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diff --git a/raylib/src/raudio.c b/raylib/src/raudio.c
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+/**********************************************************************************************
+*
+*   raudio v1.1 - A simple and easy-to-use audio library based on miniaudio
+*
+*   FEATURES:
+*       - Manage audio device (init/close)
+*       - Manage raw audio context
+*       - Manage mixing channels
+*       - Load and unload audio files
+*       - Format wave data (sample rate, size, channels)
+*       - Play/Stop/Pause/Resume loaded audio
+*
+*   CONFIGURATION:
+*       #define SUPPORT_MODULE_RAUDIO
+*           raudio module is included in the build
+*
+*       #define RAUDIO_STANDALONE
+*           Define to use the module as standalone library (independently of raylib).
+*           Required types and functions are defined in the same module.
+*
+*       #define SUPPORT_FILEFORMAT_WAV
+*       #define SUPPORT_FILEFORMAT_OGG
+*       #define SUPPORT_FILEFORMAT_MP3
+*       #define SUPPORT_FILEFORMAT_QOA
+*       #define SUPPORT_FILEFORMAT_FLAC
+*       #define SUPPORT_FILEFORMAT_XM
+*       #define SUPPORT_FILEFORMAT_MOD
+*           Selected desired fileformats to be supported for loading. Some of those formats are
+*           supported by default, to remove support, just comment unrequired #define in this module
+*
+*   DEPENDENCIES:
+*       miniaudio.h  - Audio device management lib (https://github.com/mackron/miniaudio)
+*       stb_vorbis.h - Ogg audio files loading (http://www.nothings.org/stb_vorbis/)
+*       dr_wav.h     - WAV audio files loading (http://github.com/mackron/dr_libs)
+*       dr_mp3.h     - MP3 audio file loading (https://github.com/mackron/dr_libs)
+*       dr_flac.h    - FLAC audio file loading (https://github.com/mackron/dr_libs)
+*       jar_xm.h     - XM module file loading
+*       jar_mod.h    - MOD audio file loading
+*
+*   CONTRIBUTORS:
+*       David Reid (github: @mackron) (Nov. 2017):
+*           - Complete port to miniaudio library
+*
+*       Joshua Reisenauer (github: @kd7tck) (2015):
+*           - XM audio module support (jar_xm)
+*           - MOD audio module support (jar_mod)
+*           - Mixing channels support
+*           - Raw audio context support
+*
+*
+*   LICENSE: zlib/libpng
+*
+*   Copyright (c) 2013-2024 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
+#define MA_NO_RESOURCE_MANAGER
+#define MA_NO_NODE_GRAPH
+#define MA_NO_ENGINE
+#define MA_NO_GENERATION
+
+// Threading model: Default: [0] COINIT_MULTITHREADED: COM calls objects on any thread (free threading)
+#define MA_COINIT_VALUE  2              // [2] COINIT_APARTMENTTHREADED: Each object has its own thread (apartment model)
+
+#define MINIAUDIO_IMPLEMENTATION
+//#define MA_DEBUG_OUTPUT
+#include "external/miniaudio.h"         // Audio device initialization and management
+#undef PlaySound                        // Win32 API: windows.h > mmsystem.h defines PlaySound macro
+
+#include <stdlib.h>                     // Required for: malloc(), free()
+#include <stdio.h>                      // Required for: FILE, fopen(), fclose(), fread()
+#include <string.h>                     // Required for: strcmp() [Used in IsFileExtension(), LoadWaveFromMemory(), LoadMusicStreamFromMemory()]
+
+#if defined(RAUDIO_STANDALONE)
+    #ifndef TRACELOG
+        #define TRACELOG(level, ...)    printf(__VA_ARGS__)
+    #endif
+
+    // Allow custom memory allocators
+    #ifndef RL_MALLOC
+        #define RL_MALLOC(sz)           malloc(sz)
+    #endif
+    #ifndef RL_CALLOC
+        #define RL_CALLOC(n,sz)         calloc(n,sz)
+    #endif
+    #ifndef RL_REALLOC
+        #define RL_REALLOC(ptr,sz)      realloc(ptr,sz)
+    #endif
+    #ifndef RL_FREE
+        #define RL_FREE(ptr)            free(ptr)
+    #endif
+#endif
+
+#if defined(SUPPORT_FILEFORMAT_WAV)
+    #define DRWAV_MALLOC RL_MALLOC
+    #define DRWAV_REALLOC RL_REALLOC
+    #define DRWAV_FREE RL_FREE
+
+    #define DR_WAV_IMPLEMENTATION
+    #include "external/dr_wav.h"        // WAV loading functions
+#endif
+
+#if defined(SUPPORT_FILEFORMAT_OGG)
+    // TODO: Remap stb_vorbis malloc()/free() calls to RL_MALLOC/RL_FREE
+    #include "external/stb_vorbis.c"    // OGG loading functions
+#endif
+
+#if defined(SUPPORT_FILEFORMAT_MP3)
+    #define DRMP3_MALLOC RL_MALLOC
+    #define DRMP3_REALLOC RL_REALLOC
+    #define DRMP3_FREE RL_FREE
+
+    #define DR_MP3_IMPLEMENTATION
+    #include "external/dr_mp3.h"        // MP3 loading functions
+#endif
+
+#if defined(SUPPORT_FILEFORMAT_QOA)
+    #define QOA_MALLOC RL_MALLOC
+    #define QOA_FREE RL_FREE
+
+    #if defined(_MSC_VER)           // Disable some MSVC warning
+        #pragma warning(push)
+        #pragma warning(disable : 4018)
+        #pragma warning(disable : 4267)
+        #pragma warning(disable : 4244)
+    #endif
+
+    #define QOA_IMPLEMENTATION
+    #include "external/qoa.h"           // QOA loading and saving functions
+    #include "external/qoaplay.c"       // QOA stream playing helper functions
+
+    #if defined(_MSC_VER)
+        #pragma warning(pop)        // Disable MSVC warning suppression
+    #endif
+#endif
+
+#if defined(SUPPORT_FILEFORMAT_FLAC)
+    #define DRFLAC_MALLOC RL_MALLOC
+    #define DRFLAC_REALLOC RL_REALLOC
+    #define DRFLAC_FREE RL_FREE
+
+    #define DR_FLAC_IMPLEMENTATION
+    #define DR_FLAC_NO_WIN32_IO
+    #include "external/dr_flac.h"       // FLAC loading functions
+#endif
+
+#if defined(SUPPORT_FILEFORMAT_XM)
+    #define JARXM_MALLOC RL_MALLOC
+    #define JARXM_FREE RL_FREE
+
+    #if defined(_MSC_VER)           // Disable some MSVC warning
+        #pragma warning(push)
+        #pragma warning(disable : 4244)
+    #endif
+
+    #define JAR_XM_IMPLEMENTATION
+    #include "external/jar_xm.h"        // XM loading functions
+
+    #if defined(_MSC_VER)
+        #pragma warning(pop)        // Disable MSVC warning suppression
+    #endif
+#endif
+
+#if defined(SUPPORT_FILEFORMAT_MOD)
+    #define JARMOD_MALLOC RL_MALLOC
+    #define JARMOD_FREE RL_FREE
+
+    #define JAR_MOD_IMPLEMENTATION
+    #include "external/jar_mod.h"       // MOD loading functions
+#endif
+
+//----------------------------------------------------------------------------------
+// Defines and Macros
+//----------------------------------------------------------------------------------
+#ifndef AUDIO_DEVICE_FORMAT
+    #define AUDIO_DEVICE_FORMAT    ma_format_f32    // Device output format (float-32bit)
+#endif
+#ifndef AUDIO_DEVICE_CHANNELS
+    #define AUDIO_DEVICE_CHANNELS              2    // Device output channels: stereo
+#endif
+#ifndef AUDIO_DEVICE_SAMPLE_RATE
+    #define AUDIO_DEVICE_SAMPLE_RATE           0    // Device output sample rate
+#endif
+
+#ifndef MAX_AUDIO_BUFFER_POOL_CHANNELS
+    #define MAX_AUDIO_BUFFER_POOL_CHANNELS    16    // Audio pool channels
+#endif
+
+//----------------------------------------------------------------------------------
+// Types and Structures Definition
+//----------------------------------------------------------------------------------
+#if defined(RAUDIO_STANDALONE)
+// Trace log level
+// NOTE: Organized by priority level
+typedef enum {
+    LOG_ALL = 0,        // Display all logs
+    LOG_TRACE,          // Trace logging, intended for internal use only
+    LOG_DEBUG,          // Debug logging, used for internal debugging, it should be disabled on release builds
+    LOG_INFO,           // Info logging, used for program execution info
+    LOG_WARNING,        // Warning logging, used on recoverable failures
+    LOG_ERROR,          // Error logging, used on unrecoverable failures
+    LOG_FATAL,          // Fatal logging, used to abort program: exit(EXIT_FAILURE)
+    LOG_NONE            // Disable logging
+} TraceLogLevel;
+#endif
+
+// Music context type
+// NOTE: Depends on data structure provided by the library
+// in charge of reading the different file types
+typedef enum {
+    MUSIC_AUDIO_NONE = 0,   // No audio context loaded
+    MUSIC_AUDIO_WAV,        // WAV audio context
+    MUSIC_AUDIO_OGG,        // OGG audio context
+    MUSIC_AUDIO_FLAC,       // FLAC audio context
+    MUSIC_AUDIO_MP3,        // MP3 audio context
+    MUSIC_AUDIO_QOA,        // QOA audio context
+    MUSIC_MODULE_XM,        // XM module audio context
+    MUSIC_MODULE_MOD        // MOD module audio context
+} MusicContextType;
+
+// NOTE: Different logic is used when feeding data to the playback device
+// depending on whether data is streamed (Music vs Sound)
+typedef enum {
+    AUDIO_BUFFER_USAGE_STATIC = 0,
+    AUDIO_BUFFER_USAGE_STREAM
+} AudioBufferUsage;
+
+// Audio buffer struct
+struct rAudioBuffer {
+    ma_data_converter converter;    // Audio data converter
+
+    AudioCallback callback;         // Audio buffer callback for buffer filling on audio threads
+    rAudioProcessor *processor;     // Audio processor
+
+    float volume;                   // Audio buffer volume
+    float pitch;                    // Audio buffer pitch
+    float pan;                      // Audio buffer pan (0.0f to 1.0f)
+
+    bool playing;                   // Audio buffer state: AUDIO_PLAYING
+    bool paused;                    // Audio buffer state: AUDIO_PAUSED
+    bool looping;                   // Audio buffer looping, default to true for AudioStreams
+    int usage;                      // Audio buffer usage mode: STATIC or STREAM
+
+    bool isSubBufferProcessed[2];   // SubBuffer processed (virtual double buffer)
+    unsigned int sizeInFrames;      // Total buffer size in frames
+    unsigned int frameCursorPos;    // Frame cursor position
+    unsigned int framesProcessed;   // Total frames processed in this buffer (required for play timing)
+
+    unsigned char *data;            // Data buffer, on music stream keeps filling
+
+    rAudioBuffer *next;             // Next audio buffer on the list
+    rAudioBuffer *prev;             // Previous audio buffer on the list
+};
+
+// Audio processor struct
+// NOTE: Useful to apply effects to an AudioBuffer
+struct rAudioProcessor {
+    AudioCallback process;          // Processor callback function
+    rAudioProcessor *next;          // Next audio processor on the list
+    rAudioProcessor *prev;          // Previous audio processor on the list
+};
+
+#define AudioBuffer rAudioBuffer    // HACK: To avoid CoreAudio (macOS) symbol collision
+
+// Audio data context
+typedef struct AudioData {
+    struct {
+        ma_context context;         // miniaudio context data
+        ma_device device;           // miniaudio device
+        ma_mutex lock;              // miniaudio mutex lock
+        bool isReady;               // Check if audio device is ready
+        size_t pcmBufferSize;       // Pre-allocated buffer size
+        void *pcmBuffer;            // Pre-allocated buffer to read audio data from file/memory
+    } System;
+    struct {
+        AudioBuffer *first;         // Pointer to first AudioBuffer in the list
+        AudioBuffer *last;          // Pointer to last AudioBuffer in the list
+        int defaultSize;            // Default audio buffer size for audio streams
+    } Buffer;
+    rAudioProcessor *mixedProcessor;
+} AudioData;
+
+//----------------------------------------------------------------------------------
+// Global Variables Definition
+//----------------------------------------------------------------------------------
+static AudioData AUDIO = {          // Global AUDIO context
+
+    // NOTE: Music buffer size is defined by number of samples, independent of sample size and channels number
+    // After some math, considering a sampleRate of 48000, a buffer refill rate of 1/60 seconds and a
+    // standard double-buffering system, a 4096 samples buffer has been chosen, it should be enough
+    // In case of music-stalls, just increase this number
+    .Buffer.defaultSize = 0,
+    .mixedProcessor = NULL
+};
+
+//----------------------------------------------------------------------------------
+// Module specific Functions Declaration
+//----------------------------------------------------------------------------------
+static void OnLog(void *pUserData, ma_uint32 level, const char *pMessage);
+
+// Reads audio data from an AudioBuffer object in internal/device formats
+static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer, void *framesOut, ma_uint32 frameCount);
+static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, float *framesOut, ma_uint32 frameCount);
+
+static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount);
+static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, AudioBuffer *buffer);
+
+static bool IsAudioBufferPlayingInLockedState(AudioBuffer *buffer);
+static void StopAudioBufferInLockedState(AudioBuffer *buffer);
+static void UpdateAudioStreamInLockedState(AudioStream stream, const void *data, int frameCount);
+
+#if defined(RAUDIO_STANDALONE)
+static bool IsFileExtension(const char *fileName, const char *ext); // Check file extension
+static const char *GetFileExtension(const char *fileName);          // Get pointer to extension for a filename string (includes the dot: .png)
+static const char *GetFileName(const char *filePath);               // Get pointer to filename for a path string
+static const char *GetFileNameWithoutExt(const char *filePath);     // Get filename string without extension (uses static string)
+
+static unsigned char *LoadFileData(const char *fileName, int *dataSize);    // Load file data as byte array (read)
+static bool SaveFileData(const char *fileName, void *data, int dataSize);   // Save data to file from byte array (write)
+static bool SaveFileText(const char *fileName, char *text);         // Save text data to file (write), string must be '\0' terminated
+#endif
+
+//----------------------------------------------------------------------------------
+// AudioBuffer management functions declaration
+// NOTE: Those functions are not exposed by raylib... for the moment
+//----------------------------------------------------------------------------------
+AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage);
+void UnloadAudioBuffer(AudioBuffer *buffer);
+
+bool IsAudioBufferPlaying(AudioBuffer *buffer);
+void PlayAudioBuffer(AudioBuffer *buffer);
+void StopAudioBuffer(AudioBuffer *buffer);
+void PauseAudioBuffer(AudioBuffer *buffer);
+void ResumeAudioBuffer(AudioBuffer *buffer);
+void SetAudioBufferVolume(AudioBuffer *buffer, float volume);
+void SetAudioBufferPitch(AudioBuffer *buffer, float pitch);
+void SetAudioBufferPan(AudioBuffer *buffer, float pan);
+void TrackAudioBuffer(AudioBuffer *buffer);
+void UntrackAudioBuffer(AudioBuffer *buffer);
+
+
+//----------------------------------------------------------------------------------
+// Module Functions Definition - Audio Device initialization and Closing
+//----------------------------------------------------------------------------------
+// Initialize audio device
+void InitAudioDevice(void)
+{
+    // Init audio context
+    ma_context_config ctxConfig = ma_context_config_init();
+    ma_log_callback_init(OnLog, NULL);
+
+    ma_result result = ma_context_init(NULL, 0, &ctxConfig, &AUDIO.System.context);
+    if (result != MA_SUCCESS)
+    {
+        TRACELOG(LOG_WARNING, "AUDIO: Failed to initialize context");
+        return;
+    }
+
+    // Init audio device
+    // NOTE: Using the default device. Format is floating point because it simplifies mixing
+    ma_device_config config = ma_device_config_init(ma_device_type_playback);
+    config.playback.pDeviceID = NULL;  // NULL for the default playback AUDIO.System.device
+    config.playback.format = AUDIO_DEVICE_FORMAT;
+    config.playback.channels = AUDIO_DEVICE_CHANNELS;
+    config.capture.pDeviceID = NULL;  // NULL for the default capture AUDIO.System.device
+    config.capture.format = ma_format_s16;
+    config.capture.channels = 1;
+    config.sampleRate = AUDIO_DEVICE_SAMPLE_RATE;
+    config.dataCallback = OnSendAudioDataToDevice;
+    config.pUserData = NULL;
+
+    result = ma_device_init(&AUDIO.System.context, &config, &AUDIO.System.device);
+    if (result != MA_SUCCESS)
+    {
+        TRACELOG(LOG_WARNING, "AUDIO: Failed to initialize playback device");
+        ma_context_uninit(&AUDIO.System.context);
+        return;
+    }
+
+    // Mixing happens on a separate thread which means we need to synchronize. I'm using a mutex here to make things simple, but may
+    // want to look at something a bit smarter later on to keep everything real-time, if that's necessary
+    if (ma_mutex_init(&AUDIO.System.lock) != MA_SUCCESS)
+    {
+        TRACELOG(LOG_WARNING, "AUDIO: Failed to create mutex for mixing");
+        ma_device_uninit(&AUDIO.System.device);
+        ma_context_uninit(&AUDIO.System.context);
+        return;
+    }
+
+    // Keep the device running the whole time. May want to consider doing something a bit smarter and only have the device running
+    // while there's at least one sound being played
+    result = ma_device_start(&AUDIO.System.device);
+    if (result != MA_SUCCESS)
+    {
+        TRACELOG(LOG_WARNING, "AUDIO: Failed to start playback device");
+        ma_device_uninit(&AUDIO.System.device);
+        ma_context_uninit(&AUDIO.System.context);
+        return;
+    }
+
+    TRACELOG(LOG_INFO, "AUDIO: Device initialized successfully");
+    TRACELOG(LOG_INFO, "    > Backend:       miniaudio | %s", ma_get_backend_name(AUDIO.System.context.backend));
+    TRACELOG(LOG_INFO, "    > Format:        %s -> %s", ma_get_format_name(AUDIO.System.device.playback.format), ma_get_format_name(AUDIO.System.device.playback.internalFormat));
+    TRACELOG(LOG_INFO, "    > Channels:      %d -> %d", AUDIO.System.device.playback.channels, AUDIO.System.device.playback.internalChannels);
+    TRACELOG(LOG_INFO, "    > Sample rate:   %d -> %d", AUDIO.System.device.sampleRate, AUDIO.System.device.playback.internalSampleRate);
+    TRACELOG(LOG_INFO, "    > Periods size:  %d", AUDIO.System.device.playback.internalPeriodSizeInFrames*AUDIO.System.device.playback.internalPeriods);
+
+    AUDIO.System.isReady = true;
+}
+
+// Close the audio device for all contexts
+void CloseAudioDevice(void)
+{
+    if (AUDIO.System.isReady)
+    {
+        ma_mutex_uninit(&AUDIO.System.lock);
+        ma_device_uninit(&AUDIO.System.device);
+        ma_context_uninit(&AUDIO.System.context);
+
+        AUDIO.System.isReady = false;
+        RL_FREE(AUDIO.System.pcmBuffer);
+        AUDIO.System.pcmBuffer = NULL;
+        AUDIO.System.pcmBufferSize = 0;
+
+        TRACELOG(LOG_INFO, "AUDIO: Device closed successfully");
+    }
+    else TRACELOG(LOG_WARNING, "AUDIO: Device could not be closed, not currently initialized");
+}
+
+// Check if device has been initialized successfully
+bool IsAudioDeviceReady(void)
+{
+    return AUDIO.System.isReady;
+}
+
+// Set master volume (listener)
+void SetMasterVolume(float volume)
+{
+    ma_device_set_master_volume(&AUDIO.System.device, volume);
+}
+
+// Get master volume (listener)
+float GetMasterVolume(void)
+{
+    float volume = 0.0f;
+    ma_device_get_master_volume(&AUDIO.System.device, &volume);
+    return volume;
+}
+
+//----------------------------------------------------------------------------------
+// Module Functions Definition - Audio Buffer management
+//----------------------------------------------------------------------------------
+
+// Initialize a new audio buffer (filled with silence)
+AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage)
+{
+    AudioBuffer *audioBuffer = (AudioBuffer *)RL_CALLOC(1, sizeof(AudioBuffer));
+
+    if (audioBuffer == NULL)
+    {
+        TRACELOG(LOG_WARNING, "AUDIO: Failed to allocate memory for buffer");
+        return NULL;
+    }
+
+    if (sizeInFrames > 0) audioBuffer->data = 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)
+    {
+        UntrackAudioBuffer(buffer);
+        ma_data_converter_uninit(&buffer->converter, NULL);
+        RL_FREE(buffer->data);
+        RL_FREE(buffer);
+    }
+}
+
+// Check if an audio buffer is playing from a program state without lock
+bool IsAudioBufferPlaying(AudioBuffer *buffer)
+{
+    bool result = false;
+    ma_mutex_lock(&AUDIO.System.lock);
+    result = IsAudioBufferPlayingInLockedState(buffer);
+    ma_mutex_unlock(&AUDIO.System.lock);
+    return result;
+}
+
+// Play an audio buffer
+// NOTE: Buffer is restarted to the start
+// Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position should be maintained
+void PlayAudioBuffer(AudioBuffer *buffer)
+{
+    if (buffer != NULL)
+    {
+        ma_mutex_lock(&AUDIO.System.lock);
+        buffer->playing = true;
+        buffer->paused = false;
+        buffer->frameCursorPos = 0;
+        ma_mutex_unlock(&AUDIO.System.lock);
+    }
+}
+
+// Stop an audio buffer from a program state without lock
+void StopAudioBuffer(AudioBuffer *buffer)
+{
+    ma_mutex_lock(&AUDIO.System.lock);
+    StopAudioBufferInLockedState(buffer);
+    ma_mutex_unlock(&AUDIO.System.lock);
+}
+
+// Pause an audio buffer
+void PauseAudioBuffer(AudioBuffer *buffer)
+{
+    if (buffer != NULL)
+    {
+        ma_mutex_lock(&AUDIO.System.lock);
+        buffer->paused = true;
+        ma_mutex_unlock(&AUDIO.System.lock);
+    }
+}
+
+// Resume an audio buffer
+void ResumeAudioBuffer(AudioBuffer *buffer)
+{
+    if (buffer != NULL)
+    {
+        ma_mutex_lock(&AUDIO.System.lock);
+        buffer->paused = false;
+        ma_mutex_unlock(&AUDIO.System.lock);
+    }
+}
+
+// Set volume for an audio buffer
+void SetAudioBufferVolume(AudioBuffer *buffer, float volume)
+{
+    if (buffer != NULL)
+    {
+        ma_mutex_lock(&AUDIO.System.lock);
+        buffer->volume = volume;
+        ma_mutex_unlock(&AUDIO.System.lock);
+    }
+}
+
+// Set pitch for an audio buffer
+void SetAudioBufferPitch(AudioBuffer *buffer, float pitch)
+{
+    if ((buffer != NULL) && (pitch > 0.0f))
+    {
+        ma_mutex_lock(&AUDIO.System.lock);
+        // Pitching is just an adjustment of the sample rate
+        // Note that this changes the duration of the sound:
+        //  - higher pitches will make the sound faster
+        //  - lower pitches make it slower
+        ma_uint32 outputSampleRate = (ma_uint32)((float)buffer->converter.sampleRateOut/pitch);
+        ma_data_converter_set_rate(&buffer->converter, buffer->converter.sampleRateIn, outputSampleRate);
+
+        buffer->pitch = pitch;
+        ma_mutex_unlock(&AUDIO.System.lock);
+    }
+}
+
+// Set pan for an audio buffer
+void SetAudioBufferPan(AudioBuffer *buffer, float pan)
+{
+    if (pan < 0.0f) pan = 0.0f;
+    else if (pan > 1.0f) pan = 1.0f;
+
+    if (buffer != NULL)
+    {
+        ma_mutex_lock(&AUDIO.System.lock);
+        buffer->pan = pan;
+        ma_mutex_unlock(&AUDIO.System.lock);
+    }
+}
+
+// Track audio buffer to linked list next position
+void TrackAudioBuffer(AudioBuffer *buffer)
+{
+    ma_mutex_lock(&AUDIO.System.lock);
+    {
+        if (AUDIO.Buffer.first == NULL) AUDIO.Buffer.first = buffer;
+        else
+        {
+            AUDIO.Buffer.last->next = buffer;
+            buffer->prev = AUDIO.Buffer.last;
+        }
+
+        AUDIO.Buffer.last = buffer;
+    }
+    ma_mutex_unlock(&AUDIO.System.lock);
+}
+
+// Untrack audio buffer from linked list
+void UntrackAudioBuffer(AudioBuffer *buffer)
+{
+    ma_mutex_lock(&AUDIO.System.lock);
+    {
+        if (buffer->prev == NULL) AUDIO.Buffer.first = buffer->next;
+        else buffer->prev->next = buffer->next;
+
+        if (buffer->next == NULL) AUDIO.Buffer.last = buffer->prev;
+        else buffer->next->prev = buffer->prev;
+
+        buffer->prev = NULL;
+        buffer->next = NULL;
+    }
+    ma_mutex_unlock(&AUDIO.System.lock);
+}
+
+//----------------------------------------------------------------------------------
+// Module Functions Definition - Sounds loading and playing (.WAV)
+//----------------------------------------------------------------------------------
+
+// Load wave data from file
+Wave LoadWave(const char *fileName)
+{
+    Wave wave = { 0 };
+
+    // Loading file to memory
+    int dataSize = 0;
+    unsigned char *fileData = LoadFileData(fileName, &dataSize);
+
+    // Loading wave from memory data
+    if (fileData != NULL) wave = LoadWaveFromMemory(GetFileExtension(fileName), fileData, dataSize);
+
+    UnloadFileData(fileData);
+
+    return wave;
+}
+
+// Load wave from memory buffer, fileType refers to extension: i.e. ".wav"
+// WARNING: File extension must be provided in lower-case
+Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int dataSize)
+{
+    Wave wave = { 0 };
+
+    if (false) { }
+#if defined(SUPPORT_FILEFORMAT_WAV)
+    else if ((strcmp(fileType, ".wav") == 0) || (strcmp(fileType, ".WAV") == 0))
+    {
+        drwav wav = { 0 };
+        bool success = drwav_init_memory(&wav, fileData, dataSize, NULL);
+
+        if (success)
+        {
+            wave.frameCount = (unsigned int)wav.totalPCMFrameCount;
+            wave.sampleRate = wav.sampleRate;
+            wave.sampleSize = 16;
+            wave.channels = wav.channels;
+            wave.data = (short *)RL_MALLOC(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 valid (data loaded and parameters)
+bool IsWaveValid(Wave wave)
+{
+    bool result = false;
+
+    if ((wave.data != NULL) &&      // Validate wave data available
+        (wave.frameCount > 0) &&    // Validate frame count
+        (wave.sampleRate > 0) &&    // Validate sample rate is supported
+        (wave.sampleSize > 0) &&    // Validate sample size is supported
+        (wave.channels > 0)) result = true; // Validate number of channels supported
+
+    return result;
+}
+
+// Load sound from file
+// NOTE: The entire file is loaded to memory to be played (no-streaming)
+Sound LoadSound(const char *fileName)
+{
+    Wave wave = LoadWave(fileName);
+
+    Sound sound = LoadSoundFromWave(wave);
+
+    UnloadWave(wave);       // Sound is loaded, we can unload wave
+
+    return sound;
+}
+
+// Load sound from wave data
+// NOTE: Wave data must be unallocated manually
+Sound LoadSoundFromWave(Wave wave)
+{
+    Sound sound = { 0 };
+
+    if (wave.data != NULL)
+    {
+        // When using miniaudio we need to do our own mixing
+        // To simplify this we need convert the format of each sound to be consistent with
+        // the format used to open the playback AUDIO.System.device. We can do this two ways:
+        //
+        //   1) Convert the whole sound in one go at load time (here)
+        //   2) Convert the audio data in chunks at mixing time
+        //
+        // First option has been selected, format conversion is done on the loading stage
+        // The downside is that it uses more memory if the original sound is u8 or s16
+        ma_format formatIn = ((wave.sampleSize == 8)? ma_format_u8 : ((wave.sampleSize == 16)? ma_format_s16 : ma_format_f32));
+        ma_uint32 frameCountIn = wave.frameCount;
+
+        ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, NULL, frameCountIn, formatIn, wave.channels, wave.sampleRate);
+        if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed to get frame count for format conversion");
+
+        AudioBuffer *audioBuffer = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, frameCount, AUDIO_BUFFER_USAGE_STATIC);
+        if (audioBuffer == NULL)
+        {
+            TRACELOG(LOG_WARNING, "SOUND: Failed to create buffer");
+            return sound; // early return to avoid dereferencing the audioBuffer null pointer
+        }
+
+        frameCount = (ma_uint32)ma_convert_frames(audioBuffer->data, frameCount, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, wave.data, frameCountIn, formatIn, wave.channels, wave.sampleRate);
+        if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed format conversion");
+
+        sound.frameCount = frameCount;
+        sound.stream.sampleRate = AUDIO.System.device.sampleRate;
+        sound.stream.sampleSize = 32;
+        sound.stream.channels = AUDIO_DEVICE_CHANNELS;
+        sound.stream.buffer = audioBuffer;
+    }
+
+    return sound;
+}
+
+// Clone sound from existing sound data, clone does not own wave data
+// NOTE: Wave data must be unallocated manually and will be shared across all clones
+Sound LoadSoundAlias(Sound source)
+{
+    Sound sound = { 0 };
+
+    if (source.stream.buffer->data != NULL)
+    {
+        AudioBuffer *audioBuffer = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, 0, AUDIO_BUFFER_USAGE_STATIC);
+
+        if (audioBuffer == NULL)
+        {
+            TRACELOG(LOG_WARNING, "SOUND: Failed to create buffer");
+            return sound; // Early return to avoid dereferencing the audioBuffer null pointer
+        }
+
+        audioBuffer->sizeInFrames = source.stream.buffer->sizeInFrames;
+        audioBuffer->volume = source.stream.buffer->volume;
+        audioBuffer->data = source.stream.buffer->data;
+
+        sound.frameCount = source.frameCount;
+        sound.stream.sampleRate = AUDIO.System.device.sampleRate;
+        sound.stream.sampleSize = 32;
+        sound.stream.channels = AUDIO_DEVICE_CHANNELS;
+        sound.stream.buffer = audioBuffer;
+    }
+
+    return sound;
+}
+
+
+// Checks if a sound is valid (data loaded and buffers initialized)
+bool IsSoundValid(Sound sound)
+{
+    bool result = false;
+
+    if ((sound.frameCount > 0) &&           // Validate frame count
+        (sound.stream.buffer != NULL) &&    // Validate stream buffer
+        (sound.stream.sampleRate > 0) &&    // Validate sample rate is supported
+        (sound.stream.sampleSize > 0) &&    // Validate sample size is supported
+        (sound.stream.channels > 0)) result = true; // Validate number of channels supported
+
+    return result;
+}
+
+// Unload wave data
+void UnloadWave(Wave wave)
+{
+    RL_FREE(wave.data);
+    //TRACELOG(LOG_INFO, "WAVE: Unloaded wave data from RAM");
+}
+
+// Unload sound
+void UnloadSound(Sound sound)
+{
+    UnloadAudioBuffer(sound.stream.buffer);
+    //TRACELOG(LOG_INFO, "SOUND: Unloaded sound data from RAM");
+}
+
+void UnloadSoundAlias(Sound alias)
+{
+    // Untrack and unload just the sound buffer, not the sample data, it is shared with the source for the alias
+    if (alias.stream.buffer != NULL)
+    {
+        UntrackAudioBuffer(alias.stream.buffer);
+        ma_data_converter_uninit(&alias.stream.buffer->converter, NULL);
+        RL_FREE(alias.stream.buffer);
+    }
+}
+
+// Update sound buffer with new data
+void UpdateSound(Sound sound, const void *data, int frameCount)
+{
+    if (sound.stream.buffer != NULL)
+    {
+        StopAudioBuffer(sound.stream.buffer);
+
+        memcpy(sound.stream.buffer->data, data, frameCount*ma_get_bytes_per_frame(sound.stream.buffer->converter.formatIn, sound.stream.buffer->converter.channelsIn));
+    }
+}
+
+// Export wave data to file
+bool ExportWave(Wave wave, const char *fileName)
+{
+    bool success = false;
+
+    if (false) { }
+#if defined(SUPPORT_FILEFORMAT_WAV)
+    else if (IsFileExtension(fileName, ".wav"))
+    {
+        drwav wav = { 0 };
+        drwav_data_format format = { 0 };
+        format.container = drwav_container_riff;
+        if (wave.sampleSize == 32) format.format = DR_WAVE_FORMAT_IEEE_FLOAT;
+        else format.format = DR_WAVE_FORMAT_PCM;
+        format.channels = wave.channels;
+        format.sampleRate = wave.sampleRate;
+        format.bitsPerSample = wave.sampleSize;
+
+        void *fileData = NULL;
+        size_t fileDataSize = 0;
+        success = drwav_init_memory_write(&wav, &fileData, &fileDataSize, &format, NULL);
+        if (success) success = (int)drwav_write_pcm_frames(&wav, wave.frameCount, wave.data);
+        drwav_result result = drwav_uninit(&wav);
+
+        if (result == DRWAV_SUCCESS) success = SaveFileData(fileName, (unsigned char *)fileData, (unsigned int)fileDataSize);
+
+        drwav_free(fileData, NULL);
+    }
+#endif
+#if defined(SUPPORT_FILEFORMAT_QOA)
+    else if (IsFileExtension(fileName, ".qoa"))
+    {
+        if (wave.sampleSize == 16)
+        {
+            qoa_desc qoa = { 0 };
+            qoa.channels = wave.channels;
+            qoa.samplerate = wave.sampleRate;
+            qoa.samples = wave.frameCount;
+
+            int bytesWritten = qoa_write(fileName, wave.data, &qoa);
+            if (bytesWritten > 0) success = true;
+        }
+        else TRACELOG(LOG_WARNING, "AUDIO: Wave data must be 16 bit per sample for QOA format export");
+    }
+#endif
+    else if (IsFileExtension(fileName, ".raw"))
+    {
+        // Export raw sample data (without header)
+        // NOTE: It's up to the user to track wave parameters
+        success = SaveFileData(fileName, wave.data, wave.frameCount*wave.channels*wave.sampleSize/8);
+    }
+
+    if (success) TRACELOG(LOG_INFO, "FILEIO: [%s] Wave data exported successfully", fileName);
+    else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to export wave data", fileName);
+
+    return success;
+}
+
+// Export wave sample data to code (.h)
+bool ExportWaveAsCode(Wave wave, const char *fileName)
+{
+    bool success = false;
+
+#ifndef TEXT_BYTES_PER_LINE
+    #define TEXT_BYTES_PER_LINE     20
+#endif
+
+    int waveDataSize = wave.frameCount*wave.channels*wave.sampleSize/8;
+
+    // NOTE: Text data buffer size is estimated considering wave data size in bytes
+    // and requiring 12 char bytes for every byte; the actual size varies, but
+    // the longest possible char being appended is "%.4ff,\n    ", which is 12 bytes.
+    char *txtData = (char *)RL_CALLOC(waveDataSize*12 + 2000, sizeof(char));
+
+    int byteCount = 0;
+    byteCount += sprintf(txtData + byteCount, "\n//////////////////////////////////////////////////////////////////////////////////\n");
+    byteCount += sprintf(txtData + byteCount, "//                                                                              //\n");
+    byteCount += sprintf(txtData + byteCount, "// WaveAsCode exporter v1.1 - Wave data exported as an array of bytes           //\n");
+    byteCount += sprintf(txtData + byteCount, "//                                                                              //\n");
+    byteCount += sprintf(txtData + byteCount, "// more info and bugs-report:  github.com/raysan5/raylib                        //\n");
+    byteCount += sprintf(txtData + byteCount, "// feedback and support:       ray[at]raylib.com                                //\n");
+    byteCount += sprintf(txtData + byteCount, "//                                                                              //\n");
+    byteCount += sprintf(txtData + byteCount, "// Copyright (c) 2018-2024 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)
+{
+    bool result = false;
+
+    if (IsAudioBufferPlaying(sound.stream.buffer)) result = true;
+
+    return result;
+}
+
+// Set volume for a sound
+void SetSoundVolume(Sound sound, float volume)
+{
+    SetAudioBufferVolume(sound.stream.buffer, volume);
+}
+
+// Set pitch for a sound
+void SetSoundPitch(Sound sound, float pitch)
+{
+    SetAudioBufferPitch(sound.stream.buffer, pitch);
+}
+
+// Set pan for a sound
+void SetSoundPan(Sound sound, float pan)
+{
+    SetAudioBufferPan(sound.stream.buffer, pan);
+}
+
+// Convert wave data to desired format
+void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
+{
+    ma_format formatIn = ((wave->sampleSize == 8)? ma_format_u8 : ((wave->sampleSize == 16)? ma_format_s16 : ma_format_f32));
+    ma_format formatOut = ((sampleSize == 8)? ma_format_u8 : ((sampleSize == 16)? ma_format_s16 : ma_format_f32));
+
+    ma_uint32 frameCountIn = wave->frameCount;
+    ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, formatOut, channels, sampleRate, NULL, frameCountIn, formatIn, wave->channels, wave->sampleRate);
+
+    if (frameCount == 0)
+    {
+        TRACELOG(LOG_WARNING, "WAVE: Failed to get frame count for format conversion");
+        return;
+    }
+
+    void *data = RL_MALLOC(frameCount*channels*(sampleSize/8));
+
+    frameCount = (ma_uint32)ma_convert_frames(data, frameCount, formatOut, channels, sampleRate, wave->data, frameCountIn, formatIn, wave->channels, wave->sampleRate);
+    if (frameCount == 0)
+    {
+        TRACELOG(LOG_WARNING, "WAVE: Failed format conversion");
+        return;
+    }
+
+    wave->frameCount = frameCount;
+    wave->sampleSize = sampleSize;
+    wave->sampleRate = sampleRate;
+    wave->channels = channels;
+
+    RL_FREE(wave->data);
+    wave->data = data;
+}
+
+// Copy a wave to a new wave
+Wave WaveCopy(Wave wave)
+{
+    Wave newWave = { 0 };
+
+    newWave.data = RL_MALLOC(wave.frameCount*wave.channels*wave.sampleSize/8);
+
+    if (newWave.data != NULL)
+    {
+        // NOTE: Size must be provided in bytes
+        memcpy(newWave.data, wave.data, wave.frameCount*wave.channels*wave.sampleSize/8);
+
+        newWave.frameCount = wave.frameCount;
+        newWave.sampleRate = wave.sampleRate;
+        newWave.sampleSize = wave.sampleSize;
+        newWave.channels = wave.channels;
+    }
+
+    return newWave;
+}
+
+// Crop a wave to defined frames range
+// NOTE: Security check in case of out-of-range
+void WaveCrop(Wave *wave, int initFrame, int finalFrame)
+{
+    if ((initFrame >= 0) && (initFrame < finalFrame) && ((unsigned int)finalFrame <= wave->frameCount))
+    {
+        int frameCount = finalFrame - initFrame;
+
+        void *data = RL_MALLOC(frameCount*wave->channels*wave->sampleSize/8);
+
+        memcpy(data, (unsigned char *)wave->data + (initFrame*wave->channels*wave->sampleSize/8), frameCount*wave->channels*wave->sampleSize/8);
+
+        RL_FREE(wave->data);
+        wave->data = data;
+        wave->frameCount = (unsigned int)frameCount;
+    }
+    else TRACELOG(LOG_WARNING, "WAVE: Crop range out of bounds");
+}
+
+// Load samples data from wave as a floats array
+// NOTE 1: Returned sample values are normalized to range [-1..1]
+// NOTE 2: Sample data allocated should be freed with UnloadWaveSamples()
+float *LoadWaveSamples(Wave wave)
+{
+    float *samples = (float *)RL_MALLOC(wave.frameCount*wave.channels*sizeof(float));
+
+    // NOTE: sampleCount is the total number of interlaced samples (including channels)
+
+    for (unsigned int i = 0; i < wave.frameCount*wave.channels; i++)
+    {
+        if (wave.sampleSize == 8) samples[i] = (float)(((unsigned char *)wave.data)[i] - 128)/128.0f;
+        else if (wave.sampleSize == 16) samples[i] = (float)(((short *)wave.data)[i])/32768.0f;
+        else if (wave.sampleSize == 32) samples[i] = ((float *)wave.data)[i];
+    }
+
+    return samples;
+}
+
+// Unload samples data loaded with LoadWaveSamples()
+void UnloadWaveSamples(float *samples)
+{
+    RL_FREE(samples);
+}
+
+//----------------------------------------------------------------------------------
+// Module Functions Definition - Music loading and stream playing
+//----------------------------------------------------------------------------------
+
+// Load music stream from file
+Music LoadMusicStream(const char *fileName)
+{
+    Music music = { 0 };
+    bool musicLoaded = false;
+
+    if (false) { }
+#if defined(SUPPORT_FILEFORMAT_WAV)
+    else if (IsFileExtension(fileName, ".wav"))
+    {
+        drwav *ctxWav = RL_CALLOC(1, sizeof(drwav));
+        bool success = drwav_init_file(ctxWav, fileName, NULL);
+
+        if (success)
+        {
+            music.ctxType = MUSIC_AUDIO_WAV;
+            music.ctxData = ctxWav;
+            int sampleSize = ctxWav->bitsPerSample;
+            if (ctxWav->bitsPerSample == 24) sampleSize = 16;   // Forcing conversion to s16 on UpdateMusicStream()
+
+            music.stream = LoadAudioStream(ctxWav->sampleRate, sampleSize, ctxWav->channels);
+            music.frameCount = (unsigned int)ctxWav->totalPCMFrameCount;
+            music.looping = true;   // Looping enabled by default
+            musicLoaded = true;
+        }
+        else
+        {
+            RL_FREE(ctxWav);
+        }
+    }
+#endif
+#if defined(SUPPORT_FILEFORMAT_OGG)
+    else if (IsFileExtension(fileName, ".ogg"))
+    {
+        // Open ogg audio stream
+        stb_vorbis *ctxOgg = stb_vorbis_open_filename(fileName, NULL, NULL);
+
+        if (ctxOgg != NULL)
+        {
+            music.ctxType = MUSIC_AUDIO_OGG;
+            music.ctxData = ctxOgg;
+            stb_vorbis_info info = stb_vorbis_get_info((stb_vorbis *)music.ctxData);  // Get Ogg file info
+
+            // OGG bit rate defaults to 16 bit, it's enough for compressed format
+            music.stream = LoadAudioStream(info.sample_rate, 16, info.channels);
+
+            // WARNING: It seems this function returns length in frames, not samples, so we multiply by channels
+            music.frameCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData);
+            music.looping = true;   // Looping enabled by default
+            musicLoaded = true;
+        }
+        else
+        {
+            stb_vorbis_close(ctxOgg);
+        }
+    }
+#endif
+#if defined(SUPPORT_FILEFORMAT_MP3)
+    else if (IsFileExtension(fileName, ".mp3"))
+    {
+        drmp3 *ctxMp3 = RL_CALLOC(1, sizeof(drmp3));
+        int result = drmp3_init_file(ctxMp3, fileName, NULL);
+
+        if (result > 0)
+        {
+            music.ctxType = MUSIC_AUDIO_MP3;
+            music.ctxData = ctxMp3;
+            music.stream = LoadAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
+            music.frameCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3);
+            music.looping = true;   // Looping enabled by default
+            musicLoaded = true;
+        }
+        else
+        {
+            RL_FREE(ctxMp3);
+        }
+    }
+#endif
+#if defined(SUPPORT_FILEFORMAT_QOA)
+    else if (IsFileExtension(fileName, ".qoa"))
+    {
+        qoaplay_desc *ctxQoa = qoaplay_open(fileName);
+
+        if (ctxQoa != NULL)
+        {
+            music.ctxType = MUSIC_AUDIO_QOA;
+            music.ctxData = ctxQoa;
+            // NOTE: We are loading samples are 32bit float normalized data, so,
+            // we configure the output audio stream to also use float 32bit
+            music.stream = LoadAudioStream(ctxQoa->info.samplerate, 32, ctxQoa->info.channels);
+            music.frameCount = ctxQoa->info.samples;
+            music.looping = true;   // Looping enabled by default
+            musicLoaded = true;
+        }
+        else{} //No uninit required
+    }
+#endif
+#if defined(SUPPORT_FILEFORMAT_FLAC)
+    else if (IsFileExtension(fileName, ".flac"))
+    {
+        drflac *ctxFlac = drflac_open_file(fileName, NULL);
+
+        if (ctxFlac != NULL)
+        {
+            music.ctxType = MUSIC_AUDIO_FLAC;
+            music.ctxData = ctxFlac;
+            int sampleSize = ctxFlac->bitsPerSample;
+            if (ctxFlac->bitsPerSample == 24) sampleSize = 16;   // Forcing conversion to s16 on UpdateMusicStream()
+            music.stream = LoadAudioStream(ctxFlac->sampleRate, sampleSize, ctxFlac->channels);
+            music.frameCount = (unsigned int)ctxFlac->totalPCMFrameCount;
+            music.looping = true;   // Looping enabled by default
+            musicLoaded = true;
+        }
+        else
+        {
+            drflac_free(ctxFlac, NULL);
+        }
+    }
+#endif
+#if defined(SUPPORT_FILEFORMAT_XM)
+    else if (IsFileExtension(fileName, ".xm"))
+    {
+        jar_xm_context_t *ctxXm = NULL;
+        int result = jar_xm_create_context_from_file(&ctxXm, AUDIO.System.device.sampleRate, fileName);
+
+        if (result == 0)    // XM AUDIO.System.context created successfully
+        {
+            music.ctxType = MUSIC_MODULE_XM;
+            music.ctxData = ctxXm;
+            jar_xm_set_max_loop_count(ctxXm, 0);    // Set infinite number of loops
+
+            unsigned int bits = 32;
+            if (AUDIO_DEVICE_FORMAT == ma_format_s16) bits = 16;
+            else if (AUDIO_DEVICE_FORMAT == ma_format_u8) bits = 8;
+
+            // NOTE: Only stereo is supported for XM
+            music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, bits, AUDIO_DEVICE_CHANNELS);
+            music.frameCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm);    // NOTE: Always 2 channels (stereo)
+            music.looping = true;   // Looping enabled by default
+            jar_xm_reset(ctxXm);    // Make sure we start at the beginning of the song
+            musicLoaded = true;
+        }
+        else
+        {
+            jar_xm_free_context(ctxXm);
+        }
+    }
+#endif
+#if defined(SUPPORT_FILEFORMAT_MOD)
+    else if (IsFileExtension(fileName, ".mod"))
+    {
+        jar_mod_context_t *ctxMod = RL_CALLOC(1, sizeof(jar_mod_context_t));
+        jar_mod_init(ctxMod);
+        int result = jar_mod_load_file(ctxMod, fileName);
+
+        if (result > 0)
+        {
+            music.ctxType = MUSIC_MODULE_MOD;
+            music.ctxData = ctxMod;
+            // NOTE: Only stereo is supported for MOD
+            music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, 16, AUDIO_DEVICE_CHANNELS);
+            music.frameCount = (unsigned int)jar_mod_max_samples(ctxMod);    // NOTE: Always 2 channels (stereo)
+            music.looping = true;   // Looping enabled by default
+            musicLoaded = true;
+        }
+        else
+        {
+            jar_mod_unload(ctxMod);
+            RL_FREE(ctxMod);
+        }
+    }
+#endif
+    else TRACELOG(LOG_WARNING, "STREAM: [%s] File format not supported", fileName);
+
+    if (!musicLoaded)
+    {
+        TRACELOG(LOG_WARNING, "FILEIO: [%s] Music file could not be opened", fileName);
+    }
+    else
+    {
+        // Show some music stream info
+        TRACELOG(LOG_INFO, "FILEIO: [%s] Music file loaded successfully", fileName);
+        TRACELOG(LOG_INFO, "    > Sample rate:   %i Hz", music.stream.sampleRate);
+        TRACELOG(LOG_INFO, "    > Sample size:   %i bits", music.stream.sampleSize);
+        TRACELOG(LOG_INFO, "    > Channels:      %i (%s)", music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi");
+        TRACELOG(LOG_INFO, "    > Total frames:  %i", music.frameCount);
+    }
+
+    return music;
+}
+
+// Load music stream from memory buffer, fileType refers to extension: i.e. ".wav"
+// WARNING: File extension must be provided in lower-case
+Music LoadMusicStreamFromMemory(const char *fileType, const unsigned char *data, int dataSize)
+{
+    Music music = { 0 };
+    bool musicLoaded = false;
+
+    if (false) { }
+#if defined(SUPPORT_FILEFORMAT_WAV)
+    else if ((strcmp(fileType, ".wav") == 0) || (strcmp(fileType, ".WAV") == 0))
+    {
+        drwav *ctxWav = RL_CALLOC(1, sizeof(drwav));
+
+        bool success = drwav_init_memory(ctxWav, (const void *)data, dataSize, NULL);
+
+        if (success)
+        {
+            music.ctxType = MUSIC_AUDIO_WAV;
+            music.ctxData = ctxWav;
+            int sampleSize = ctxWav->bitsPerSample;
+            if (ctxWav->bitsPerSample == 24) sampleSize = 16;   // Forcing conversion to s16 on UpdateMusicStream()
+
+            music.stream = LoadAudioStream(ctxWav->sampleRate, sampleSize, ctxWav->channels);
+            music.frameCount = (unsigned int)ctxWav->totalPCMFrameCount;
+            music.looping = true;   // Looping enabled by default
+            musicLoaded = true;
+        }
+        else {
+            drwav_uninit(ctxWav);
+            RL_FREE(ctxWav);
+        }
+    }
+#endif
+#if defined(SUPPORT_FILEFORMAT_OGG)
+    else if ((strcmp(fileType, ".ogg") == 0) || (strcmp(fileType, ".OGG") == 0))
+    {
+        // Open ogg audio stream
+        stb_vorbis* ctxOgg = stb_vorbis_open_memory((const unsigned char *)data, dataSize, NULL, NULL);
+
+        if (ctxOgg != NULL)
+        {
+            music.ctxType = MUSIC_AUDIO_OGG;
+            music.ctxData = ctxOgg;
+            stb_vorbis_info info = stb_vorbis_get_info((stb_vorbis *)music.ctxData);  // Get Ogg file info
+
+            // OGG bit rate defaults to 16 bit, it's enough for compressed format
+            music.stream = LoadAudioStream(info.sample_rate, 16, info.channels);
+
+            // WARNING: It seems this function returns length in frames, not samples, so we multiply by channels
+            music.frameCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData);
+            music.looping = true;   // Looping enabled by default
+            musicLoaded = true;
+        }
+        else
+        {
+            stb_vorbis_close(ctxOgg);
+        }
+    }
+#endif
+#if defined(SUPPORT_FILEFORMAT_MP3)
+    else if ((strcmp(fileType, ".mp3") == 0) || (strcmp(fileType, ".MP3") == 0))
+    {
+        drmp3 *ctxMp3 = RL_CALLOC(1, sizeof(drmp3));
+        int success = drmp3_init_memory(ctxMp3, (const void*)data, dataSize, NULL);
+
+        if (success)
+        {
+            music.ctxType = MUSIC_AUDIO_MP3;
+            music.ctxData = ctxMp3;
+            music.stream = LoadAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
+            music.frameCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3);
+            music.looping = true;   // Looping enabled by default
+            musicLoaded = true;
+        }
+        else
+        {
+            drmp3_uninit(ctxMp3);
+            RL_FREE(ctxMp3);
+        }
+    }
+#endif
+#if defined(SUPPORT_FILEFORMAT_QOA)
+    else if ((strcmp(fileType, ".qoa") == 0) || (strcmp(fileType, ".QOA") == 0))
+    {
+        qoaplay_desc *ctxQoa = NULL;
+        if ((data != NULL) && (dataSize > 0))
+        {
+            ctxQoa = qoaplay_open_memory(data, dataSize);
+        }
+
+        if (ctxQoa != NULL)
+        {
+            music.ctxType = MUSIC_AUDIO_QOA;
+            music.ctxData = ctxQoa;
+            // NOTE: We are loading samples are 32bit float normalized data, so,
+            // we configure the output audio stream to also use float 32bit
+            music.stream = LoadAudioStream(ctxQoa->info.samplerate, 32, ctxQoa->info.channels);
+            music.frameCount = ctxQoa->info.samples;
+            music.looping = true;   // Looping enabled by default
+            musicLoaded = true;
+        }
+        else{} //No uninit required
+    }
+#endif
+#if defined(SUPPORT_FILEFORMAT_FLAC)
+    else if ((strcmp(fileType, ".flac") == 0) || (strcmp(fileType, ".FLAC") == 0))
+    {
+        drflac *ctxFlac = drflac_open_memory((const void*)data, dataSize, NULL);
+
+        if (ctxFlac != NULL)
+        {
+            music.ctxType = MUSIC_AUDIO_FLAC;
+            music.ctxData = ctxFlac;
+            int sampleSize = ctxFlac->bitsPerSample;
+            if (ctxFlac->bitsPerSample == 24) sampleSize = 16;   // Forcing conversion to s16 on UpdateMusicStream()
+            music.stream = LoadAudioStream(ctxFlac->sampleRate, sampleSize, ctxFlac->channels);
+            music.frameCount = (unsigned int)ctxFlac->totalPCMFrameCount;
+            music.looping = true;   // Looping enabled by default
+            musicLoaded = true;
+        }
+        else
+        {
+            drflac_free(ctxFlac, NULL);
+        }
+    }
+#endif
+#if defined(SUPPORT_FILEFORMAT_XM)
+    else if ((strcmp(fileType, ".xm") == 0) || (strcmp(fileType, ".XM") == 0))
+    {
+        jar_xm_context_t *ctxXm = NULL;
+        int result = jar_xm_create_context_safe(&ctxXm, (const char *)data, dataSize, AUDIO.System.device.sampleRate);
+        if (result == 0)    // XM AUDIO.System.context created successfully
+        {
+            music.ctxType = MUSIC_MODULE_XM;
+            music.ctxData = ctxXm;
+            jar_xm_set_max_loop_count(ctxXm, 0);    // Set infinite number of loops
+
+            unsigned int bits = 32;
+            if (AUDIO_DEVICE_FORMAT == ma_format_s16) bits = 16;
+            else if (AUDIO_DEVICE_FORMAT == ma_format_u8) bits = 8;
+
+            // NOTE: Only stereo is supported for XM
+            music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, bits, 2);
+            music.frameCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm);    // NOTE: Always 2 channels (stereo)
+            music.looping = true;   // Looping enabled by default
+            jar_xm_reset(ctxXm);    // Make sure we start at the beginning of the song
+
+            musicLoaded = true;
+        }
+        else
+        {
+            jar_xm_free_context(ctxXm);
+        }
+    }
+#endif
+#if defined(SUPPORT_FILEFORMAT_MOD)
+    else if ((strcmp(fileType, ".mod") == 0) || (strcmp(fileType, ".MOD") == 0))
+    {
+        jar_mod_context_t *ctxMod = (jar_mod_context_t *)RL_MALLOC(sizeof(jar_mod_context_t));
+        int result = 0;
+
+        jar_mod_init(ctxMod);
+
+        // Copy data to allocated memory for default UnloadMusicStream
+        unsigned char *newData = (unsigned char *)RL_MALLOC(dataSize);
+        int it = dataSize/sizeof(unsigned char);
+        for (int i = 0; i < it; i++) newData[i] = data[i];
+
+        // Memory loaded version for jar_mod_load_file()
+        if (dataSize && (dataSize < 32*1024*1024))
+        {
+            ctxMod->modfilesize = dataSize;
+            ctxMod->modfile = newData;
+            if (jar_mod_load(ctxMod, (void *)ctxMod->modfile, dataSize)) result = dataSize;
+        }
+
+        if (result > 0)
+        {
+            music.ctxType = MUSIC_MODULE_MOD;
+            music.ctxData = ctxMod;
+
+            // NOTE: Only stereo is supported for MOD
+            music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, 16, 2);
+            music.frameCount = (unsigned int)jar_mod_max_samples(ctxMod);    // NOTE: Always 2 channels (stereo)
+            music.looping = true;   // Looping enabled by default
+            musicLoaded = true;
+        }
+        else
+        {
+            jar_mod_unload(ctxMod);
+            RL_FREE(ctxMod);
+        }
+    }
+#endif
+    else TRACELOG(LOG_WARNING, "STREAM: Data format not supported");
+
+    if (!musicLoaded)
+    {
+        TRACELOG(LOG_WARNING, "FILEIO: Music data could not be loaded");
+    }
+    else
+    {
+        // Show some music stream info
+        TRACELOG(LOG_INFO, "FILEIO: Music data loaded successfully");
+        TRACELOG(LOG_INFO, "    > Sample rate:   %i Hz", music.stream.sampleRate);
+        TRACELOG(LOG_INFO, "    > Sample size:   %i bits", music.stream.sampleSize);
+        TRACELOG(LOG_INFO, "    > Channels:      %i (%s)", music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi");
+        TRACELOG(LOG_INFO, "    > Total frames:  %i", music.frameCount);
+    }
+
+    return music;
+}
+
+// Checks if a music stream is valid (context and buffers initialized)
+bool IsMusicValid(Music music)
+{
+    return ((music.ctxData != NULL) &&          // Validate context loaded
+            (music.frameCount > 0) &&           // Validate audio frame count
+            (music.stream.sampleRate > 0) &&    // Validate sample rate is supported
+            (music.stream.sampleSize > 0) &&    // Validate sample size is supported
+            (music.stream.channels > 0));       // Validate number of channels supported
+}
+
+// Unload music stream
+void UnloadMusicStream(Music music)
+{
+    UnloadAudioStream(music.stream);
+
+    if (music.ctxData != NULL)
+    {
+        if (false) { }
+#if defined(SUPPORT_FILEFORMAT_WAV)
+        else if (music.ctxType == MUSIC_AUDIO_WAV) drwav_uninit((drwav *)music.ctxData);
+#endif
+#if defined(SUPPORT_FILEFORMAT_OGG)
+        else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData);
+#endif
+#if defined(SUPPORT_FILEFORMAT_MP3)
+        else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
+#endif
+#if defined(SUPPORT_FILEFORMAT_QOA)
+        else if (music.ctxType == MUSIC_AUDIO_QOA) qoaplay_close((qoaplay_desc *)music.ctxData);
+#endif
+#if defined(SUPPORT_FILEFORMAT_FLAC)
+        else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL);
+#endif
+#if defined(SUPPORT_FILEFORMAT_XM)
+        else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData);
+#endif
+#if defined(SUPPORT_FILEFORMAT_MOD)
+        else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); }
+#endif
+    }
+}
+
+// Start music playing (open stream) from beginning
+void PlayMusicStream(Music music)
+{
+    PlayAudioStream(music.stream);
+}
+
+// Pause music playing
+void PauseMusicStream(Music music)
+{
+    PauseAudioStream(music.stream);
+}
+
+// Resume music playing
+void ResumeMusicStream(Music music)
+{
+    ResumeAudioStream(music.stream);
+}
+
+// Stop music playing (close stream)
+void StopMusicStream(Music music)
+{
+    StopAudioStream(music.stream);
+
+    switch (music.ctxType)
+    {
+#if defined(SUPPORT_FILEFORMAT_WAV)
+        case MUSIC_AUDIO_WAV: drwav_seek_to_first_pcm_frame((drwav *)music.ctxData); break;
+#endif
+#if defined(SUPPORT_FILEFORMAT_OGG)
+        case MUSIC_AUDIO_OGG: stb_vorbis_seek_start((stb_vorbis *)music.ctxData); break;
+#endif
+#if defined(SUPPORT_FILEFORMAT_MP3)
+        case MUSIC_AUDIO_MP3: drmp3_seek_to_start_of_stream((drmp3 *)music.ctxData); break;
+#endif
+#if defined(SUPPORT_FILEFORMAT_QOA)
+        case MUSIC_AUDIO_QOA: qoaplay_rewind((qoaplay_desc *)music.ctxData); break;
+#endif
+#if defined(SUPPORT_FILEFORMAT_FLAC)
+        case MUSIC_AUDIO_FLAC: drflac__seek_to_first_frame((drflac *)music.ctxData); break;
+#endif
+#if defined(SUPPORT_FILEFORMAT_XM)
+        case MUSIC_MODULE_XM: jar_xm_reset((jar_xm_context_t *)music.ctxData); break;
+#endif
+#if defined(SUPPORT_FILEFORMAT_MOD)
+        case MUSIC_MODULE_MOD: jar_mod_seek_start((jar_mod_context_t *)music.ctxData); break;
+#endif
+        default: break;
+    }
+}
+
+// Seek music to a certain position (in seconds)
+void SeekMusicStream(Music music, float position)
+{
+    // Seeking is not supported in module formats
+    if ((music.ctxType == MUSIC_MODULE_XM) || (music.ctxType == MUSIC_MODULE_MOD)) return;
+
+    unsigned int positionInFrames = (unsigned int)(position*music.stream.sampleRate);
+
+    switch (music.ctxType)
+    {
+#if defined(SUPPORT_FILEFORMAT_WAV)
+        case MUSIC_AUDIO_WAV: drwav_seek_to_pcm_frame((drwav *)music.ctxData, positionInFrames); break;
+#endif
+#if defined(SUPPORT_FILEFORMAT_OGG)
+        case MUSIC_AUDIO_OGG: stb_vorbis_seek_frame((stb_vorbis *)music.ctxData, positionInFrames); break;
+#endif
+#if defined(SUPPORT_FILEFORMAT_MP3)
+        case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame((drmp3 *)music.ctxData, positionInFrames); break;
+#endif
+#if defined(SUPPORT_FILEFORMAT_QOA)
+        case MUSIC_AUDIO_QOA:
+        {
+            int qoaFrame = positionInFrames/QOA_FRAME_LEN;
+            qoaplay_seek_frame((qoaplay_desc *)music.ctxData, qoaFrame); // Seeks to QOA frame, not PCM frame
+
+            // We need to compute QOA frame number and update positionInFrames
+            positionInFrames = ((qoaplay_desc *)music.ctxData)->sample_position;
+        } break;
+#endif
+#if defined(SUPPORT_FILEFORMAT_FLAC)
+        case MUSIC_AUDIO_FLAC: drflac_seek_to_pcm_frame((drflac *)music.ctxData, positionInFrames); break;
+#endif
+        default: break;
+    }
+
+    ma_mutex_lock(&AUDIO.System.lock);
+    music.stream.buffer->framesProcessed = positionInFrames;
+    ma_mutex_unlock(&AUDIO.System.lock);
+}
+
+// Update (re-fill) music buffers if data already processed
+void UpdateMusicStream(Music music)
+{
+    if (music.stream.buffer == NULL) return;
+
+    ma_mutex_lock(&AUDIO.System.lock);
+
+    unsigned int subBufferSizeInFrames = music.stream.buffer->sizeInFrames/2;
+
+    // On first call of this function we lazily pre-allocated a temp buffer to read audio files/memory data in
+    int frameSize = music.stream.channels*music.stream.sampleSize/8;
+    unsigned int pcmSize = subBufferSizeInFrames*frameSize;
+
+    if (AUDIO.System.pcmBufferSize < pcmSize)
+    {
+        RL_FREE(AUDIO.System.pcmBuffer);
+        AUDIO.System.pcmBuffer = RL_CALLOC(1, pcmSize);
+        AUDIO.System.pcmBufferSize = pcmSize;
+    }
+
+    // Check both sub-buffers to check if they require refilling
+    for (int i = 0; i < 2; i++)
+    {
+        if (!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 = (int)drflac_read_pcm_frames_s16((drflac *)music.ctxData, frameCountStillNeeded, (short *)((char *)AUDIO.System.pcmBuffer + frameCountReadTotal*frameSize));
+                    frameCountReadTotal += frameCountRead;
+                    frameCountStillNeeded -= frameCountRead;
+                    if (frameCountStillNeeded == 0) break;
+                    else drflac__seek_to_first_frame((drflac *)music.ctxData);
+                }
+            } break;
+        #endif
+        #if defined(SUPPORT_FILEFORMAT_XM)
+            case MUSIC_MODULE_XM:
+            {
+                // NOTE: Internally we consider 2 channels generation, so sampleCount/2
+                if (AUDIO_DEVICE_FORMAT == ma_format_f32) jar_xm_generate_samples((jar_xm_context_t *)music.ctxData, (float *)AUDIO.System.pcmBuffer, framesToStream);
+                else if (AUDIO_DEVICE_FORMAT == ma_format_s16) jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)AUDIO.System.pcmBuffer, framesToStream);
+                else if (AUDIO_DEVICE_FORMAT == ma_format_u8) jar_xm_generate_samples_8bit((jar_xm_context_t *)music.ctxData, (char *)AUDIO.System.pcmBuffer, framesToStream);
+                //jar_xm_reset((jar_xm_context_t *)music.ctxData);
+
+            } break;
+        #endif
+        #if defined(SUPPORT_FILEFORMAT_MOD)
+            case MUSIC_MODULE_MOD:
+            {
+                // NOTE: 3rd parameter (nbsample) specify the number of stereo 16bits samples you want, so sampleCount/2
+                jar_mod_fillbuffer((jar_mod_context_t *)music.ctxData, (short *)AUDIO.System.pcmBuffer, framesToStream, 0);
+                //jar_mod_seek_start((jar_mod_context_t *)music.ctxData);
+
+            } break;
+        #endif
+            default: break;
+        }
+
+        UpdateAudioStreamInLockedState(music.stream, AUDIO.System.pcmBuffer, framesToStream);
+
+        music.stream.buffer->framesProcessed = music.stream.buffer->framesProcessed%music.frameCount;
+
+        if (framesLeft <= subBufferSizeInFrames)
+        {
+            if (!music.looping)
+            {
+                ma_mutex_unlock(&AUDIO.System.lock);
+                // Streaming is ending, we filled latest frames from input
+                StopMusicStream(music);
+                return;
+            }
+        }
+    }
+
+    ma_mutex_unlock(&AUDIO.System.lock);
+}
+
+// Check if any music is playing
+bool IsMusicStreamPlaying(Music music)
+{
+    return IsAudioStreamPlaying(music.stream);
+}
+
+// Set volume for music
+void SetMusicVolume(Music music, float volume)
+{
+    SetAudioStreamVolume(music.stream, volume);
+}
+
+// Set pitch for music
+void SetMusicPitch(Music music, float pitch)
+{
+    SetAudioBufferPitch(music.stream.buffer, pitch);
+}
+
+// Set pan for a music
+void SetMusicPan(Music music, float pan)
+{
+    SetAudioBufferPan(music.stream.buffer, pan);
+}
+
+// Get music time length (in seconds)
+float GetMusicTimeLength(Music music)
+{
+    float totalSeconds = 0.0f;
+
+    totalSeconds = (float)music.frameCount/music.stream.sampleRate;
+
+    return totalSeconds;
+}
+
+// Get current music time played (in seconds)
+float GetMusicTimePlayed(Music music)
+{
+    float secondsPlayed = 0.0f;
+    if (music.stream.buffer != NULL)
+    {
+#if defined(SUPPORT_FILEFORMAT_XM)
+        if (music.ctxType == MUSIC_MODULE_XM)
+        {
+            uint64_t framesPlayed = 0;
+
+            jar_xm_get_position(music.ctxData, NULL, NULL, NULL, &framesPlayed);
+            secondsPlayed = (float)framesPlayed/music.stream.sampleRate;
+        }
+        else
+#endif
+        {
+            ma_mutex_lock(&AUDIO.System.lock);
+            //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
+            int framesProcessed = (int)music.stream.buffer->framesProcessed;
+            int subBufferSize = (int)music.stream.buffer->sizeInFrames/2;
+            int framesInFirstBuffer = music.stream.buffer->isSubBufferProcessed[0]? 0 : subBufferSize;
+            int framesInSecondBuffer = music.stream.buffer->isSubBufferProcessed[1]? 0 : subBufferSize;
+            int 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;
+            ma_mutex_unlock(&AUDIO.System.lock);
+        }
+    }
+
+    return secondsPlayed;
+}
+
+// Load audio stream (to stream audio pcm data)
+AudioStream LoadAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels)
+{
+    AudioStream stream = { 0 };
+
+    stream.sampleRate = sampleRate;
+    stream.sampleSize = sampleSize;
+    stream.channels = channels;
+
+    ma_format formatIn = ((stream.sampleSize == 8)? ma_format_u8 : ((stream.sampleSize == 16)? ma_format_s16 : ma_format_f32));
+
+    // The size of a streaming buffer must be at least double the size of a period
+    unsigned int periodSize = AUDIO.System.device.playback.internalPeriodSizeInFrames;
+
+    // If the buffer is not set, compute one that would give us a buffer good enough for a decent frame rate
+    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 valid (buffers initialized)
+bool IsAudioStreamValid(AudioStream stream)
+{
+    return ((stream.buffer != NULL) &&    // Validate stream buffer
+            (stream.sampleRate > 0) &&    // Validate sample rate is supported
+            (stream.sampleSize > 0) &&    // Validate sample size is supported
+            (stream.channels > 0));       // Validate number of channels supported
+}
+
+// Unload audio stream and free memory
+void UnloadAudioStream(AudioStream stream)
+{
+    UnloadAudioBuffer(stream.buffer);
+
+    TRACELOG(LOG_INFO, "STREAM: Unloaded audio stream data from RAM");
+}
+
+// Update audio stream buffers with data
+// NOTE 1: Only updates one buffer of the stream source: dequeue -> update -> queue
+// NOTE 2: To dequeue a buffer it needs to be processed: IsAudioStreamProcessed()
+void UpdateAudioStream(AudioStream stream, const void *data, int frameCount)
+{
+    ma_mutex_lock(&AUDIO.System.lock);
+    UpdateAudioStreamInLockedState(stream, data, frameCount);
+    ma_mutex_unlock(&AUDIO.System.lock);
+}
+
+// Check if any audio stream buffers requires refill
+bool IsAudioStreamProcessed(AudioStream stream)
+{
+    if (stream.buffer == NULL) return false;
+
+    bool result = false;
+    ma_mutex_lock(&AUDIO.System.lock);
+    result = stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1];
+    ma_mutex_unlock(&AUDIO.System.lock);
+    return result;
+}
+
+// Play audio stream
+void PlayAudioStream(AudioStream stream)
+{
+    PlayAudioBuffer(stream.buffer);
+}
+
+// Play audio stream
+void PauseAudioStream(AudioStream stream)
+{
+    PauseAudioBuffer(stream.buffer);
+}
+
+// Resume audio stream playing
+void ResumeAudioStream(AudioStream stream)
+{
+    ResumeAudioBuffer(stream.buffer);
+}
+
+// Check if audio stream is playing
+bool IsAudioStreamPlaying(AudioStream stream)
+{
+    return IsAudioBufferPlaying(stream.buffer);
+}
+
+// Stop audio stream
+void StopAudioStream(AudioStream stream)
+{
+    StopAudioBuffer(stream.buffer);
+}
+
+// Set volume for audio stream (1.0 is max level)
+void SetAudioStreamVolume(AudioStream stream, float volume)
+{
+    SetAudioBufferVolume(stream.buffer, volume);
+}
+
+// Set pitch for audio stream (1.0 is base level)
+void SetAudioStreamPitch(AudioStream stream, float pitch)
+{
+    SetAudioBufferPitch(stream.buffer, pitch);
+}
+
+// Set pan for audio stream
+void SetAudioStreamPan(AudioStream stream, float pan)
+{
+    SetAudioBufferPan(stream.buffer, pan);
+}
+
+// Default size for new audio streams
+void SetAudioStreamBufferSizeDefault(int size)
+{
+    AUDIO.Buffer.defaultSize = size;
+}
+
+// Audio thread callback to request new data
+void SetAudioStreamCallback(AudioStream stream, AudioCallback callback)
+{
+    if (stream.buffer != NULL)
+    {
+        ma_mutex_lock(&AUDIO.System.lock);
+        stream.buffer->callback = callback;
+        ma_mutex_unlock(&AUDIO.System.lock);
+    }
+}
+
+// Add processor to audio stream. Contrary to buffers, the order of processors is important
+// The new processor must be added at the end. As there aren't supposed to be a lot of processors attached to
+// a given stream, we iterate through the list to find the end. That way we don't need a pointer to the last element
+void AttachAudioStreamProcessor(AudioStream stream, AudioCallback process)
+{
+    ma_mutex_lock(&AUDIO.System.lock);
+
+    rAudioProcessor *processor = (rAudioProcessor *)RL_CALLOC(1, sizeof(rAudioProcessor));
+    processor->process = process;
+
+    rAudioProcessor *last = stream.buffer->processor;
+
+    while (last && last->next)
+    {
+        last = last->next;
+    }
+    if (last)
+    {
+        processor->prev = last;
+        last->next = processor;
+    }
+    else stream.buffer->processor = processor;
+
+    ma_mutex_unlock(&AUDIO.System.lock);
+}
+
+// Remove processor from audio stream
+void DetachAudioStreamProcessor(AudioStream stream, AudioCallback process)
+{
+    ma_mutex_lock(&AUDIO.System.lock);
+
+    rAudioProcessor *processor = stream.buffer->processor;
+
+    while (processor)
+    {
+        rAudioProcessor *next = processor->next;
+        rAudioProcessor *prev = processor->prev;
+
+        if (processor->process == process)
+        {
+            if (stream.buffer->processor == processor) stream.buffer->processor = next;
+            if (prev) prev->next = next;
+            if (next) next->prev = prev;
+
+            RL_FREE(processor);
+        }
+
+        processor = next;
+    }
+
+    ma_mutex_unlock(&AUDIO.System.lock);
+}
+
+// Add processor to audio pipeline. Order of processors is important
+// Works the same way as {Attach,Detach}AudioStreamProcessor() functions, except
+// these two work on the already mixed output just before sending it to the sound hardware
+void AttachAudioMixedProcessor(AudioCallback process)
+{
+    ma_mutex_lock(&AUDIO.System.lock);
+
+    rAudioProcessor *processor = (rAudioProcessor *)RL_CALLOC(1, sizeof(rAudioProcessor));
+    processor->process = process;
+
+    rAudioProcessor *last = AUDIO.mixedProcessor;
+
+    while (last && last->next)
+    {
+        last = last->next;
+    }
+    if (last)
+    {
+        processor->prev = last;
+        last->next = processor;
+    }
+    else AUDIO.mixedProcessor = processor;
+
+    ma_mutex_unlock(&AUDIO.System.lock);
+}
+
+// Remove processor from audio pipeline
+void DetachAudioMixedProcessor(AudioCallback process)
+{
+    ma_mutex_lock(&AUDIO.System.lock);
+
+    rAudioProcessor *processor = AUDIO.mixedProcessor;
+
+    while (processor)
+    {
+        rAudioProcessor *next = processor->next;
+        rAudioProcessor *prev = processor->prev;
+
+        if (processor->process == process)
+        {
+            if (AUDIO.mixedProcessor == processor) AUDIO.mixedProcessor = next;
+            if (prev) prev->next = next;
+            if (next) next->prev = prev;
+
+            RL_FREE(processor);
+        }
+
+        processor = next;
+    }
+
+    ma_mutex_unlock(&AUDIO.System.lock);
+}
+
+
+//----------------------------------------------------------------------------------
+// Module 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)
+            {
+                StopAudioBufferInLockedState(audioBuffer);
+                break;
+            }
+        }
+    }
+
+    // Zero-fill excess
+    ma_uint32 totalFramesRemaining = (frameCount - framesRead);
+    if (totalFramesRemaining > 0)
+    {
+        memset((unsigned char *)framesOut + (framesRead*frameSizeInBytes), 0, totalFramesRemaining*frameSizeInBytes);
+
+        // For static buffers we can fill the remaining frames with silence for safety, but we don't want
+        // to report those frames as "read". The reason for this is that the caller uses the return value
+        // to know whether a non-looping sound has finished playback
+        if (audioBuffer->usage != AUDIO_BUFFER_USAGE_STATIC) framesRead += totalFramesRemaining;
+    }
+
+    return framesRead;
+}
+
+// Reads audio data from an AudioBuffer object in device format, returned data will be in a format appropriate for mixing
+static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, float *framesOut, ma_uint32 frameCount)
+{
+    // What's going on here is that we're continuously converting data from the AudioBuffer's internal format to the mixing format, which
+    // should be defined by the output format of the data converter. We do this until frameCount frames have been output. The important
+    // detail to remember here is that we never, ever attempt to read more input data than is required for the specified number of output
+    // frames. This can be achieved with ma_data_converter_get_required_input_frame_count()
+    ma_uint8 inputBuffer[4096] = { 0 };
+    ma_uint32 inputBufferFrameCap = sizeof(inputBuffer)/ma_get_bytes_per_frame(audioBuffer->converter.formatIn, audioBuffer->converter.channelsIn);
+
+    ma_uint32 totalOutputFramesProcessed = 0;
+    while (totalOutputFramesProcessed < frameCount)
+    {
+        ma_uint64 outputFramesToProcessThisIteration = frameCount - totalOutputFramesProcessed;
+        ma_uint64 inputFramesToProcessThisIteration = 0;
+
+        (void)ma_data_converter_get_required_input_frame_count(&audioBuffer->converter, outputFramesToProcessThisIteration, &inputFramesToProcessThisIteration);
+        if (inputFramesToProcessThisIteration > inputBufferFrameCap)
+        {
+            inputFramesToProcessThisIteration = inputBufferFrameCap;
+        }
+
+        float *runningFramesOut = framesOut + (totalOutputFramesProcessed*audioBuffer->converter.channelsOut);
+
+        /* At this point we can convert the data to our mixing format. */
+        ma_uint64 inputFramesProcessedThisIteration = ReadAudioBufferFramesInInternalFormat(audioBuffer, inputBuffer, (ma_uint32)inputFramesToProcessThisIteration);    /* 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)
+                        {
+                            StopAudioBufferInLockedState(audioBuffer);
+                            break;
+                        }
+                        else
+                        {
+                            // Should never get here, but just for safety,
+                            // move the cursor position back to the start and continue the loop
+                            audioBuffer->frameCursorPos = 0;
+                            continue;
+                        }
+                    }
+                }
+
+                // If for some reason we weren't able to read every frame we'll need to break from the loop
+                // Not doing this could theoretically put us into an infinite loop
+                if (framesToRead > 0) break;
+            }
+        }
+    }
+
+    rAudioProcessor *processor = AUDIO.mixedProcessor;
+    while (processor)
+    {
+        processor->process(pFramesOut, frameCount);
+        processor = processor->next;
+    }
+
+    ma_mutex_unlock(&AUDIO.System.lock);
+}
+
+// Main mixing function, pretty simple in this project, just an accumulation
+// NOTE: framesOut is both an input and an output, it is initially filled with zeros outside of this function
+static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, AudioBuffer *buffer)
+{
+    const float localVolume = buffer->volume;
+    const ma_uint32 channels = AUDIO.System.device.playback.channels;
+
+    if (channels == 2)  // We consider panning
+    {
+        const float 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);
+            }
+        }
+    }
+}
+
+// Check if an audio buffer is playing, assuming the audio system mutex has been locked
+static bool IsAudioBufferPlayingInLockedState(AudioBuffer *buffer)
+{
+    bool result = false;
+
+    if (buffer != NULL) result = (buffer->playing && !buffer->paused);
+
+    return result;
+}
+
+// Stop an audio buffer, assuming the audio system mutex has been locked
+static void StopAudioBufferInLockedState(AudioBuffer *buffer)
+{
+    if (buffer != NULL)
+    {
+        if (IsAudioBufferPlayingInLockedState(buffer))
+        {
+            buffer->playing = false;
+            buffer->paused = false;
+            buffer->frameCursorPos = 0;
+            buffer->framesProcessed = 0;
+            buffer->isSubBufferProcessed[0] = true;
+            buffer->isSubBufferProcessed[1] = true;
+        }
+    }
+}
+
+// Update audio stream, assuming the audio system mutex has been locked
+static void UpdateAudioStreamInLockedState(AudioStream stream, const void *data, int frameCount)
+{
+    if (stream.buffer != NULL)
+    {
+        if (stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1])
+        {
+            ma_uint32 subBufferToUpdate = 0;
+
+            if (stream.buffer->isSubBufferProcessed[0] && stream.buffer->isSubBufferProcessed[1])
+            {
+                // Both buffers are available for updating
+                // Update the first one and make sure the cursor is moved back to the front
+                subBufferToUpdate = 0;
+                stream.buffer->frameCursorPos = 0;
+            }
+            else
+            {
+                // Just update whichever sub-buffer is processed
+                subBufferToUpdate = (stream.buffer->isSubBufferProcessed[0])? 0 : 1;
+            }
+
+            ma_uint32 subBufferSizeInFrames = stream.buffer->sizeInFrames/2;
+            unsigned char *subBuffer = stream.buffer->data + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate);
+
+            // 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");
+    }
+}
+
+// Some required functions for audio standalone module version
+#if defined(RAUDIO_STANDALONE)
+// Check file extension
+static bool IsFileExtension(const char *fileName, const char *ext)
+{
+    bool result = false;
+    const char *fileExt;
+
+    if ((fileExt = strrchr(fileName, '.')) != NULL)
+    {
+        if (strcmp(fileExt, ext) == 0) result = true;
+    }
+
+    return result;
+}
+
+// Get pointer to extension for a filename string (includes the dot: .png)
+static const char *GetFileExtension(const char *fileName)
+{
+    const char *dot = strrchr(fileName, '.');
+
+    if (!dot || dot == fileName) return NULL;
+
+    return dot;
+}
+
+// String pointer reverse break: returns right-most occurrence of charset in s
+static const char *strprbrk(const char *s, const char *charset)
+{
+    const char *latestMatch = NULL;
+    for (; s = strpbrk(s, charset), s != NULL; latestMatch = s++) { }
+    return latestMatch;
+}
+
+// Get pointer to filename for a path string
+static const char *GetFileName(const char *filePath)
+{
+    const char *fileName = NULL;
+    if (filePath != NULL) fileName = strprbrk(filePath, "\\/");
+
+    if (!fileName) return filePath;
+
+    return fileName + 1;
+}
+
+// Get filename string without extension (uses static string)
+static const char *GetFileNameWithoutExt(const char *filePath)
+{
+    #define MAX_FILENAMEWITHOUTEXT_LENGTH   256
+
+    static char fileName[MAX_FILENAMEWITHOUTEXT_LENGTH] = { 0 };
+    memset(fileName, 0, MAX_FILENAMEWITHOUTEXT_LENGTH);
+
+    if (filePath != NULL) strcpy(fileName, GetFileName(filePath));   // Get filename with extension
+
+    int size = (int)strlen(fileName);   // Get size in bytes
+
+    for (int i = 0; (i < size) && (i < MAX_FILENAMEWITHOUTEXT_LENGTH); i++)
+    {
+        if (fileName[i] == '.')
+        {
+            // NOTE: We break on first '.' found
+            fileName[i] = '\0';
+            break;
+        }
+    }
+
+    return fileName;
+}
+
+// Load data from file into a buffer
+static unsigned char *LoadFileData(const char *fileName, int *dataSize)
+{
+    unsigned char *data = NULL;
+    *dataSize = 0;
+
+    if (fileName != NULL)
+    {
+        FILE *file = fopen(fileName, "rb");
+
+        if (file != NULL)
+        {
+            // WARNING: On binary streams SEEK_END could not be found,
+            // using fseek() and ftell() could not work in some (rare) cases
+            fseek(file, 0, SEEK_END);
+            int size = ftell(file);
+            fseek(file, 0, SEEK_SET);
+
+            if (size > 0)
+            {
+                data = (unsigned char *)RL_MALLOC(size*sizeof(unsigned char));
+
+                // NOTE: fread() returns number of read elements instead of bytes, so we read [1 byte, size elements]
+                unsigned int count = (unsigned int)fread(data, sizeof(unsigned char), size, file);
+                *dataSize = count;
+
+                if (count != size) TRACELOG(LOG_WARNING, "FILEIO: [%s] File partially loaded", fileName);
+                else TRACELOG(LOG_INFO, "FILEIO: [%s] File loaded successfully", fileName);
+            }
+            else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to read file", fileName);
+
+            fclose(file);
+        }
+        else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open file", fileName);
+    }
+    else TRACELOG(LOG_WARNING, "FILEIO: File name provided is not valid");
+
+    return data;
+}
+
+// Save data to file from buffer
+static bool SaveFileData(const char *fileName, void *data, int dataSize)
+{
+    if (fileName != NULL)
+    {
+        FILE *file = fopen(fileName, "wb");
+
+        if (file != NULL)
+        {
+            unsigned int count = (unsigned int)fwrite(data, sizeof(unsigned char), dataSize, file);
+
+            if (count == 0) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to write file", fileName);
+            else if (count != dataSize) TRACELOG(LOG_WARNING, "FILEIO: [%s] File partially written", fileName);
+            else TRACELOG(LOG_INFO, "FILEIO: [%s] File saved successfully", fileName);
+
+            fclose(file);
+        }
+        else
+        {
+            TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open file", fileName);
+            return false;
+        }
+    }
+    else
+    {
+        TRACELOG(LOG_WARNING, "FILEIO: File name provided is not valid");
+        return false;
+    }
+
+    return true;
+}
+
+// Save text data to file (write), string must be '\0' terminated
+static bool SaveFileText(const char *fileName, char *text)
+{
+    if (fileName != NULL)
+    {
+        FILE *file = fopen(fileName, "wt");
+
+        if (file != NULL)
+        {
+            int count = fprintf(file, "%s", text);
+
+            if (count == 0) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to write text file", fileName);
+            else TRACELOG(LOG_INFO, "FILEIO: [%s] Text file saved successfully", fileName);
+
+            fclose(file);
+        }
+        else
+        {
+            TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open text file", fileName);
+            return false;
+        }
+    }
+    else
+    {
+        TRACELOG(LOG_WARNING, "FILEIO: File name provided is not valid");
+        return false;
+    }
+
+    return true;
+}
+#endif
+
+#undef AudioBuffer
+
+#endif      // SUPPORT_MODULE_RAUDIO