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diff --git a/raylib/raudio.c b/raylib/raudio.c
deleted file mode 100644
index d0f6963..0000000
--- a/raylib/raudio.c
+++ /dev/null
@@ -1,2668 +0,0 @@
-/**********************************************************************************************
-*
-*   raudio v1.1 - A simple and easy-to-use audio library based on miniaudio
-*
-*   FEATURES:
-*       - Manage audio device (init/close)
-*       - Manage raw audio context
-*       - Manage mixing channels
-*       - Load and unload audio files
-*       - Format wave data (sample rate, size, channels)
-*       - Play/Stop/Pause/Resume loaded audio
-*
-*   CONFIGURATION:
-*       #define SUPPORT_MODULE_RAUDIO
-*           raudio module is included in the build
-*
-*       #define RAUDIO_STANDALONE
-*           Define to use the module as standalone library (independently of raylib).
-*           Required types and functions are defined in the same module.
-*
-*       #define SUPPORT_FILEFORMAT_WAV
-*       #define SUPPORT_FILEFORMAT_OGG
-*       #define SUPPORT_FILEFORMAT_MP3
-*       #define SUPPORT_FILEFORMAT_QOA
-*       #define SUPPORT_FILEFORMAT_FLAC
-*       #define SUPPORT_FILEFORMAT_XM
-*       #define SUPPORT_FILEFORMAT_MOD
-*           Selected desired fileformats to be supported for loading. Some of those formats are
-*           supported by default, to remove support, just comment unrequired #define in this module
-*
-*   DEPENDENCIES:
-*       miniaudio.h  - Audio device management lib (https://github.com/mackron/miniaudio)
-*       stb_vorbis.h - Ogg audio files loading (http://www.nothings.org/stb_vorbis/)
-*       dr_wav.h     - WAV audio files loading (http://github.com/mackron/dr_libs)
-*       dr_mp3.h     - MP3 audio file loading (https://github.com/mackron/dr_libs)
-*       dr_flac.h    - FLAC audio file loading (https://github.com/mackron/dr_libs)
-*       jar_xm.h     - XM module file loading
-*       jar_mod.h    - MOD audio file loading
-*
-*   CONTRIBUTORS:
-*       David Reid (github: @mackron) (Nov. 2017):
-*           - Complete port to miniaudio library
-*
-*       Joshua Reisenauer (github: @kd7tck) (2015):
-*           - XM audio module support (jar_xm)
-*           - MOD audio module support (jar_mod)
-*           - Mixing channels support
-*           - Raw audio context support
-*
-*
-*   LICENSE: zlib/libpng
-*
-*   Copyright (c) 2013-2023 Ramon Santamaria (@raysan5)
-*
-*   This software is provided "as-is", without any express or implied warranty. In no event
-*   will the authors be held liable for any damages arising from the use of this software.
-*
-*   Permission is granted to anyone to use this software for any purpose, including commercial
-*   applications, and to alter it and redistribute it freely, subject to the following restrictions:
-*
-*     1. The origin of this software must not be misrepresented; you must not claim that you
-*     wrote the original software. If you use this software in a product, an acknowledgment
-*     in the product documentation would be appreciated but is not required.
-*
-*     2. Altered source versions must be plainly marked as such, and must not be misrepresented
-*     as being the original software.
-*
-*     3. This notice may not be removed or altered from any source distribution.
-*
-**********************************************************************************************/
-
-#if defined(RAUDIO_STANDALONE)
-    #include "raudio.h"
-#else
-    #include "raylib.h"         // Declares module functions
-
-    // Check if config flags have been externally provided on compilation line
-    #if !defined(EXTERNAL_CONFIG_FLAGS)
-        #include "config.h"     // Defines module configuration flags
-    #endif
-    #include "utils.h"          // Required for: fopen() Android mapping
-#endif
-
-#if defined(SUPPORT_MODULE_RAUDIO)
-
-#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
-
-// Threading model: Default: [0] COINIT_MULTITHREADED: COM calls objects on any thread (free threading)
-#define MA_COINIT_VALUE  2              // [2] COINIT_APARTMENTTHREADED: Each object has its own thread (apartment model)
-
-#define MINIAUDIO_IMPLEMENTATION
-//#define MA_DEBUG_OUTPUT
-#include "external/miniaudio.h"         // Audio device initialization and management
-#undef PlaySound                        // Win32 API: windows.h > mmsystem.h defines PlaySound macro
-
-#include <stdlib.h>                     // Required for: malloc(), free()
-#include <stdio.h>                      // Required for: FILE, fopen(), fclose(), fread()
-#include <string.h>                     // Required for: strcmp() [Used in IsFileExtension(), LoadWaveFromMemory(), LoadMusicStreamFromMemory()]
-
-#if defined(RAUDIO_STANDALONE)
-    #ifndef TRACELOG
-        #define TRACELOG(level, ...)    printf(__VA_ARGS__)
-    #endif
-
-    // Allow custom memory allocators
-    #ifndef RL_MALLOC
-        #define RL_MALLOC(sz)           malloc(sz)
-    #endif
-    #ifndef RL_CALLOC
-        #define RL_CALLOC(n,sz)         calloc(n,sz)
-    #endif
-    #ifndef RL_REALLOC
-        #define RL_REALLOC(ptr,sz)      realloc(ptr,sz)
-    #endif
-    #ifndef RL_FREE
-        #define RL_FREE(ptr)            free(ptr)
-    #endif
-#endif
-
-#if defined(SUPPORT_FILEFORMAT_WAV)
-    #define DRWAV_MALLOC RL_MALLOC
-    #define DRWAV_REALLOC RL_REALLOC
-    #define DRWAV_FREE RL_FREE
-
-    #define DR_WAV_IMPLEMENTATION
-    #include "external/dr_wav.h"        // WAV loading functions
-#endif
-
-#if defined(SUPPORT_FILEFORMAT_OGG)
-    // TODO: Remap stb_vorbis malloc()/free() calls to RL_MALLOC/RL_FREE
-    #include "external/stb_vorbis.c"    // OGG loading functions
-#endif
-
-#if defined(SUPPORT_FILEFORMAT_MP3)
-    #define DRMP3_MALLOC RL_MALLOC
-    #define DRMP3_REALLOC RL_REALLOC
-    #define DRMP3_FREE RL_FREE
-
-    #define DR_MP3_IMPLEMENTATION
-    #include "external/dr_mp3.h"        // MP3 loading functions
-#endif
-
-#if defined(SUPPORT_FILEFORMAT_QOA)
-    #define QOA_MALLOC RL_MALLOC
-    #define QOA_FREE RL_FREE
-
-#if defined(_MSC_VER ) // par shapes has 2 warnings on windows, so disable them just fof this file
-#pragma warning( push )
-#pragma warning( disable : 4018)
-#pragma warning( disable : 4267)
-#pragma warning( disable : 4244)
-#endif
-
-
-    #define QOA_IMPLEMENTATION
-    #include "external/qoa.h"           // QOA loading and saving functions
-    #include "external/qoaplay.c"       // QOA stream playing helper functions
-#endif
-
-#if defined(SUPPORT_FILEFORMAT_FLAC)
-    #define DRFLAC_MALLOC RL_MALLOC
-    #define DRFLAC_REALLOC RL_REALLOC
-    #define DRFLAC_FREE RL_FREE
-
-    #define DR_FLAC_IMPLEMENTATION
-    #define DR_FLAC_NO_WIN32_IO
-    #include "external/dr_flac.h"       // FLAC loading functions
-#endif
-
-#if defined(SUPPORT_FILEFORMAT_XM)
-    #define JARXM_MALLOC RL_MALLOC
-    #define JARXM_FREE RL_FREE
-
-    #if defined(_MSC_VER )              // jar_xm has warnings on windows, so disable them just for this file
-        #pragma warning( push )
-        #pragma warning( disable : 4244)
-    #endif
-
-    #define JAR_XM_IMPLEMENTATION
-    #include "external/jar_xm.h"        // XM loading functions
-
-    #if defined(_MSC_VER )
-        #pragma warning( pop )
-    #endif
-#endif
-
-#if defined(SUPPORT_FILEFORMAT_MOD)
-    #define JARMOD_MALLOC RL_MALLOC
-    #define JARMOD_FREE RL_FREE
-
-    #define JAR_MOD_IMPLEMENTATION
-    #include "external/jar_mod.h"       // MOD loading functions
-#endif
-
-//----------------------------------------------------------------------------------
-// Defines and Macros
-//----------------------------------------------------------------------------------
-#ifndef AUDIO_DEVICE_FORMAT
-    #define AUDIO_DEVICE_FORMAT    ma_format_f32    // Device output format (float-32bit)
-#endif
-#ifndef AUDIO_DEVICE_CHANNELS
-    #define AUDIO_DEVICE_CHANNELS              2    // Device output channels: stereo
-#endif
-#ifndef AUDIO_DEVICE_SAMPLE_RATE
-    #define AUDIO_DEVICE_SAMPLE_RATE           0    // Device output sample rate
-#endif
-
-#ifndef MAX_AUDIO_BUFFER_POOL_CHANNELS
-    #define MAX_AUDIO_BUFFER_POOL_CHANNELS    16    // Audio pool channels
-#endif
-
-//----------------------------------------------------------------------------------
-// Types and Structures Definition
-//----------------------------------------------------------------------------------
-#if defined(RAUDIO_STANDALONE)
-// Trace log level
-// NOTE: Organized by priority level
-typedef enum {
-    LOG_ALL = 0,        // Display all logs
-    LOG_TRACE,          // Trace logging, intended for internal use only
-    LOG_DEBUG,          // Debug logging, used for internal debugging, it should be disabled on release builds
-    LOG_INFO,           // Info logging, used for program execution info
-    LOG_WARNING,        // Warning logging, used on recoverable failures
-    LOG_ERROR,          // Error logging, used on unrecoverable failures
-    LOG_FATAL,          // Fatal logging, used to abort program: exit(EXIT_FAILURE)
-    LOG_NONE            // Disable logging
-} TraceLogLevel;
-#endif
-
-// Music context type
-// NOTE: Depends on data structure provided by the library
-// in charge of reading the different file types
-typedef enum {
-    MUSIC_AUDIO_NONE = 0,   // No audio context loaded
-    MUSIC_AUDIO_WAV,        // WAV audio context
-    MUSIC_AUDIO_OGG,        // OGG audio context
-    MUSIC_AUDIO_FLAC,       // FLAC audio context
-    MUSIC_AUDIO_MP3,        // MP3 audio context
-    MUSIC_AUDIO_QOA,        // QOA audio context
-    MUSIC_MODULE_XM,        // XM module audio context
-    MUSIC_MODULE_MOD        // MOD module audio context
-} MusicContextType;
-
-// NOTE: Different logic is used when feeding data to the playback device
-// depending on whether data is streamed (Music vs Sound)
-typedef enum {
-    AUDIO_BUFFER_USAGE_STATIC = 0,
-    AUDIO_BUFFER_USAGE_STREAM
-} AudioBufferUsage;
-
-// Audio buffer struct
-struct rAudioBuffer {
-    ma_data_converter converter;    // Audio data converter
-
-    AudioCallback callback;         // Audio buffer callback for buffer filling on audio threads
-    rAudioProcessor *processor;     // Audio processor
-
-    float volume;                   // Audio buffer volume
-    float pitch;                    // Audio buffer pitch
-    float pan;                      // Audio buffer pan (0.0f to 1.0f)
-
-    bool playing;                   // Audio buffer state: AUDIO_PLAYING
-    bool paused;                    // Audio buffer state: AUDIO_PAUSED
-    bool looping;                   // Audio buffer looping, default to true for AudioStreams
-    int usage;                      // Audio buffer usage mode: STATIC or STREAM
-
-    bool isSubBufferProcessed[2];   // SubBuffer processed (virtual double buffer)
-    unsigned int sizeInFrames;      // Total buffer size in frames
-    unsigned int frameCursorPos;    // Frame cursor position
-    unsigned int framesProcessed;   // Total frames processed in this buffer (required for play timing)
-
-    unsigned char *data;            // Data buffer, on music stream keeps filling
-
-    rAudioBuffer *next;             // Next audio buffer on the list
-    rAudioBuffer *prev;             // Previous audio buffer on the list
-};
-
-// Audio processor struct
-// NOTE: Useful to apply effects to an AudioBuffer
-struct rAudioProcessor {
-    AudioCallback process;          // Processor callback function
-    rAudioProcessor *next;          // Next audio processor on the list
-    rAudioProcessor *prev;          // Previous audio processor on the list
-};
-
-#define AudioBuffer rAudioBuffer    // HACK: To avoid CoreAudio (macOS) symbol collision
-
-// Audio data context
-typedef struct AudioData {
-    struct {
-        ma_context context;         // miniaudio context data
-        ma_device device;           // miniaudio device
-        ma_mutex lock;              // miniaudio mutex lock
-        bool isReady;               // Check if audio device is ready
-        size_t pcmBufferSize;       // Pre-allocated buffer size
-        void *pcmBuffer;            // Pre-allocated buffer to read audio data from file/memory
-    } System;
-    struct {
-        AudioBuffer *first;         // Pointer to first AudioBuffer in the list
-        AudioBuffer *last;          // Pointer to last AudioBuffer in the list
-        int defaultSize;            // Default audio buffer size for audio streams
-    } Buffer;
-    rAudioProcessor *mixedProcessor;
-} AudioData;
-
-//----------------------------------------------------------------------------------
-// Global Variables Definition
-//----------------------------------------------------------------------------------
-static AudioData AUDIO = {          // Global AUDIO context
-
-    // NOTE: Music buffer size is defined by number of samples, independent of sample size and channels number
-    // After some math, considering a sampleRate of 48000, a buffer refill rate of 1/60 seconds and a
-    // standard double-buffering system, a 4096 samples buffer has been chosen, it should be enough
-    // In case of music-stalls, just increase this number
-    .Buffer.defaultSize = 0,
-    .mixedProcessor = NULL
-};
-
-//----------------------------------------------------------------------------------
-// Module 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 separate thread which means we need to synchronize. I'm using a mutex here to make things simple, but may
-    // want to look at something a bit smarter later on to keep everything real-time, if that's necessary.
-    if (ma_mutex_init(&AUDIO.System.lock) != MA_SUCCESS)
-    {
-        TRACELOG(LOG_WARNING, "AUDIO: Failed to create mutex for mixing");
-        ma_device_uninit(&AUDIO.System.device);
-        ma_context_uninit(&AUDIO.System.context);
-        return;
-    }
-
-    TRACELOG(LOG_INFO, "AUDIO: Device initialized successfully");
-    TRACELOG(LOG_INFO, "    > Backend:       miniaudio / %s", ma_get_backend_name(AUDIO.System.context.backend));
-    TRACELOG(LOG_INFO, "    > Format:        %s -> %s", ma_get_format_name(AUDIO.System.device.playback.format), ma_get_format_name(AUDIO.System.device.playback.internalFormat));
-    TRACELOG(LOG_INFO, "    > Channels:      %d -> %d", AUDIO.System.device.playback.channels, AUDIO.System.device.playback.internalChannels);
-    TRACELOG(LOG_INFO, "    > Sample rate:   %d -> %d", AUDIO.System.device.sampleRate, AUDIO.System.device.playback.internalSampleRate);
-    TRACELOG(LOG_INFO, "    > Periods size:  %d", AUDIO.System.device.playback.internalPeriodSizeInFrames*AUDIO.System.device.playback.internalPeriods);
-
-    AUDIO.System.isReady = true;
-}
-
-// Close the audio device for all contexts
-void CloseAudioDevice(void)
-{
-    if (AUDIO.System.isReady)
-    {
-        ma_mutex_uninit(&AUDIO.System.lock);
-        ma_device_uninit(&AUDIO.System.device);
-        ma_context_uninit(&AUDIO.System.context);
-
-        AUDIO.System.isReady = false;
-        RL_FREE(AUDIO.System.pcmBuffer);
-        AUDIO.System.pcmBuffer = NULL;
-        AUDIO.System.pcmBufferSize = 0;
-
-        TRACELOG(LOG_INFO, "AUDIO: Device closed successfully");
-    }
-    else TRACELOG(LOG_WARNING, "AUDIO: Device could not be closed, not currently initialized");
-}
-
-// Check if device has been initialized successfully
-bool IsAudioDeviceReady(void)
-{
-    return AUDIO.System.isReady;
-}
-
-// Set master volume (listener)
-void SetMasterVolume(float volume)
-{
-    ma_device_set_master_volume(&AUDIO.System.device, volume);
-}
-
-//----------------------------------------------------------------------------------
-// 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) || (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) || (strcmp(fileType, ".OGG") == 0))
-    {
-        stb_vorbis *oggData = stb_vorbis_open_memory((unsigned char *)fileData, dataSize, NULL, NULL);
-
-        if (oggData != NULL)
-        {
-            stb_vorbis_info info = stb_vorbis_get_info(oggData);
-
-            wave.sampleRate = info.sample_rate;
-            wave.sampleSize = 16;       // By default, ogg data is 16 bit per sample (short)
-            wave.channels = info.channels;
-            wave.frameCount = (unsigned int)stb_vorbis_stream_length_in_samples(oggData);  // NOTE: It returns frames!
-            wave.data = (short *)RL_MALLOC(wave.frameCount*wave.channels*sizeof(short));
-
-            // NOTE: Get the number of samples to process (be careful! we ask for number of shorts, not bytes!)
-            stb_vorbis_get_samples_short_interleaved(oggData, info.channels, (short *)wave.data, wave.frameCount*wave.channels);
-            stb_vorbis_close(oggData);
-        }
-        else TRACELOG(LOG_WARNING, "WAVE: Failed to load OGG data");
-    }
-#endif
-#if defined(SUPPORT_FILEFORMAT_MP3)
-    else if ((strcmp(fileType, ".mp3") == 0) || (strcmp(fileType, ".MP3") == 0))
-    {
-        drmp3_config config = { 0 };
-        unsigned long long int totalFrameCount = 0;
-
-        // NOTE: We are forcing conversion to 32bit float sample size on reading
-        wave.data = drmp3_open_memory_and_read_pcm_frames_f32(fileData, dataSize, &config, &totalFrameCount, NULL);
-        wave.sampleSize = 32;
-
-        if (wave.data != NULL)
-        {
-            wave.channels = config.channels;
-            wave.sampleRate = config.sampleRate;
-            wave.frameCount = (int)totalFrameCount;
-        }
-        else TRACELOG(LOG_WARNING, "WAVE: Failed to load MP3 data");
-
-    }
-#endif
-#if defined(SUPPORT_FILEFORMAT_QOA)
-    else if ((strcmp(fileType, ".qoa") == 0) || (strcmp(fileType, ".QOA") == 0))
-    {
-        qoa_desc qoa = { 0 };
-
-        // NOTE: Returned sample data is always 16 bit?
-        wave.data = qoa_decode(fileData, dataSize, &qoa);
-        wave.sampleSize = 16;
-
-        if (wave.data != NULL)
-        {
-            wave.channels = qoa.channels;
-            wave.sampleRate = qoa.samplerate;
-            wave.frameCount = qoa.samples;
-        }
-        else TRACELOG(LOG_WARNING, "WAVE: Failed to load QOA data");
-
-    }
-#endif
-#if defined(SUPPORT_FILEFORMAT_FLAC)
-    else if ((strcmp(fileType, ".flac") == 0) || (strcmp(fileType, ".FLAC") == 0))
-    {
-        unsigned long long int totalFrameCount = 0;
-
-        // NOTE: We are forcing conversion to 16bit sample size on reading
-        wave.data = drflac_open_memory_and_read_pcm_frames_s16(fileData, dataSize, &wave.channels, &wave.sampleRate, &totalFrameCount, NULL);
-        wave.sampleSize = 16;
-
-        if (wave.data != NULL) wave.frameCount = (unsigned int)totalFrameCount;
-        else TRACELOG(LOG_WARNING, "WAVE: Failed to load FLAC data");
-    }
-#endif
-    else TRACELOG(LOG_WARNING, "WAVE: Data format not supported");
-
-    TRACELOG(LOG_INFO, "WAVE: Data loaded successfully (%i Hz, %i bit, %i channels)", wave.sampleRate, wave.sampleSize, wave.channels);
-
-    return wave;
-}
-
-// Checks if wave data is ready
-bool IsWaveReady(Wave wave)
-{
-    return ((wave.data != NULL) &&      // Validate wave data available
-            (wave.frameCount > 0) &&    // Validate frame count
-            (wave.sampleRate > 0) &&    // Validate sample rate is supported
-            (wave.sampleSize > 0) &&    // Validate sample size is supported
-            (wave.channels > 0));       // Validate number of channels supported
-}
-
-// 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;
-}
-
-// Checks if a sound is ready
-bool IsSoundReady(Sound sound)
-{
-    return ((sound.frameCount > 0) &&           // Validate frame count
-            (sound.stream.buffer != NULL) &&    // Validate stream buffer
-            (sound.stream.sampleRate > 0) &&    // Validate sample rate is supported
-            (sound.stream.sampleSize > 0) &&    // Validate sample size is supported
-            (sound.stream.channels > 0));       // Validate number of channels supported
-}
-
-// 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
-#if defined(SUPPORT_FILEFORMAT_QOA)
-    else if (IsFileExtension(fileName, ".qoa"))
-    {
-        if (wave.sampleSize == 16)
-        {
-            qoa_desc qoa = { 0 };
-            qoa.channels = wave.channels;
-            qoa.samplerate = wave.sampleRate;
-            qoa.samples = wave.frameCount;
-
-            int bytesWritten = qoa_write(fileName, wave.data, &qoa);
-            if (bytesWritten > 0) success = true;
-        }
-        else TRACELOG(LOG_WARNING, "AUDIO: Wave data must be 16 bit per sample for QOA format export");
-    }
-#endif
-    else if (IsFileExtension(fileName, ".raw"))
-    {
-        // Export raw sample data (without header)
-        // NOTE: It's up to the user to track wave parameters
-        success = SaveFileData(fileName, wave.data, wave.frameCount*wave.channels*wave.sampleSize/8);
-    }
-
-    if (success) TRACELOG(LOG_INFO, "FILEIO: [%s] Wave data exported successfully", fileName);
-    else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to export wave data", fileName);
-
-    return success;
-}
-
-// Export wave sample data to code (.h)
-bool ExportWaveAsCode(Wave wave, const char *fileName)
-{
-    bool success = false;
-
-#ifndef TEXT_BYTES_PER_LINE
-    #define TEXT_BYTES_PER_LINE     20
-#endif
-
-    int waveDataSize = wave.frameCount*wave.channels*wave.sampleSize/8;
-
-    // NOTE: Text data buffer size is estimated considering wave data size in bytes
-    // and requiring 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) 2018-2023 Ramon Santamaria (@raysan5)                          //\n");
-    byteCount += sprintf(txtData + byteCount, "//                                                                              //\n");
-    byteCount += sprintf(txtData + byteCount, "//////////////////////////////////////////////////////////////////////////////////\n\n");
-
-    // Get file name from path and convert variable name to uppercase
-    char varFileName[256] = { 0 };
-    strcpy(varFileName, GetFileNameWithoutExt(fileName));
-    for (int i = 0; varFileName[i] != '\0'; i++) if (varFileName[i] >= 'a' && varFileName[i] <= 'z') { varFileName[i] = varFileName[i] - 32; }
-
-    //Add wave information
-    byteCount += sprintf(txtData + byteCount, "// Wave data information\n");
-    byteCount += sprintf(txtData + byteCount, "#define %s_FRAME_COUNT      %u\n", varFileName, wave.frameCount);
-    byteCount += sprintf(txtData + byteCount, "#define %s_SAMPLE_RATE      %u\n", varFileName, wave.sampleRate);
-    byteCount += sprintf(txtData + byteCount, "#define %s_SAMPLE_SIZE      %u\n", varFileName, wave.sampleSize);
-    byteCount += sprintf(txtData + byteCount, "#define %s_CHANNELS         %u\n\n", varFileName, wave.channels);
-
-    // Write wave data as an array of values
-    // Wave data is exported as byte array for 8/16bit and float array for 32bit float data
-    // NOTE: Frame data exported is channel-interlaced: frame01[sampleChannel1, sampleChannel2, ...], frame02[], frame03[]
-    if (wave.sampleSize == 32)
-    {
-        byteCount += sprintf(txtData + byteCount, "static float %s_DATA[%i] = {\n", varFileName, waveDataSize/4);
-        for (int i = 1; i < waveDataSize/4; i++) byteCount += sprintf(txtData + byteCount, ((i%TEXT_BYTES_PER_LINE == 0)? "%.4ff,\n    " : "%.4ff, "), ((float *)wave.data)[i - 1]);
-        byteCount += sprintf(txtData + byteCount, "%.4ff };\n", ((float *)wave.data)[waveDataSize/4 - 1]);
-    }
-    else
-    {
-        byteCount += sprintf(txtData + byteCount, "static unsigned char %s_DATA[%i] = { ", varFileName, waveDataSize);
-        for (int i = 1; i < waveDataSize; i++) byteCount += sprintf(txtData + byteCount, ((i%TEXT_BYTES_PER_LINE == 0)? "0x%x,\n    " : "0x%x, "), ((unsigned char *)wave.data)[i - 1]);
-        byteCount += sprintf(txtData + byteCount, "0x%x };\n", ((unsigned char *)wave.data)[waveDataSize - 1]);
-    }
-
-    // NOTE: Text data length exported is determined by '\0' (NULL) character
-    success = SaveFileText(fileName, txtData);
-
-    RL_FREE(txtData);
-
-    if (success != 0) TRACELOG(LOG_INFO, "FILEIO: [%s] Wave as code exported successfully", fileName);
-    else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to export wave as code", fileName);
-
-    return success;
-}
-
-// Play a sound
-void PlaySound(Sound sound)
-{
-    PlayAudioBuffer(sound.stream.buffer);
-}
-
-// Pause a sound
-void PauseSound(Sound sound)
-{
-    PauseAudioBuffer(sound.stream.buffer);
-}
-
-// Resume a paused sound
-void ResumeSound(Sound sound)
-{
-    ResumeAudioBuffer(sound.stream.buffer);
-}
-
-// Stop reproducing a sound
-void StopSound(Sound sound)
-{
-    StopAudioBuffer(sound.stream.buffer);
-}
-
-// Check if a sound is playing
-bool IsSoundPlaying(Sound sound)
-{
-    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) && ((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
-//----------------------------------------------------------------------------------
-
-// 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_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_QOA)
-    else if (IsFileExtension(fileName, ".qoa"))
-    {
-        qoaplay_desc *ctxQoa = qoaplay_open(fileName);
-        music.ctxType = MUSIC_AUDIO_QOA;
-        music.ctxData = ctxQoa;
-
-        if (ctxQoa->file != NULL)
-        {
-            // NOTE: We are loading samples are 32bit float normalized data, so,
-            // we configure the output audio stream to also use float 32bit
-            music.stream = LoadAudioStream(ctxQoa->info.samplerate, 32, ctxQoa->info.channels);
-            music.frameCount = ctxQoa->info.samples;
-            music.looping = true;   // Looping enabled by default
-            musicLoaded = true;
-        }
-    }
-#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_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_MP3)
-        else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
-    #endif
-    #if defined(SUPPORT_FILEFORMAT_QOA)
-        else if (music.ctxType == MUSIC_AUDIO_QOA) qoaplay_close((qoaplay_desc *)music.ctxData);
-    #endif
-    #if defined(SUPPORT_FILEFORMAT_FLAC)
-        else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL);
-    #endif
-    #if defined(SUPPORT_FILEFORMAT_XM)
-        else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData);
-    #endif
-    #if defined(SUPPORT_FILEFORMAT_MOD)
-        else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); }
-    #endif
-
-        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) || (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_OGG)
-    else if ((strcmp(fileType, ".ogg") == 0) || (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_MP3)
-    else if ((strcmp(fileType, ".mp3") == 0) || (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_QOA)
-    else if ((strcmp(fileType, ".qoa") == 0) || (strcmp(fileType, ".QOA") == 0))
-    {
-        qoaplay_desc *ctxQoa = qoaplay_open_memory(data, dataSize);
-        music.ctxType = MUSIC_AUDIO_QOA;
-        music.ctxData = ctxQoa;
-
-        if ((ctxQoa->file_data != NULL) && (ctxQoa->file_data_size != 0))
-        {
-            // NOTE: We are loading samples are 32bit float normalized data, so,
-            // we configure the output audio stream to also use float 32bit
-            music.stream = LoadAudioStream(ctxQoa->info.samplerate, 32, ctxQoa->info.channels);
-            music.frameCount = ctxQoa->info.samples;
-            music.looping = true;   // Looping enabled by default
-            musicLoaded = true;
-        }
-    }
-#endif
-#if defined(SUPPORT_FILEFORMAT_FLAC)
-    else if ((strcmp(fileType, ".flac") == 0) || (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_XM)
-    else if ((strcmp(fileType, ".xm") == 0) || (strcmp(fileType, ".XM") == 0))
-    {
-        jar_xm_context_t *ctxXm = NULL;
-        int result = jar_xm_create_context_safe(&ctxXm, (const char *)data, dataSize, AUDIO.System.device.sampleRate);
-        if (result == 0)    // XM AUDIO.System.context created successfully
-        {
-            music.ctxType = MUSIC_MODULE_XM;
-            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) || (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_OGG)
-        else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData);
-#endif
-#if defined(SUPPORT_FILEFORMAT_MP3)
-        else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
-#endif
-#if defined(SUPPORT_FILEFORMAT_QOA)
-        else if (music.ctxType == MUSIC_AUDIO_QOA) qoaplay_close((qoaplay_desc *)music.ctxData);
-#endif
-#if defined(SUPPORT_FILEFORMAT_FLAC)
-        else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL);
-#endif
-#if defined(SUPPORT_FILEFORMAT_XM)
-        else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData);
-#endif
-#if defined(SUPPORT_FILEFORMAT_MOD)
-        else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); }
-#endif
-
-        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;
-}
-
-// Checks if a music stream is ready
-bool IsMusicReady(Music music)
-{
-    return ((music.ctxData != NULL) &&          // Validate context loaded
-            (music.frameCount > 0) &&           // Validate audio frame count
-            (music.stream.sampleRate > 0) &&    // Validate sample rate is supported
-            (music.stream.sampleSize > 0) &&    // Validate sample size is supported
-            (music.stream.channels > 0));       // Validate number of channels supported
-}
-
-// Unload music stream
-void UnloadMusicStream(Music music)
-{
-    UnloadAudioStream(music.stream);
-
-    if (music.ctxData != NULL)
-    {
-        if (false) { }
-#if defined(SUPPORT_FILEFORMAT_WAV)
-        else if (music.ctxType == MUSIC_AUDIO_WAV) drwav_uninit((drwav *)music.ctxData);
-#endif
-#if defined(SUPPORT_FILEFORMAT_OGG)
-        else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData);
-#endif
-#if defined(SUPPORT_FILEFORMAT_MP3)
-        else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
-#endif
-#if defined(SUPPORT_FILEFORMAT_QOA)
-        else if (music.ctxType == MUSIC_AUDIO_QOA) qoaplay_close((qoaplay_desc *)music.ctxData);
-#endif
-#if defined(SUPPORT_FILEFORMAT_FLAC)
-        else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL);
-#endif
-#if defined(SUPPORT_FILEFORMAT_XM)
-        else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData);
-#endif
-#if defined(SUPPORT_FILEFORMAT_MOD)
-        else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); }
-#endif
-    }
-}
-
-// Start music playing (open stream)
-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_first_pcm_frame((drwav *)music.ctxData); break;
-#endif
-#if defined(SUPPORT_FILEFORMAT_OGG)
-        case MUSIC_AUDIO_OGG: stb_vorbis_seek_start((stb_vorbis *)music.ctxData); break;
-#endif
-#if defined(SUPPORT_FILEFORMAT_MP3)
-        case MUSIC_AUDIO_MP3: drmp3_seek_to_start_of_stream((drmp3 *)music.ctxData); break;
-#endif
-#if defined(SUPPORT_FILEFORMAT_QOA)
-        case MUSIC_AUDIO_QOA: qoaplay_rewind((qoaplay_desc *)music.ctxData); break;
-#endif
-#if defined(SUPPORT_FILEFORMAT_FLAC)
-        case MUSIC_AUDIO_FLAC: drflac__seek_to_first_frame((drflac *)music.ctxData); break;
-#endif
-#if defined(SUPPORT_FILEFORMAT_XM)
-        case MUSIC_MODULE_XM: jar_xm_reset((jar_xm_context_t *)music.ctxData); break;
-#endif
-#if defined(SUPPORT_FILEFORMAT_MOD)
-        case MUSIC_MODULE_MOD: jar_mod_seek_start((jar_mod_context_t *)music.ctxData); break;
-#endif
-        default: break;
-    }
-}
-
-// Seek music to a certain position (in seconds)
-void SeekMusicStream(Music music, float position)
-{
-    // Seeking is not supported in module formats
-    if ((music.ctxType == MUSIC_MODULE_XM) || (music.ctxType == MUSIC_MODULE_MOD)) return;
-
-    unsigned int positionInFrames = (unsigned int)(position*music.stream.sampleRate);
-
-    switch (music.ctxType)
-    {
-#if defined(SUPPORT_FILEFORMAT_WAV)
-        case MUSIC_AUDIO_WAV: drwav_seek_to_pcm_frame((drwav *)music.ctxData, positionInFrames); break;
-#endif
-#if defined(SUPPORT_FILEFORMAT_OGG)
-        case MUSIC_AUDIO_OGG: stb_vorbis_seek_frame((stb_vorbis *)music.ctxData, positionInFrames); break;
-#endif
-#if defined(SUPPORT_FILEFORMAT_MP3)
-        case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame((drmp3 *)music.ctxData, positionInFrames); break;
-#endif
-#if defined(SUPPORT_FILEFORMAT_QOA)
-        case MUSIC_AUDIO_QOA: qoaplay_seek_frame((qoaplay_desc *)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
-        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;
-
-    unsigned int subBufferSizeInFrames = music.stream.buffer->sizeInFrames/2;
-
-    // On first call of this function we lazily pre-allocated a temp buffer to read audio files/memory data in
-    int frameSize = music.stream.channels*music.stream.sampleSize/8;
-    unsigned int pcmSize = subBufferSizeInFrames*frameSize;
-
-    if (AUDIO.System.pcmBufferSize < pcmSize)
-    {
-        RL_FREE(AUDIO.System.pcmBuffer);
-        AUDIO.System.pcmBuffer = RL_CALLOC(1, pcmSize);
-        AUDIO.System.pcmBufferSize = pcmSize;
-    }
-
-    // Check both sub-buffers to check if they require refilling
-    for (int i = 0; i < 2; i++)
-    {
-        if ((music.stream.buffer != NULL) && !music.stream.buffer->isSubBufferProcessed[i]) continue; // No refilling required, move to next sub-buffer
-
-        unsigned int framesLeft = music.frameCount - music.stream.buffer->framesProcessed;  // Frames left to be processed
-        unsigned int framesToStream = 0;                 // Total frames to be streamed
-
-        if ((framesLeft >= subBufferSizeInFrames) || music.looping) framesToStream = subBufferSizeInFrames;
-        else framesToStream = framesLeft;
-
-        int frameCountStillNeeded = framesToStream;
-        int frameCountReadTotal = 0;
-
-        switch (music.ctxType)
-        {
-        #if defined(SUPPORT_FILEFORMAT_WAV)
-            case MUSIC_AUDIO_WAV:
-            {
-                if (music.stream.sampleSize == 16)
-                {
-                    while (true)
-                    {
-                        int frameCountRead = (int)drwav_read_pcm_frames_s16((drwav *)music.ctxData, frameCountStillNeeded, (short *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize));
-                        frameCountReadTotal += frameCountRead;
-                        frameCountStillNeeded -= frameCountRead;
-                        if (frameCountStillNeeded == 0) break;
-                        else drwav_seek_to_first_pcm_frame((drwav *)music.ctxData);
-                    }
-                }
-                else if (music.stream.sampleSize == 32)
-                {
-                    while (true)
-                    {
-                        int frameCountRead = (int)drwav_read_pcm_frames_f32((drwav *)music.ctxData, frameCountStillNeeded, (float *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize));
-                        frameCountReadTotal += frameCountRead;
-                        frameCountStillNeeded -= frameCountRead;
-                        if (frameCountStillNeeded == 0) break;
-                        else drwav_seek_to_first_pcm_frame((drwav *)music.ctxData);
-                    }
-                }
-            } break;
-        #endif
-        #if defined(SUPPORT_FILEFORMAT_OGG)
-            case MUSIC_AUDIO_OGG:
-            {
-                while (true)
-                {
-                    int frameCountRead = stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize), frameCountStillNeeded*music.stream.channels);
-                    frameCountReadTotal += frameCountRead;
-                    frameCountStillNeeded -= frameCountRead;
-                    if (frameCountStillNeeded == 0) break;
-                    else stb_vorbis_seek_start((stb_vorbis *)music.ctxData);
-                }
-            } break;
-        #endif
-        #if defined(SUPPORT_FILEFORMAT_MP3)
-            case MUSIC_AUDIO_MP3:
-            {
-                while (true)
-                {
-                    int frameCountRead = (int)drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, frameCountStillNeeded, (float *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize));
-                    frameCountReadTotal += frameCountRead;
-                    frameCountStillNeeded -= frameCountRead;
-                    if (frameCountStillNeeded == 0) break;
-                    else drmp3_seek_to_start_of_stream((drmp3 *)music.ctxData);
-                }
-            } break;
-        #endif
-        #if defined(SUPPORT_FILEFORMAT_QOA)
-            case MUSIC_AUDIO_QOA:
-            {
-                unsigned int frameCountRead = qoaplay_decode((qoaplay_desc *)music.ctxData, (float *)AUDIO.System.pcmBuffer, framesToStream);
-                frameCountReadTotal += frameCountRead;
-                /*
-                while (true)
-                {
-                    int frameCountRead = (int)qoaplay_decode((qoaplay_desc *)music.ctxData, (float *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize),  frameCountStillNeeded);
-                    frameCountReadTotal += frameCountRead;
-                    frameCountStillNeeded -= frameCountRead;
-                    if (frameCountStillNeeded == 0) break;
-                    else qoaplay_rewind((qoaplay_desc *)music.ctxData);
-                }
-                */
-            } break;
-        #endif
-        #if defined(SUPPORT_FILEFORMAT_FLAC)
-            case MUSIC_AUDIO_FLAC:
-            {
-                while (true)
-                {
-                    int frameCountRead = drflac_read_pcm_frames_s16((drflac *)music.ctxData, frameCountStillNeeded, (short *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize));
-                    frameCountReadTotal += frameCountRead;
-                    frameCountStillNeeded -= frameCountRead;
-                    if (frameCountStillNeeded == 0) break;
-                    else drflac__seek_to_first_frame((drflac *)music.ctxData);
-                }
-            } break;
-        #endif
-        #if defined(SUPPORT_FILEFORMAT_XM)
-            case MUSIC_MODULE_XM:
-            {
-                // NOTE: Internally we consider 2 channels generation, so sampleCount/2
-                if (AUDIO_DEVICE_FORMAT == ma_format_f32) jar_xm_generate_samples((jar_xm_context_t *)music.ctxData, (float *)AUDIO.System.pcmBuffer, framesToStream);
-                else if (AUDIO_DEVICE_FORMAT == ma_format_s16) jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)AUDIO.System.pcmBuffer, framesToStream);
-                else if (AUDIO_DEVICE_FORMAT == ma_format_u8) jar_xm_generate_samples_8bit((jar_xm_context_t *)music.ctxData, (char *)AUDIO.System.pcmBuffer, framesToStream);
-                //jar_xm_reset((jar_xm_context_t *)music.ctxData);
-
-            } break;
-        #endif
-        #if defined(SUPPORT_FILEFORMAT_MOD)
-            case MUSIC_MODULE_MOD:
-            {
-                // NOTE: 3rd parameter (nbsample) specify the number of stereo 16bits samples you want, so sampleCount/2
-                jar_mod_fillbuffer((jar_mod_context_t *)music.ctxData, (short *)AUDIO.System.pcmBuffer, framesToStream, 0);
-                //jar_mod_seek_start((jar_mod_context_t *)music.ctxData);
-
-            } break;
-        #endif
-            default: break;
-        }
-
-        UpdateAudioStream(music.stream, AUDIO.System.pcmBuffer, framesToStream);
-
-        music.stream.buffer->framesProcessed = music.stream.buffer->framesProcessed%music.frameCount;
-
-        if (framesLeft <= subBufferSizeInFrames)
-        {
-            if (!music.looping)
-            {
-                // Streaming is ending, we filled latest frames from input
-                StopMusicStream(music);
-                return;
-            }
-			if (music.loopPoint != 0.0f) SeekMusicStream(music, music.loopPoint);
-        }
-    }
-
-    // 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;
-            int framesProcessed = (int)music.stream.buffer->framesProcessed;
-            int subBufferSize = (int)music.stream.buffer->sizeInFrames/2;
-            int framesInFirstBuffer = music.stream.buffer->isSubBufferProcessed[0]? 0 : subBufferSize;
-            int framesInSecondBuffer = music.stream.buffer->isSubBufferProcessed[1]? 0 : subBufferSize;
-            int framesSentToMix = music.stream.buffer->frameCursorPos%subBufferSize;
-            int framesPlayed = (framesProcessed - framesInFirstBuffer - framesInSecondBuffer + framesSentToMix)%(int)music.frameCount;
-            if (framesPlayed < 0) framesPlayed += music.frameCount;
-            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;
-}
-
-// Checks if an audio stream is ready
-bool IsAudioStreamReady(AudioStream stream)
-{
-    return ((stream.buffer != NULL) &&    // Validate stream buffer
-            (stream.sampleRate > 0) &&    // Validate sample rate is supported
-            (stream.sampleSize > 0) &&    // Validate sample size is supported
-            (stream.channels > 0));       // Validate number of channels supported
-}
-
-// Unload audio stream and free memory
-void UnloadAudioStream(AudioStream stream)
-{
-    UnloadAudioBuffer(stream.buffer);
-
-    TRACELOG(LOG_INFO, "STREAM: Unloaded audio stream data from RAM");
-}
-
-// Update audio stream buffers with data
-// NOTE 1: Only updates one buffer of the stream source: dequeue -> update -> queue
-// NOTE 2: To dequeue a buffer it needs to be processed: IsAudioStreamProcessed()
-void UpdateAudioStream(AudioStream stream, const void *data, int frameCount)
-{
-    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);
-
-            // Total frames processed in buffer is always the complete size, filled with 0 if required
-            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 = (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);
-}
-
-// Remove processor from audio stream
-void DetachAudioStreamProcessor(AudioStream stream, AudioCallback process)
-{
-    ma_mutex_lock(&AUDIO.System.lock);
-
-    rAudioProcessor *processor = stream.buffer->processor;
-
-    while (processor)
-    {
-        rAudioProcessor *next = processor->next;
-        rAudioProcessor *prev = processor->prev;
-
-        if (processor->process == process)
-        {
-            if (stream.buffer->processor == processor) stream.buffer->processor = next;
-            if (prev) prev->next = next;
-            if (next) next->prev = prev;
-
-            RL_FREE(processor);
-        }
-
-        processor = next;
-    }
-
-    ma_mutex_unlock(&AUDIO.System.lock);
-}
-
-// Add processor to audio pipeline. Order of processors is important
-// Works the same way as {Attach,Detach}AudioStreamProcessor() functions, except
-// these two work on the already mixed output just before sending it to the sound hardware
-void AttachAudioMixedProcessor(AudioCallback process)
-{
-    ma_mutex_lock(&AUDIO.System.lock);
-
-    rAudioProcessor *processor = (rAudioProcessor *)RL_CALLOC(1, sizeof(rAudioProcessor));
-    processor->process = process;
-
-    rAudioProcessor *last = AUDIO.mixedProcessor;
-
-    while (last && last->next)
-    {
-        last = last->next;
-    }
-    if (last)
-    {
-        processor->prev = last;
-        last->next = processor;
-    }
-    else AUDIO.mixedProcessor = processor;
-
-    ma_mutex_unlock(&AUDIO.System.lock);
-}
-
-// Remove processor from audio pipeline
-void DetachAudioMixedProcessor(AudioCallback process)
-{
-    ma_mutex_lock(&AUDIO.System.lock);
-
-    rAudioProcessor *processor = AUDIO.mixedProcessor;
-
-    while (processor)
-    {
-        rAudioProcessor *next = processor->next;
-        rAudioProcessor *prev = processor->prev;
-
-        if (processor->process == process)
-        {
-            if (AUDIO.mixedProcessor == processor) AUDIO.mixedProcessor = next;
-            if (prev) prev->next = next;
-            if (next) next->prev = prev;
-
-            RL_FREE(processor);
-        }
-
-        processor = next;
-    }
-
-    ma_mutex_unlock(&AUDIO.System.lock);
-}
-
-
-//----------------------------------------------------------------------------------
-// Module 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 half of the buffer that are processed
-        //    Unprocessed halves must keep their audio data in-tact
-        if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
-        {
-            if (framesRead >= frameCount) break;
-        }
-        else
-        {
-            if (isSubBufferProcessed[currentSubBufferIndex]) break;
-        }
-
-        ma_uint32 totalFramesRemaining = (frameCount - framesRead);
-        if (totalFramesRemaining == 0) break;
-
-        ma_uint32 framesRemainingInOutputBuffer;
-        if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
-        {
-            framesRemainingInOutputBuffer = audioBuffer->sizeInFrames - audioBuffer->frameCursorPos;
-        }
-        else
-        {
-            ma_uint32 firstFrameIndexOfThisSubBuffer = subBufferSizeInFrames*currentSubBufferIndex;
-            framesRemainingInOutputBuffer = subBufferSizeInFrames - (audioBuffer->frameCursorPos - firstFrameIndexOfThisSubBuffer);
-        }
-
-        ma_uint32 framesToRead = totalFramesRemaining;
-        if (framesToRead > framesRemainingInOutputBuffer) framesToRead = framesRemainingInOutputBuffer;
-
-        memcpy((unsigned char *)framesOut + (framesRead*frameSizeInBytes), audioBuffer->data + (audioBuffer->frameCursorPos*frameSizeInBytes), framesToRead*frameSizeInBytes);
-        audioBuffer->frameCursorPos = (audioBuffer->frameCursorPos + framesToRead)%audioBuffer->sizeInFrames;
-        framesRead += framesToRead;
-
-        // If we've read to the end of the buffer, mark it as processed
-        if (framesToRead == framesRemainingInOutputBuffer)
-        {
-            audioBuffer->isSubBufferProcessed[currentSubBufferIndex] = true;
-            isSubBufferProcessed[currentSubBufferIndex] = true;
-
-            currentSubBufferIndex = (currentSubBufferIndex + 1)%2;
-
-            // We need to break from this loop if we're not looping
-            if (!audioBuffer->looping)
-            {
-                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 a non-looping sound has finished playback.
-        if (audioBuffer->usage != AUDIO_BUFFER_USAGE_STATIC) framesRead += totalFramesRemaining;
-    }
-
-    return framesRead;
-}
-
-// Reads audio data from an AudioBuffer object in device format. Returned data will be in a format appropriate for mixing.
-static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, float *framesOut, ma_uint32 frameCount)
-{
-    // What's going on here is that we're continuously converting data from the AudioBuffer's internal format to the mixing format, which
-    // should be defined by the output format of the data converter. We do this until frameCount frames have been output. The important
-    // detail to remember here is that we never, ever attempt to read more input data than is required for the specified number of output
-    // frames. This can be achieved with ma_data_converter_get_required_input_frame_count().
-    ma_uint8 inputBuffer[4096] = { 0 };
-    ma_uint32 inputBufferFrameCap = sizeof(inputBuffer)/ma_get_bytes_per_frame(audioBuffer->converter.formatIn, audioBuffer->converter.channelsIn);
-
-    ma_uint32 totalOutputFramesProcessed = 0;
-    while (totalOutputFramesProcessed < frameCount)
-    {
-        ma_uint64 outputFramesToProcessThisIteration = frameCount - totalOutputFramesProcessed;
-        ma_uint64 inputFramesToProcessThisIteration = 0;
-
-        (void)ma_data_converter_get_required_input_frame_count(&audioBuffer->converter, outputFramesToProcessThisIteration, &inputFramesToProcessThisIteration);
-        if (inputFramesToProcessThisIteration > inputBufferFrameCap)
-        {
-            inputFramesToProcessThisIteration = inputBufferFrameCap;
-        }
-
-        float *runningFramesOut = framesOut + (totalOutputFramesProcessed*audioBuffer->converter.channelsOut);
-
-        /* At this point we can convert the data to our mixing format. */
-        ma_uint64 inputFramesProcessedThisIteration = ReadAudioBufferFramesInInternalFormat(audioBuffer, inputBuffer, (ma_uint32)inputFramesToProcessThisIteration);    /* 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;
-            }
-        }
-    }
-
-    rAudioProcessor *processor = AUDIO.mixedProcessor;
-    while (processor)
-    {
-        processor->process(pFramesOut, frameCount);
-        processor = processor->next;
-    }
-
-    ma_mutex_unlock(&AUDIO.System.lock);
-}
-
-// Main mixing function, pretty simple in this project, just an accumulation
-// NOTE: framesOut is both an input and an output, it is initially filled with zeros outside of this function
-static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, AudioBuffer *buffer)
-{
-    const float localVolume = buffer->volume;
-    const ma_uint32 channels = AUDIO.System.device.playback.channels;
-
-    if (channels == 2)  // We consider panning
-    {
-        const float 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