text: /**********************************************************************************************
  *
  *   raudio - A simple and easy-to-use audio library based on miniaudio
  *
  *   FEATURES:
  *       - Manage audio device (init/close)
  *       - Manage raw audio context
  *       - Manage mixing channels
  *       - Load and unload audio files
  *       - Format wave data (sample rate, size, channels)
  *       - Play/Stop/Pause/Resume loaded audio
  *
  *   CONFIGURATION:
  *
  *   #define RAUDIO_STANDALONE
  *       Define to use the module as standalone library (independently of raylib).
  *       Required types and functions are defined in the same module.
  *
  *   #define SUPPORT_FILEFORMAT_WAV
  *   #define SUPPORT_FILEFORMAT_OGG
  *   #define SUPPORT_FILEFORMAT_XM
  *   #define SUPPORT_FILEFORMAT_MOD
  *   #define SUPPORT_FILEFORMAT_FLAC
  *   #define SUPPORT_FILEFORMAT_MP3
  *       Selected desired fileformats to be supported for loading. Some of those formats are
  *       supported by default, to remove support, just comment unrequired #define in this module
  *
  *   DEPENDENCIES:
  *       miniaudio.h  - Audio device management lib (https://github.com/dr-soft/miniaudio)
  *       stb_vorbis.h - Ogg audio files loading (http://www.nothings.org/stb_vorbis/)
  *       dr_mp3.h     - MP3 audio file loading (https://github.com/mackron/dr_libs)
  *       dr_flac.h    - FLAC audio file loading (https://github.com/mackron/dr_libs)
  *       jar_xm.h     - XM module file loading
  *       jar_mod.h    - MOD audio file loading
  *
  *   CONTRIBUTORS:
  *       David Reid (github: @mackron) (Nov. 2017):
  *           - Complete port to miniaudio library
  *
  *       Joshua Reisenauer (github: @kd7tck) (2015)
  *           - XM audio module support (jar_xm)
  *           - MOD audio module support (jar_mod)
  *           - Mixing channels support
  *           - Raw audio context support
  *
  *
  *   LICENSE: zlib/libpng
  *
  *   Copyright (c) 2013-2020 Ramon Santamaria (@raysan5)
  *
  *   This software is provided "as-is", without any express or implied warranty. In no event
  *   will the authors be held liable for any damages arising from the use of this software.
  *
  *   Permission is granted to anyone to use this software for any purpose, including commercial
  *   applications, and to alter it and redistribute it freely, subject to the following restrictions:
  *
  *     1. The origin of this software must not be misrepresented; you must not claim that you
  *     wrote the original software. If you use this software in a product, an acknowledgment
  *     in the product documentation would be appreciated but is not required.
  *
  *     2. Altered source versions must be plainly marked as such, and must not be misrepresented
  *     as being the original software.
  *
  *     3. This notice may not be removed or altered from any source distribution.
  *
  **********************************************************************************************/
  
  #if defined(RAUDIO_STANDALONE)
      #include "raudio.h"
      #include <stdarg.h>         // Required for: va_list, va_start(), vfprintf(), va_end()
  #else
      #include "raylib.h"         // Declares module functions
  
  // Check if config flags have been externally provided on compilation line
  #if !defined(EXTERNAL_CONFIG_FLAGS)
      #include "config.h"         // Defines module configuration flags
  #endif
      #include "utils.h"          // Required for: fopen() Android mapping
  #endif
  
  #if defined(_WIN32)
  // To avoid conflicting windows.h symbols with raylib, some flags are defined
  // WARNING: Those flags avoid inclusion of some Win32 headers that could be required
  // by user at some point and won't be included...
  //-------------------------------------------------------------------------------------
  
  // If defined, the following flags inhibit definition of the indicated items.
  #define NOGDICAPMASKS     // CC_*, LC_*, PC_*, CP_*, TC_*, RC_
  #define NOVIRTUALKEYCODES // VK_*
  #define NOWINMESSAGES     // WM_*, EM_*, LB_*, CB_*
  #define NOWINSTYLES       // WS_*, CS_*, ES_*, LBS_*, SBS_*, CBS_*
  #define NOSYSMETRICS      // SM_*
  #define NOMENUS           // MF_*
  #define NOICONS           // IDI_*
  #define NOKEYSTATES       // MK_*
  #define NOSYSCOMMANDS     // SC_*
  #define NORASTEROPS       // Binary and Tertiary raster ops
  #define NOSHOWWINDOW      // SW_*
  #define OEMRESOURCE       // OEM Resource values
  #define NOATOM            // Atom Manager routines
  #define NOCLIPBOARD       // Clipboard routines
  #define NOCOLOR           // Screen colors
  #define NOCTLMGR          // Control and Dialog routines
  #define NODRAWTEXT        // DrawText() and DT_*
  #define NOGDI             // All GDI defines and routines
  #define NOKERNEL          // All KERNEL defines and routines
  #define NOUSER            // All USER defines and routines
  //#define NONLS             // All NLS defines and routines
  #define NOMB              // MB_* and MessageBox()
  #define NOMEMMGR          // GMEM_*, LMEM_*, GHND, LHND, associated routines
  #define NOMETAFILE        // typedef METAFILEPICT
  #define NOMINMAX          // Macros min(a,b) and max(a,b)
  #define NOMSG             // typedef MSG and associated routines
  #define NOOPENFILE        // OpenFile(), OemToAnsi, AnsiToOem, and OF_*
  #define NOSCROLL          // SB_* and scrolling routines
  #define NOSERVICE         // All Service Controller routines, SERVICE_ equates, etc.
  #define NOSOUND           // Sound driver routines
  #define NOTEXTMETRIC      // typedef TEXTMETRIC and associated routines
  #define NOWH              // SetWindowsHook and WH_*
  #define NOWINOFFSETS      // GWL_*, GCL_*, associated routines
  #define NOCOMM            // COMM driver routines
  #define NOKANJI           // Kanji support stuff.
  #define NOHELP            // Help engine interface.
  #define NOPROFILER        // Profiler interface.
  #define NODEFERWINDOWPOS  // DeferWindowPos routines
  #define NOMCX             // Modem Configuration Extensions
  
  // Type required before windows.h inclusion
  typedef struct tagMSG *LPMSG;
  
  #include <windows.h>
  
  // Type required by some unused function...
  typedef struct tagBITMAPINFOHEADER {
    DWORD biSize;
    LONG  biWidth;
    LONG  biHeight;
    WORD  biPlanes;
    WORD  biBitCount;
    DWORD biCompression;
    DWORD biSizeImage;
    LONG  biXPelsPerMeter;
    LONG  biYPelsPerMeter;
    DWORD biClrUsed;
    DWORD biClrImportant;
  } BITMAPINFOHEADER, *PBITMAPINFOHEADER;
  
  #include <objbase.h>
  #include <mmreg.h>
  #include <mmsystem.h>
  
  // Some required types defined for MSVC/TinyC compiler
  #if defined(_MSC_VER) || defined(__TINYC__)
      #include "propidl.h"
  #endif
  #endif
  
  #define MA_MALLOC RL_MALLOC
  #define MA_FREE RL_FREE
  
  #define MA_NO_JACK
  #define MINIAUDIO_IMPLEMENTATION
  #include "external/miniaudio.h"         // miniaudio library
  #undef PlaySound                        // Win32 API: windows.h > mmsystem.h defines PlaySound macro
  
  #include <stdlib.h>                     // Required for: malloc(), free()
  #include <stdio.h>                      // Required for: FILE, fopen(), fclose(), fread()
  
  #if defined(RAUDIO_STANDALONE)
      #include <string.h>                 // Required for: strcmp() [Used in IsFileExtension()]
  
      #if !defined(TRACELOG)
          #define TRACELOG(level, ...) (void)0
      #endif
  #endif
  
  #if defined(SUPPORT_FILEFORMAT_OGG)
      // TODO: Remap malloc()/free() calls to RL_MALLOC/RL_FREE
  
      #define STB_VORBIS_IMPLEMENTATION
      #include "external/stb_vorbis.h"    // OGG loading functions
  #endif
  
  #if defined(SUPPORT_FILEFORMAT_XM)
      #define JARXM_MALLOC RL_MALLOC
      #define JARXM_FREE RL_FREE
  
      #define JAR_XM_IMPLEMENTATION
      #include "external/jar_xm.h"        // XM loading functions
  #endif
  
  #if defined(SUPPORT_FILEFORMAT_MOD)
      #define JARMOD_MALLOC RL_MALLOC
      #define JARMOD_FREE RL_FREE
  
      #define JAR_MOD_IMPLEMENTATION
      #include "external/jar_mod.h"       // MOD loading functions
  #endif
  
  #if defined(SUPPORT_FILEFORMAT_WAV)
      #define DRWAV_MALLOC RL_MALLOC
      #define DRWAV_REALLOC RL_REALLOC
      #define DRWAV_FREE RL_FREE
  
      #define DR_WAV_IMPLEMENTATION
      #include "external/dr_wav.h"        // WAV loading functions
  #endif
  
  #if defined(SUPPORT_FILEFORMAT_MP3)
      #define DRMP3_MALLOC RL_MALLOC
      #define DRMP3_REALLOC RL_REALLOC
      #define DRMP3_FREE RL_FREE
  
      #define DR_MP3_IMPLEMENTATION
      #include "external/dr_mp3.h"        // MP3 loading functions
  #endif
  
  #if defined(SUPPORT_FILEFORMAT_FLAC)
      #define DRFLAC_MALLOC RL_MALLOC
      #define DRFLAC_REALLOC RL_REALLOC
      #define DRFLAC_FREE RL_FREE
  
      #define DR_FLAC_IMPLEMENTATION
      #define DR_FLAC_NO_WIN32_IO
      #include "external/dr_flac.h"       // FLAC loading functions
  #endif
  
  #if defined(_MSC_VER)
      #undef bool
  #endif
  
  //----------------------------------------------------------------------------------
  // Defines and Macros
  //----------------------------------------------------------------------------------
  #ifndef AUDIO_DEVICE_FORMAT
      #define AUDIO_DEVICE_FORMAT    ma_format_f32    // Device output format (float-32bit)
  #endif
  #ifndef AUDIO_DEVICE_CHANNELS
      #define AUDIO_DEVICE_CHANNELS              2    // Device output channels: stereo
  #endif
  #ifndef AUDIO_DEVICE_SAMPLE_RATE
      #define AUDIO_DEVICE_SAMPLE_RATE       44100    // Device output sample rate
  #endif
  #ifndef MAX_AUDIO_BUFFER_POOL_CHANNELS
      #define MAX_AUDIO_BUFFER_POOL_CHANNELS    16    // Audio pool channels
  #endif
  #ifndef DEFAULT_AUDIO_BUFFER_SIZE
      #define DEFAULT_AUDIO_BUFFER_SIZE       4096    // Default audio buffer size
  #endif
  
  
  //----------------------------------------------------------------------------------
  // Types and Structures Definition
  //----------------------------------------------------------------------------------
  
  // Music context type
  // NOTE: Depends on data structure provided by the library
  // in charge of reading the different file types
  typedef enum {
      MUSIC_AUDIO_WAV = 0,
      MUSIC_AUDIO_OGG,
      MUSIC_AUDIO_FLAC,
      MUSIC_AUDIO_MP3,
      MUSIC_MODULE_XM,
      MUSIC_MODULE_MOD
  } MusicContextType;
  
  #if defined(RAUDIO_STANDALONE)
  typedef enum {
      LOG_ALL,
      LOG_TRACE,
      LOG_DEBUG,
      LOG_INFO,
      LOG_WARNING,
      LOG_ERROR,
      LOG_FATAL,
      LOG_NONE
  } TraceLogType;
  #endif
  
  // NOTE: Different logic is used when feeding data to the playback device
  // depending on whether or not data is streamed (Music vs Sound)
  typedef enum {
      AUDIO_BUFFER_USAGE_STATIC = 0,
      AUDIO_BUFFER_USAGE_STREAM
  } AudioBufferUsage;
  
  // Audio buffer structure
  struct rAudioBuffer {
      ma_data_converter converter;    // Audio data converter
  
      float volume;                   // Audio buffer volume
      float pitch;                    // Audio buffer pitch
  
      bool playing;                   // Audio buffer state: AUDIO_PLAYING
      bool paused;                    // Audio buffer state: AUDIO_PAUSED
      bool looping;                   // Audio buffer looping, always true for AudioStreams
      int usage;                      // Audio buffer usage mode: STATIC or STREAM
  
      bool isSubBufferProcessed[2];   // SubBuffer processed (virtual double buffer)
      unsigned int sizeInFrames;      // Total buffer size in frames
      unsigned int frameCursorPos;    // Frame cursor position
      unsigned int totalFramesProcessed;  // Total frames processed in this buffer (required for play timing)
  
      unsigned char *data;            // Data buffer, on music stream keeps filling
  
      rAudioBuffer *next;             // Next audio buffer on the list
      rAudioBuffer *prev;             // Previous audio buffer on the list
  };
  
  #define AudioBuffer rAudioBuffer    // HACK: To avoid CoreAudio (macOS) symbol collision
  
  // Audio data context
  typedef struct AudioData {
      struct {
          ma_context context;         // miniaudio context data
          ma_device device;           // miniaudio device
          ma_mutex lock;              // miniaudio mutex lock
          bool isReady;               // Check if audio device is ready
      } System;
      struct {
          AudioBuffer *first;         // Pointer to first AudioBuffer in the list
          AudioBuffer *last;          // Pointer to last AudioBuffer in the list
          int defaultSize;            // Default audio buffer size for audio streams
      } Buffer;
      struct {
          AudioBuffer *pool[MAX_AUDIO_BUFFER_POOL_CHANNELS];      // Multichannel AudioBuffer pointers pool
          unsigned int poolCounter;                               // AudioBuffer pointers pool counter
          unsigned int channels[MAX_AUDIO_BUFFER_POOL_CHANNELS];  // AudioBuffer pool channels
      } MultiChannel;
  } AudioData;
  
  //----------------------------------------------------------------------------------
  // Global Variables Definition
  //----------------------------------------------------------------------------------
  static AudioData AUDIO = {          // Global AUDIO context
  
      // NOTE: Music buffer size is defined by number of samples, independent of sample size and channels number
      // After some math, considering a sampleRate of 48000, a buffer refill rate of 1/60 seconds and a
      // standard double-buffering system, a 4096 samples buffer has been chosen, it should be enough
      // In case of music-stalls, just increase this number
      .Buffer.defaultSize = DEFAULT_AUDIO_BUFFER_SIZE
  };
  
  //----------------------------------------------------------------------------------
  // Module specific Functions Declaration
  //----------------------------------------------------------------------------------
  static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message);
  static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount);
  static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume);
  
  static void InitAudioBufferPool(void);                  // Initialise the multichannel buffer pool
  static void CloseAudioBufferPool(void);                 // Close the audio buffers pool
  
  #if defined(SUPPORT_FILEFORMAT_WAV)
  static Wave LoadWAV(const char *fileName);              // Load WAV file
  static int SaveWAV(Wave wave, const char *fileName);    // Save wave data as WAV file
  #endif
  #if defined(SUPPORT_FILEFORMAT_OGG)
  static Wave LoadOGG(const char *fileName);              // Load OGG file
  #endif
  #if defined(SUPPORT_FILEFORMAT_FLAC)
  static Wave LoadFLAC(const char *fileName);             // Load FLAC file
  #endif
  #if defined(SUPPORT_FILEFORMAT_MP3)
  static Wave LoadMP3(const char *fileName);              // Load MP3 file
  #endif
  
  #if defined(RAUDIO_STANDALONE)
  static bool IsFileExtension(const char *fileName, const char *ext); // Check file extension
  static void 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 TrackAudioBuffer(AudioBuffer *buffer);
  void UntrackAudioBuffer(AudioBuffer *buffer);
  
  //----------------------------------------------------------------------------------
  // Module Functions Definition - Audio Device initialization and Closing
  //----------------------------------------------------------------------------------
  // Initialize audio device
  void InitAudioDevice(void)
  {
      // TODO: Load AUDIO context memory dynamically?
  
      // Init audio context
      ma_context_config ctxConfig = ma_context_config_init();
      ctxConfig.logCallback = OnLog;
  
      ma_result result = ma_context_init(NULL, 0, &ctxConfig, &AUDIO.System.context);
      if (result != MA_SUCCESS)
      {
          TRACELOG(LOG_ERROR, "AUDIO: Failed to initialize context");
          return;
      }
  
      // Init audio device
      // NOTE: Using the default device. Format is floating point because it simplifies mixing.
      ma_device_config config = ma_device_config_init(ma_device_type_playback);
      config.playback.pDeviceID = NULL;  // NULL for the default playback AUDIO.System.device.
      config.playback.format = AUDIO_DEVICE_FORMAT;
      config.playback.channels = AUDIO_DEVICE_CHANNELS;
      config.capture.pDeviceID = NULL;  // NULL for the default capture AUDIO.System.device.
      config.capture.format = ma_format_s16;
      config.capture.channels = 1;
      config.sampleRate = AUDIO_DEVICE_SAMPLE_RATE;
      config.dataCallback = OnSendAudioDataToDevice;
      config.pUserData = NULL;
  
      result = ma_device_init(&AUDIO.System.context, &config, &AUDIO.System.device);
      if (result != MA_SUCCESS)
      {
          TRACELOG(LOG_ERROR, "AUDIO: Failed to initialize playback device");
          ma_context_uninit(&AUDIO.System.context);
          return;
      }
  
      // Keep the device running the whole time. May want to consider doing something a bit smarter and only have the device running
      // while there's at least one sound being played.
      result = ma_device_start(&AUDIO.System.device);
      if (result != MA_SUCCESS)
      {
          TRACELOG(LOG_ERROR, "AUDIO: Failed to start playback device");
          ma_device_uninit(&AUDIO.System.device);
          ma_context_uninit(&AUDIO.System.context);
          return;
      }
  
      // Mixing happens on a seperate thread which means we need to synchronize. I'm using a mutex here to make things simple, but may
      // want to look at something a bit smarter later on to keep everything real-time, if that's necessary.
      if (ma_mutex_init(&AUDIO.System.context, &AUDIO.System.lock) != MA_SUCCESS)
      {
          TRACELOG(LOG_ERROR, "AUDIO: Failed to create mutex for mixing");
          ma_device_uninit(&AUDIO.System.device);
          ma_context_uninit(&AUDIO.System.context);
          return;
      }
  
      TRACELOG(LOG_INFO, "AUDIO: Device initialized successfully");
      TRACELOG(LOG_INFO, "    > Backend:       miniaudio / %s", ma_get_backend_name(AUDIO.System.context.backend));
      TRACELOG(LOG_INFO, "    > Format:        %s -> %s", ma_get_format_name(AUDIO.System.device.playback.format), ma_get_format_name(AUDIO.System.device.playback.internalFormat));
      TRACELOG(LOG_INFO, "    > Channels:      %d -> %d", AUDIO.System.device.playback.channels, AUDIO.System.device.playback.internalChannels);
      TRACELOG(LOG_INFO, "    > Sample rate:   %d -> %d", AUDIO.System.device.sampleRate, AUDIO.System.device.playback.internalSampleRate);
      TRACELOG(LOG_INFO, "    > Periods size:  %d", AUDIO.System.device.playback.internalPeriodSizeInFrames*AUDIO.System.device.playback.internalPeriods);
  
      InitAudioBufferPool();
  
      AUDIO.System.isReady = true;
  }
  
  // Close the audio device for all contexts
  void CloseAudioDevice(void)
  {
      if (AUDIO.System.isReady)
      {
          ma_mutex_uninit(&AUDIO.System.lock);
          ma_device_uninit(&AUDIO.System.device);
          ma_context_uninit(&AUDIO.System.context);
  
          CloseAudioBufferPool();
  
          TRACELOG(LOG_INFO, "AUDIO: Device closed successfully");
      }
      else TRACELOG(LOG_WARNING, "AUDIO: Device could not be closed, not currently initialized");
  }
  
  // Check if device has been initialized successfully
  bool IsAudioDeviceReady(void)
  {
      return AUDIO.System.isReady;
  }
  
  // Set master volume (listener)
  void SetMasterVolume(float volume)
  {
      ma_device_set_master_volume(&AUDIO.System.device, volume);
  }
  
  //----------------------------------------------------------------------------------
  // Module Functions Definition - Audio Buffer management
  //----------------------------------------------------------------------------------
  
  // Initialize a new audio buffer (filled with silence)
  AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 sizeInFrames, int usage)
  {
      AudioBuffer *audioBuffer = (AudioBuffer *)RL_CALLOC(1, sizeof(AudioBuffer));
  
      if (audioBuffer == NULL)
      {
          TRACELOG(LOG_ERROR, "AUDIO: Failed to allocate memory for buffer");
          return NULL;
      }
  
      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_DEVICE_SAMPLE_RATE);
      converterConfig.resampling.allowDynamicSampleRate = true;        // Required for pitch shifting
  
      ma_result result = ma_data_converter_init(&converterConfig, &audioBuffer->converter);
  
      if (result != MA_SUCCESS)
      {
          TRACELOG(LOG_ERROR, "AUDIO: Failed to create data conversion pipeline");
          RL_FREE(audioBuffer);
          return NULL;
      }
  
      // Init audio buffer values
      audioBuffer->volume = 1.0f;
      audioBuffer->pitch = 1.0f;
      audioBuffer->playing = false;
      audioBuffer->paused = false;
      audioBuffer->looping = false;
      audioBuffer->usage = usage;
      audioBuffer->frameCursorPos = 0;
      audioBuffer->sizeInFrames = sizeInFrames;
  
      // Buffers should be marked as processed by default so that a call to
      // UpdateAudioStream() immediately after initialization works correctly
      audioBuffer->isSubBufferProcessed[0] = true;
      audioBuffer->isSubBufferProcessed[1] = true;
  
      // Track audio buffer to linked list next position
      TrackAudioBuffer(audioBuffer);
  
      return audioBuffer;
  }
  
  // Delete an audio buffer
  void UnloadAudioBuffer(AudioBuffer *buffer)
  {
      if (buffer != NULL)
      {
          ma_data_converter_uninit(&buffer->converter);
          UntrackAudioBuffer(buffer);
          RL_FREE(buffer->data);
          RL_FREE(buffer);
      }
  }
  
  // Check if an audio buffer is playing
  bool IsAudioBufferPlaying(AudioBuffer *buffer)
  {
      bool result = false;
  
      if (buffer != NULL) result = (buffer->playing && !buffer->paused);
  
      return result;
  }
  
  // Play an audio buffer
  // NOTE: Buffer is restarted to the start.
  // Use PauseAudioBuffer() and ResumeAudioBuffer() if the playback position should be maintained.
  void PlayAudioBuffer(AudioBuffer *buffer)
  {
      if (buffer != NULL)
      {
          buffer->playing = true;
          buffer->paused = false;
          buffer->frameCursorPos = 0;
      }
  }
  
  // Stop an audio buffer
  void StopAudioBuffer(AudioBuffer *buffer)
  {
      if (buffer != NULL)
      {
          if (IsAudioBufferPlaying(buffer))
          {
              buffer->playing = false;
              buffer->paused = false;
              buffer->frameCursorPos = 0;
              buffer->totalFramesProcessed = 0;
              buffer->isSubBufferProcessed[0] = true;
              buffer->isSubBufferProcessed[1] = true;
          }
      }
  }
  
  // Pause an audio buffer
  void PauseAudioBuffer(AudioBuffer *buffer)
  {
      if (buffer != NULL) buffer->paused = true;
  }
  
  // Resume an audio buffer
  void ResumeAudioBuffer(AudioBuffer *buffer)
  {
      if (buffer != NULL) buffer->paused = false;
  }
  
  // Set volume for an audio buffer
  void SetAudioBufferVolume(AudioBuffer *buffer, float volume)
  {
      if (buffer != NULL) buffer->volume = volume;
  }
  
  // Set pitch for an audio buffer
  void SetAudioBufferPitch(AudioBuffer *buffer, float pitch)
  {
      if (buffer != NULL)
      {
          float pitchMul = pitch/buffer->pitch;
  
          // Pitching is just an adjustment of the sample rate.
          // Note that this changes the duration of the sound:
          //  - higher pitches will make the sound faster
          //  - lower pitches make it slower
          ma_uint32 newOutputSampleRate = (ma_uint32)((float)buffer->converter.config.sampleRateOut/pitchMul);
          buffer->pitch *= (float)buffer->converter.config.sampleRateOut/newOutputSampleRate;
  
          ma_data_converter_set_rate(&buffer->converter, buffer->converter.config.sampleRateIn, newOutputSampleRate);
      }
  }
  
  // Track audio buffer to linked list next position
  void TrackAudioBuffer(AudioBuffer *buffer)
  {
      ma_mutex_lock(&AUDIO.System.lock);
      {
          if (AUDIO.Buffer.first == NULL) AUDIO.Buffer.first = buffer;
          else
          {
              AUDIO.Buffer.last->next = buffer;
              buffer->prev = AUDIO.Buffer.last;
          }
  
          AUDIO.Buffer.last = buffer;
      }
      ma_mutex_unlock(&AUDIO.System.lock);
  }
  
  // Untrack audio buffer from linked list
  void UntrackAudioBuffer(AudioBuffer *buffer)
  {
      ma_mutex_lock(&AUDIO.System.lock);
      {
          if (buffer->prev == NULL) AUDIO.Buffer.first = buffer->next;
          else buffer->prev->next = buffer->next;
  
          if (buffer->next == NULL) AUDIO.Buffer.last = buffer->prev;
          else buffer->next->prev = buffer->prev;
  
          buffer->prev = NULL;
          buffer->next = NULL;
      }
      ma_mutex_unlock(&AUDIO.System.lock);
  }
  
  //----------------------------------------------------------------------------------
  // Module Functions Definition - Sounds loading and playing (.WAV)
  //----------------------------------------------------------------------------------
  
  // Load wave data from file
  Wave LoadWave(const char *fileName)
  {
      Wave wave = { 0 };
  
      if (false) { }
  #if defined(SUPPORT_FILEFORMAT_WAV)
      else if (IsFileExtension(fileName, ".wav")) wave = LoadWAV(fileName);
  #endif
  #if defined(SUPPORT_FILEFORMAT_OGG)
      else if (IsFileExtension(fileName, ".ogg")) wave = LoadOGG(fileName);
  #endif
  #if defined(SUPPORT_FILEFORMAT_FLAC)
      else if (IsFileExtension(fileName, ".flac")) wave = LoadFLAC(fileName);
  #endif
  #if defined(SUPPORT_FILEFORMAT_MP3)
      else if (IsFileExtension(fileName, ".mp3")) wave = LoadMP3(fileName);
  #endif
      else TRACELOG(LOG_WARNING, "FILEIO: [%s] File format not supported", fileName);
  
      return wave;
  }
  
  // Load sound from file
  // NOTE: The entire file is loaded to memory to be played (no-streaming)
  Sound LoadSound(const char *fileName)
  {
      Wave wave = LoadWave(fileName);
  
      Sound sound = LoadSoundFromWave(wave);
  
      UnloadWave(wave);       // Sound is loaded, we can unload wave
  
      return sound;
  }
  
  // Load sound from wave data
  // NOTE: Wave data must be unallocated manually
  Sound LoadSoundFromWave(Wave wave)
  {
      Sound sound = { 0 };
  
      if (wave.data != NULL)
      {
          // When using miniaudio we need to do our own mixing.
          // To simplify this we need convert the format of each sound to be consistent with
          // the format used to open the playback AUDIO.System.device. We can do this two ways:
          //
          //   1) Convert the whole sound in one go at load time (here).
          //   2) Convert the audio data in chunks at mixing time.
          //
          // First option has been selected, format conversion is done on the loading stage.
          // The downside is that it uses more memory if the original sound is u8 or s16.
          ma_format formatIn  = ((wave.sampleSize == 8)? ma_format_u8 : ((wave.sampleSize == 16)? ma_format_s16 : ma_format_f32));
          ma_uint32 frameCountIn = wave.sampleCount/wave.channels;
  
          ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, NULL, frameCountIn, formatIn, wave.channels, wave.sampleRate);
          if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed to get frame count for format conversion");
  
          AudioBuffer *audioBuffer = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, frameCount, AUDIO_BUFFER_USAGE_STATIC);
          if (audioBuffer == NULL) TRACELOG(LOG_WARNING, "SOUND: Failed to create buffer");
  
          frameCount = (ma_uint32)ma_convert_frames(audioBuffer->data, frameCount, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, wave.data, frameCountIn, formatIn, wave.channels, wave.sampleRate);
          if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed format conversion");
  
          sound.sampleCount = frameCount*AUDIO_DEVICE_CHANNELS;
          sound.stream.sampleRate = AUDIO_DEVICE_SAMPLE_RATE;
          sound.stream.sampleSize = 32;
          sound.stream.channels = AUDIO_DEVICE_CHANNELS;
          sound.stream.buffer = audioBuffer;
      }
  
      return sound;
  }
  
  // Unload wave data
  void UnloadWave(Wave wave)
  {
      if (wave.data != NULL) RL_FREE(wave.data);
  
      TRACELOG(LOG_INFO, "WAVE: Unloaded wave data from RAM");
  }
  
  // Unload sound
  void UnloadSound(Sound sound)
  {
      UnloadAudioBuffer(sound.stream.buffer);
  
      TRACELOG(LOG_INFO, "WAVE: Unloaded sound data from RAM");
  }
  
  // Update sound buffer with new data
  void UpdateSound(Sound sound, const void *data, int samplesCount)
  {
      if (sound.stream.buffer != NULL)
      {
          StopAudioBuffer(sound.stream.buffer);
  
          // TODO: May want to lock/unlock this since this data buffer is read at mixing time
          memcpy(sound.stream.buffer->data, data, samplesCount*ma_get_bytes_per_frame(sound.stream.buffer->converter.config.formatIn, sound.stream.buffer->converter.config.channelsIn));
      }
  }
  
  // Export wave data to file
  void ExportWave(Wave wave, const char *fileName)
  {
      bool success = false;
  
      if (false) { }
  #if defined(SUPPORT_FILEFORMAT_WAV)
      else if (IsFileExtension(fileName, ".wav")) success = SaveWAV(wave, fileName);
  #endif
      else if (IsFileExtension(fileName, ".raw"))
      {
          // Export raw sample data (without header)
          // NOTE: It's up to the user to track wave parameters
          SaveFileData(fileName, wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8);
          success = true;
      }
  
      if (success) TRACELOG(LOG_INFO, "FILEIO: [%s] Wave data exported successfully", fileName);
      else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to export wave data", fileName);
  }
  
  // Export wave sample data to code (.h)
  void ExportWaveAsCode(Wave wave, const char *fileName)
  {
  #ifndef TEXT_BYTES_PER_LINE
      #define TEXT_BYTES_PER_LINE     20
  #endif
  
      int waveDataSize = wave.sampleCount*wave.channels*wave.sampleSize/8;
  
      // NOTE: Text data buffer size is estimated considering wave data size in bytes
      // and requiring 6 char bytes for every byte: "0x00, "
      char *txtData = (char *)RL_CALLOC(6*waveDataSize + 2000, sizeof(char));
  
      int bytesCount = 0;
      bytesCount += sprintf(txtData + bytesCount, "\n//////////////////////////////////////////////////////////////////////////////////\n");
      bytesCount += sprintf(txtData + bytesCount, "//                                                                              //\n");
      bytesCount += sprintf(txtData + bytesCount, "// WaveAsCode exporter v1.0 - Wave data exported as an array of bytes           //\n");
      bytesCount += sprintf(txtData + bytesCount, "//                                                                              //\n");
      bytesCount += sprintf(txtData + bytesCount, "// more info and bugs-report:  github.com/raysan5/raylib                        //\n");
      bytesCount += sprintf(txtData + bytesCount, "// feedback and support:       ray[at]raylib.com                                //\n");
      bytesCount += sprintf(txtData + bytesCount, "//                                                                              //\n");
      bytesCount += sprintf(txtData + bytesCount, "// Copyright (c) 2018 Ramon Santamaria (@raysan5)                               //\n");
      bytesCount += sprintf(txtData + bytesCount, "//                                                                              //\n");
      bytesCount += sprintf(txtData + bytesCount, "//////////////////////////////////////////////////////////////////////////////////\n\n");
  
      char varFileName[256] = { 0 };
  #if !defined(RAUDIO_STANDALONE)
      // Get file name from path and convert variable name to uppercase
      strcpy(varFileName, GetFileNameWithoutExt(fileName));
      for (int i = 0; varFileName[i] != '\0'; i++) if (varFileName[i] >= 'a' && varFileName[i] <= 'z') { varFileName[i] = varFileName[i] - 32; }
  #else
      strcpy(varFileName, fileName);
  #endif
  
      bytesCount += sprintf(txtData + bytesCount, "// Wave data information\n");
      bytesCount += sprintf(txtData + bytesCount, "#define %s_SAMPLE_COUNT     %u\n", varFileName, wave.sampleCount);
      bytesCount += sprintf(txtData + bytesCount, "#define %s_SAMPLE_RATE      %u\n", varFileName, wave.sampleRate);
      bytesCount += sprintf(txtData + bytesCount, "#define %s_SAMPLE_SIZE      %u\n", varFileName, wave.sampleSize);
      bytesCount += sprintf(txtData + bytesCount, "#define %s_CHANNELS         %u\n\n", varFileName, wave.channels);
  
      // Write byte data as hexadecimal text
      bytesCount += sprintf(txtData + bytesCount, "static unsigned char %s_DATA[%i] = { ", varFileName, waveDataSize);
      for (int i = 0; i < waveDataSize - 1; i++) bytesCount += sprintf(txtData + bytesCount, ((i%TEXT_BYTES_PER_LINE == 0)? "0x%x,\n" : "0x%x, "), ((unsigned char *)wave.data)[i]);
      bytesCount += sprintf(txtData + bytesCount, "0x%x };\n", ((unsigned char *)wave.data)[waveDataSize - 1]);
  
      // NOTE: Text data length exported is determined by '\0' (NULL) character
      SaveFileText(fileName, txtData);
  
      RL_FREE(txtData);
  }
  
  // Play a sound
  void PlaySound(Sound sound)
  {
      PlayAudioBuffer(sound.stream.buffer);
  }
  
  // Play a sound in the multichannel buffer pool
  void PlaySoundMulti(Sound sound)
  {
      int index = -1;
      unsigned int oldAge = 0;
      int oldIndex = -1;
  
      // find the first non playing pool entry
      for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
      {
          if (AUDIO.MultiChannel.channels[i] > oldAge)
          {
              oldAge = AUDIO.MultiChannel.channels[i];
              oldIndex = i;
          }
  
          if (!IsAudioBufferPlaying(AUDIO.MultiChannel.pool[i]))
          {
              index = i;
              break;
          }
      }
  
      // If no none playing pool members can be index choose the oldest
      if (index == -1)
      {
          TRACELOG(LOG_WARNING, "SOUND: Buffer pool is already full, count: %i", AUDIO.MultiChannel.poolCounter);
  
          if (oldIndex == -1)
          {
              // Shouldn't be able to get here... but just in case something odd happens!
              TRACELOG(LOG_WARNING, "SOUND: Buffer pool could not determine oldest buffer not playing sound");
              return;
          }
  
          index = oldIndex;
  
          // Just in case...
          StopAudioBuffer(AUDIO.MultiChannel.pool[index]);
      }
  
      // Experimentally mutex lock doesn't seem to be needed this makes sense
      // as pool[index] isn't playing and the only stuff we're copying
      // shouldn't be changing...
  
      AUDIO.MultiChannel.channels[index] = AUDIO.MultiChannel.poolCounter;
      AUDIO.MultiChannel.poolCounter++;
  
      AUDIO.MultiChannel.pool[index]->volume = sound.stream.buffer->volume;
      AUDIO.MultiChannel.pool[index]->pitch = sound.stream.buffer->pitch;
      AUDIO.MultiChannel.pool[index]->looping = sound.stream.buffer->looping;
      AUDIO.MultiChannel.pool[index]->usage = sound.stream.buffer->usage;
      AUDIO.MultiChannel.pool[index]->isSubBufferProcessed[0] = false;
      AUDIO.MultiChannel.pool[index]->isSubBufferProcessed[1] = false;
      AUDIO.MultiChannel.pool[index]->sizeInFrames = sound.stream.buffer->sizeInFrames;
      AUDIO.MultiChannel.pool[index]->data = sound.stream.buffer->data;
  
      PlayAudioBuffer(AUDIO.MultiChannel.pool[index]);
  }
  
  // Stop any sound played with PlaySoundMulti()
  void StopSoundMulti(void)
  {
      for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) StopAudioBuffer(AUDIO.MultiChannel.pool[i]);
  }
  
  // Get number of sounds playing in the multichannel buffer pool
  int GetSoundsPlaying(void)
  {
      int counter = 0;
  
      for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
      {
          if (IsAudioBufferPlaying(AUDIO.MultiChannel.pool[i])) counter++;
      }
  
      return counter;
  }
  
  // Pause a sound
  void PauseSound(Sound sound)
  {
      PauseAudioBuffer(sound.stream.buffer);
  }
  
  // Resume a paused sound
  void ResumeSound(Sound sound)
  {
      ResumeAudioBuffer(sound.stream.buffer);
  }
  
  // Stop reproducing a sound
  void StopSound(Sound sound)
  {
      StopAudioBuffer(sound.stream.buffer);
  }
  
  // Check if a sound is playing
  bool IsSoundPlaying(Sound sound)
  {
      return IsAudioBufferPlaying(sound.stream.buffer);
  }
  
  // Set volume for a sound
  void SetSoundVolume(Sound sound, float volume)
  {
      SetAudioBufferVolume(sound.stream.buffer, volume);
  }
  
  // Set pitch for a sound
  void SetSoundPitch(Sound sound, float pitch)
  {
      SetAudioBufferPitch(sound.stream.buffer, pitch);
  }
  
  // Convert wave data to desired format
  void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
  {
      ma_format formatIn  = ((wave->sampleSize == 8)? ma_format_u8 : ((wave->sampleSize == 16)? ma_format_s16 : ma_format_f32));
      ma_format formatOut = ((      sampleSize == 8)? ma_format_u8 : ((      sampleSize == 16)? ma_format_s16 : ma_format_f32));
  
      ma_uint32 frameCountIn = wave->sampleCount;  // Is wave->sampleCount actually the frame count? That terminology needs to change, if so.
  
      ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, formatOut, channels, sampleRate, NULL, frameCountIn, formatIn, wave->channels, wave->sampleRate);
      if (frameCount == 0)
      {
          TRACELOG(LOG_WARNING, "WAVE: Failed to get frame count for format conversion");
          return;
      }
  
      void *data = RL_MALLOC(frameCount*channels*(sampleSize/8));
  
      frameCount = (ma_uint32)ma_convert_frames(data, frameCount, formatOut, channels, sampleRate, wave->data, frameCountIn, formatIn, wave->channels, wave->sampleRate);
      if (frameCount == 0)
      {
          TRACELOG(LOG_WARNING, "WAVE: Failed format conversion");
          return;
      }
  
      wave->sampleCount = frameCount;
      wave->sampleSize = sampleSize;
      wave->sampleRate = sampleRate;
      wave->channels = channels;
      RL_FREE(wave->data);
      wave->data = data;
  }
  
  // Copy a wave to a new wave
  Wave WaveCopy(Wave wave)
  {
      Wave newWave = { 0 };
  
      newWave.data = RL_MALLOC(wave.sampleCount*wave.sampleSize/8*wave.channels);
  
      if (newWave.data != NULL)
      {
          // NOTE: Size must be provided in bytes
          memcpy(newWave.data, wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8);
  
          newWave.sampleCount = wave.sampleCount;
          newWave.sampleRate = wave.sampleRate;
          newWave.sampleSize = wave.sampleSize;
          newWave.channels = wave.channels;
      }
  
      return newWave;
  }
  
  // Crop a wave to defined samples range
  // NOTE: Security check in case of out-of-range
  void WaveCrop(Wave *wave, int initSample, int finalSample)
  {
      if ((initSample >= 0) && (initSample < finalSample) &&
          (finalSample > 0) && ((unsigned int)finalSample < wave->sampleCount))
      {
          int sampleCount = finalSample - initSample;
  
          void *data = RL_MALLOC(sampleCount*wave->sampleSize/8*wave->channels);
  
          memcpy(data, (unsigned char *)wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->channels*wave->sampleSize/8);
  
          RL_FREE(wave->data);
          wave->data = data;
      }
      else TRACELOG(LOG_WARNING, "WAVE: Crop range out of bounds");
  }
  
  // Get samples data from wave as a floats array
  // NOTE: Returned sample values are normalized to range [-1..1]
  float *GetWaveData(Wave wave)
  {
      float *samples = (float *)RL_MALLOC(wave.sampleCount*wave.channels*sizeof(float));
  
      for (unsigned int i = 0; i < wave.sampleCount; i++)
      {
          for (unsigned int j = 0; j < wave.channels; j++)
          {
              if (wave.sampleSize == 8) samples[wave.channels*i + j] = (float)(((unsigned char *)wave.data)[wave.channels*i + j] - 127)/256.0f;
              else if (wave.sampleSize == 16) samples[wave.channels*i + j] = (float)((short *)wave.data)[wave.channels*i + j]/32767.0f;
              else if (wave.sampleSize == 32) samples[wave.channels*i + j] = ((float *)wave.data)[wave.channels*i + j];
          }
      }
  
      return samples;
  }
  
  //----------------------------------------------------------------------------------
  // Module Functions Definition - Music loading and stream playing (.OGG)
  //----------------------------------------------------------------------------------
  
  // Load music stream from file
  Music LoadMusicStream(const char *fileName)
  {
      Music music = { 0 };
      bool musicLoaded = false;
  
      if (false) { }
  #if defined(SUPPORT_FILEFORMAT_WAV)
      else if (IsFileExtension(fileName, ".wav"))
      {
          drwav *ctxWav = RL_MALLOC(sizeof(drwav));
          bool success = drwav_init_file(ctxWav, fileName, NULL);
  
          if (success)
          {
              music.ctxType = MUSIC_AUDIO_WAV;
              music.ctxData = ctxWav;
  
              music.stream = InitAudioStream(ctxWav->sampleRate, ctxWav->bitsPerSample, ctxWav->channels);
              music.sampleCount = (unsigned int)ctxWav->totalPCMFrameCount*ctxWav->channels;
              music.looping = true;   // Looping enabled by default
              musicLoaded = true;
          }
      }
  #endif
  #if defined(SUPPORT_FILEFORMAT_OGG)
      else if (IsFileExtension(fileName, ".ogg"))
      {
          // Open ogg audio stream
          music.ctxData = stb_vorbis_open_filename(fileName, NULL, NULL);
  
          if (music.ctxData != NULL)
          {
              music.ctxType = MUSIC_AUDIO_OGG;
              stb_vorbis_info info = stb_vorbis_get_info((stb_vorbis *)music.ctxData);  // Get Ogg file info
  
              // OGG bit rate defaults to 16 bit, it's enough for compressed format
              music.stream = InitAudioStream(info.sample_rate, 16, info.channels);
              music.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData)*info.channels;
              music.looping = true;   // Looping enabled by default
              musicLoaded = true;
          }
      }
  #endif
  #if defined(SUPPORT_FILEFORMAT_FLAC)
      else if (IsFileExtension(fileName, ".flac"))
      {
          music.ctxData = drflac_open_file(fileName);
  
          if (music.ctxData != NULL)
          {
              music.ctxType = MUSIC_AUDIO_FLAC;
              drflac *ctxFlac = (drflac *)music.ctxData;
  
              music.stream = InitAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
              music.sampleCount = (unsigned int)ctxFlac->totalSampleCount;
              music.looping = true;   // Looping enabled by default
              musicLoaded = true;
          }
      }
  #endif
  #if defined(SUPPORT_FILEFORMAT_MP3)
      else if (IsFileExtension(fileName, ".mp3"))
      {
          drmp3 *ctxMp3 = RL_MALLOC(sizeof(drmp3));
          music.ctxData = ctxMp3;
  
          int result = drmp3_init_file(ctxMp3, fileName, NULL);
  
          if (result > 0)
          {
              music.ctxType = MUSIC_AUDIO_MP3;
  
              music.stream = InitAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
              music.sampleCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels;
              music.looping = true;   // Looping enabled by default
              musicLoaded = true;
          }
      }
  #endif
  #if defined(SUPPORT_FILEFORMAT_XM)
      else if (IsFileExtension(fileName, ".xm"))
      {
          jar_xm_context_t *ctxXm = NULL;
  
          int result = jar_xm_create_context_from_file(&ctxXm, 48000, fileName);
  
          if (result == 0)    // XM AUDIO.System.context created successfully
          {
              music.ctxType = MUSIC_MODULE_XM;
              jar_xm_set_max_loop_count(ctxXm, 0);    // Set infinite number of loops
  
              // NOTE: Only stereo is supported for XM
              music.stream = InitAudioStream(48000, 16, 2);
              music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)*2;
              music.looping = true;   // Looping enabled by default
              jar_xm_reset(ctxXm);   // make sure we start at the beginning of the song
              musicLoaded = true;
  
              music.ctxData = ctxXm;
          }
      }
  #endif
  #if defined(SUPPORT_FILEFORMAT_MOD)
      else if (IsFileExtension(fileName, ".mod"))
      {
          jar_mod_context_t *ctxMod = RL_MALLOC(sizeof(jar_mod_context_t));
  
          jar_mod_init(ctxMod);
          int result = jar_mod_load_file(ctxMod, fileName);
  
          if (result > 0)
          {
              music.ctxType = MUSIC_MODULE_MOD;
  
              // NOTE: Only stereo is supported for MOD
              music.stream = InitAudioStream(48000, 16, 2);
              music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)*2;
              music.looping = true;   // Looping enabled by default
              musicLoaded = true;
  
              music.ctxData = ctxMod;
          }
      }
  #endif
      else TRACELOG(LOG_WARNING, "STREAM: [%s] Fileformat not supported", fileName);
      
      if (!musicLoaded)
      {
          if (false) { }
      #if defined(SUPPORT_FILEFORMAT_WAV)
          else if (music.ctxType == MUSIC_AUDIO_WAV) drwav_uninit((drwav *)music.ctxData);
      #endif
      #if defined(SUPPORT_FILEFORMAT_OGG)
          else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData);
      #endif
      #if defined(SUPPORT_FILEFORMAT_FLAC)
          else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData);
      #endif
      #if defined(SUPPORT_FILEFORMAT_MP3)
          else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
      #endif
      #if defined(SUPPORT_FILEFORMAT_XM)
          else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData);
      #endif
      #if defined(SUPPORT_FILEFORMAT_MOD)
          else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); }
      #endif
  
          TRACELOG(LOG_WARNING, "FILEIO: [%s] Music file could not be opened", fileName);
      }
      else
      {
          // Show some music stream info
          TRACELOG(LOG_INFO, "FILEIO: [%s] Music file successfully loaded:", fileName);
          TRACELOG(LOG_INFO, "    > Total samples: %i", music.sampleCount);
          TRACELOG(LOG_INFO, "    > Sample rate:   %i Hz", music.stream.sampleRate);
          TRACELOG(LOG_INFO, "    > Sample size:   %i bits", music.stream.sampleSize);
          TRACELOG(LOG_INFO, "    > Channels:      %i (%s)", music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi");
      }
  
      return music;
  }
  
  // Unload music stream
  void UnloadMusicStream(Music music)
  {
      CloseAudioStream(music.stream);
  
      if (false) { }
  #if defined(SUPPORT_FILEFORMAT_WAV)
      else if (music.ctxType == MUSIC_AUDIO_WAV) drwav_uninit((drwav *)music.ctxData);
  #endif
  #if defined(SUPPORT_FILEFORMAT_OGG)
      else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData);
  #endif
  #if defined(SUPPORT_FILEFORMAT_FLAC)
      else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData);
  #endif
  #if defined(SUPPORT_FILEFORMAT_MP3)
      else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
  #endif
  #if defined(SUPPORT_FILEFORMAT_XM)
      else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData);
  #endif
  #if defined(SUPPORT_FILEFORMAT_MOD)
      else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); }
  #endif
  }
  
  // Start music playing (open stream)
  void PlayMusicStream(Music music)
  {
      if (music.stream.buffer != NULL)
      {
          // For music streams, we need to make sure we maintain the frame cursor position
          // This is a hack for this section of code in UpdateMusicStream()
          // NOTE: In case window is minimized, music stream is stopped, just make sure to
          // play again on window restore: if (IsMusicPlaying(music)) PlayMusicStream(music);
          ma_uint32 frameCursorPos = music.stream.buffer->frameCursorPos;
          PlayAudioStream(music.stream);  // WARNING: This resets the cursor position.
          music.stream.buffer->frameCursorPos = frameCursorPos;
      }
  }
  
  // Pause music playing
  void PauseMusicStream(Music music)
  {
      PauseAudioStream(music.stream);
  }
  
  // Resume music playing
  void ResumeMusicStream(Music music)
  {
      ResumeAudioStream(music.stream);
  }
  
  // Stop music playing (close stream)
  void StopMusicStream(Music music)
  {
      StopAudioStream(music.stream);
  
      switch (music.ctxType)
      {
  #if defined(SUPPORT_FILEFORMAT_WAV)
          case MUSIC_AUDIO_WAV: drwav_seek_to_pcm_frame((drwav *)music.ctxData, 0); break;
  #endif
  #if defined(SUPPORT_FILEFORMAT_OGG)
          case MUSIC_AUDIO_OGG: stb_vorbis_seek_start((stb_vorbis *)music.ctxData); break;
  #endif
  #if defined(SUPPORT_FILEFORMAT_FLAC)
          case MUSIC_AUDIO_FLAC: drflac_seek_to_pcm_frame((drflac *)music.ctxData, 0); break;
  #endif
  #if defined(SUPPORT_FILEFORMAT_MP3)
          case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame((drmp3 *)music.ctxData, 0); break;
  #endif
  #if defined(SUPPORT_FILEFORMAT_XM)
          case MUSIC_MODULE_XM: jar_xm_reset((jar_xm_context_t *)music.ctxData); break;
  #endif
  #if defined(SUPPORT_FILEFORMAT_MOD)
          case MUSIC_MODULE_MOD: jar_mod_seek_start((jar_mod_context_t *)music.ctxData); break;
  #endif
          default: break;
      }
  }
  
  // Update (re-fill) music buffers if data already processed
  void UpdateMusicStream(Music music)
  {
      if (music.stream.buffer == NULL) return;
  
      bool streamEnding = false;
  
      unsigned int subBufferSizeInFrames = music.stream.buffer->sizeInFrames/2;
  
      // NOTE: Using dynamic allocation because it could require more than 16KB
      void *pcm = RL_CALLOC(subBufferSizeInFrames*music.stream.channels*music.stream.sampleSize/8, 1);
  
      int samplesCount = 0;    // Total size of data streamed in L+R samples for xm floats, individual L or R for ogg shorts
  
      // TODO: Get the sampleLeft using totalFramesProcessed... but first, get total frames processed correctly...
      //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
      int sampleLeft = music.sampleCount - (music.stream.buffer->totalFramesProcessed*music.stream.channels);
  
      while (IsAudioStreamProcessed(music.stream))
      {
          if ((sampleLeft/music.stream.channels) >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames*music.stream.channels;
          else samplesCount = sampleLeft;
  
          switch (music.ctxType)
          {
          #if defined(SUPPORT_FILEFORMAT_WAV)
              case MUSIC_AUDIO_WAV:
              {
                  // NOTE: Returns the number of samples to process (not required)
                  drwav_read_pcm_frames_s16((drwav *)music.ctxData, samplesCount/music.stream.channels, (short *)pcm);
  
              } break;
          #endif
          #if defined(SUPPORT_FILEFORMAT_OGG)
              case MUSIC_AUDIO_OGG:
              {
                  // NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
                  stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)pcm, samplesCount);
  
              } break;
          #endif
          #if defined(SUPPORT_FILEFORMAT_FLAC)
              case MUSIC_AUDIO_FLAC:
              {
                  // NOTE: Returns the number of samples to process (not required)
                  drflac_read_pcm_frames_s16((drflac *)music.ctxData, samplesCount, (short *)pcm);
  
              } break;
          #endif
          #if defined(SUPPORT_FILEFORMAT_MP3)
              case MUSIC_AUDIO_MP3:
              {
                  // NOTE: samplesCount, actually refers to framesCount and returns the number of frames processed
                  drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, samplesCount/music.stream.channels, (float *)pcm);
  
              } break;
          #endif
          #if defined(SUPPORT_FILEFORMAT_XM)
              case MUSIC_MODULE_XM:
              {
                  // NOTE: Internally this function considers 2 channels generation, so samplesCount/2
                  jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)pcm, samplesCount/2);
              } break;
          #endif
          #if defined(SUPPORT_FILEFORMAT_MOD)
              case MUSIC_MODULE_MOD:
              {
                  // NOTE: 3rd parameter (nbsample) specify the number of stereo 16bits samples you want, so sampleCount/2
                  jar_mod_fillbuffer((jar_mod_context_t *)music.ctxData, (short *)pcm, samplesCount/2, 0);
              } break;
          #endif
              default: break;
          }
  
          UpdateAudioStream(music.stream, pcm, samplesCount);
  
          if ((music.ctxType == MUSIC_MODULE_XM) || (music.ctxType == MUSIC_MODULE_MOD))
          {
              if (samplesCount > 1) sampleLeft -= samplesCount/2;
              else sampleLeft -= samplesCount;
          }
          else sampleLeft -= samplesCount;
  
          if (sampleLeft <= 0)
          {
              streamEnding = true;
              break;
          }
      }
  
      // Free allocated pcm data
      RL_FREE(pcm);
  
      // Reset audio stream for looping
      if (streamEnding)
      {
          StopMusicStream(music);                     // Stop music (and reset)
          if (music.looping) PlayMusicStream(music);  // Play again
      }
      else
      {
          // NOTE: In case window is minimized, music stream is stopped,
          // just make sure to play again on window restore
          if (IsMusicPlaying(music)) PlayMusicStream(music);
      }
  }
  
  // Check if any music is playing
  bool IsMusicPlaying(Music music)
  {
      return IsAudioStreamPlaying(music.stream);
  }
  
  // Set volume for music
  void SetMusicVolume(Music music, float volume)
  {
      SetAudioStreamVolume(music.stream, volume);
  }
  
  // Set pitch for music
  void SetMusicPitch(Music music, float pitch)
  {
      SetAudioStreamPitch(music.stream, pitch);
  }
  
  // Get music time length (in seconds)
  float GetMusicTimeLength(Music music)
  {
      float totalSeconds = 0.0f;
  
      totalSeconds = (float)music.sampleCount/(music.stream.sampleRate*music.stream.channels);
  
      return totalSeconds;
  }
  
  // Get current music time played (in seconds)
  float GetMusicTimePlayed(Music music)
  {
      float secondsPlayed = 0.0f;
  
      if (music.stream.buffer != NULL)
      {
          //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
          unsigned int samplesPlayed = music.stream.buffer->totalFramesProcessed*music.stream.channels;
          secondsPlayed = (float)samplesPlayed / (music.stream.sampleRate*music.stream.channels);
      }
  
      return secondsPlayed;
  }
  
  // Init audio stream (to stream audio pcm data)
  AudioStream InitAudioStream(unsigned int sampleRate, unsigned int sampleSize, unsigned int channels)
  {
      AudioStream stream = { 0 };
  
      stream.sampleRate = sampleRate;
      stream.sampleSize = sampleSize;
      stream.channels = channels;
  
      ma_format formatIn = ((stream.sampleSize == 8)? ma_format_u8 : ((stream.sampleSize == 16)? ma_format_s16 : ma_format_f32));
  
      // The size of a streaming buffer must be at least double the size of a period
      unsigned int periodSize = AUDIO.System.device.playback.internalPeriodSizeInFrames;
      unsigned int subBufferSize = AUDIO.Buffer.defaultSize;     // Default buffer size (audio stream)
  
      if (subBufferSize < periodSize) subBufferSize = periodSize;
  
      // Create a double audio buffer of defined size
      stream.buffer = LoadAudioBuffer(formatIn, stream.channels, stream.sampleRate, subBufferSize*2, AUDIO_BUFFER_USAGE_STREAM);
  
      if (stream.buffer != NULL)
      {
          stream.buffer->looping = true;    // Always loop for streaming buffers
          TRACELOG(LOG_INFO, "STREAM: Initialized successfully (%i Hz, %i bit, %s)", stream.sampleRate, stream.sampleSize, (stream.channels == 1)? "Mono" : "Stereo");
      }
      else TRACELOG(LOG_WARNING, "STREAM: Failed to load audio buffer, stream could not be created");
  
      return stream;
  }
  
  // Close audio stream and free memory
  void CloseAudioStream(AudioStream stream)
  {
      UnloadAudioBuffer(stream.buffer);
  
      TRACELOG(LOG_INFO, "STREAM: Unloaded audio stream data from RAM");
  }
  
  // Update audio stream buffers with data
  // NOTE 1: Only updates one buffer of the stream source: unqueue -> update -> queue
  // NOTE 2: To unqueue a buffer it needs to be processed: IsAudioStreamProcessed()
  void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
  {
      if (stream.buffer != NULL)
      {
          if (stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1])
          {
              ma_uint32 subBufferToUpdate = 0;
  
              if (stream.buffer->isSubBufferProcessed[0] && stream.buffer->isSubBufferProcessed[1])
              {
                  // Both buffers are available for updating.
                  // Update the first one and make sure the cursor is moved back to the front.
                  subBufferToUpdate = 0;
                  stream.buffer->frameCursorPos = 0;
              }
              else
              {
                  // Just update whichever sub-buffer is processed.
                  subBufferToUpdate = (stream.buffer->isSubBufferProcessed[0])? 0 : 1;
              }
  
              ma_uint32 subBufferSizeInFrames = stream.buffer->sizeInFrames/2;
              unsigned char *subBuffer = stream.buffer->data + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate);
  
              // TODO: Get total frames processed on this buffer... DOES NOT WORK.
              stream.buffer->totalFramesProcessed += subBufferSizeInFrames;
  
              // Does this API expect a whole buffer to be updated in one go?
              // Assuming so, but if not will need to change this logic.
              if (subBufferSizeInFrames >= (ma_uint32)samplesCount/stream.channels)
              {
                  ma_uint32 framesToWrite = subBufferSizeInFrames;
  
                  if (framesToWrite > ((ma_uint32)samplesCount/stream.channels)) framesToWrite = (ma_uint32)samplesCount/stream.channels;
  
                  ma_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8);
                  memcpy(subBuffer, data, bytesToWrite);
  
                  // Any leftover frames should be filled with zeros.
                  ma_uint32 leftoverFrameCount = subBufferSizeInFrames - framesToWrite;
  
                  if (leftoverFrameCount > 0) memset(subBuffer + bytesToWrite, 0, leftoverFrameCount*stream.channels*(stream.sampleSize/8));
  
                  stream.buffer->isSubBufferProcessed[subBufferToUpdate] = false;
              }
              else TRACELOG(LOG_WARNING, "STREAM: Attempting to write too many frames to buffer");
          }
          else TRACELOG(LOG_WARNING, "STREAM: Buffer not available for updating");
      }
  }
  
  // Check if any audio stream buffers requires refill
  bool IsAudioStreamProcessed(AudioStream stream)
  {
      if (stream.buffer == NULL) return false;
  
      return (stream.buffer->isSubBufferProcessed[0] || stream.buffer->isSubBufferProcessed[1]);
  }
  
  // Play audio stream
  void PlayAudioStream(AudioStream stream)
  {
      PlayAudioBuffer(stream.buffer);
  }
  
  // Play audio stream
  void PauseAudioStream(AudioStream stream)
  {
      PauseAudioBuffer(stream.buffer);
  }
  
  // Resume audio stream playing
  void ResumeAudioStream(AudioStream stream)
  {
      ResumeAudioBuffer(stream.buffer);
  }
  
  // Check if audio stream is playing.
  bool IsAudioStreamPlaying(AudioStream stream)
  {
      return IsAudioBufferPlaying(stream.buffer);
  }
  
  // Stop audio stream
  void StopAudioStream(AudioStream stream)
  {
      StopAudioBuffer(stream.buffer);
  }
  
  // Set volume for audio stream (1.0 is max level)
  void SetAudioStreamVolume(AudioStream stream, float volume)
  {
      SetAudioBufferVolume(stream.buffer, volume);
  }
  
  // Set pitch for audio stream (1.0 is base level)
  void SetAudioStreamPitch(AudioStream stream, float pitch)
  {
      SetAudioBufferPitch(stream.buffer, pitch);
  }
  
  // Default size for new audio streams
  void SetAudioStreamBufferSizeDefault(int size)
  {
      AUDIO.Buffer.defaultSize = size;
  }
  
  //----------------------------------------------------------------------------------
  // Module specific Functions Definition
  //----------------------------------------------------------------------------------
  
  // Log callback function
  static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message)
  {
      (void)pContext;
      (void)pDevice;
  
      TRACELOG(LOG_ERROR, "miniaudio: %s", message);   // All log messages from miniaudio are errors
  }
  
  // Reads audio data from an AudioBuffer object in internal format.
  static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer, void *framesOut, ma_uint32 frameCount)
  {
      ma_uint32 subBufferSizeInFrames = (audioBuffer->sizeInFrames > 1)? audioBuffer->sizeInFrames/2 : audioBuffer->sizeInFrames;
      ma_uint32 currentSubBufferIndex = audioBuffer->frameCursorPos/subBufferSizeInFrames;
  
      if (currentSubBufferIndex > 1) return 0;
  
      // Another thread can update the processed state of buffers so
      // we just take a copy here to try and avoid potential synchronization problems
      bool isSubBufferProcessed[2];
      isSubBufferProcessed[0] = audioBuffer->isSubBufferProcessed[0];
      isSubBufferProcessed[1] = audioBuffer->isSubBufferProcessed[1];
  
      ma_uint32 frameSizeInBytes = ma_get_bytes_per_frame(audioBuffer->converter.config.formatIn, audioBuffer->converter.config.channelsIn);
  
      // Fill out every frame until we find a buffer that's marked as processed. Then fill the remainder with 0
      ma_uint32 framesRead = 0;
      while (1)
      {
          // We break from this loop differently depending on the buffer's usage
          //  - For static buffers, we simply fill as much data as we can
          //  - For streaming buffers we only fill the halves of the buffer that are processed
          //    Unprocessed halves must keep their audio data in-tact
          if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
          {
              if (framesRead >= frameCount) break;
          }
          else
          {
              if (isSubBufferProcessed[currentSubBufferIndex]) break;
          }
  
          ma_uint32 totalFramesRemaining = (frameCount - framesRead);
          if (totalFramesRemaining == 0) break;
  
          ma_uint32 framesRemainingInOutputBuffer;
          if (audioBuffer->usage == AUDIO_BUFFER_USAGE_STATIC)
          {
              framesRemainingInOutputBuffer = audioBuffer->sizeInFrames - audioBuffer->frameCursorPos;
          }
          else
          {
              ma_uint32 firstFrameIndexOfThisSubBuffer = subBufferSizeInFrames*currentSubBufferIndex;
              framesRemainingInOutputBuffer = subBufferSizeInFrames - (audioBuffer->frameCursorPos - firstFrameIndexOfThisSubBuffer);
          }
  
          ma_uint32 framesToRead = totalFramesRemaining;
          if (framesToRead > framesRemainingInOutputBuffer) framesToRead = framesRemainingInOutputBuffer;
  
          memcpy((unsigned char *)framesOut + (framesRead*frameSizeInBytes), audioBuffer->data + (audioBuffer->frameCursorPos*frameSizeInBytes), framesToRead*frameSizeInBytes);
          audioBuffer->frameCursorPos = (audioBuffer->frameCursorPos + framesToRead)%audioBuffer->sizeInFrames;
          framesRead += framesToRead;
  
          // If we've read to the end of the buffer, mark it as processed
          if (framesToRead == framesRemainingInOutputBuffer)
          {
              audioBuffer->isSubBufferProcessed[currentSubBufferIndex] = true;
              isSubBufferProcessed[currentSubBufferIndex] = true;
  
              currentSubBufferIndex = (currentSubBufferIndex + 1)%2;
  
              // We need to break from this loop if we're not looping
              if (!audioBuffer->looping)
              {
                  StopAudioBuffer(audioBuffer);
                  break;
              }
          }
      }
  
      // Zero-fill excess
      ma_uint32 totalFramesRemaining = (frameCount - framesRead);
      if (totalFramesRemaining > 0)
      {
          memset((unsigned char *)framesOut + (framesRead*frameSizeInBytes), 0, totalFramesRemaining*frameSizeInBytes);
  
          // For static buffers we can fill the remaining frames with silence for safety, but we don't want
          // to report those frames as "read". The reason for this is that the caller uses the return value
          // to know whether or not a non-looping sound has finished playback.
          if (audioBuffer->usage != AUDIO_BUFFER_USAGE_STATIC) framesRead += totalFramesRemaining;
      }
  
      return framesRead;
  }
  
  // Reads audio data from an AudioBuffer object in device format. Returned data will be in a format appropriate for mixing.
  static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, float *framesOut, ma_uint32 frameCount)
  {
      // What's going on here is that we're continuously converting data from the AudioBuffer's internal format to the mixing format, which
      // should be defined by the output format of the data converter. We do this until frameCount frames have been output. The important
      // detail to remember here is that we never, ever attempt to read more input data than is required for the specified number of output
      // frames. This can be achieved with ma_data_converter_get_required_input_frame_count().
      ma_uint8 inputBuffer[4096];
      ma_uint32 inputBufferFrameCap = sizeof(inputBuffer)/ma_get_bytes_per_frame(audioBuffer->converter.config.formatIn, audioBuffer->converter.config.channelsIn);
  
      ma_uint32 totalOutputFramesProcessed = 0;
      while (totalOutputFramesProcessed < frameCount)
      {
          ma_uint64 outputFramesToProcessThisIteration = frameCount - totalOutputFramesProcessed;
  
          ma_uint64 inputFramesToProcessThisIteration = ma_data_converter_get_required_input_frame_count(&audioBuffer->converter, outputFramesToProcessThisIteration);
          if (inputFramesToProcessThisIteration > inputBufferFrameCap)
          {
              inputFramesToProcessThisIteration = inputBufferFrameCap;
          }
  
          float *runningFramesOut = framesOut + (totalOutputFramesProcessed * audioBuffer->converter.config.channelsOut);
  
          /* At this point we can convert the data to our mixing format. */
          ma_uint64 inputFramesProcessedThisIteration = ReadAudioBufferFramesInInternalFormat(audioBuffer, inputBuffer, (ma_uint32)inputFramesToProcessThisIteration);    /* Safe cast. */
          ma_uint64 outputFramesProcessedThisIteration = outputFramesToProcessThisIteration;
          ma_data_converter_process_pcm_frames(&audioBuffer->converter, inputBuffer, &inputFramesProcessedThisIteration, runningFramesOut, &outputFramesProcessedThisIteration);
  
          totalOutputFramesProcessed += (ma_uint32)outputFramesProcessedThisIteration; /* Safe cast. */
  
          if (inputFramesProcessedThisIteration < inputFramesToProcessThisIteration)
          {
              break;  /* Ran out of input data. */
          }
  
          /* This should never be hit, but will add it here for safety. Ensures we get out of the loop when no input nor output frames are processed. */
          if (inputFramesProcessedThisIteration == 0 && outputFramesProcessedThisIteration == 0)
          {
              break;
          }
      }
  
      return totalOutputFramesProcessed;
  }
  
  
  // Sending audio data to device callback function
  // NOTE: All the mixing takes place here
  static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount)
  {
      (void)pDevice;
  
      // Mixing is basically just an accumulation, we need to initialize the output buffer to 0
      memset(pFramesOut, 0, frameCount*pDevice->playback.channels*ma_get_bytes_per_sample(pDevice->playback.format));
  
      // Using a mutex here for thread-safety which makes things not real-time
      // This is unlikely to be necessary for this project, but may want to consider how you might want to avoid this
      ma_mutex_lock(&AUDIO.System.lock);
      {
          for (AudioBuffer *audioBuffer = AUDIO.Buffer.first; audioBuffer != NULL; audioBuffer = audioBuffer->next)
          {
              // Ignore stopped or paused sounds
              if (!audioBuffer->playing || audioBuffer->paused) continue;
  
              ma_uint32 framesRead = 0;
  
              while (1)
              {
                  if (framesRead >= frameCount) break;
  
                  // Just read as much data as we can from the stream
                  ma_uint32 framesToRead = (frameCount - framesRead);
  
                  while (framesToRead > 0)
                  {
                      float tempBuffer[1024]; // 512 frames for stereo
  
                      ma_uint32 framesToReadRightNow = framesToRead;
                      if (framesToReadRightNow > sizeof(tempBuffer)/sizeof(tempBuffer[0])/AUDIO_DEVICE_CHANNELS)
                      {
                          framesToReadRightNow = sizeof(tempBuffer)/sizeof(tempBuffer[0])/AUDIO_DEVICE_CHANNELS;
                      }
  
                      ma_uint32 framesJustRead = ReadAudioBufferFramesInMixingFormat(audioBuffer, tempBuffer, framesToReadRightNow);
                      if (framesJustRead > 0)
                      {
                          float *framesOut = (float *)pFramesOut + (framesRead*AUDIO.System.device.playback.channels);
                          float *framesIn  = tempBuffer;
  
                          MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume);
  
                          framesToRead -= framesJustRead;
                          framesRead += framesJustRead;
                      }
  
                      if (!audioBuffer->playing)
                      {
                          framesRead = frameCount;
                          break;
                      }
  
                      // If we weren't able to read all the frames we requested, break
                      if (framesJustRead < framesToReadRightNow)
                      {
                          if (!audioBuffer->looping)
                          {
                              StopAudioBuffer(audioBuffer);
                              break;
                          }
                          else
                          {
                              // Should never get here, but just for safety,
                              // move the cursor position back to the start and continue the loop
                              audioBuffer->frameCursorPos = 0;
                              continue;
                          }
                      }
                  }
  
                  // If for some reason we weren't able to read every frame we'll need to break from the loop
                  // Not doing this could theoretically put us into an infinite loop
                  if (framesToRead > 0) break;
              }
          }
      }
  
      ma_mutex_unlock(&AUDIO.System.lock);
  }
  
  // This is the main mixing function. Mixing is pretty simple in this project - it's just an accumulation.
  // NOTE: framesOut is both an input and an output. It will be initially filled with zeros outside of this function.
  static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume)
  {
      for (ma_uint32 iFrame = 0; iFrame < frameCount; ++iFrame)
      {
          for (ma_uint32 iChannel = 0; iChannel < AUDIO.System.device.playback.channels; ++iChannel)
          {
              float *frameOut = framesOut + (iFrame*AUDIO.System.device.playback.channels);
              const float *frameIn  = framesIn  + (iFrame*AUDIO.System.device.playback.channels);
  
              frameOut[iChannel] += (frameIn[iChannel]*localVolume);
          }
      }
  }
  
  // Initialise the multichannel buffer pool
  static void InitAudioBufferPool(void)
  {
      // Dummy buffers
      for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
      {
          // WARNING: An empty audioBuffer is created (data = 0)
          AUDIO.MultiChannel.pool[i] = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO_DEVICE_SAMPLE_RATE, 0, AUDIO_BUFFER_USAGE_STATIC);
      }
      
      // TODO: Verification required for log
      TRACELOG(LOG_INFO, "AUDIO: Multichannel pool size: %i", MAX_AUDIO_BUFFER_POOL_CHANNELS);
  }
  
  // Close the audio buffers pool
  static void CloseAudioBufferPool(void)
  {
      for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) RL_FREE(AUDIO.MultiChannel.pool[i]);
  }
  
  #if defined(SUPPORT_FILEFORMAT_WAV)
  // Load WAV file into Wave structure
  static Wave LoadWAV(const char *fileName)
  {
      Wave wave = { 0 };
  
      // Decode an entire WAV file in one go
      unsigned long long int totalPCMFrameCount = 0;
      wave.data = drwav_open_file_and_read_pcm_frames_s16(fileName, &wave.channels, &wave.sampleRate, &totalPCMFrameCount, NULL);
  
      if (wave.data == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to load WAV data", fileName);
      else
      {
          wave.sampleCount = (unsigned int)totalPCMFrameCount*wave.channels;
          wave.sampleSize = 16;
  
          TRACELOG(LOG_INFO, "WAVE: [%s] WAV file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
      }
      
  /*
      // Loading WAV from memory to avoid FILE accesses
      unsigned int fileSize = 0;
      unsigned char *fileData = LoadFileData(fileName, &fileSize);
      
      drwav wav = { 0 };
      
      bool success = drwav_init_memory(&wav, fileData, fileSize, NULL);
      
      if (success)
      {
          wave.sampleCount = wav.totalPCMFrameCount*wav.channels;
          wave.sampleRate = wav.sampleRate;
          wave.sampleSize = 16;   // NOTE: We are forcing conversion to 16bit
          wave.channels = wav.channels;
          wave.data = (short *)RL_MALLOC(wave.sampleCount*sizeof(short));
          drwav_read_pcm_frames_s16(&wav, wav.totalPCMFrameCount, wave.data);
      }
      else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to load WAV data", fileName);
      
      drwav_uninit(&wav);
      RL_FREE(fileData);
  */
      return wave;
  }
  
  // Save wave data as WAV file
  static int SaveWAV(Wave wave, const char *fileName)
  {
      drwav wav = { 0 };
      drwav_data_format format = { 0 };
      format.container = drwav_container_riff;     // <-- drwav_container_riff = normal WAV files, drwav_container_w64 = Sony Wave64.
      format.format = DR_WAVE_FORMAT_PCM;          // <-- Any of the DR_WAVE_FORMAT_* codes.
      format.channels = wave.channels;
      format.sampleRate = wave.sampleRate;
      format.bitsPerSample = wave.sampleSize;
      
      drwav_init_file_write(&wav, fileName, &format, NULL);
      //drwav_init_memory_write(&wav, &fileData, &fileDataSize, &format, NULL);       // Memory version
      drwav_write_pcm_frames(&wav, wave.sampleCount/wave.channels, wave.data);
      
      drwav_uninit(&wav);
      
      // SaveFileData(fileName, fileData, fileDataSize);
      //drwav_free(fileData, NULL);
      
      return true;
  }
  #endif
  
  #if defined(SUPPORT_FILEFORMAT_OGG)
  // Load OGG file into Wave structure
  // NOTE: Using stb_vorbis library
  static Wave LoadOGG(const char *fileName)
  {
      Wave wave = { 0 };
  
      stb_vorbis *oggFile = stb_vorbis_open_filename(fileName, NULL, NULL);
  
      if (oggFile == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open OGG file", fileName);
      else
      {
          stb_vorbis_info info = stb_vorbis_get_info(oggFile);
  
          wave.sampleRate = info.sample_rate;
          wave.sampleSize = 16;                   // 16 bit per sample (short)
          wave.channels = info.channels;
          wave.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples(oggFile)*info.channels;  // Independent by channel
  
          float totalSeconds = stb_vorbis_stream_length_in_seconds(oggFile);
          if (totalSeconds > 10) TRACELOG(LOG_WARNING, "WAVE: [%s] Ogg audio length larger than 10 seconds (%f), that's a big file in memory, consider music streaming", fileName, totalSeconds);
  
          wave.data = (short *)RL_MALLOC(wave.sampleCount*wave.channels*sizeof(short));
  
          // NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
          stb_vorbis_get_samples_short_interleaved(oggFile, info.channels, (short *)wave.data, wave.sampleCount*wave.channels);
          TRACELOG(LOG_INFO, "WAVE: [%s] OGG file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
  
          stb_vorbis_close(oggFile);
      }
  
      return wave;
  }
  #endif
  
  #if defined(SUPPORT_FILEFORMAT_FLAC)
  // Load FLAC file into Wave structure
  // NOTE: Using dr_flac library
  static Wave LoadFLAC(const char *fileName)
  {
      Wave wave = { 0 };
  
      // Decode an entire FLAC file in one go
      unsigned long long int totalSampleCount = 0;
      wave.data = drflac_open_file_and_read_pcm_frames_s16(fileName, &wave.channels, &wave.sampleRate, &totalSampleCount);
  
      if (wave.data == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to load FLAC data", fileName);
      else
      {
          wave.sampleCount = (unsigned int)totalSampleCount;
          wave.sampleSize = 16;
  
          TRACELOG(LOG_INFO, "WAVE: [%s] FLAC file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
      }
   
      return wave;
  }
  #endif
  
  #if defined(SUPPORT_FILEFORMAT_MP3)
  // Load MP3 file into Wave structure
  // NOTE: Using dr_mp3 library
  static Wave LoadMP3(const char *fileName)
  {
      Wave wave = { 0 };
  
      // Decode an entire MP3 file in one go
      unsigned long long int totalFrameCount = 0;
      drmp3_config config = { 0 };
      wave.data = drmp3_open_file_and_read_f32(fileName, &config, &totalFrameCount);
  
      if (wave.data == NULL) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to load MP3 data", fileName);
      else
      {
          wave.channels = config.outputChannels;
          wave.sampleRate = config.outputSampleRate;
          wave.sampleCount = (int)totalFrameCount*wave.channels;
          wave.sampleSize = 32;
  
          // NOTE: Only support up to 2 channels (mono, stereo)
          if (wave.channels > 2) TRACELOG(LOG_WARNING, "WAVE: [%s] MP3 channels number (%i) not supported", fileName, wave.channels);
  
          TRACELOG(LOG_INFO, "WAVE: [%s] MP3 file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
      }
      
      return wave;
  }
  #endif
  
  // Some required functions for audio standalone module version
  #if defined(RAUDIO_STANDALONE)
  // Check file extension
  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;
  }
  
  // Save text data to file (write), string must be '\0' terminated
  static void SaveFileText(const char *fileName, char *text)
  {
      if (fileName != NULL)
      {
          FILE *file = fopen(fileName, "wt");
  
          if (file != NULL)
          {
              int count = fprintf(file, "%s", text);
  
              if (count == 0) TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to write text file", fileName);
              else TRACELOG(LOG_INFO, "FILEIO: [%s] Text file saved successfully", fileName);
  
              fclose(file);
          }
          else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to open text file", fileName);
      }
      else TRACELOG(LOG_WARNING, "FILEIO: File name provided is not valid");
  }
  #endif
  
  #undef AudioBuffer
  