, ((i%BYTES_TEXT_PER_LINE == 0)? "0x%x,\n" : "0x%x, "), ((unsigned char *)wave.data)[i]);
-        fprintf(txtFile, "0x%x };\n", ((unsigned char *)wave.data)[dataSize - 1]);
-
-        fclose(txtFile);
-    }
-}
-
-// Play a sound
-void PlaySound(Sound sound)
-{
-    PlayAudioBuffer(sound.stream.buffer);
-}
-
-// Play a sound in the multichannel buffer pool
-void PlaySoundMulti(Sound sound)
-{
-    int index = -1;
-    unsigned int oldAge = 0;
-    int oldIndex = -1;
-
-    // find the first non playing pool entry
-    for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
-    {
-        if (AUDIO.MultiChannel.channels[i] > oldAge)
-        {
-            oldAge = AUDIO.MultiChannel.channels[i];
-            oldIndex = i;
-        }
-
-        if (!IsAudioBufferPlaying(AUDIO.MultiChannel.pool[i]))
-        {
-            index = i;
-            break;
-        }
-    }
-
-    // If no none playing pool members can be index choose the oldest
-    if (index == -1)
-    {
-        TRACELOG(LOG_WARNING, "SOUND: Buffer pool is already full, count: %i", AUDIO.MultiChannel.poolCounter);
-
-        if (oldIndex == -1)
-        {
-            // Shouldn't be able to get here... but just in case something odd happens!
-            TRACELOG(LOG_WARNING, "SOUND: Buffer pool could not determine oldest buffer not playing sound");
-            return;
-        }
-
-        index = oldIndex;
-
-        // Just in case...
-        StopAudioBuffer(AUDIO.MultiChannel.pool[index]);
-    }
-
-    // Experimentally mutex lock doesn't seem to be needed this makes sense
-    // as pool[index] isn't playing and the only stuff we're copying
-    // shouldn't be changing...
-
-    AUDIO.MultiChannel.channels[index] = AUDIO.MultiChannel.poolCounter;
-    AUDIO.MultiChannel.poolCounter++;
-
-    AUDIO.MultiChannel.pool[index]->volume = sound.stream.buffer->volume;
-    AUDIO.MultiChannel.pool[index]->pitch = sound.stream.buffer->pitch;
-    AUDIO.MultiChannel.pool[index]->looping = sound.stream.buffer->looping;
-    AUDIO.MultiChannel.pool[index]->usage = sound.stream.buffer->usage;
-    AUDIO.MultiChannel.pool[index]->isSubBufferProcessed[0] = false;
-    AUDIO.MultiChannel.pool[index]->isSubBufferProcessed[1] = false;
-    AUDIO.MultiChannel.pool[index]->sizeInFrames = sound.stream.buffer->sizeInFrames;
-    AUDIO.MultiChannel.pool[index]->data = sound.stream.buffer->data;
-
-    PlayAudioBuffer(AUDIO.MultiChannel.pool[index]);
-}
-
-// Stop any sound played with PlaySoundMulti()
-void StopSoundMulti(void)
-{
-    for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) StopAudioBuffer(AUDIO.MultiChannel.pool[i]);
-}
-
-// Get number of sounds playing in the multichannel buffer pool
-int GetSoundsPlaying(void)
-{
-    int counter = 0;
-
-    for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
-    {
-        if (IsAudioBufferPlaying(AUDIO.MultiChannel.pool[i])) counter++;
-    }
-
-    return counter;
-}
-
-// Pause a sound
-void PauseSound(Sound sound)
-{
-    PauseAudioBuffer(sound.stream.buffer);
-}
-
-// Resume a paused sound
-void ResumeSound(Sound sound)
-{
-    ResumeAudioBuffer(sound.stream.buffer);
-}
-
-// Stop reproducing a sound
-void StopSound(Sound sound)
-{
-    StopAudioBuffer(sound.stream.buffer);
-}
-
-// Check if a sound is playing
-bool IsSoundPlaying(Sound sound)
-{
-    return IsAudioBufferPlaying(sound.stream.buffer);
-}
-
-// Set volume for a sound
-void SetSoundVolume(Sound sound, float volume)
-{
-    SetAudioBufferVolume(sound.stream.buffer, volume);
-}
-
-// Set pitch for a sound
-void SetSoundPitch(Sound sound, float pitch)
-{
-    SetAudioBufferPitch(sound.stream.buffer, pitch);
-}
-
-// Convert wave data to desired format
-void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
-{
-    ma_format formatIn  = ((wave->sampleSize == 8)? ma_format_u8 : ((wave->sampleSize == 16)? ma_format_s16 : ma_format_f32));
-    ma_format formatOut = ((      sampleSize == 8)? ma_format_u8 : ((      sampleSize == 16)? ma_format_s16 : ma_format_f32));
-
-    ma_uint32 frameCountIn = wave->sampleCount;  // Is wave->sampleCount actually the frame count? That terminology needs to change, if so.
-
-    ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, formatOut, channels, sampleRate, NULL, frameCountIn, formatIn, wave->channels, wave->sampleRate);
-    if (frameCount == 0)
-    {
-        TRACELOG(LOG_WARNING, "WAVE: Failed to get frame count for format conversion");
-        return;
-    }
-
-    void *data = RL_MALLOC(frameCount*channels*(sampleSize/8));
-
-    frameCount = (ma_uint32)ma_convert_frames(data, frameCount, formatOut, channels, sampleRate, wave->data, frameCountIn, formatIn, wave->channels, wave->sampleRate);
-    if (frameCount == 0)
-    {
-        TRACELOG(LOG_WARNING, "WAVE: Failed format conversion");
-        return;
-    }
-
-    wave->sampleCount = frameCount;
-    wave->sampleSize = sampleSize;
-    wave->sampleRate = sampleRate;
-    wave->channels = channels;
-    RL_FREE(wave->data);
-    wave->data = data;
-}
-
-// Copy a wave to a new wave
-Wave WaveCopy(Wave wave)
-{
-    Wave newWave = { 0 };
-
-    newWave.data = RL_MALLOC(wave.sampleCount*wave.sampleSize/8*wave.channels);
-
-    if (newWave.data != NULL)
-    {
-        // NOTE: Size must be provided in bytes
-        memcpy(newWave.data, wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8);
-
-        newWave.sampleCount = wave.sampleCount;
-        newWave.sampleRate = wave.sampleRate;
-        newWave.sampleSize = wave.sampleSize;
-        newWave.channels = wave.channels;
-    }
-
-    return newWave;
-}
-
-// Crop a wave to defined samples range
-// NOTE: Security check in case of out-of-range
-void WaveCrop(Wave *wave, int initSample, int finalSample)
-{
-    if ((initSample >= 0) && (initSample < finalSample) &&
-        (finalSample > 0) && ((unsigned int)finalSample < wave->sampleCount))
-    {
-        int sampleCount = finalSample - initSample;
-
-        void *data = RL_MALLOC(sampleCount*wave->sampleSize/8*wave->channels);
-
-        memcpy(data, (unsigned char *)wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->channels*wave->sampleSize/8);
-
-        RL_FREE(wave->data);
-        wave->data = data;
-    }
-    else TRACELOG(LOG_WARNING, "WAVE: Crop range out of bounds");
-}
-
-// Get samples data from wave as a floats array
-// NOTE: Returned sample values are normalized to range [-1..1]
-float *GetWaveData(Wave wave)
-{
-    float *samples = (float *)RL_MALLOC(wave.sampleCount*wave.channels*sizeof(float));
-
-    for (unsigned int i = 0; i < wave.sampleCount; i++)
-    {
-        for (unsigned int j = 0; j < wave.channels; j++)
-        {
-            if (wave.sampleSize == 8) samples[wave.channels*i + j] = (float)(((unsigned char *)wave.data)[wave.channels*i + j] - 127)/256.0f;
-            else if (wave.sampleSize == 16) samples[wave.channels*i + j] = (float)((short *)wave.data)[wave.channels*i + j]/32767.0f;
-            else if (wave.sampleSize == 32) samples[wave.channels*i + j] = ((float *)wave.data)[wave.channels*i + j];
-        }
-    }
-
-    return samples;
-}
-
-//----------------------------------------------------------------------------------
-// Module Functions Definition - Music loading and stream playing (.OGG)
-//----------------------------------------------------------------------------------
-
-// Load music stream from file
-Music LoadMusicStream(const char *fileName)
-{
-    Music music = { 0 };
-    bool musicLoaded = false;
-
-    if (false) { }
-#if defined(SUPPORT_FILEFORMAT_OGG)
-    else if (IsFileExtension(fileName, ".ogg"))
-    {
-        // Open ogg audio stream
-        music.ctxData = stb_vorbis_open_filename(fileName, NULL, NULL);
-
-        if (music.ctxData != NULL)
-        {
-            music.ctxType = MUSIC_AUDIO_OGG;
-            stb_vorbis_info info = stb_vorbis_get_info((stb_vorbis *)music.ctxData);  // Get Ogg file info
-
-            // OGG bit rate defaults to 16 bit, it's enough for compressed format
-            music.stream = InitAudioStream(info.sample_rate, 16, info.channels);
-            music.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData)*info.channels;
-            music.loopCount = 0;   // Infinite loop by default
-            musicLoaded = true;
-        }
-    }
-#endif
-#if defined(SUPPORT_FILEFORMAT_FLAC)
-    else if (IsFileExtension(fileName, ".flac"))
-    {
-        music.ctxData = drflac_open_file(fileName);
-
-        if (music.ctxData != NULL)
-        {
-            music.ctxType = MUSIC_AUDIO_FLAC;
-            drflac *ctxFlac = (drflac *)music.ctxData;
-
-            music.stream = InitAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
-            music.sampleCount = (unsigned int)ctxFlac->totalSampleCount;
-            music.loopCount = 0;   // Infinite loop by default
-            musicLoaded = true;
-        }
-    }
-#endif
-#if defined(SUPPORT_FILEFORMAT_MP3)
-    else if (IsFileExtension(fileName, ".mp3"))
-    {
-        drmp3 *ctxMp3 = RL_MALLOC(sizeof(drmp3));
-        music.ctxData = ctxMp3;
-
-        int result = drmp3_init_file(ctxMp3, fileName, NULL);
-
-        if (result > 0)
-        {
-            music.ctxType = MUSIC_AUDIO_MP3;
-
-            music.stream = InitAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
-            music.sampleCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels;
-            music.loopCount = 0;   // Infinite loop by default
-            musicLoaded = true;
-        }
-    }
-#endif
-#if defined(SUPPORT_FILEFORMAT_XM)
-    else if (IsFileExtension(fileName, ".xm"))
-    {
-        jar_xm_context_t *ctxXm = NULL;
-
-        int result = jar_xm_create_context_from_file(&ctxXm, 48000, fileName);
-
-        if (result == 0)    // XM AUDIO.System.context created successfully
-        {
-            music.ctxType = MUSIC_MODULE_XM;
-            jar_xm_set_max_loop_count(ctxXm, 0);    // Set infinite number of loops
-
-            // NOTE: Only stereo is supported for XM
-            music.stream = InitAudioStream(48000, 16, 2);
-            music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)*2;
-            music.loopCount = 0;   // Infinite loop by default
-            jar_xm_reset(ctxXm);   // make sure we start at the beginning of the song
-            musicLoaded = true;
-
-            music.ctxData = ctxXm;
-        }
-    }
-#endif
-#if defined(SUPPORT_FILEFORMAT_MOD)
-    else if (IsFileExtension(fileName, ".mod"))
-    {
-        jar_mod_context_t *ctxMod = RL_MALLOC(sizeof(jar_mod_context_t));
-
-        jar_mod_init(ctxMod);
-        int result = jar_mod_load_file(ctxMod, fileName);
-
-        if (result > 0)
-        {
-            music.ctxType = MUSIC_MODULE_MOD;
-
-            // NOTE: Only stereo is supported for MOD
-            music.stream = InitAudioStream(48000, 16, 2);
-            music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)*2;
-            music.loopCount = 0;   // Infinite loop by default
-            musicLoaded = true;
-
-            music.ctxData = ctxMod;
-        }
-    }
-#endif
-
-    if (!musicLoaded)
-    {
-        if (false) { }
-    #if defined(SUPPORT_FILEFORMAT_OGG)
-        else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData);
-    #endif
-    #if defined(SUPPORT_FILEFORMAT_FLAC)
-        else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData);
-    #endif
-    #if defined(SUPPORT_FILEFORMAT_MP3)
-        else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
-    #endif
-    #if defined(SUPPORT_FILEFORMAT_XM)
-        else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData);
-    #endif
-    #if defined(SUPPORT_FILEFORMAT_MOD)
-        else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); }
-    #endif
-
-        TRACELOG(LOG_WARNING, "FILEIO: [%s] Music file could not be opened", fileName);
-    }
-    else
-    {
-        // Show some music stream info
-        TRACELOG(LOG_INFO, "FILEIO: [%s] Music file successfully loaded:", fileName);
-        TRACELOG(LOG_INFO, "    > Total samples: %i", music.sampleCount);
-        TRACELOG(LOG_INFO, "    > Sample rate:   %i Hz", music.stream.sampleRate);
-        TRACELOG(LOG_INFO, "    > Sample size:   %i bits", music.stream.sampleSize);
-        TRACELOG(LOG_INFO, "    > Channels:      %i (%s)", music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi");
-    }
-
-    return music;
-}
-
-// Unload music stream
-void UnloadMusicStream(Music music)
-{
-    CloseAudioStream(music.stream);
-
-    if (false) { }
-#if defined(SUPPORT_FILEFORMAT_OGG)
-    else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData);
-#endif
-#if defined(SUPPORT_FILEFORMAT_FLAC)
-    else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData);
-#endif
-#if defined(SUPPORT_FILEFORMAT_MP3)
-    else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
-#endif
-#if defined(SUPPORT_FILEFORMAT_XM)
-    else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData);
-#endif
-#if defined(SUPPORT_FILEFORMAT_MOD)
-    else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); }
-#endif
-}
-
-// Start music playing (open stream)
-void PlayMusicStream(Music music)
-{
-    if (music.stream.buffer != NULL)
-    {
-        // For music streams, we need to make sure we maintain the frame cursor position
-        // This is a hack for this section of code in UpdateMusicStream()
-        // NOTE: In case window is minimized, music stream is stopped, just make sure to
-        // play again on window restore: if (IsMusicPlaying(music)) PlayMusicStream(music);
-        ma_uint32 frameCursorPos = music.stream.buffer->frameCursorPos;
-        PlayAudioStream(music.stream);  // WARNING: This resets the cursor position.
-        music.stream.buffer->frameCursorPos = frameCursorPos;
-    }
-}
-
-// Pause music playing
-void PauseMusicStream(Music music)
-{
-    PauseAudioStream(music.stream);
-}
-
-// Resume music playing
-void ResumeMusicStream(Music music)
-{
-    ResumeAudioStream(music.stream);
-}
-
-// Stop music playing (close stream)
-void StopMusicStream(Music music)
-{
-    StopAudioStream(music.stream);
-
-    switch (music.ctxType)
-    {
-#if defined(SUPPORT_FILEFORMAT_OGG)
-        case MUSIC_AUDIO_OGG: stb_vorbis_seek_start((stb_vorbis *)music.ctxData); break;
-#endif
-#if defined(SUPPORT_FILEFORMAT_FLAC)
-        case MUSIC_AUDIO_FLAC: drflac_seek_to_pcm_frame((drflac *)music.ctxData, 0); break;
-#endif
-#if defined(SUPPORT_FILEFORMAT_MP3)
-        case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame((drmp3 *)music.ctxData, 0); break;
-#endif
-#if defined(SUPPORT_FILEFORMAT_XM)
-        case MUSIC_MODULE_XM: jar_xm_reset((jar_xm_context_t *)music.ctxData); break;
-#endif
-#if defined(SUPPORT_FILEFORMAT_MOD)
-        case MUSIC_MODULE_MOD: jar_mod_seek_start((jar_mod_context_t *)music.ctxData); break;
-#endif
-        default: break;
-    }
-}
-
-// Update (re-fill) music buffers if data already processed
-void UpdateMusicStream(Music music)
-{
-    if (music.stream.buffer == NULL) return;
-
-    bool streamEnding = false;
-
-    unsigned int subBufferSizeInFrames = music.stream.buffer->sizeInFrames/2;
-
-    // NOTE: Using dynamic allocation because it could require more than 16KB
-    void *pcm = RL_CALLOC(subBufferSizeInFrames*music.stream.channels*music.stream.sampleSize/8, 1);
-
-    int samplesCount = 0;    // Total size of data streamed in L+R samples for xm floats, individual L or R for ogg shorts
-
-    // TODO: Get the sampleLeft using totalFramesProcessed... but first, get total frames processed correctly...
-    //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
-    int sampleLeft = music.sampleCount - (music.stream.buffer->totalFramesProcessed*music.stream.channels);
-
-    while (IsAudioStreamProcessed(music.stream))
-    {
-        if ((sampleLeft/music.stream.channels) >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames*music.stream.channels;
-        else samplesCount = sampleLeft;
-
-        switch (music.ctxType)
-        {
-        #if defined(SUPPORT_FILEFORMAT_OGG)
-            case MUSIC_AUDIO_OGG:
-            {
-                // NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
-                stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)pcm, samplesCount);
-
-            } break;
-        #endif
-        #if defined(SUPPORT_FILEFORMAT_FLAC)
-            case MUSIC_AUDIO_FLAC:
-            {
-                // NOTE: Returns the number of samples to process (not required)
-                drflac_read_pcm_frames_s16((drflac *)music.ctxData, samplesCount, (short *)pcm);
-
-            } break;
-        #endif
-        #if defined(SUPPORT_FILEFORMAT_MP3)
-            case MUSIC_AUDIO_MP3:
-            {
-                // NOTE: samplesCount, actually refers to framesCount and returns the number of frames processed
-                drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, samplesCount/music.stream.channels, (float *)pcm);
-
-            } break;
-        #endif
-        #if defined(SUPPORT_FILEFORMAT_XM)
-            case MUSIC_MODULE_XM:
-            {
-                // NOTE: Internally this function considers 2 channels generation, so samplesCount/2
-                jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)pcm, samplesCount/2);
-            } break;
-        #endif
-        #if defined(SUPPORT_FILEFORMAT_MOD)
-            case MUSIC_MODULE_MOD:
-            {
-                // NOTE: 3rd parameter (nbsample) specify the number of stereo 16bits samples you want, so sampleCount/2
-                jar_mod_fillbuffer((jar_mod_context_t *)music.ctxData, (short *)pcm, samplesCount/2, 0);
-            } break;
-        #endif
-            default: break;
-        }
-
-        UpdateAudioStream(music.stream, pcm, samplesCount);
-
-        if ((music.ctxType == MUSIC_MODULE_XM) || (music.ctxType == MUSIC_MODULE_MOD))
-        {
-            if (samplesCount > 1) sampleLeft -= samplesCount/2;
-            else sampleLeft -= samplesCount;
-        }
-        else sampleLeft -= samplesCount;
-
-        if (sampleLeft <= 0)
-        {
-            streamEnding = true;
-            break;
-        }
-    }
-
-    // Free allocated pcm data
-    RL_FREE(pcm);
-
-    // Reset audio stream for looping
-    if (streamEnding)
-    {
-        StopMusicStream(music);        // Stop music (and reset)
-
-        // Decrease loopCount to stop when required
-        if (music.loopCount > 1)
-        {
-            music.loopCount--;         // Decrease loop count
-            PlayMusicStream(music);    // Play again
-        }
-        else if (music.loopCount == 0) PlayMusicStream(music);
-    }
-    else
-    {
-        // NOTE: In case window is minimized, music stream is stopped,
-        // just make sure to play again on window restore
-        if (IsMusicPlaying(music)) PlayMusicStream(music);
-    }
-}
-
-// Check if any music is playing
-bool IsMusicPlaying(Music music)
-{
-    return IsAudioStreamPlaying(music.stream);
-}
-
-// Set volume for music
-void SetMusicVolume(Music music, float volume)
-{
-    SetAudioStreamVolume(music.stream, volume);
-}
-
-// Set pitch for music
-void SetMusicPitch(Music music, float pitch)
-{
-    SetAudioStreamPitch(music.stream, pitch);
-}
-
-// Set music loop count (loop repeats)
-// NOTE: If set to 0, means infinite loop
-void SetMusicLoopCount(Music music, int count)
-{
-    music.loopCount = count;
-}
-
-// Get music time length (in seconds)
-float GetMusicTimeLength(Music music)
-{
-    float totalSeconds = 0.0f;
-
-    totalSeconds = (float)music.sampleCount/(music.stream.sampleRate*music.stream.channels);
-
-    return totalSeconds;
-}
-
-// Get current music time played (in seconds)
-float GetMusicTimePlayed(Music music)
-{
-    float secondsPlayed = 0.0f;
-
-    if (music.stream.buffer != NULL)
-    {
-        //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
-        unsigned int samplesPlayed = music.stream.buffer->totalFramesProcessed*music.stream.channels;
-        secondsPlayed = (float)samplesPlayed / (music.stream.sampleRate*music.stream.channels);
-    }
-
-    return secondsPlayed;
-}
-
-// Init audio stream (to stream audio pcm data)
-AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels)
-{
-    AudioStream stream = { 0 };
-
-    stream.sampleRate = sampleRate;
-    stream.sampleSize = sampleSize;
-    stream.channels = channels;
-
-    ma_format formatIn = ((stream.sampleSize == 8)? ma_format_u8 : ((stream.sampleSize == 16)? ma_format_s16 : ma_format_f32));
-
-    // The size of a streaming buffer must be at least double the size of a period
-    unsigned int periodSize = AUDIO.System.device.playback.internalPeriodSizeInFrames;
-    unsigned int subBufferSize = AUDIO.Buffer.defaultSize;     // Default buffer size (audio stream)
-
-    if (subBufferSize < periodSize) subBufferSize = periodSize;
-
-    // Create a double audio buffer of defined size
-    stream.buffer = LoadAudioBuffer(formatIn, stream.channels, stream.sampleRate, subBufferSize*2, AUDIO_BUFFER_USAGE_STREAM);
-
-    if (stream.buffer != NULL)
-    {
-        stream.buffer->looping = true;    // Always loop for streaming buffers
-        TRACELOG(LOG_INFO, "STREAM: Initialized successfully (%i Hz, %i bit, %s)", stream.sampleRate, stream.sampleSize, (stream.channels == 1)? "Mono" : "Stereo");
-    }
-    else TRACELOG(LOG_WARNING, "STREAM: Failed to load audio buffer, stream could not be created");
-
-    return stream;
-}
-
-// Close audio stream and free memory
-void CloseAudioStream(AudioStream stream)
-{
-    UnloadAudioBuffer(stream.buffer);
-
-    TRACELOG(LOG_INFO, "STREAM: Unloaded audio stream data from RAM");
-}
-
-// Update audio stream buffers with data
-// NOTE 1: Only updates one buffer of the stream source: unqueue -> update -> queue
-// NOTE 2: To unqueue a buffer it needs to be processed: IsAudioStreamProcessed()
-void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
-{
-    if (stream.buffer != NULL)
-    {
-        if (stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1])
-        {
-            ma_uint32 subBufferToUpdate = 0;
-
-            if (stream.buffer->isSubBufferProcessed[0] && stream.buffer->isSubBufferProcessed[1])
-            {
-                // Both buffers are available for updating.
-                // Update the first one and make sure the cursor is moved back to the front.
-                subBufferToUpdate = 0;
-                stream.buffer->frameCursorPos = 0;
-            }
-            else
-            {
-                // Just update whichever sub-buffer is processed.
-                subBufferToUpdate = (stream.buffer->isSubBufferProcessed[0])? 0 : 1;
-            }
-
-            ma_uint32 subBufferSizeInFrames = stream.buffer->sizeInFrames/2;
-            unsigned char *subBuffer = stream.buffer->data + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate);
-
-            // TODO: Get total frames processed on this buffer... DOES NOT WORK.
-            stream.buffer->totalFramesProcessed += subBufferSizeInFrames;
-
-            // Does this API expect a whole buffer to be updated in one go?
-            // Assuming so, but if not will need to change this logic.
-            if (subBufferSizeInFrames >= (ma_uint32)samplesCount/stream.channels)
-            {
-                ma_uint32 framesToWrite = subBufferSizeInFrames;
-
-                if (framesToWrite > ((ma_uint32)samplesCount/stream.channels)) framesToWrite = (ma_uint32)samplesCount/stream.channels;
-
-                ma_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8);
-                memcpy(subBuffer, data, bytesToWrite);
-
-                // Any leftover frames should be filled with zeros.
-                ma_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite;
-
-                if (leftoverFrameCount > 0) memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8));
-
-                stream.buffer->isSubBufferProcessed[subBufferToUpdate] = false;
-            }
-            else TRACELOG(LOG_WARNING, "STREAM: Attempting to write too many frames to buffer");
-        }
-        else TRACELOG(LOG_WARNING, "STREAM: Buffer not available for updating");
-    }
-}
-
-// Check if any audio stream buffers requires refill
-bool IsAudioStreamProcessed(AudioStream stream)
-{
-    if (stream.buffer == NULL) return false;
-
-    return (stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1]);
-}
-
-// Play audio stream
-void PlayAudioStream(AudioStream stream)
-{
-    PlayAudioBuffer(stream.buffer);
-}
-
-// Play audio stream
-void PauseAudioStream(AudioStream stream)
-{
-    PauseAudioBuffer(stream.buffer);
-}
-
-// Resume audio stream playing
-void ResumeAudioStream(AudioStream stream)
-{
-    ResumeAudioBuffer(stream.buffer);
-}
-
-// Check if audio stream is playing.
-bool IsAudioStreamPlaying(AudioStream stream)
-{
-    return IsAudioBufferPlaying(stream.buffer);
-}
-
-// Stop audio stream
-void StopAudioStream(AudioStream stream)
-{
-    StopAudioBuffer(stream.buffer);
-}
-
-// Set volume for audio stream (1.0 is max level)
-void SetAudioStreamVolume(AudioStream stream, float volume)
-{
-    SetAudioBufferVolume(stream.buffer, volume);
-}
-
-// Set pitch for audio stream (1.0 is base level)
-void SetAudioStreamPitch(AudioStream stream, float pitch)
-{
-    SetAudioBufferPitch(stream.buffer, pitch);
-}
-
-// Default size for new audio streams
-void SetAudioStreamBufferSizeDefault(int size)
-{
-    AUDIO.Buffer.defaultSize = size;
-}
-
-//----------------------------------------------------------------------------------
-// Module specific Functions Definition
-//----------------------------------------------------------------------------------
-
-// Log callback function
-static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message)
-{
-    (void)pContext;
-    (void)pDevice;
-
-    TRACELOG(LOG_ERROR, "miniaudio: %s", message);   // All log messages from miniaudio are errors
-}
-
-// Reads audio data from an AudioBuffer object in internal format.
-static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer, void *framesOut, ma_uint32 frameCount)
-{
-    ma_uint32 subBufferSizeInFrames = (audioBuffer->sizeInFrames > 1)? audioBuffer->sizeInFrames/2 : audioBuffer->sizeInFrames;
-    ma_uint32 currentSubBufferIndex = audioBuffer->frameCursorPos/subBufferSizeInFrames;
-
-    if (currentSubBufferIndex > 1) return 0;
-
-    // Another thread can update the processed state of buffers so
-    // we just take a copy here to try and avoid potential synchronization problems
-    bool isSubBufferProcessed[2];
-    isSubBufferProcessed[0] = audioBuffer->isSubBufferProcessed[0];
-    isSubBufferProcessed[1] = audioBuffer->isSubBufferProcessed[1];
-
-    ma_uint32 frameSizeInBytes = ma_get_bytes_per_frame(audioBuffer->converter.config.formatIn, audioBuffer->converter.config.channelsIn);
-
-    // Fill out every frame until we find a buffer that's marked as processed. Then fill the remainder with 0
-    ma_uint32 framesRead = 0;
-    while (1)
-    {
-        // We break from this loop differently depending on the buffer's usage
-        //  - For static buffers, we simply fill as much data as we can
-        //  - For streaming buffers we only fill the halves of the buffer that are processed
-        //    Unprocessed halves must keep their audio data in-tact
-        if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
-        {
-            if (framesRead >= frameCount) break;
-        }
-        else
-        {
-            if (isSubBufferProcessed[currentSubBufferIndex]) break;
-        }
-
-        ma_uint32 totalFramesRemaining = (frameCount - framesRead);
-        if (totalFramesRemaining == 0) break;
-
-        ma_uint32 framesRemainingInOutputBuffer;
-        if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
-        {
-            framesRemainingInOutputBuffer = audioBuffer->sizeInFrames - audioBuffer->frameCursorPos;
-        }
-        else
-        {
-            ma_uint32 firstFrameIndexOfThisSubBuffer = subBufferSizeInFrames*currentSubBufferIndex;
-            framesRemainingInOutputBuffer = subBufferSizeInFrames - (audioBuffer->frameCursorPos - firstFrameIndexOfThisSubBuffer);
-        }
-
-        ma_uint32 framesToRead = totalFramesRemaining;
-        if (framesToRead > framesRemainingInOutputBuffer) framesToRead = framesRemainingInOutputBuffer;
-
-        memcpy((unsigned char *)framesOut + (framesRead*frameSizeInBytes), audioBuffer->data + (audioBuffer->frameCursorPos*frameSizeInBytes), framesToRead*frameSizeInBytes);
-        audioBuffer->frameCursorPos = (audioBuffer->frameCursorPos + framesToRead)%audioBuffer->sizeInFrames;
-        framesRead += framesToRead;
-
-        // If we've read to the end of the buffer, mark it as processed
-        if (framesToRead == framesRemainingInOutputBuffer)
-        {
-            audioBuffer->isSubBufferProcessed[currentSubBufferIndex] = true;
-            isSubBufferProcessed[currentSubBufferIndex] = true;
-
-            currentSubBufferIndex = (currentSubBufferIndex + 1)%2;
-
-            // We need to break from this loop if we're not looping
-            if (!audioBuffer->looping)
-            {
-                StopAudioBuffer(audioBuffer);
-                break;
-            }
-        }
-    }
-
-    // Zero-fill excess
-    ma_uint32 totalFramesRemaining = (frameCount - framesRead);
-    if (totalFramesRemaining > 0)
-    {
-        memset((unsigned char *)framesOut + (framesRead*frameSizeInBytes), 0, totalFramesRemaining*frameSizeInBytes);
-
-        // For static buffers we can fill the remaining frames with silence for safety, but we don't want
-        // to report those frames as "read". The reason for this is that the caller uses the return value
-        // to know whether or not a non-looping sound has finished playback.
-        if (audioBuffer->usage != AUDIO_BUFFER_USAGE_STATIC) framesRead += totalFramesRemaining;
-    }
-
-    return framesRead;
-}
-
-// Reads audio data from an AudioBuffer object in device format. Returned data will be in a format appropriate for mixing.
-static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, float *framesOut, ma_uint32 frameCount)
-{
-    // What's going on here is that we're continuously converting data from the AudioBuffer's internal format to the mixing format, which
-    // should be defined by the output format of the data converter. We do this until frameCount frames have been output. The important
-    // detail to remember here is that we never, ever attempt to read more input data than is required for the specified number of output
-    // frames. This can be achieved with ma_data_converter_get_required_input_frame_count().
-    ma_uint8 inputBuffer[4096];
-    ma_uint32 inputBufferFrameCap = sizeof(inputBuffer) / ma_get_bytes_per_frame(audioBuffer->converter.config.formatIn, audioBuffer->converter.config.channelsIn);
-
-    ma_uint32 totalOutputFramesProcessed = 0;
-    while (totalOutputFramesProcessed < frameCount)
-    {
-        ma_uint64 outputFramesToProcessThisIteration = frameCount - totalOutputFramesProcessed;
-
-        ma_uint64 inputFramesToProcessThisIteration = ma_data_converter_get_required_input_frame_count(&audioBuffer->converter, outputFramesToProcessThisIteration);
-        if (inputFramesToProcessThisIteration > inputBufferFrameCap)
-        {
-            inputFramesToProcessThisIteration = inputBufferFrameCap;
-        }
-
-        float *runningFramesOut = framesOut + (totalOutputFramesProcessed * audioBuffer->converter.config.channelsOut);
-
-        /* At this point we can convert the data to our mixing format. */
-        ma_uint64 inputFramesProcessedThisIteration = ReadAudioBufferFramesInInternalFormat(audioBuffer, inputBuffer, (ma_uint32)inputFramesToProcessThisIteration);    /* Safe cast. */
-        ma_uint64 outputFramesProcessedThisIteration = outputFramesToProcessThisIteration;
-        ma_data_converter_process_pcm_frames(&audioBuffer->converter, inputBuffer, &inputFramesProcessedThisIteration, runningFramesOut, &outputFramesProcessedThisIteration);
-
-        totalOutputFramesProcessed += (ma_uint32)outputFramesProcessedThisIteration; /* Safe cast. */
-
-        if (inputFramesProcessedThisIteration < inputFramesToProcessThisIteration)
-        {
-            break;  /* Ran out of input data. */
-        }
-
-        /* This should never be hit, but will add it here for safety. Ensures we get out of the loop when no input nor output frames are processed. */
-        if (inputFramesProcessedThisIteration == 0 && outputFramesProcessedThisIteration == 0)
-        {
-            break;
-        }
-    }
-
-    return totalOutputFramesProcessed;
-}
-
-
-// Sending audio data to device callback function
-// NOTE: All the mixing takes place here
-static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount)
-{
-    (void)pDevice;
-
-    // Mixing is basically just an accumulation, we need to initialize the output buffer to 0
-    memset(pFramesOut, 0, frameCount*pDevice->playback.channels*ma_get_bytes_per_sample(pDevice->playback.format));
-
-    // Using a mutex here for thread-safety which makes things not real-time
-    // This is unlikely to be necessary for this project, but may want to consider how you might want to avoid this
-    ma_mutex_lock(&AUDIO.System.lock);
-    {
-        for (AudioBuffer *audioBuffer = AUDIO.Buffer.first; audioBuffer != NULL; audioBuffer = audioBuffer->next)
-        {
-            // Ignore stopped or paused sounds
-            if (!audioBuffer->playing || audioBuffer->paused) continue;
-
-            ma_uint32 framesRead = 0;
-
-            while (1)
-            {
-                if (framesRead >= frameCount) break;
-
-                // Just read as much data as we can from the stream
-                ma_uint32 framesToRead = (frameCount - framesRead);
-
-                while (framesToRead > 0)
-                {
-                    float tempBuffer[1024]; // 512 frames for stereo
-
-                    ma_uint32 framesToReadRightNow = framesToRead;
-                    if (framesToReadRightNow > sizeof(tempBuffer)/sizeof(tempBuffer[0])/AUDIO_DEVICE_CHANNELS)
-                    {
-                        framesToReadRightNow = sizeof(tempBuffer)/sizeof(tempBuffer[0])/AUDIO_DEVICE_CHANNELS;
-                    }
-
-                    ma_uint32 framesJustRead = ReadAudioBufferFramesInMixingFormat(audioBuffer, tempBuffer, framesToReadRightNow);
-                    if (framesJustRead > 0)
-                    {
-                        float *framesOut = (float *)pFramesOut + (framesRead*AUDIO.System.device.playback.channels);
-                        float *framesIn  = tempBuffer;
-
-                        MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume);
-
-                        framesToRead -= framesJustRead;
-                        framesRead += framesJustRead;
-                    }
-
-                    if (!audioBuffer->playing)
-                    {
-                        framesRead = frameCount;
-                        break;
-                    }
-
-                    // If we weren't able to read all the frames we requested, break
-                    if (framesJustRead < framesToReadRightNow)
-                    {
-                        if (!audioBuffer->looping)
-                        {
-                            StopAudioBuffer(audioBuffer);
-                            break;
-                        }
-                        else
-                        {
-                            // Should never get here, but just for safety,
-                            // move the cursor position back to the start and continue the loop
-                            audioBuffer->frameCursorPos = 0;
-                            continue;
-                        }
-                    }
-                }
-
-                // If for some reason we weren't able to read every frame we'll need to break from the loop
-                // Not doing this could theoretically put us into an infinite loop
-                if (framesToRead > 0) break;
-            }
-        }
-    }
-
-    ma_mutex_unlock(&AUDIO.System.lock);
-}
-
-// This is the main mixing function. Mixing is pretty simple in this project - it's just an accumulation.
-// NOTE: framesOut is both an input and an output. It will be initially filled with zeros outside of this function.
-static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume)
-{
-    for (ma_uint32 iFrame = 0; iFrame < frameCount; ++iFrame)
-    {
-        for (ma_uint32 iChannel = 0; iChannel < AUDIO.System.device.playback.channels; ++iChannel)
-        {
-            float *frameOut = framesOut + (iFrame*AUDIO.System.device.playback.channels);
-            const float *frameIn  = framesIn  + (iFrame*AUDIO.System.device.playback.channels);
-
-            frameOut[iChannel] += (frameIn[iChannel]*localVolume);
-        }
-    }
-}
-
-// Initialise the multichannel buffer pool
-static void InitAudioBufferPool(void)
-{
-    // Dummy buffers
-    for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
-    {
-        AUDIO.MultiChannel.pool[i] = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, 0, AUDIO_BUFFER_USAGE_STATIC);
-    }
-    
-    // TODO: Verification required for log
-    TRACELOG(LOG_INFO, "AUDIO: Multichannel pool size: %i", MAX_AUDIO_BUFFER_POOL_CHANNELS);
-}
-
-// Close the audio buffers pool
-static void CloseAudioBufferPool(void)
-{
-    for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
-    {
-        RL_FREE(AUDIO.MultiChannel.pool[i]->data);
-        RL_FREE(AUDIO.MultiChannel.pool[i]);
-    }
-}
-
-#if defined(SUPPORT_FILEFORMAT_WAV)
-// Load WAV file into Wave structure
-static Wave LoadWAV(const char *fileName)
-{
-    // Basic WAV headers structs
-    typedef struct {
-        char chunkID[4];
-        int chunkSize;
-        char format[4];
-    } WAVRiffHeader;
-
-    typedef struct {
-        char subChunkID[4];
-        int subChunkSize;
-        short audioFormat;
-        short numChannels;
-        int sampleRate;
-        int byteRate;
-        short blockAlign;
-        short bitsPerSample;
-    } WAVFormat;
-
-    typedef struct {
-        char subChunkID[4];
-        int subChunkSize;
-    } WAVData;
-
-    WAVRiffHeader wavRiffHeader = { 0 };
-    WAVFormat wavFormat = { 0 };
-    WAVData wavData = { 0 };
-
-    Wave wave = { 0 };
-    FILE *wavFile = NULL;
-
-    wavFile = fopen(fileName, "rb");
-
-    if (wavFile == NULL)
-    {
-        TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open WAV file", fileName);
-        wave.data = NULL;
-    }
-    else
-    {
-        // Read in the first chunk into the struct
-        fread(&wavRiffHeader, sizeof(WAVRiffHeader), 1, wavFile);
-
-        // Check for RIFF and WAVE tags
-        if ((wavRiffHeader.chunkID[0] != 'R') ||
-            (wavRiffHeader.chunkID[1] != 'I') ||
-            (wavRiffHeader.chunkID[2] != 'F') ||
-            (wavRiffHeader.chunkID[3] != 'F') ||
-            (wavRiffHeader.format[0] != 'W') ||
-            (wavRiffHeader.format[1] != 'A') ||
-            (wavRiffHeader.format[2] != 'V') ||
-            (wavRiffHeader.format[3] != 'E'))
-        {
-            TRACELOG(LOG_WARNING, "WAVE: [%s] RIFF or WAVE header are not valid", fileName);
-        }
-        else
-        {
-            // Read in the 2nd chunk for the wave info
-            fread(&wavFormat, sizeof(WAVFormat), 1, wavFile);
-
-            // Check for fmt tag
-            if ((wavFormat.subChunkID[0] != 'f') || (wavFormat.subChunkID[1] != 'm') ||
-                (wavFormat.subChunkID[2] != 't') || (wavFormat.subChunkID[3] != ' '))
-            {
-                TRACELOG(LOG_WARNING, "WAVE: [%s] Wave format header is not valid", fileName);
-            }
-            else
-            {
-                // Check for extra parameters;
-                if (wavFormat.subChunkSize > 16) fseek(wavFile, sizeof(short), SEEK_CUR);
-
-                // Read in the the last byte of data before the sound file
-                fread(&wavData, sizeof(WAVData), 1, wavFile);
-
-                // Check for data tag
-                if ((wavData.subChunkID[0] != 'd') || (wavData.subChunkID[1] != 'a') ||
-                    (wavData.subChunkID[2] != 't') || (wavData.subChunkID[3] != 'a'))
-                {
-                    TRACELOG(LOG_WARNING, "WAVE: [%s] Data header is not valid", fileName);
-                }
-                else
-                {
-                    // Allocate memory for data
-                    wave.data = RL_MALLOC(wavData.subChunkSize);
-
-                    // Read in the sound data into the soundData variable
-                    fread(wave.data, wavData.subChunkSize, 1, wavFile);
-
-                    // Store wave parameters
-                    wave.sampleRate = wavFormat.sampleRate;
-                    wave.sampleSize = wavFormat.bitsPerSample;
-                    wave.channels = wavFormat.numChannels;
-
-                    // NOTE: Only support 8 bit, 16 bit and 32 bit sample sizes
-                    if ((wave.sampleSize != 8) && (wave.sampleSize != 16) && (wave.sampleSize != 32))
-                    {
-                        TRACELOG(LOG_WARNING, "WAVE: [%s] Sample size (%ibit) not supported, converted to 16bit", fileName, wave.sampleSize);
-                        WaveFormat(&wave, wave.sampleRate, 16, wave.channels);
-                    }
-
-                    // NOTE: Only support up to 2 channels (mono, stereo)
-                    if (wave.channels > 2)
-                    {
-                        WaveFormat(&wave, wave.sampleRate, wave.sampleSize, 2);
-                        TRACELOG(LOG_WARNING, "WAVE: [%s] Channels number (%i) not supported, converted to 2 channels", fileName, wave.channels);
-                    }
-
-                    // NOTE: subChunkSize comes in bytes, we need to translate it to number of samples
-                    wave.sampleCount = (wavData.subChunkSize/(wave.sampleSize/8))/wave.channels;
-
-                    TRACELOG(LOG_INFO, "WAVE: [%s] File loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
-                }
-            }
-        }
-
-        fclose(wavFile);
-    }
-
-    return wave;
-}
-
-// Save wave data as WAV file
-static int SaveWAV(Wave wave, const char *fileName)
-{
-    int success = 0;
-    int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8;
-
-    // Basic WAV headers structs
-    typedef struct {
-        char chunkID[4];
-        int chunkSize;
-        char format[4];
-    } RiffHeader;
-
-    typedef struct {
-        char subChunkID[4];
-        int subChunkSize;
-        short audioFormat;
-        short numChannels;
-        int sampleRate;
-        int byteRate;
-        short blockAlign;
-        short bitsPerSample;
-    } WaveFormat;
-
-    typedef struct {
-        char subChunkID[4];
-        int subChunkSize;
-    } WaveData;
-
-    FILE *wavFile = fopen(fileName, "wb");
-
-    if (wavFile == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open audio file", fileName);
-    else
-    {
-        RiffHeader riffHeader;
-        WaveFormat waveFormat;
-        WaveData waveData;
-
-        // Fill structs with data
-        riffHeader.chunkID[0] = 'R';
-        riffHeader.chunkID[1] = 'I';
-        riffHeader.chunkID[2] = 'F';
-        riffHeader.chunkID[3] = 'F';
-        riffHeader.chunkSize = 44 - 4 + wave.sampleCount*wave.sampleSize/8;
-        riffHeader.format[0] = 'W';
-        riffHeader.format[1] = 'A';
-        riffHeader.format[2] = 'V';
-        riffHeader.format[3] = 'E';
-
-        waveFormat.subChunkID[0] = 'f';
-        waveFormat.subChunkID[1] = 'm';
-        waveFormat.subChunkID[2] = 't';
-        waveFormat.subChunkID[3] = ' ';
-        waveFormat.subChunkSize = 16;
-        waveFormat.audioFormat = 1;
-        waveFormat.numChannels = wave.channels;
-        waveFormat.sampleRate = wave.sampleRate;
-        waveFormat.byteRate = wave.sampleRate*wave.sampleSize/8;
-        waveFormat.blockAlign = wave.sampleSize/8;
-        waveFormat.bitsPerSample = wave.sampleSize;
-
-        waveData.subChunkID[0] = 'd';
-        waveData.subChunkID[1] = 'a';
-        waveData.subChunkID[2] = 't';
-        waveData.subChunkID[3] = 'a';
-        waveData.subChunkSize = dataSize;
-
-        fwrite(&riffHeader, sizeof(RiffHeader), 1, wavFile);
-        fwrite(&waveFormat, sizeof(WaveFormat), 1, wavFile);
-        fwrite(&waveData, sizeof(WaveData), 1, wavFile);
-
-        success = fwrite(wave.data, dataSize, 1, wavFile);
-
-        fclose(wavFile);
-    }
-
-    // If all data has been written correctly to file, success = 1
-    return success;
-}
-#endif
-
-#if defined(SUPPORT_FILEFORMAT_OGG)
-// Load OGG file into Wave structure
-// NOTE: Using stb_vorbis library
-static Wave LoadOGG(const char *fileName)
-{
-    Wave wave = { 0 };
-
-    stb_vorbis *oggFile = stb_vorbis_open_filename(fileName, NULL, NULL);
-
-    if (oggFile == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open OGG file", fileName);
-    else
-    {
-        stb_vorbis_info info = stb_vorbis_get_info(oggFile);
-
-        wave.sampleRate = info.sample_rate;
-        wave.sampleSize = 16;                   // 16 bit per sample (short)
-        wave.channels = info.channels;
-        wave.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples(oggFile)*info.channels;  // Independent by channel
-
-        float totalSeconds = stb_vorbis_stream_length_in_seconds(oggFile);
-        if (totalSeconds > 10) TRACELOG(LOG_WARNING, "WAVE: [%s] Ogg audio length larger than 10 seconds (%f), that's a big file in memory, consider music streaming", fileName, totalSeconds);
-
-        wave.data = (short *)RL_MALLOC(wave.sampleCount*wave.channels*sizeof(short));
-
-        // NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
-        stb_vorbis_get_samples_short_interleaved(oggFile, info.channels, (short *)wave.data, wave.sampleCount*wave.channels);
-        TRACELOG(LOG_INFO, "WAVE: [%s] OGG file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
-
-        stb_vorbis_close(oggFile);
-    }
-
-    return wave;
-}
-#endif
-
-#if defined(SUPPORT_FILEFORMAT_FLAC)
-// Load FLAC file into Wave structure
-// NOTE: Using dr_flac library
-static Wave LoadFLAC(const char *fileName)
-{
-    Wave wave = { 0 };
-
-    // Decode an entire FLAC file in one go
-    unsigned long long int totalSampleCount = 0;
-    wave.data = drflac_open_file_and_read_pcm_frames_s16(fileName, &wave.channels, &wave.sampleRate, &totalSampleCount);
-
-    if (wave.data == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to load FLAC data", fileName);
-    else
-    {
-        wave.sampleCount = (unsigned int)totalSampleCount;
-        wave.sampleSize = 16;
-
-        // NOTE: Only support up to 2 channels (mono, stereo)
-        if (wave.channels > 2) TRACELOG(LOG_WARNING, "WAVE: [%s] FLAC channels number (%i) not supported", fileName, wave.channels);
-
-        TRACELOG(LOG_INFO, "WAVE: [%s] FLAC file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
-    }
- 
-    return wave;
-}
-#endif
-
-#if defined(SUPPORT_FILEFORMAT_MP3)
-// Load MP3 file into Wave structure
-// NOTE: Using dr_mp3 library
-static Wave LoadMP3(const char *fileName)
-{
-    Wave wave = { 0 };
-
-    // Decode an entire MP3 file in one go
-    unsigned long long int totalFrameCount = 0;
-    drmp3_config config = { 0 };
-    wave.data = drmp3_open_file_and_read_f32(fileName, &config, &totalFrameCount);
-
-    if (wave.data == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to load MP3 data", fileName);
-    else
-    {
-        wave.channels = config.outputChannels;
-        wave.sampleRate = config.outputSampleRate;
-        wave.sampleCount = (int)totalFrameCount*wave.channels;
-        wave.sampleSize = 32;
-
-        // NOTE: Only support up to 2 channels (mono, stereo)
-        if (wave.channels > 2) TRACELOG(LOG_WARNING, "WAVE: [%s] MP3 channels number (%i) not supported", fileName, wave.channels);
-
-        TRACELOG(LOG_INFO, "WAVE: [%s] MP3 file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
-    }
-    
-    return wave;
-}
-#endif
-
-// Some required functions for audio standalone module version
-#if defined(RAUDIO_STANDALONE)
-// Check file extension
-bool IsFileExtension(const char *fileName, const char *ext)
-{
-    bool result = false;
-    const char *fileExt;
-
-    if ((fileExt = strrchr(fileName, '.')) != NULL)
-    {
-        if (strcmp(fileExt, ext) == 0) result = true;
-    }
-
-    return result;
-}
-#endif
-
-#undef AudioBuffer