Setup the project

1 files changed, 2028 insertions, 0 deletions

diff --git a/libs/raylib/src/raudio.c b/libs/raylib/src/raudio.c
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+/**********************************************************************************************
+*
+*   raudio - A simple and easy-to-use audio library based on miniaudio
+*
+*   FEATURES:
+*       - Manage audio device (init/close)
+*       - Load and unload audio files
+*       - Format wave data (sample rate, size, channels)
+*       - Play/Stop/Pause/Resume loaded audio
+*       - Manage mixing channels
+*       - Manage raw audio context
+*
+*   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
+
+#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 <string.h>             // Required for: strcmp(), strncmp()
+#include <stdio.h>              // Required for: FILE, fopen(), fclose(), fread()
+
+#if defined(SUPPORT_FILEFORMAT_OGG)
+    #define STB_VORBIS_IMPLEMENTATION
+    #include "external/stb_vorbis.h"    // OGG loading functions
+#endif
+
+#if defined(SUPPORT_FILEFORMAT_XM)
+    #define JAR_XM_IMPLEMENTATION
+    #include "external/jar_xm.h"        // XM loading functions
+#endif
+
+#if defined(SUPPORT_FILEFORMAT_MOD)
+    #define JAR_MOD_IMPLEMENTATION
+    #include "external/jar_mod.h"       // MOD loading functions
+#endif
+
+#if defined(SUPPORT_FILEFORMAT_FLAC)
+    #define DR_FLAC_IMPLEMENTATION
+    #define DR_FLAC_NO_WIN32_IO
+    #include "external/dr_flac.h"       // FLAC loading functions
+#endif
+
+#if defined(SUPPORT_FILEFORMAT_MP3)
+    #define DR_MP3_IMPLEMENTATION
+    #include "external/dr_mp3.h"        // MP3 loading functions
+#endif
+
+#if defined(_MSC_VER)
+    #undef bool
+#endif
+
+//----------------------------------------------------------------------------------
+// Defines and Macros
+//----------------------------------------------------------------------------------
+// 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
+#define AUDIO_BUFFER_SIZE        4096       // PCM data samples (i.e. 16bit, Mono: 8Kb)
+
+//----------------------------------------------------------------------------------
+// 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
+
+//----------------------------------------------------------------------------------
+// Global Variables Definition
+//----------------------------------------------------------------------------------
+// ...
+
+//----------------------------------------------------------------------------------
+// Module specific Functions Declaration
+//----------------------------------------------------------------------------------
+#if defined(SUPPORT_FILEFORMAT_WAV)
+static Wave LoadWAV(const char *fileName);              // Load WAV file
+static int SaveWAV(Wave wave, const char *fileName);    // Save wave data as WAV file
+#endif
+#if defined(SUPPORT_FILEFORMAT_OGG)
+static Wave LoadOGG(const char *fileName);              // Load OGG file
+#endif
+#if defined(SUPPORT_FILEFORMAT_FLAC)
+static Wave LoadFLAC(const char *fileName);             // Load FLAC file
+#endif
+#if defined(SUPPORT_FILEFORMAT_MP3)
+static Wave LoadMP3(const char *fileName);              // Load MP3 file
+#endif
+
+#if defined(RAUDIO_STANDALONE)
+bool IsFileExtension(const char *fileName, const char *ext);    // Check file extension
+void TraceLog(int msgType, const char *text, ...);              // Show trace log messages (LOG_INFO, LOG_WARNING, LOG_ERROR, LOG_DEBUG)
+#endif
+
+//----------------------------------------------------------------------------------
+// AudioBuffer Functionality
+//----------------------------------------------------------------------------------
+#define DEVICE_FORMAT       ma_format_f32
+#define DEVICE_CHANNELS     2
+#define DEVICE_SAMPLE_RATE  44100
+
+#define MAX_AUDIO_BUFFER_POOL_CHANNELS 16
+
+typedef enum { AUDIO_BUFFER_USAGE_STATIC = 0, AUDIO_BUFFER_USAGE_STREAM } AudioBufferUsage;
+
+// Audio buffer structure
+// NOTE: Slightly different logic is used when feeding data to the
+// playback device depending on whether or not data is streamed
+struct rAudioBuffer {
+    ma_pcm_converter dsp;   // PCM 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 frameCursorPos;        // Frame cursor position
+    unsigned int bufferSizeInFrames;    // Total buffer size in frames
+    unsigned int totalFramesProcessed;  // Total frames processed in this buffer (required for play timming)
+
+    unsigned char *buffer;              // 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 buffers are tracked in a linked list
+static AudioBuffer *firstAudioBuffer = NULL;    // Pointer to first AudioBuffer in the list
+static AudioBuffer *lastAudioBuffer = NULL;     // Pointer to last AudioBuffer in the list
+
+// miniaudio global variables
+static ma_context context;                      // miniaudio context data
+static ma_device device;                        // miniaudio device
+static ma_mutex audioLock;                      // miniaudio mutex lock
+static bool isAudioInitialized = false;         // Check if audio device is initialized
+static float masterVolume = 1.0f;               // Master volume (multiplied on output mixing)
+
+// Multi channel playback global variables
+static AudioBuffer *audioBufferPool[MAX_AUDIO_BUFFER_POOL_CHANNELS] = { 0 };         // Multichannel AudioBuffer pointers pool
+static unsigned int audioBufferPoolCounter = 0;                                      // AudioBuffer pointers pool counter
+static unsigned int audioBufferPoolChannels[MAX_AUDIO_BUFFER_POOL_CHANNELS] = { 0 }; // AudioBuffer pool channels
+
+// miniaudio 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 ma_uint32 OnAudioBufferDSPRead(ma_pcm_converter *pDSP, void *pFramesOut, ma_uint32 frameCount, void *pUserData);
+static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume);
+
+// AudioBuffer management functions declaration
+// NOTE: Those functions are not exposed by raylib... for the moment
+AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 bufferSizeInFrames, int usage);
+void CloseAudioBuffer(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);
+
+
+//----------------------------------------------------------------------------------
+// miniaudio functions definitions
+//----------------------------------------------------------------------------------
+
+// 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, message);   // All log messages from miniaudio are errors
+}
+
+// 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(&audioLock);
+    {
+        for (AudioBuffer *audioBuffer = firstAudioBuffer; 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)
+                {
+                    TraceLog(LOG_DEBUG, "Mixed too many frames from audio buffer");
+                    break;
+                }
+
+                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])/DEVICE_CHANNELS)
+                    {
+                        framesToReadRightNow = sizeof(tempBuffer)/sizeof(tempBuffer[0])/DEVICE_CHANNELS;
+                    }
+
+                    ma_uint32 framesJustRead = (ma_uint32)ma_pcm_converter_read(&audioBuffer->dsp, tempBuffer, framesToReadRightNow);
+                    if (framesJustRead > 0)
+                    {
+                        float *framesOut = (float *)pFramesOut + (framesRead*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(&audioLock);
+}
+
+// DSP read from audio buffer callback function
+static ma_uint32 OnAudioBufferDSPRead(ma_pcm_converter *pDSP, void *pFramesOut, ma_uint32 frameCount, void *pUserData)
+{
+    AudioBuffer *audioBuffer = (AudioBuffer *)pUserData;
+
+    ma_uint32 subBufferSizeInFrames = (audioBuffer->bufferSizeInFrames > 1)? audioBuffer->bufferSizeInFrames/2 : audioBuffer->bufferSizeInFrames;
+    ma_uint32 currentSubBufferIndex = audioBuffer->frameCursorPos/subBufferSizeInFrames;
+
+    if (currentSubBufferIndex > 1)
+    {
+        TraceLog(LOG_DEBUG, "Frame cursor position moved too far forward in audio stream");
+        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_sample(audioBuffer->dsp.formatConverterIn.config.formatIn)*audioBuffer->dsp.formatConverterIn.config.channels;
+
+    // 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->bufferSizeInFrames - 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 *)pFramesOut + (framesRead*frameSizeInBytes), audioBuffer->buffer + (audioBuffer->frameCursorPos*frameSizeInBytes), framesToRead*frameSizeInBytes);
+        audioBuffer->frameCursorPos = (audioBuffer->frameCursorPos + framesToRead)%audioBuffer->bufferSizeInFrames;
+        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 *)pFramesOut + (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;
+}
+
+// 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 < device.playback.channels; ++iChannel)
+        {
+            float *frameOut = framesOut + (iFrame*device.playback.channels);
+            const float *frameIn  = framesIn  + (iFrame*device.playback.channels);
+
+            frameOut[iChannel] += (frameIn[iChannel]*masterVolume*localVolume);
+        }
+    }
+}
+
+// Initialise the multichannel buffer pool
+static void InitAudioBufferPool()
+{
+    // Dummy buffers
+    for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
+    {
+        audioBufferPool[i] = InitAudioBuffer(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, 0, AUDIO_BUFFER_USAGE_STATIC);
+    }
+}
+
+// Close the audio buffers pool
+static void CloseAudioBufferPool()
+{
+    for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) 
+    {
+        RL_FREE(audioBufferPool[i]->buffer);
+        RL_FREE(audioBufferPool[i]);
+    }
+}
+
+//----------------------------------------------------------------------------------
+// Module Functions Definition - Audio Device initialization and Closing
+//----------------------------------------------------------------------------------
+// Initialize audio device
+void InitAudioDevice(void)
+{
+    // Init audio context
+    ma_context_config contextConfig = ma_context_config_init();
+    contextConfig.logCallback = OnLog;
+
+    ma_result result = ma_context_init(NULL, 0, &contextConfig, &context);
+    if (result != MA_SUCCESS)
+    {
+        TraceLog(LOG_ERROR, "Failed to initialize audio 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 device.
+    config.playback.format    = DEVICE_FORMAT;
+    config.playback.channels  = DEVICE_CHANNELS;
+    config.capture.pDeviceID  = NULL;  // NULL for the default capture device.
+    config.capture.format     = ma_format_s16;
+    config.capture.channels   = 1;
+    config.sampleRate         = DEVICE_SAMPLE_RATE;
+    config.dataCallback       = OnSendAudioDataToDevice;
+    config.pUserData          = NULL;
+
+    result = ma_device_init(&context, &config, &device);
+    if (result != MA_SUCCESS)
+    {
+        TraceLog(LOG_ERROR, "Failed to initialize audio playback device");
+        ma_context_uninit(&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(&device);
+    if (result != MA_SUCCESS)
+    {
+        TraceLog(LOG_ERROR, "Failed to start audio playback device");
+        ma_device_uninit(&device);
+        ma_context_uninit(&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(&context, &audioLock) != MA_SUCCESS)
+    {
+        TraceLog(LOG_ERROR, "Failed to create mutex for audio mixing");
+        ma_device_uninit(&device);
+        ma_context_uninit(&context);
+        return;
+    }
+
+    TraceLog(LOG_INFO, "Audio device initialized successfully");
+    TraceLog(LOG_INFO, "Audio backend: miniaudio / %s", ma_get_backend_name(context.backend));
+    TraceLog(LOG_INFO, "Audio format: %s -> %s", ma_get_format_name(device.playback.format), ma_get_format_name(device.playback.internalFormat));
+    TraceLog(LOG_INFO, "Audio channels: %d -> %d", device.playback.channels, device.playback.internalChannels);
+    TraceLog(LOG_INFO, "Audio sample rate: %d -> %d", device.sampleRate, device.playback.internalSampleRate);
+    TraceLog(LOG_INFO, "Audio buffer size: %d", device.playback.internalBufferSizeInFrames);
+
+    InitAudioBufferPool();
+    TraceLog(LOG_INFO, "Audio multichannel pool size: %i", MAX_AUDIO_BUFFER_POOL_CHANNELS);
+
+    isAudioInitialized = true;
+}
+
+// Close the audio device for all contexts
+void CloseAudioDevice(void)
+{
+    if (isAudioInitialized)
+    {
+        ma_mutex_uninit(&audioLock);
+        ma_device_uninit(&device);
+        ma_context_uninit(&context);
+
+        CloseAudioBufferPool();
+
+        TraceLog(LOG_INFO, "Audio device closed successfully");
+    }
+    else TraceLog(LOG_WARNING, "Could not close audio device because it is not currently initialized");
+}
+
+// Check if device has been initialized successfully
+bool IsAudioDeviceReady(void)
+{
+    return isAudioInitialized;
+}
+
+// Set master volume (listener)
+void SetMasterVolume(float volume)
+{
+    if (volume < 0.0f) volume = 0.0f;
+    else if (volume > 1.0f) volume = 1.0f;
+
+    masterVolume = volume;
+}
+
+//----------------------------------------------------------------------------------
+// Module Functions Definition - Audio Buffer management
+//----------------------------------------------------------------------------------
+
+// Initialize a new audio buffer (filled with silence)
+AudioBuffer *InitAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 bufferSizeInFrames, int usage)
+{
+    AudioBuffer *audioBuffer = (AudioBuffer *)RL_CALLOC(1, sizeof(AudioBuffer));
+
+    if (audioBuffer == NULL)
+    {
+        TraceLog(LOG_ERROR, "InitAudioBuffer() : Failed to allocate memory for audio buffer");
+        return NULL;
+    }
+    
+    audioBuffer->buffer = RL_CALLOC(bufferSizeInFrames*channels*ma_get_bytes_per_sample(format), 1);
+
+    // Audio data runs through a format converter
+    ma_pcm_converter_config dspConfig;
+    memset(&dspConfig, 0, sizeof(dspConfig));
+    dspConfig.formatIn = format;
+    dspConfig.formatOut = DEVICE_FORMAT;
+    dspConfig.channelsIn = channels;
+    dspConfig.channelsOut = DEVICE_CHANNELS;
+    dspConfig.sampleRateIn = sampleRate;
+    dspConfig.sampleRateOut = DEVICE_SAMPLE_RATE;
+    dspConfig.onRead = OnAudioBufferDSPRead;        // Callback on data reading
+    dspConfig.pUserData = audioBuffer;              // Audio data pointer
+    dspConfig.allowDynamicSampleRate = true;        // Required for pitch shifting
+
+    ma_result result = ma_pcm_converter_init(&dspConfig, &audioBuffer->dsp);
+
+    if (result != MA_SUCCESS)
+    {
+        TraceLog(LOG_ERROR, "InitAudioBuffer() : 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->bufferSizeInFrames = bufferSizeInFrames;
+
+    // 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 CloseAudioBuffer(AudioBuffer *buffer)
+{
+    if (buffer != NULL)
+    {
+        UntrackAudioBuffer(buffer);
+        RL_FREE(buffer->buffer);
+        RL_FREE(buffer);
+    }
+    else TraceLog(LOG_ERROR, "CloseAudioBuffer() : No audio buffer");
+}
+
+// Check if an audio buffer is playing
+bool IsAudioBufferPlaying(AudioBuffer *buffer)
+{
+    bool result = false;
+
+    if (buffer != NULL) result = (buffer->playing && !buffer->paused);
+    else TraceLog(LOG_WARNING, "IsAudioBufferPlaying() : No audio buffer");
+
+    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;
+    }
+    else TraceLog(LOG_ERROR, "PlayAudioBuffer() : No audio buffer");
+}
+
+// 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;
+        }
+    }
+    else TraceLog(LOG_ERROR, "StopAudioBuffer() : No audio buffer");
+}
+
+// Pause an audio buffer
+void PauseAudioBuffer(AudioBuffer *buffer)
+{
+    if (buffer != NULL) buffer->paused = true;
+    else TraceLog(LOG_ERROR, "PauseAudioBuffer() : No audio buffer");
+}
+
+// Resume an audio buffer
+void ResumeAudioBuffer(AudioBuffer *buffer)
+{
+    if (buffer != NULL) buffer->paused = false;
+    else TraceLog(LOG_ERROR, "ResumeAudioBuffer() : No audio buffer");
+}
+
+// Set volume for an audio buffer
+void SetAudioBufferVolume(AudioBuffer *buffer, float volume)
+{
+    if (buffer != NULL) buffer->volume = volume;
+    else TraceLog(LOG_WARNING, "SetAudioBufferVolume() : No audio buffer");
+}
+
+// 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->dsp.src.config.sampleRateOut/pitchMul);
+        buffer->pitch *= (float)buffer->dsp.src.config.sampleRateOut/newOutputSampleRate;
+
+        ma_pcm_converter_set_output_sample_rate(&buffer->dsp, newOutputSampleRate);
+    }
+    else TraceLog(LOG_WARNING, "SetAudioBufferPitch() : No audio buffer");
+}
+
+// Track audio buffer to linked list next position
+void TrackAudioBuffer(AudioBuffer *buffer)
+{
+    ma_mutex_lock(&audioLock);
+    {
+        if (firstAudioBuffer == NULL) firstAudioBuffer = buffer;
+        else
+        {
+            lastAudioBuffer->next = buffer;
+            buffer->prev = lastAudioBuffer;
+        }
+
+        lastAudioBuffer = buffer;
+    }
+    ma_mutex_unlock(&audioLock);
+}
+
+// Untrack audio buffer from linked list
+void UntrackAudioBuffer(AudioBuffer *buffer)
+{
+    ma_mutex_lock(&audioLock);
+    {
+        if (buffer->prev == NULL) firstAudioBuffer = buffer->next;
+        else buffer->prev->next = buffer->next;
+
+        if (buffer->next == NULL) lastAudioBuffer = buffer->prev;
+        else buffer->next->prev = buffer->prev;
+
+        buffer->prev = NULL;
+        buffer->next = NULL;
+    }
+    ma_mutex_unlock(&audioLock);
+}
+
+//----------------------------------------------------------------------------------
+// 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, "[%s] Audio fileformat not supported, it can't be loaded", 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 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, DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, NULL, formatIn, wave.channels, wave.sampleRate, frameCountIn);
+        if (frameCount == 0) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Failed to get frame count for format conversion");
+
+        AudioBuffer *audioBuffer = InitAudioBuffer(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, frameCount, AUDIO_BUFFER_USAGE_STATIC);
+        if (audioBuffer == NULL) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Failed to create audio buffer");
+
+        frameCount = (ma_uint32)ma_convert_frames(audioBuffer->buffer, audioBuffer->dsp.formatConverterIn.config.formatIn, audioBuffer->dsp.formatConverterIn.config.channels, audioBuffer->dsp.src.config.sampleRateIn, wave.data, formatIn, wave.channels, wave.sampleRate, frameCountIn);
+        if (frameCount == 0) TraceLog(LOG_WARNING, "LoadSoundFromWave() : Format conversion failed");
+
+        sound.sampleCount = frameCount*DEVICE_CHANNELS;
+        sound.stream.sampleRate = DEVICE_SAMPLE_RATE;
+        sound.stream.sampleSize = 32;
+        sound.stream.channels = 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, "Unloaded wave data from RAM");
+}
+
+// Unload sound
+void UnloadSound(Sound sound)
+{
+    CloseAudioBuffer(sound.stream.buffer);
+
+    TraceLog(LOG_INFO, "Unloaded sound data from RAM");
+}
+
+// Update sound buffer with new data
+void UpdateSound(Sound sound, const void *data, int samplesCount)
+{
+    AudioBuffer *audioBuffer = sound.stream.buffer;
+
+    if (audioBuffer != NULL)
+    {
+        StopAudioBuffer(audioBuffer);
+
+        // TODO: May want to lock/unlock this since this data buffer is read at mixing time
+        memcpy(audioBuffer->buffer, data, samplesCount*audioBuffer->dsp.formatConverterIn.config.channels*ma_get_bytes_per_sample(audioBuffer->dsp.formatConverterIn.config.formatIn));
+    }
+    else TraceLog(LOG_ERROR, "UpdateSound() : Invalid sound - no audio buffer");
+}
+
+// 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
+        FILE *rawFile = fopen(fileName, "wb");
+        success = fwrite(wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8, 1, rawFile);
+        fclose(rawFile);
+    }
+
+    if (success) TraceLog(LOG_INFO, "Wave exported successfully: %s", fileName);
+    else TraceLog(LOG_WARNING, "Wave could not be exported.");
+}
+
+// Export wave sample data to code (.h)
+void ExportWaveAsCode(Wave wave, const char *fileName)
+{
+    #define BYTES_TEXT_PER_LINE     20
+
+    char varFileName[256] = { 0 };
+    int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8;
+
+    FILE *txtFile = fopen(fileName, "wt");
+
+    if (txtFile != NULL)
+    {
+        fprintf(txtFile, "\n//////////////////////////////////////////////////////////////////////////////////\n");
+        fprintf(txtFile, "//                                                                              //\n");
+        fprintf(txtFile, "// WaveAsCode exporter v1.0 - Wave data exported as an array of bytes           //\n");
+        fprintf(txtFile, "//                                                                              //\n");
+        fprintf(txtFile, "// more info and bugs-report:  github.com/raysan5/raylib                        //\n");
+        fprintf(txtFile, "// feedback and support:       ray[at]raylib.com                                //\n");
+        fprintf(txtFile, "//                                                                              //\n");
+        fprintf(txtFile, "// Copyright (c) 2018 Ramon Santamaria (@raysan5)                               //\n");
+        fprintf(txtFile, "//                                                                              //\n");
+        fprintf(txtFile, "//////////////////////////////////////////////////////////////////////////////////\n\n");
+
+#if !defined(RAUDIO_STANDALONE)
+        // Get file name from path and convert variable name to uppercase
+        strcpy(varFileName, GetFileNameWithoutExt(fileName));
+        for (int i = 0; varFileName[i] != '\0'; i++) if (varFileName[i] >= 'a' && varFileName[i] <= 'z') { varFileName[i] = varFileName[i] - 32; }
+#else
+        strcpy(varFileName, fileName);
+#endif
+
+        fprintf(txtFile, "// Wave data information\n");
+        fprintf(txtFile, "#define %s_SAMPLE_COUNT     %i\n", varFileName, wave.sampleCount);
+        fprintf(txtFile, "#define %s_SAMPLE_RATE      %i\n", varFileName, wave.sampleRate);
+        fprintf(txtFile, "#define %s_SAMPLE_SIZE      %i\n", varFileName, wave.sampleSize);
+        fprintf(txtFile, "#define %s_CHANNELS         %i\n\n", varFileName, wave.channels);
+
+        // Write byte data as hexadecimal text
+        fprintf(txtFile, "static unsigned char %s_DATA[%i] = { ", varFileName, dataSize);
+        for (int i = 0; i < dataSize - 1; i++) fprintf(txtFile, ((i%BYTES_TEXT_PER_LINE == 0)? "0x%x,\n" : "0x%x, "), ((unsigned char *)wave.data)[i]);
+        fprintf(txtFile, "0x%x };\n", ((unsigned char *)wave.data)[dataSize - 1]);
+
+        fclose(txtFile);
+    }
+}
+
+// Play a sound
+void PlaySound(Sound sound)
+{
+    PlayAudioBuffer(sound.stream.buffer);
+}
+
+// Play a sound in the multichannel buffer pool
+void PlaySoundMulti(Sound sound)
+{
+    int index = -1;
+    unsigned int oldAge = 0;
+    int oldIndex = -1;
+
+    // find the first non playing pool entry
+    for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
+    {
+        if (audioBufferPoolChannels[i] > oldAge)
+        {
+            oldAge = audioBufferPoolChannels[i];
+            oldIndex = i;
+        }
+
+        if (!IsAudioBufferPlaying(audioBufferPool[i]))
+        {
+            index = i;
+            break;
+        }
+    }
+
+    // If no none playing pool members can be index choose the oldest
+    if (index == -1)
+    {
+        TraceLog(LOG_WARNING,"pool age %i ended a sound early no room in buffer pool", audioBufferPoolCounter);
+
+        if (oldIndex == -1)
+        {
+            // Shouldn't be able to get here... but just in case something odd happens!
+            TraceLog(LOG_ERROR,"sound buffer pool couldn't determine oldest buffer not playing sound");
+
+            return;
+        }
+
+        index = oldIndex;
+
+        // Just in case...
+        StopAudioBuffer(audioBufferPool[index]);
+    }
+
+    // Experimentally mutex lock doesn't seem to be needed this makes sense
+    // as audioBufferPool[index] isn't playing and the only stuff we're copying
+    // shouldn't be changing...
+
+    audioBufferPoolChannels[index] = audioBufferPoolCounter;
+    audioBufferPoolCounter++;
+
+    audioBufferPool[index]->volume = sound.stream.buffer->volume;
+    audioBufferPool[index]->pitch = sound.stream.buffer->pitch;
+    audioBufferPool[index]->looping = sound.stream.buffer->looping;
+    audioBufferPool[index]->usage = sound.stream.buffer->usage;
+    audioBufferPool[index]->isSubBufferProcessed[0] = false;
+    audioBufferPool[index]->isSubBufferProcessed[1] = false;
+    audioBufferPool[index]->bufferSizeInFrames = sound.stream.buffer->bufferSizeInFrames;
+    audioBufferPool[index]->buffer = sound.stream.buffer->buffer;
+
+    PlayAudioBuffer(audioBufferPool[index]);
+}
+
+// Stop any sound played with PlaySoundMulti()
+void StopSoundMulti(void)
+{
+    for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++) StopAudioBuffer(audioBufferPool[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(audioBufferPool[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, formatOut, channels, sampleRate, NULL, formatIn, wave->channels, wave->sampleRate, frameCountIn);
+    if (frameCount == 0)
+    {
+        TraceLog(LOG_ERROR, "WaveFormat() : Failed to get frame count for format conversion.");
+        return;
+    }
+
+    void *data = RL_MALLOC(frameCount*channels*(sampleSize/8));
+
+    frameCount = (ma_uint32)ma_convert_frames(data, formatOut, channels, sampleRate, wave->data, formatIn, wave->channels, wave->sampleRate, frameCountIn);
+    if (frameCount == 0)
+    {
+        TraceLog(LOG_ERROR, "WaveFormat() : Format conversion failed.");
+        return;
+    }
+
+    wave->sampleCount = frameCount;
+    wave->sampleSize = sampleSize;
+    wave->sampleRate = sampleRate;
+    wave->channels = channels;
+    RL_FREE(wave->data);
+    wave->data = data;
+}
+
+// Copy a wave to a new wave
+Wave WaveCopy(Wave wave)
+{
+    Wave newWave = { 0 };
+
+    newWave.data = RL_MALLOC(wave.sampleCount*wave.sampleSize/8*wave.channels);
+
+    if (newWave.data != NULL)
+    {
+        // NOTE: Size must be provided in bytes
+        memcpy(newWave.data, wave.data, wave.sampleCount*wave.channels*wave.sampleSize/8);
+
+        newWave.sampleCount = wave.sampleCount;
+        newWave.sampleRate = wave.sampleRate;
+        newWave.sampleSize = wave.sampleSize;
+        newWave.channels = wave.channels;
+    }
+
+    return newWave;
+}
+
+// Crop a wave to defined samples range
+// NOTE: Security check in case of out-of-range
+void WaveCrop(Wave *wave, int initSample, int finalSample)
+{
+    if ((initSample >= 0) && (initSample < finalSample) &&
+        (finalSample > 0) && ((unsigned int)finalSample < wave->sampleCount))
+    {
+        int sampleCount = finalSample - initSample;
+
+        void *data = RL_MALLOC(sampleCount*wave->sampleSize/8*wave->channels);
+
+        memcpy(data, (unsigned char *)wave->data + (initSample*wave->channels*wave->sampleSize/8), sampleCount*wave->channels*wave->sampleSize/8);
+
+        RL_FREE(wave->data);
+        wave->data = data;
+    }
+    else TraceLog(LOG_WARNING, "Wave crop range out of bounds");
+}
+
+// Get samples data from wave as a floats array
+// NOTE: Returned sample values are normalized to range [-1..1]
+float *GetWaveData(Wave wave)
+{
+    float *samples = (float *)RL_MALLOC(wave.sampleCount*wave.channels*sizeof(float));
+
+    for (unsigned int i = 0; i < wave.sampleCount; i++)
+    {
+        for (unsigned int j = 0; j < wave.channels; j++)
+        {
+            if (wave.sampleSize == 8) samples[wave.channels*i + j] = (float)(((unsigned char *)wave.data)[wave.channels*i + j] - 127)/256.0f;
+            else if (wave.sampleSize == 16) samples[wave.channels*i + j] = (float)((short *)wave.data)[wave.channels*i + j]/32767.0f;
+            else if (wave.sampleSize == 32) samples[wave.channels*i + j] = ((float *)wave.data)[wave.channels*i + j];
+        }
+    }
+
+    return samples;
+}
+
+//----------------------------------------------------------------------------------
+// Module Functions Definition - Music loading and stream playing (.OGG)
+//----------------------------------------------------------------------------------
+
+// Load music stream from file
+Music LoadMusicStream(const char *fileName)
+{
+    Music music = { 0 };
+    bool musicLoaded = false;
+
+    if (false) { }
+#if defined(SUPPORT_FILEFORMAT_OGG)
+    else if (IsFileExtension(fileName, ".ogg"))
+    {
+        // Open ogg audio stream
+        music.ctxData = stb_vorbis_open_filename(fileName, NULL, NULL);
+
+        if (music.ctxData != NULL)
+        {
+            music.ctxType = MUSIC_AUDIO_OGG;
+            stb_vorbis_info info = stb_vorbis_get_info((stb_vorbis *)music.ctxData);  // Get Ogg file info
+
+            // OGG bit rate defaults to 16 bit, it's enough for compressed format
+            music.stream = InitAudioStream(info.sample_rate, 16, info.channels);
+            music.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData)*info.channels;
+            music.loopCount = 0;   // Infinite loop by default
+            musicLoaded = true;
+        }
+    }
+#endif
+#if defined(SUPPORT_FILEFORMAT_FLAC)
+    else if (IsFileExtension(fileName, ".flac"))
+    {
+        music.ctxData = drflac_open_file(fileName);
+
+        if (music.ctxData != NULL)
+        {
+            music.ctxType = MUSIC_AUDIO_FLAC;
+            drflac *ctxFlac = (drflac *)music.ctxData;
+
+            music.stream = InitAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
+            music.sampleCount = (unsigned int)ctxFlac->totalSampleCount;
+            music.loopCount = 0;   // Infinite loop by default
+            musicLoaded = true;
+        }
+    }
+#endif
+#if defined(SUPPORT_FILEFORMAT_MP3)
+    else if (IsFileExtension(fileName, ".mp3"))
+    {
+        drmp3 *ctxMp3 = RL_MALLOC(sizeof(drmp3));
+        music.ctxData = ctxMp3;
+
+        int result = drmp3_init_file(ctxMp3, fileName, NULL);
+
+        if (result > 0)
+        {
+            music.ctxType = MUSIC_AUDIO_MP3;
+
+            music.stream = InitAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
+            music.sampleCount = drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels;
+            music.loopCount = 0;   // Infinite loop by default
+            musicLoaded = true;
+        }
+    }
+#endif
+#if defined(SUPPORT_FILEFORMAT_XM)
+    else if (IsFileExtension(fileName, ".xm"))
+    {
+        jar_xm_context_t *ctxXm = NULL;
+
+        int result = jar_xm_create_context_from_file(&ctxXm, 48000, fileName);
+
+        if (result == 0)    // XM 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);
+            music.loopCount = 0;   // Infinite loop by default
+            jar_xm_reset(ctxXm);   // make sure we start at the beginning of the song
+            musicLoaded = true;
+
+            music.ctxData = ctxXm;
+        }
+    }
+#endif
+#if defined(SUPPORT_FILEFORMAT_MOD)
+    else if (IsFileExtension(fileName, ".mod"))
+    {
+        jar_mod_context_t *ctxMod = RL_MALLOC(sizeof(jar_mod_context_t));
+        music.ctxData = ctxMod;
+
+        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);
+            music.loopCount = 0;   // Infinite loop by default
+            musicLoaded = true;
+        }
+    }
+#endif
+
+    if (!musicLoaded)
+    {
+        if (false) { }
+    #if defined(SUPPORT_FILEFORMAT_OGG)
+        else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData);
+    #endif
+    #if defined(SUPPORT_FILEFORMAT_FLAC)
+        else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData);
+    #endif
+    #if defined(SUPPORT_FILEFORMAT_MP3)
+        else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
+    #endif
+    #if defined(SUPPORT_FILEFORMAT_XM)
+        else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData);
+    #endif
+    #if defined(SUPPORT_FILEFORMAT_MOD)
+        else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); }
+    #endif
+
+        TraceLog(LOG_WARNING, "[%s] Music file could not be opened", fileName);
+    }
+    else
+    {
+        // Show some music stream info
+        TraceLog(LOG_INFO, "[%s] Music file successfully loaded:", fileName);
+        TraceLog(LOG_INFO, "   Total samples: %i", music.sampleCount);
+        TraceLog(LOG_INFO, "   Sample rate: %i Hz", music.stream.sampleRate);
+        TraceLog(LOG_INFO, "   Sample size: %i bits", music.stream.sampleSize);
+        TraceLog(LOG_INFO, "   Channels: %i (%s)", music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi");
+    }
+
+    return music;
+}
+
+// Unload music stream
+void UnloadMusicStream(Music music)
+{
+    CloseAudioStream(music.stream);
+
+    if (false) { }
+#if defined(SUPPORT_FILEFORMAT_OGG)
+    else if (music.ctxType == MUSIC_AUDIO_OGG) stb_vorbis_close((stb_vorbis *)music.ctxData);
+#endif
+#if defined(SUPPORT_FILEFORMAT_FLAC)
+    else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData);
+#endif
+#if defined(SUPPORT_FILEFORMAT_MP3)
+    else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
+#endif
+#if defined(SUPPORT_FILEFORMAT_XM)
+    else if (music.ctxType == MUSIC_MODULE_XM) jar_xm_free_context((jar_xm_context_t *)music.ctxData);
+#endif
+#if defined(SUPPORT_FILEFORMAT_MOD)
+    else if (music.ctxType == MUSIC_MODULE_MOD) { jar_mod_unload((jar_mod_context_t *)music.ctxData); RL_FREE(music.ctxData); }
+#endif
+}
+
+// Start music playing (open stream)
+void PlayMusicStream(Music music)
+{
+    AudioBuffer *audioBuffer = music.stream.buffer;
+
+    if (audioBuffer != 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 = audioBuffer->frameCursorPos;
+        PlayAudioStream(music.stream);  // WARNING: This resets the cursor position.
+        audioBuffer->frameCursorPos = frameCursorPos;
+    }
+    else TraceLog(LOG_ERROR, "PlayMusicStream() : No audio buffer");
+
+}
+
+// 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);
+
+    // Restart music context
+    switch (music.ctxType)
+    {
+#if defined(SUPPORT_FILEFORMAT_OGG)
+        case MUSIC_AUDIO_OGG: stb_vorbis_seek_start((stb_vorbis *)music.ctxData); break;
+#endif
+#if defined(SUPPORT_FILEFORMAT_FLAC)
+        case MUSIC_AUDIO_FLAC: drflac_seek_to_pcm_frame((drflac *)music.ctxData, 0); break;
+#endif
+#if defined(SUPPORT_FILEFORMAT_MP3)
+        case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame((drmp3 *)music.ctxData, 0); break;
+#endif
+#if defined(SUPPORT_FILEFORMAT_XM)
+        case MUSIC_MODULE_XM: jar_xm_reset((jar_xm_context_t *)music.ctxData); break;
+#endif
+#if defined(SUPPORT_FILEFORMAT_MOD)
+        case MUSIC_MODULE_MOD: jar_mod_seek_start((jar_mod_context_t *)music.ctxData); break;
+#endif
+        default: break;
+    }
+}
+
+// Update (re-fill) music buffers if data already processed
+void UpdateMusicStream(Music music)
+{
+    bool streamEnding = false;
+
+    unsigned int subBufferSizeInFrames = music.stream.buffer->bufferSizeInFrames/2;
+
+    // NOTE: Using dynamic allocation because it could require more than 16KB
+    void *pcm = RL_CALLOC(subBufferSizeInFrames*music.stream.channels*music.stream.sampleSize/8, 1);
+
+    int samplesCount = 0;    // Total size of data streamed in L+R samples for xm floats, individual L or R for ogg shorts
+
+    // TODO: Get the sampleLeft using totalFramesProcessed... but first, get total frames processed correctly...
+    //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
+    int sampleLeft = music.sampleCount - (music.stream.buffer->totalFramesProcessed*music.stream.channels);
+
+    while (IsAudioStreamProcessed(music.stream))
+    {
+        if ((sampleLeft/music.stream.channels) >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames*music.stream.channels;
+        else samplesCount = sampleLeft;
+
+        switch (music.ctxType)
+        {
+        #if defined(SUPPORT_FILEFORMAT_OGG)
+            case MUSIC_AUDIO_OGG:
+            {
+                // NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
+                stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)pcm, samplesCount);
+
+            } break;
+        #endif
+        #if defined(SUPPORT_FILEFORMAT_FLAC)
+            case MUSIC_AUDIO_FLAC:
+            {
+                // NOTE: Returns the number of samples to process (not required)
+                drflac_read_pcm_frames_s16((drflac *)music.ctxData, samplesCount, (short *)pcm);
+
+            } break;
+        #endif
+        #if defined(SUPPORT_FILEFORMAT_MP3)
+            case MUSIC_AUDIO_MP3:
+            {
+                // NOTE: samplesCount, actually refers to framesCount and returns the number of frames processed
+                drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, samplesCount/music.stream.channels, (float *)pcm);
+
+            } break;
+        #endif
+        #if defined(SUPPORT_FILEFORMAT_XM)
+            case MUSIC_MODULE_XM:
+            {
+                // NOTE: Internally this function considers 2 channels generation, so samplesCount/2
+                jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)pcm, samplesCount/2);
+            } break;
+        #endif
+        #if defined(SUPPORT_FILEFORMAT_MOD)
+            case MUSIC_MODULE_MOD:
+            {
+                // NOTE: 3rd parameter (nbsample) specify the number of stereo 16bits samples you want, so sampleCount/2
+                jar_mod_fillbuffer((jar_mod_context_t *)music.ctxData, (short *)pcm, samplesCount/2, 0);
+            } break;
+        #endif
+            default: break;
+        }
+
+        UpdateAudioStream(music.stream, pcm, samplesCount);
+
+        if ((music.ctxType == MUSIC_MODULE_XM) || (music.ctxType == MUSIC_MODULE_MOD))
+        {
+            if (samplesCount > 1) sampleLeft -= samplesCount/2;
+            else sampleLeft -= samplesCount;
+        }
+        else sampleLeft -= samplesCount;
+
+        if (sampleLeft <= 0)
+        {
+            streamEnding = true;
+            break;
+        }
+    }
+
+    // Free allocated pcm data
+    RL_FREE(pcm);
+
+    // Reset audio stream for looping
+    if (streamEnding)
+    {
+        StopMusicStream(music);        // Stop music (and reset)
+
+        // Decrease loopCount to stop when required
+        if (music.loopCount > 1)
+        {
+            music.loopCount--;         // Decrease loop count
+            PlayMusicStream(music);    // Play again
+        }
+        else if (music.loopCount == 0) PlayMusicStream(music);
+    }
+    else
+    {
+        // NOTE: In case window is minimized, music stream is stopped,
+        // just make sure to play again on window restore
+        if (IsMusicPlaying(music)) PlayMusicStream(music);
+    }
+}
+
+// Check if any music is playing
+bool IsMusicPlaying(Music music)
+{
+    return IsAudioStreamPlaying(music.stream);
+}
+
+// Set volume for music
+void SetMusicVolume(Music music, float volume)
+{
+    SetAudioStreamVolume(music.stream, volume);
+}
+
+// Set pitch for music
+void SetMusicPitch(Music music, float pitch)
+{
+    SetAudioStreamPitch(music.stream, pitch);
+}
+
+// Set music loop count (loop repeats)
+// NOTE: If set to 0, means infinite loop
+void SetMusicLoopCount(Music music, int count)
+{
+    music.loopCount = count;
+}
+
+// Get music time length (in seconds)
+float GetMusicTimeLength(Music music)
+{
+    float totalSeconds = 0.0f;
+
+    totalSeconds = (float)music.sampleCount/(music.stream.sampleRate*music.stream.channels);
+
+    return totalSeconds;
+}
+
+// Get current music time played (in seconds)
+float GetMusicTimePlayed(Music music)
+{
+    float secondsPlayed = 0.0f;
+
+    //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 = device.playback.internalBufferSizeInFrames/device.playback.internalPeriods;
+    unsigned int subBufferSize = AUDIO_BUFFER_SIZE;
+
+    if (subBufferSize < periodSize) subBufferSize = periodSize;
+
+    stream.buffer = InitAudioBuffer(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, "Audio stream loaded successfully (%i Hz, %i bit, %s)", stream.sampleRate, stream.sampleSize, (stream.channels == 1)? "Mono" : "Stereo");
+    }
+    else TraceLog(LOG_ERROR, "InitAudioStream() : Failed to create audio buffer");
+
+    return stream;
+}
+
+// Close audio stream and free memory
+void CloseAudioStream(AudioStream stream)
+{
+    CloseAudioBuffer(stream.buffer);
+
+    TraceLog(LOG_INFO, "Unloaded audio stream data");
+}
+
+// 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)
+{
+    AudioBuffer *audioBuffer = stream.buffer;
+
+    if (audioBuffer != NULL)
+    {
+        if (audioBuffer->isSubBufferProcessed[0] || audioBuffer->isSubBufferProcessed[1])
+        {
+            ma_uint32 subBufferToUpdate = 0;
+
+            if (audioBuffer->isSubBufferProcessed[0] && audioBuffer->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;
+                audioBuffer->frameCursorPos = 0;
+            }
+            else
+            {
+                // Just update whichever sub-buffer is processed.
+                subBufferToUpdate = (audioBuffer->isSubBufferProcessed[0])? 0 : 1;
+            }
+
+            ma_uint32 subBufferSizeInFrames = audioBuffer->bufferSizeInFrames/2;
+            unsigned char *subBuffer = audioBuffer->buffer + ((subBufferSizeInFrames*stream.channels*(stream.sampleSize/8))*subBufferToUpdate);
+
+            // TODO: Get total frames processed on this buffer... DOES NOT WORK.
+            audioBuffer->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));
+
+                audioBuffer->isSubBufferProcessed[subBufferToUpdate] = false;
+            }
+            else TraceLog(LOG_ERROR, "UpdateAudioStream() : Attempting to write too many frames to buffer");
+        }
+        else TraceLog(LOG_ERROR, "UpdateAudioStream() : Audio buffer not available for updating");
+    }
+    else TraceLog(LOG_ERROR, "UpdateAudioStream() : No audio buffer");
+}
+
+// Check if any audio stream buffers requires refill
+bool IsAudioStreamProcessed(AudioStream stream)
+{
+    if (stream.buffer == NULL)
+    {
+        TraceLog(LOG_ERROR, "IsAudioStreamProcessed() : No audio buffer");
+        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);
+}
+
+void SetAudioStreamVolume(AudioStream stream, float volume)
+{
+    SetAudioBufferVolume(stream.buffer, volume);
+}
+
+void SetAudioStreamPitch(AudioStream stream, float pitch)
+{
+    SetAudioBufferPitch(stream.buffer, pitch);
+}
+
+//----------------------------------------------------------------------------------
+// Module specific Functions Definition
+//----------------------------------------------------------------------------------
+
+#if defined(SUPPORT_FILEFORMAT_WAV)
+// Load WAV file into Wave structure
+static Wave LoadWAV(const char *fileName)
+{
+    // Basic WAV headers structs
+    typedef struct {
+        char chunkID[4];
+        int chunkSize;
+        char format[4];
+    } WAVRiffHeader;
+
+    typedef struct {
+        char subChunkID[4];
+        int subChunkSize;
+        short audioFormat;
+        short numChannels;
+        int sampleRate;
+        int byteRate;
+        short blockAlign;
+        short bitsPerSample;
+    } WAVFormat;
+
+    typedef struct {
+        char subChunkID[4];
+        int subChunkSize;
+    } WAVData;
+
+    WAVRiffHeader wavRiffHeader;
+    WAVFormat wavFormat;
+    WAVData wavData;
+
+    Wave wave = { 0 };
+    FILE *wavFile;
+
+    wavFile = fopen(fileName, "rb");
+
+    if (wavFile == NULL)
+    {
+        TraceLog(LOG_WARNING, "[%s] WAV file could not be opened", fileName);
+        wave.data = NULL;
+    }
+    else
+    {
+        // Read in the first chunk into the struct
+        fread(&wavRiffHeader, sizeof(WAVRiffHeader), 1, wavFile);
+
+        // Check for RIFF and WAVE tags
+        if (strncmp(wavRiffHeader.chunkID, "RIFF", 4) ||
+            strncmp(wavRiffHeader.format, "WAVE", 4))
+        {
+                TraceLog(LOG_WARNING, "[%s] Invalid RIFF or WAVE Header", fileName);
+        }
+        else
+        {
+            // Read in the 2nd chunk for the wave info
+            fread(&wavFormat, sizeof(WAVFormat), 1, wavFile);
+
+            // Check for fmt tag
+            if ((wavFormat.subChunkID[0] != 'f') || (wavFormat.subChunkID[1] != 'm') ||
+                (wavFormat.subChunkID[2] != 't') || (wavFormat.subChunkID[3] != ' '))
+            {
+                TraceLog(LOG_WARNING, "[%s] Invalid Wave format", fileName);
+            }
+            else
+            {
+                // Check for extra parameters;
+                if (wavFormat.subChunkSize > 16) fseek(wavFile, sizeof(short), SEEK_CUR);
+
+                // Read in the the last byte of data before the sound file
+                fread(&wavData, sizeof(WAVData), 1, wavFile);
+
+                // Check for data tag
+                if ((wavData.subChunkID[0] != 'd') || (wavData.subChunkID[1] != 'a') ||
+                    (wavData.subChunkID[2] != 't') || (wavData.subChunkID[3] != 'a'))
+                {
+                    TraceLog(LOG_WARNING, "[%s] Invalid data header", fileName);
+                }
+                else
+                {
+                    // Allocate memory for data
+                    wave.data = RL_MALLOC(wavData.subChunkSize);
+
+                    // Read in the sound data into the soundData variable
+                    fread(wave.data, wavData.subChunkSize, 1, wavFile);
+
+                    // Store wave parameters
+                    wave.sampleRate = wavFormat.sampleRate;
+                    wave.sampleSize = wavFormat.bitsPerSample;
+                    wave.channels = wavFormat.numChannels;
+
+                    // NOTE: Only support 8 bit, 16 bit and 32 bit sample sizes
+                    if ((wave.sampleSize != 8) && (wave.sampleSize != 16) && (wave.sampleSize != 32))
+                    {
+                        TraceLog(LOG_WARNING, "[%s] WAV sample size (%ibit) not supported, converted to 16bit", fileName, wave.sampleSize);
+                        WaveFormat(&wave, wave.sampleRate, 16, wave.channels);
+                    }
+
+                    // NOTE: Only support up to 2 channels (mono, stereo)
+                    if (wave.channels > 2)
+                    {
+                        WaveFormat(&wave, wave.sampleRate, wave.sampleSize, 2);
+                        TraceLog(LOG_WARNING, "[%s] WAV channels number (%i) not supported, converted to 2 channels", fileName, wave.channels);
+                    }
+
+                    // NOTE: subChunkSize comes in bytes, we need to translate it to number of samples
+                    wave.sampleCount = (wavData.subChunkSize/(wave.sampleSize/8))/wave.channels;
+
+                    TraceLog(LOG_INFO, "[%s] WAV file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
+                }
+            }
+        }
+
+        fclose(wavFile);
+    }
+
+    return wave;
+}
+
+// Save wave data as WAV file
+static int SaveWAV(Wave wave, const char *fileName)
+{
+    int success = 0;
+    int dataSize = wave.sampleCount*wave.channels*wave.sampleSize/8;
+
+    // Basic WAV headers structs
+    typedef struct {
+        char chunkID[4];
+        int chunkSize;
+        char format[4];
+    } RiffHeader;
+
+    typedef struct {
+        char subChunkID[4];
+        int subChunkSize;
+        short audioFormat;
+        short numChannels;
+        int sampleRate;
+        int byteRate;
+        short blockAlign;
+        short bitsPerSample;
+    } WaveFormat;
+
+    typedef struct {
+        char subChunkID[4];
+        int subChunkSize;
+    } WaveData;
+
+    FILE *wavFile = fopen(fileName, "wb");
+
+    if (wavFile == NULL) TraceLog(LOG_WARNING, "[%s] WAV audio file could not be created", fileName);
+    else
+    {
+        RiffHeader riffHeader;
+        WaveFormat waveFormat;
+        WaveData waveData;
+
+        // Fill structs with data
+        riffHeader.chunkID[0] = 'R';
+        riffHeader.chunkID[1] = 'I';
+        riffHeader.chunkID[2] = 'F';
+        riffHeader.chunkID[3] = 'F';
+        riffHeader.chunkSize = 44 - 4 + wave.sampleCount*wave.sampleSize/8;
+        riffHeader.format[0] = 'W';
+        riffHeader.format[1] = 'A';
+        riffHeader.format[2] = 'V';
+        riffHeader.format[3] = 'E';
+
+        waveFormat.subChunkID[0] = 'f';
+        waveFormat.subChunkID[1] = 'm';
+        waveFormat.subChunkID[2] = 't';
+        waveFormat.subChunkID[3] = ' ';
+        waveFormat.subChunkSize = 16;
+        waveFormat.audioFormat = 1;
+        waveFormat.numChannels = wave.channels;
+        waveFormat.sampleRate = wave.sampleRate;
+        waveFormat.byteRate = wave.sampleRate*wave.sampleSize/8;
+        waveFormat.blockAlign = wave.sampleSize/8;
+        waveFormat.bitsPerSample = wave.sampleSize;
+
+        waveData.subChunkID[0] = 'd';
+        waveData.subChunkID[1] = 'a';
+        waveData.subChunkID[2] = 't';
+        waveData.subChunkID[3] = 'a';
+        waveData.subChunkSize = dataSize;
+
+        fwrite(&riffHeader, sizeof(RiffHeader), 1, wavFile);
+        fwrite(&waveFormat, sizeof(WaveFormat), 1, wavFile);
+        fwrite(&waveData, sizeof(WaveData), 1, wavFile);
+
+        success = fwrite(wave.data, dataSize, 1, wavFile);
+
+        fclose(wavFile);
+    }
+
+    // If all data has been written correctly to file, success = 1
+    return success;
+}
+#endif
+
+#if defined(SUPPORT_FILEFORMAT_OGG)
+// Load OGG file into Wave structure
+// NOTE: Using stb_vorbis library
+static Wave LoadOGG(const char *fileName)
+{
+    Wave wave = { 0 };
+
+    stb_vorbis *oggFile = stb_vorbis_open_filename(fileName, NULL, NULL);
+
+    if (oggFile == NULL) TraceLog(LOG_WARNING, "[%s] OGG file could not be opened", 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, "[%s] Ogg audio length is 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!)
+        int numSamplesOgg = stb_vorbis_get_samples_short_interleaved(oggFile, info.channels, (short *)wave.data, wave.sampleCount*wave.channels);
+
+        TraceLog(LOG_DEBUG, "[%s] Samples obtained: %i", fileName, numSamplesOgg);
+
+        TraceLog(LOG_INFO, "[%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;
+
+    // Decode an entire FLAC file in one go
+    uint64_t totalSampleCount;
+    wave.data = drflac_open_file_and_read_pcm_frames_s16(fileName, &wave.channels, &wave.sampleRate, &totalSampleCount);
+
+    wave.sampleCount = (unsigned int)totalSampleCount;
+    wave.sampleSize = 16;
+
+    // NOTE: Only support up to 2 channels (mono, stereo)
+    if (wave.channels > 2) TraceLog(LOG_WARNING, "[%s] FLAC channels number (%i) not supported", fileName, wave.channels);
+
+    if (wave.data == NULL) TraceLog(LOG_WARNING, "[%s] FLAC data could not be loaded", fileName);
+    else TraceLog(LOG_INFO, "[%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
+    uint64_t totalFrameCount = 0;
+    drmp3_config config = { 0 };
+    wave.data = drmp3_open_file_and_read_f32(fileName, &config, &totalFrameCount);
+
+    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, "[%s] MP3 channels number (%i) not supported", fileName, wave.channels);
+
+    if (wave.data == NULL) TraceLog(LOG_WARNING, "[%s] MP3 data could not be loaded", fileName);
+    else TraceLog(LOG_INFO, "[%s] MP3 file loaded successfully (%i Hz, %i bit, %s)", fileName, wave.sampleRate, wave.sampleSize, (wave.channels == 1)? "Mono" : "Stereo");
+
+    return wave;
+}
+#endif
+
+// Some required functions for audio standalone module version
+#if defined(RAUDIO_STANDALONE)
+// Check file extension
+bool IsFileExtension(const char *fileName, const char *ext)
+{
+    bool result = false;
+    const char *fileExt;
+
+    if ((fileExt = strrchr(fileName, '.')) != NULL)
+    {
+        if (strcmp(fileExt, ext) == 0) result = true;
+    }
+
+    return result;
+}
+
+// Show trace log messages (LOG_INFO, LOG_WARNING, LOG_ERROR, LOG_DEBUG)
+void TraceLog(int msgType, const char *text, ...)
+{
+    va_list args;
+    va_start(args, text);
+
+    switch (msgType)
+    {
+        case LOG_INFO: fprintf(stdout, "INFO: "); break;
+        case LOG_ERROR: fprintf(stdout, "ERROR: "); break;
+        case LOG_WARNING: fprintf(stdout, "WARNING: "); break;
+        case LOG_DEBUG: fprintf(stdout, "DEBUG: "); break;
+        default: break;
+    }
+
+    vfprintf(stdout, text, args);
+    fprintf(stdout, "\n");
+
+    va_end(args);
+
+    if (msgType == LOG_ERROR) exit(1);
+}
+#endif
+
+#undef AudioBuffer