koffi 2.1.1 → 2.1.2

package/vendor/raylib/src/raudio.c

@@ -1,6 +1,6 @@
 /**********************************************************************************************
 *
-*   raudio v1.0 - A simple and easy-to-use audio library based on miniaudio
+*   raudio v1.1 - A simple and easy-to-use audio library based on miniaudio
 *
 *   FEATURES:
 *       - Manage audio device (init/close)
@@ -12,6 +12,9 @@
 *
 *   CONFIGURATION:
 *
+*   #define SUPPORT_MODULE_RAUDIO
+*       raudio module is included in the build
+*
 *   #define RAUDIO_STANDALONE
 *       Define to use the module as standalone library (independently of raylib).
 *       Required types and functions are defined in the same module.
@@ -47,7 +50,7 @@
 *
 *   LICENSE: zlib/libpng
 *
-*   Copyright (c) 2013-2021 Ramon Santamaria (@raysan5)
+*   Copyright (c) 2013-2022 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.
@@ -68,9 +71,9 @@
 
 #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
@@ -78,6 +81,8 @@
     #include "utils.h"          // Required for: fopen() Android mapping
 #endif
 
+#if defined(SUPPORT_MODULE_RAUDIO)
+
 #if defined(_WIN32)
 // To avoid conflicting windows.h symbols with raylib, some flags are defined
 // WARNING: Those flags avoid inclusion of some Win32 headers that could be required
@@ -169,26 +174,25 @@ typedef struct tagBITMAPINFOHEADER {
 
 #include <stdlib.h>                     // Required for: malloc(), free()
 #include <stdio.h>                      // Required for: FILE, fopen(), fclose(), fread()
+#include <string.h>                     // Required for: strcmp() [Used in IsFileExtension(), LoadWaveFromMemory(), LoadMusicStreamFromMemory()]
 
 #if defined(RAUDIO_STANDALONE)
-    #include <string.h>                 // Required for: strcmp() [Used in IsFileExtension()]
-
     #ifndef TRACELOG
-        #define TRACELOG(level, ...) (void)0
+        #define TRACELOG(level, ...)    printf(__VA_ARGS__)
     #endif
 
     // Allow custom memory allocators
     #ifndef RL_MALLOC
-        #define RL_MALLOC(sz)       malloc(sz)
+        #define RL_MALLOC(sz)           malloc(sz)
     #endif
     #ifndef RL_CALLOC
-        #define RL_CALLOC(n,sz)     calloc(n,sz)
+        #define RL_CALLOC(n,sz)         calloc(n,sz)
     #endif
     #ifndef RL_REALLOC
-        #define RL_REALLOC(ptr,sz)  realloc(ptr,sz)
+        #define RL_REALLOC(ptr,sz)      realloc(ptr,sz)
     #endif
     #ifndef RL_FREE
-        #define RL_FREE(ptr)        free(ptr)
+        #define RL_FREE(ptr)            free(ptr)
     #endif
 #endif
 
@@ -243,10 +247,6 @@ typedef struct tagBITMAPINFOHEADER {
     #include "external/dr_flac.h"       // FLAC loading functions
 #endif
 
-#if defined(_MSC_VER)
-    #undef bool
-#endif
-
 //----------------------------------------------------------------------------------
 // Defines and Macros
 //----------------------------------------------------------------------------------
@@ -263,10 +263,6 @@ typedef struct tagBITMAPINFOHEADER {
 #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
@@ -307,16 +303,20 @@ typedef enum {
     AUDIO_BUFFER_USAGE_STREAM
 } AudioBufferUsage;
 
-// Audio buffer structure
+// Audio buffer struct
 struct rAudioBuffer {
     ma_data_converter converter;    // Audio data converter
 
+    AudioCallback callback;         // Audio buffer callback for buffer filling on audio threads
+    rAudioProcessor *processor;     // Audio processor
+
     float volume;                   // Audio buffer volume
     float pitch;                    // Audio buffer pitch
+    float pan;                      // Audio buffer pan (0.0f to 1.0f)
 
     bool playing;                   // Audio buffer state: AUDIO_PLAYING
     bool paused;                    // Audio buffer state: AUDIO_PAUSED
-    bool looping;                   // Audio buffer looping, always true for AudioStreams
+    bool looping;                   // Audio buffer looping, default to true for AudioStreams
     int usage;                      // Audio buffer usage mode: STATIC or STREAM
 
     bool isSubBufferProcessed[2];   // SubBuffer processed (virtual double buffer)
@@ -330,6 +330,14 @@ struct rAudioBuffer {
     rAudioBuffer *prev;             // Previous audio buffer on the list
 };
 
+// Audio processor struct
+// NOTE: Useful to apply effects to an AudioBuffer
+struct rAudioProcessor {
+    AudioCallback process;          // Processor callback function
+    rAudioProcessor *next;          // Next audio processor on the list
+    rAudioProcessor *prev;          // Previous audio processor on the list
+};
+
 #define AudioBuffer rAudioBuffer    // HACK: To avoid CoreAudio (macOS) symbol collision
 
 // Audio data context
@@ -339,6 +347,8 @@ typedef struct AudioData {
         ma_device device;           // miniaudio device
         ma_mutex lock;              // miniaudio mutex lock
         bool isReady;               // Check if audio device is ready
+        size_t pcmBufferSize;       // Pre-allocated buffer size
+        void *pcmBuffer;            // Pre-allocated buffer to read audio data from file/memory
     } System;
     struct {
         AudioBuffer *first;         // Pointer to first AudioBuffer in the list
@@ -367,15 +377,13 @@ static AudioData AUDIO = {          // Global AUDIO context
 //----------------------------------------------------------------------------------
 // Module specific Functions Declaration
 //----------------------------------------------------------------------------------
-static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message);
+static void OnLog(void *pUserData, ma_uint32 level, const char *pMessage);
 static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const void *pFramesInput, ma_uint32 frameCount);
-static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, float localVolume);
+static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, AudioBuffer *buffer);
 
 #if defined(RAUDIO_STANDALONE)
 static bool IsFileExtension(const char *fileName, const char *ext); // Check file extension
 static const char *GetFileExtension(const char *fileName);          // Get pointer to extension for a filename string (includes the dot: .png)
-static bool TextIsEqual(const char *text1, const char *text2);      // Check if two text string are equal
-static const char *TextToLower(const char *text);                   // Get lower case version of provided string
 
 static unsigned char *LoadFileData(const char *fileName, unsigned int *bytesRead);     // Load file data as byte array (read)
 static bool SaveFileData(const char *fileName, void *data, unsigned int bytesToWrite); // Save data to file from byte array (write)
@@ -396,6 +404,7 @@ void PauseAudioBuffer(AudioBuffer *buffer);
 void ResumeAudioBuffer(AudioBuffer *buffer);
 void SetAudioBufferVolume(AudioBuffer *buffer, float volume);
 void SetAudioBufferPitch(AudioBuffer *buffer, float pitch);
+void SetAudioBufferPan(AudioBuffer *buffer, float pan);
 void TrackAudioBuffer(AudioBuffer *buffer);
 void UntrackAudioBuffer(AudioBuffer *buffer);
 
@@ -407,7 +416,7 @@ void InitAudioDevice(void)
 {
     // Init audio context
     ma_context_config ctxConfig = ma_context_config_init();
-    ctxConfig.logCallback = OnLog;
+    ma_log_callback_init(OnLog, NULL);
 
     ma_result result = ma_context_init(NULL, 0, &ctxConfig, &AUDIO.System.context);
     if (result != MA_SUCCESS)
@@ -461,8 +470,7 @@ void InitAudioDevice(void)
     // Init dummy audio buffers pool for multichannel sound playing
     for (int i = 0; i < MAX_AUDIO_BUFFER_POOL_CHANNELS; i++)
     {
-        // WARNING: An empty audio buffer is created (data = 0)
-        // AudioBuffer data just points to loaded sound data
+        // WARNING: An empty audio buffer is created (data = 0) and added to list, AudioBuffer data is filled on PlaySoundMulti()
         AUDIO.MultiChannel.pool[i] = LoadAudioBuffer(AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, 0, AUDIO_BUFFER_USAGE_STATIC);
     }
 
@@ -488,7 +496,7 @@ void CloseAudioDevice(void)
             //UnloadAudioBuffer(AUDIO.MultiChannel.pool[i]);
             if (AUDIO.MultiChannel.pool[i] != NULL)
             {
-                ma_data_converter_uninit(&AUDIO.MultiChannel.pool[i]->converter);
+                ma_data_converter_uninit(&AUDIO.MultiChannel.pool[i]->converter, NULL);
                 UntrackAudioBuffer(AUDIO.MultiChannel.pool[i]);
                 //RL_FREE(buffer->data);    // Already unloaded by UnloadSound()
                 RL_FREE(AUDIO.MultiChannel.pool[i]);
@@ -500,6 +508,7 @@ void CloseAudioDevice(void)
         ma_context_uninit(&AUDIO.System.context);
 
         AUDIO.System.isReady = false;
+        RL_FREE(AUDIO.System.pcmBuffer);
 
         TRACELOG(LOG_INFO, "AUDIO: Device closed successfully");
     }
@@ -537,9 +546,9 @@ AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sam
 
     // Audio data runs through a format converter
     ma_data_converter_config converterConfig = ma_data_converter_config_init(format, AUDIO_DEVICE_FORMAT, channels, AUDIO_DEVICE_CHANNELS, sampleRate, AUDIO.System.device.sampleRate);
-    converterConfig.resampling.allowDynamicSampleRate = true;        // Pitch shifting
+    converterConfig.allowDynamicSampleRate = true;
 
-    ma_result result = ma_data_converter_init(&converterConfig, &audioBuffer->converter);
+    ma_result result = ma_data_converter_init(&converterConfig, NULL, &audioBuffer->converter);
 
     if (result != MA_SUCCESS)
     {
@@ -551,9 +560,15 @@ AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sam
     // Init audio buffer values
     audioBuffer->volume = 1.0f;
     audioBuffer->pitch = 1.0f;
+    audioBuffer->pan = 0.5f;
+
+    audioBuffer->callback = NULL;
+    audioBuffer->processor = NULL;
+
     audioBuffer->playing = false;
     audioBuffer->paused = false;
     audioBuffer->looping = false;
+
     audioBuffer->usage = usage;
     audioBuffer->frameCursorPos = 0;
     audioBuffer->sizeInFrames = sizeInFrames;
@@ -574,7 +589,7 @@ void UnloadAudioBuffer(AudioBuffer *buffer)
 {
     if (buffer != NULL)
     {
-        ma_data_converter_uninit(&buffer->converter);
+        ma_data_converter_uninit(&buffer->converter, NULL);
         UntrackAudioBuffer(buffer);
         RL_FREE(buffer->data);
         RL_FREE(buffer);
@@ -648,13 +663,22 @@ void SetAudioBufferPitch(AudioBuffer *buffer, float pitch)
         // Note that this changes the duration of the sound:
         //  - higher pitches will make the sound faster
         //  - lower pitches make it slower
-        ma_uint32 outputSampleRate = (ma_uint32)((float)buffer->converter.config.sampleRateOut/pitch);
-        ma_data_converter_set_rate(&buffer->converter, buffer->converter.config.sampleRateIn, outputSampleRate);
+        ma_uint32 outputSampleRate = (ma_uint32)((float)buffer->converter.sampleRateOut/pitch);
+        ma_data_converter_set_rate(&buffer->converter, buffer->converter.sampleRateIn, outputSampleRate);
 
         buffer->pitch = pitch;
     }
 }
 
+// Set pan for an audio buffer
+void SetAudioBufferPan(AudioBuffer *buffer, float pan)
+{
+    if (pan < 0.0f) pan = 0.0f;
+    else if (pan > 1.0f) pan = 1.0f;
+
+    if (buffer != NULL) buffer->pan = pan;
+}
+
 // Track audio buffer to linked list next position
 void TrackAudioBuffer(AudioBuffer *buffer)
 {
@@ -711,16 +735,14 @@ Wave LoadWave(const char *fileName)
 }
 
 // Load wave from memory buffer, fileType refers to extension: i.e. ".wav"
+// WARNING: File extension must be provided in lower-case
 Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int dataSize)
 {
     Wave wave = { 0 };
 
-    char fileExtLower[16] = { 0 };
-    strcpy(fileExtLower, TextToLower(fileType));
-
     if (false) { }
 #if defined(SUPPORT_FILEFORMAT_WAV)
-    else if (TextIsEqual(fileExtLower, ".wav"))
+    else if (strcmp(fileType, ".wav") == 0)
     {
         drwav wav = { 0 };
         bool success = drwav_init_memory(&wav, fileData, dataSize, NULL);
@@ -742,7 +764,7 @@ Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int
     }
 #endif
 #if defined(SUPPORT_FILEFORMAT_OGG)
-    else if (TextIsEqual(fileExtLower, ".ogg"))
+    else if (strcmp(fileType, ".ogg") == 0)
     {
         stb_vorbis *oggData = stb_vorbis_open_memory((unsigned char *)fileData, dataSize, NULL, NULL);
 
@@ -764,7 +786,7 @@ Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int
     }
 #endif
 #if defined(SUPPORT_FILEFORMAT_FLAC)
-    else if (TextIsEqual(fileExtLower, ".flac"))
+    else if (strcmp(fileType, ".flac") == 0)
     {
         unsigned long long int totalFrameCount = 0;
 
@@ -777,7 +799,7 @@ Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int
     }
 #endif
 #if defined(SUPPORT_FILEFORMAT_MP3)
-    else if (TextIsEqual(fileExtLower, ".mp3"))
+    else if (strcmp(fileType, ".mp3") == 0)
     {
         drmp3_config config = { 0 };
         unsigned long long int totalFrameCount = 0;
@@ -862,17 +884,15 @@ Sound LoadSoundFromWave(Wave wave)
 // Unload wave data
 void UnloadWave(Wave wave)
 {
-    if (wave.data != NULL) RL_FREE(wave.data);
-
-    TRACELOG(LOG_INFO, "WAVE: Unloaded wave data from RAM");
+    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");
+    //TRACELOG(LOG_INFO, "SOUND: Unloaded sound data from RAM");
 }
 
 // Update sound buffer with new data
@@ -883,7 +903,7 @@ void UpdateSound(Sound sound, const void *data, int sampleCount)
         StopAudioBuffer(sound.stream.buffer);
 
         // TODO: May want to lock/unlock this since this data buffer is read at mixing time
-        memcpy(sound.stream.buffer->data, data, sampleCount*ma_get_bytes_per_frame(sound.stream.buffer->converter.config.formatIn, sound.stream.buffer->converter.config.channelsIn));
+        memcpy(sound.stream.buffer->data, data, sampleCount*ma_get_bytes_per_frame(sound.stream.buffer->converter.formatIn, sound.stream.buffer->converter.channelsIn));
     }
 }
 
@@ -899,7 +919,8 @@ bool ExportWave(Wave wave, const char *fileName)
         drwav wav = { 0 };
         drwav_data_format format = { 0 };
         format.container = drwav_container_riff;
-        format.format = DR_WAVE_FORMAT_PCM;
+        if (wave.sampleSize == 32) format.format = DR_WAVE_FORMAT_IEEE_FLOAT;
+        else format.format = DR_WAVE_FORMAT_PCM;
         format.channels = wave.channels;
         format.sampleRate = wave.sampleRate;
         format.bitsPerSample = wave.sampleSize;
@@ -946,42 +967,50 @@ bool ExportWaveAsCode(Wave wave, const char *fileName)
     int byteCount = 0;
     byteCount += sprintf(txtData + byteCount, "\n//////////////////////////////////////////////////////////////////////////////////\n");
     byteCount += sprintf(txtData + byteCount, "//                                                                              //\n");
-    byteCount += sprintf(txtData + byteCount, "// WaveAsCode exporter v1.0 - Wave data exported as an array of bytes           //\n");
+    byteCount += sprintf(txtData + byteCount, "// WaveAsCode exporter v1.1 - Wave data exported as an array of bytes           //\n");
     byteCount += sprintf(txtData + byteCount, "//                                                                              //\n");
     byteCount += sprintf(txtData + byteCount, "// more info and bugs-report:  github.com/raysan5/raylib                        //\n");
     byteCount += sprintf(txtData + byteCount, "// feedback and support:       ray[at]raylib.com                                //\n");
     byteCount += sprintf(txtData + byteCount, "//                                                                              //\n");
-    byteCount += sprintf(txtData + byteCount, "// Copyright (c) 2018-2021 Ramon Santamaria (@raysan5)                          //\n");
+    byteCount += sprintf(txtData + byteCount, "// Copyright (c) 2018-2022 Ramon Santamaria (@raysan5)                          //\n");
     byteCount += sprintf(txtData + byteCount, "//                                                                              //\n");
     byteCount += sprintf(txtData + byteCount, "//////////////////////////////////////////////////////////////////////////////////\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
+    char fileNameLower[256] = { 0 };
+    char fileNameUpper[256] = { 0 };
+    for (int i = 0; fileName[i] != '.'; i++) { fileNameLower[i] = fileName[i]; }      // Get filename without extension
+    for (int i = 0; fileNameLower[i] != '\0'; i++) if (fileNameLower[i] >= 'a' && fileNameLower[i] <= 'z') { fileNameUpper[i] = fileNameLower[i] - 32; }
 
     byteCount += sprintf(txtData + byteCount, "// Wave data information\n");
-    byteCount += sprintf(txtData + byteCount, "#define %s_FRAME_COUNT      %u\n", varFileName, wave.frameCount);
-    byteCount += sprintf(txtData + byteCount, "#define %s_FRAME_COUNT      %u\n", varFileName, wave.frameCount);
-    byteCount += sprintf(txtData + byteCount, "#define %s_SAMPLE_RATE      %u\n", varFileName, wave.sampleRate);
-    byteCount += sprintf(txtData + byteCount, "#define %s_SAMPLE_SIZE      %u\n", varFileName, wave.sampleSize);
-    byteCount += sprintf(txtData + byteCount, "#define %s_CHANNELS         %u\n\n", varFileName, wave.channels);
-
-    // Write byte data as hexadecimal text
-    // NOTE: Frame data exported is interlaced: Frame01[Sample-Channel01, Sample-Channel02, ...], Frame02[], Frame03[]
-    byteCount += sprintf(txtData + byteCount, "static unsigned char %s_DATA[%i] = { ", varFileName, waveDataSize);
-    for (int i = 0; i < waveDataSize - 1; i++) byteCount += sprintf(txtData + byteCount, ((i%TEXT_BYTES_PER_LINE == 0)? "0x%x,\n" : "0x%x, "), ((unsigned char *)wave.data)[i]);
-    byteCount += sprintf(txtData + byteCount, "0x%x };\n", ((unsigned char *)wave.data)[waveDataSize - 1]);
+    byteCount += sprintf(txtData + byteCount, "#define %s_FRAME_COUNT      %u\n", fileNameUpper, wave.frameCount);
+    byteCount += sprintf(txtData + byteCount, "#define %s_SAMPLE_RATE      %u\n", fileNameUpper, wave.sampleRate);
+    byteCount += sprintf(txtData + byteCount, "#define %s_SAMPLE_SIZE      %u\n", fileNameUpper, wave.sampleSize);
+    byteCount += sprintf(txtData + byteCount, "#define %s_CHANNELS         %u\n\n", fileNameUpper, wave.channels);
+
+    // Write wave data as an array of values
+    // Wave data is exported as byte array for 8/16bit and float array for 32bit float data
+    // NOTE: Frame data exported is channel-interlaced: frame01[sampleChannel1, sampleChannel2, ...], frame02[], frame03[]
+    if (wave.sampleSize == 32)
+    {
+        byteCount += sprintf(txtData + byteCount, "static float %sData[%i] = {\n", fileNameLower, waveDataSize/4);
+        for (int i = 1; i < waveDataSize/4; i++) byteCount += sprintf(txtData + byteCount, ((i%TEXT_BYTES_PER_LINE == 0)? "%.4ff,\n    " : "%.4ff, "), ((float *)wave.data)[i - 1]);
+        byteCount += sprintf(txtData + byteCount, "%.4ff };\n", ((float *)wave.data)[waveDataSize/4 - 1]);
+    }
+    else
+    {
+        byteCount += sprintf(txtData + byteCount, "static unsigned char %sData[%i] = { ", fileNameLower, waveDataSize);
+        for (int i = 1; i < waveDataSize; i++) byteCount += sprintf(txtData + byteCount, ((i%TEXT_BYTES_PER_LINE == 0)? "0x%x,\n    " : "0x%x, "), ((unsigned char *)wave.data)[i - 1]);
+        byteCount += sprintf(txtData + byteCount, "0x%x };\n", ((unsigned char *)wave.data)[waveDataSize - 1]);
+    }
 
     // NOTE: Text data length exported is determined by '\0' (NULL) character
     success = SaveFileText(fileName, txtData);
 
     RL_FREE(txtData);
 
+    if (success != 0) TRACELOG(LOG_INFO, "FILEIO: [%s] Wave as code exported successfully", fileName);
+    else TRACELOG(LOG_WARNING, "FILEIO: [%s] Failed to export wave as code", fileName);
+
     return success;
 }
 
@@ -1039,14 +1068,17 @@ void PlaySoundMulti(Sound sound)
     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;
+    SetAudioBufferVolume(AUDIO.MultiChannel.pool[index], sound.stream.buffer->volume);
+    SetAudioBufferPitch(AUDIO.MultiChannel.pool[index], sound.stream.buffer->pitch);
+    SetAudioBufferPan(AUDIO.MultiChannel.pool[index], sound.stream.buffer->pan);
+
     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;
+
+    AUDIO.MultiChannel.pool[index]->data = sound.stream.buffer->data;       // Fill dummy track with data for playing
 
     PlayAudioBuffer(AUDIO.MultiChannel.pool[index]);
 }
@@ -1106,6 +1138,12 @@ void SetSoundPitch(Sound sound, float pitch)
     SetAudioBufferPitch(sound.stream.buffer, pitch);
 }
 
+// Set pan for a sound
+void SetSoundPan(Sound sound, float pan)
+{
+    SetAudioBufferPan(sound.stream.buffer, pan);
+}
+
 // Convert wave data to desired format
 void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
 {
@@ -1113,8 +1151,8 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
     ma_format formatOut = ((sampleSize == 8)? ma_format_u8 : ((sampleSize == 16)? ma_format_s16 : ma_format_f32));
 
     ma_uint32 frameCountIn = wave->frameCount;
-
     ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, formatOut, channels, sampleRate, NULL, frameCountIn, formatIn, wave->channels, wave->sampleRate);
+
     if (frameCount == 0)
     {
         TRACELOG(LOG_WARNING, "WAVE: Failed to get frame count for format conversion");
@@ -1134,6 +1172,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
     wave->sampleSize = sampleSize;
     wave->sampleRate = sampleRate;
     wave->channels = channels;
+
     RL_FREE(wave->data);
     wave->data = data;
 }
@@ -1163,8 +1202,7 @@ Wave WaveCopy(Wave wave)
 // 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->frameCount*wave->channels)))
+    if ((initSample >= 0) && (initSample < finalSample) && ((unsigned int)finalSample < (wave->frameCount*wave->channels)))
     {
         int sampleCount = finalSample - initSample;
 
@@ -1377,18 +1415,16 @@ Music LoadMusicStream(const char *fileName)
     return music;
 }
 
-// extension including period ".mod"
-Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int dataSize)
+// Load music stream from memory buffer, fileType refers to extension: i.e. ".wav"
+// WARNING: File extension must be provided in lower-case
+Music LoadMusicStreamFromMemory(const char *fileType, const unsigned char *data, int dataSize)
 {
     Music music = { 0 };
     bool musicLoaded = false;
 
-    char fileExtLower[16] = { 0 };
-    strcpy(fileExtLower, TextToLower(fileType));
-
     if (false) { }
 #if defined(SUPPORT_FILEFORMAT_WAV)
-    else if (TextIsEqual(fileExtLower, ".wav"))
+    else if (strcmp(fileType, ".wav") == 0)
     {
         drwav *ctxWav = RL_CALLOC(1, sizeof(drwav));
 
@@ -1410,7 +1446,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
     }
 #endif
 #if defined(SUPPORT_FILEFORMAT_FLAC)
-    else if (TextIsEqual(fileExtLower, ".flac"))
+    else if (strcmp(fileType, ".flac") == 0)
     {
         music.ctxType = MUSIC_AUDIO_FLAC;
         music.ctxData = drflac_open_memory((const void*)data, dataSize, NULL);
@@ -1427,7 +1463,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
     }
 #endif
 #if defined(SUPPORT_FILEFORMAT_MP3)
-    else if (TextIsEqual(fileExtLower, ".mp3"))
+    else if (strcmp(fileType, ".mp3") == 0)
     {
         drmp3 *ctxMp3 = RL_CALLOC(1, sizeof(drmp3));
         int success = drmp3_init_memory(ctxMp3, (const void*)data, dataSize, NULL);
@@ -1445,7 +1481,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
     }
 #endif
 #if defined(SUPPORT_FILEFORMAT_OGG)
-    else if (TextIsEqual(fileExtLower, ".ogg"))
+    else if (strcmp(fileType, ".ogg") == 0)
     {
         // Open ogg audio stream
         music.ctxType = MUSIC_AUDIO_OGG;
@@ -1467,7 +1503,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
     }
 #endif
 #if defined(SUPPORT_FILEFORMAT_XM)
-    else if (TextIsEqual(fileExtLower, ".xm"))
+    else if (strcmp(fileType, ".xm") == 0)
     {
         jar_xm_context_t *ctxXm = NULL;
         int result = jar_xm_create_context_safe(&ctxXm, (const char *)data, dataSize, AUDIO.System.device.sampleRate);
@@ -1477,16 +1513,14 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
             jar_xm_set_max_loop_count(ctxXm, 0);    // Set infinite number of loops
 
             unsigned int bits = 32;
-            if (AUDIO_DEVICE_FORMAT == ma_format_s16)
-                bits = 16;
-            else if (AUDIO_DEVICE_FORMAT == ma_format_u8)
-                bits = 8;
+            if (AUDIO_DEVICE_FORMAT == ma_format_s16) bits = 16;
+            else if (AUDIO_DEVICE_FORMAT == ma_format_u8) bits = 8;
 
             // NOTE: Only stereo is supported for XM
             music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, bits, 2);
             music.frameCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm);    // NOTE: Always 2 channels (stereo)
             music.looping = true;   // Looping enabled by default
-            jar_xm_reset(ctxXm);   // make sure we start at the beginning of the song
+            jar_xm_reset(ctxXm);    // make sure we start at the beginning of the song
 
             music.ctxData = ctxXm;
             musicLoaded = true;
@@ -1494,7 +1528,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
     }
 #endif
 #if defined(SUPPORT_FILEFORMAT_MOD)
-    else if (TextIsEqual(fileExtLower, ".mod"))
+    else if (strcmp(fileType, ".mod") == 0)
     {
         jar_mod_context_t *ctxMod = (jar_mod_context_t *)RL_MALLOC(sizeof(jar_mod_context_t));
         int result = 0;
@@ -1507,7 +1541,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
         for (int i = 0; i < it; i++) newData[i] = data[i];
 
         // Memory loaded version for jar_mod_load_file()
-        if (dataSize && dataSize < 32*1024*1024)
+        if (dataSize && (dataSize < 32*1024*1024))
         {
             ctxMod->modfilesize = dataSize;
             ctxMod->modfile = newData;
@@ -1587,7 +1621,7 @@ void UnloadMusicStream(Music music)
         else if (music.ctxType == MUSIC_AUDIO_FLAC) drflac_free((drflac *)music.ctxData, NULL);
 #endif
 #if defined(SUPPORT_FILEFORMAT_MP3)
-    else if (music.ctxType == MUSIC_AUDIO_MP3) { drmp3_uninit((drmp3 *)music.ctxData); RL_FREE(music.ctxData); }
+        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);
@@ -1633,16 +1667,16 @@ void StopMusicStream(Music music)
     switch (music.ctxType)
     {
 #if defined(SUPPORT_FILEFORMAT_WAV)
-        case MUSIC_AUDIO_WAV: drwav_seek_to_pcm_frame((drwav *)music.ctxData, 0); break;
+        case MUSIC_AUDIO_WAV: drwav_seek_to_first_pcm_frame((drwav *)music.ctxData); break;
 #endif
 #if defined(SUPPORT_FILEFORMAT_OGG)
         case MUSIC_AUDIO_OGG: stb_vorbis_seek_start((stb_vorbis *)music.ctxData); break;
 #endif
 #if defined(SUPPORT_FILEFORMAT_FLAC)
-        case MUSIC_AUDIO_FLAC: drflac_seek_to_pcm_frame((drflac *)music.ctxData, 0); break;
+        case MUSIC_AUDIO_FLAC: drflac__seek_to_first_frame((drflac *)music.ctxData); break;
 #endif
 #if defined(SUPPORT_FILEFORMAT_MP3)
-        case MUSIC_AUDIO_MP3: drmp3_seek_to_pcm_frame((drmp3 *)music.ctxData, 0); break;
+        case MUSIC_AUDIO_MP3: drmp3_seek_to_start_of_stream((drmp3 *)music.ctxData); break;
 #endif
 #if defined(SUPPORT_FILEFORMAT_XM)
         case MUSIC_MODULE_XM: jar_xm_reset((jar_xm_context_t *)music.ctxData); break;
@@ -1687,64 +1721,107 @@ 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 frameCountToStream = 0;    // Total size of data in frames to be streamed
-
-    // TODO: Get the framesLeft using framesProcessed... 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;
-    unsigned int framesLeft = music.frameCount - music.stream.buffer->framesProcessed;
+    // On first call of this function we lazily pre-allocated a temp buffer to read audio files/memory data in
+    int frameSize = music.stream.channels*music.stream.sampleSize/8;
+    unsigned int pcmSize = subBufferSizeInFrames*frameSize;
+    if (AUDIO.System.pcmBufferSize < pcmSize)
+    {
+        RL_FREE(AUDIO.System.pcmBuffer);
+        AUDIO.System.pcmBuffer = RL_CALLOC(1, pcmSize);
+        AUDIO.System.pcmBufferSize = pcmSize;
+    }
 
-    while (IsAudioStreamProcessed(music.stream))
+    // Check both sub-buffers to check if they require refilling
+    for (int i = 0; i < 2; i++)
     {
-        if (framesLeft >= subBufferSizeInFrames) frameCountToStream = subBufferSizeInFrames;
-        else frameCountToStream = framesLeft;
+        if ((music.stream.buffer != NULL) && !music.stream.buffer->isSubBufferProcessed[i]) continue; // No refilling required, move to next sub-buffer
 
+        unsigned int framesLeft = music.frameCount - music.stream.buffer->framesProcessed;  // Frames left to be processed
+        unsigned int framesToStream = 0;                 // Total frames to be streamed
+        if ((framesLeft >= subBufferSizeInFrames) || music.looping) framesToStream = subBufferSizeInFrames;
+        else framesToStream = framesLeft;
+
+        int frameCountStillNeeded = framesToStream;
+        int frameCountRedTotal = 0;
         switch (music.ctxType)
         {
         #if defined(SUPPORT_FILEFORMAT_WAV)
             case MUSIC_AUDIO_WAV:
             {
-                // NOTE: Returns the number of samples to process (not required)
-                if (music.stream.sampleSize == 16) drwav_read_pcm_frames_s16((drwav *)music.ctxData, frameCountToStream, (short *)pcm);
-                else if (music.stream.sampleSize == 32) drwav_read_pcm_frames_f32((drwav *)music.ctxData, frameCountToStream, (float *)pcm);
-
+                if (music.stream.sampleSize == 16)
+                {
+                    while (true)
+                    {
+                        int frameCountRed = drwav_read_pcm_frames_s16((drwav *)music.ctxData, frameCountStillNeeded, (short *)((char *)AUDIO.System.pcmBuffer + frameCountRedTotal*frameSize));
+                        frameCountRedTotal += frameCountRed;
+                        frameCountStillNeeded -= frameCountRed;
+                        if (frameCountStillNeeded == 0) break;
+                        else drwav_seek_to_first_pcm_frame((drwav *)music.ctxData);
+                    }
+                }
+                else if (music.stream.sampleSize == 32)
+                {
+                    while (true)
+                    {
+                        int frameCountRed = drwav_read_pcm_frames_f32((drwav *)music.ctxData, frameCountStillNeeded, (float *)((char *)AUDIO.System.pcmBuffer + frameCountRedTotal*frameSize));
+                        frameCountRedTotal += frameCountRed;
+                        frameCountStillNeeded -= frameCountRed;
+                        if (frameCountStillNeeded == 0) break;
+                        else drwav_seek_to_first_pcm_frame((drwav *)music.ctxData);
+                    }
+                }
             } break;
         #endif
         #if defined(SUPPORT_FILEFORMAT_OGG)
             case MUSIC_AUDIO_OGG:
             {
-                // 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, frameCountToStream*music.stream.channels);
-
+                while (true)
+                {
+                    int frameCountRed = stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)((char *)AUDIO.System.pcmBuffer + frameCountRedTotal*frameSize), frameCountStillNeeded*music.stream.channels);
+                    frameCountRedTotal += frameCountRed;
+                    frameCountStillNeeded -= frameCountRed;
+                    if (frameCountStillNeeded == 0) break;
+                    else stb_vorbis_seek_start((stb_vorbis *)music.ctxData);
+                }
             } 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, frameCountToStream*music.stream.channels, (short *)pcm);
-
+                while (true)
+                {
+                    int frameCountRed = drflac_read_pcm_frames_s16((drflac *)music.ctxData, frameCountStillNeeded, (short *)((char *)AUDIO.System.pcmBuffer + frameCountRedTotal*frameSize));
+                    frameCountRedTotal += frameCountRed;
+                    frameCountStillNeeded -= frameCountRed;
+                    if (frameCountStillNeeded == 0) break;
+                    else drflac__seek_to_first_frame((drflac *)music.ctxData);
+                }
             } break;
         #endif
         #if defined(SUPPORT_FILEFORMAT_MP3)
             case MUSIC_AUDIO_MP3:
             {
-                drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, frameCountToStream, (float *)pcm);
-
+                while (true)
+                {
+                    int frameCountRed = drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, frameCountStillNeeded, (float *)((char *)AUDIO.System.pcmBuffer + frameCountRedTotal*frameSize));
+                    frameCountRedTotal += frameCountRed;
+                    frameCountStillNeeded -= frameCountRed;
+                    if (frameCountStillNeeded == 0) break;
+                    else drmp3_seek_to_start_of_stream((drmp3 *)music.ctxData);
+                }
             } break;
         #endif
         #if defined(SUPPORT_FILEFORMAT_XM)
             case MUSIC_MODULE_XM:
             {
                 // NOTE: Internally we consider 2 channels generation, so sampleCount/2
-                if (AUDIO_DEVICE_FORMAT == ma_format_f32) jar_xm_generate_samples((jar_xm_context_t *)music.ctxData, (float *)pcm, frameCountToStream);
-                else if (AUDIO_DEVICE_FORMAT == ma_format_s16) jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)pcm, frameCountToStream);
-                else if (AUDIO_DEVICE_FORMAT == ma_format_u8) jar_xm_generate_samples_8bit((jar_xm_context_t *)music.ctxData, (char *)pcm, frameCountToStream);
+                if (AUDIO_DEVICE_FORMAT == ma_format_f32) jar_xm_generate_samples((jar_xm_context_t *)music.ctxData, (float *)AUDIO.System.pcmBuffer, framesToStream);
+                else if (AUDIO_DEVICE_FORMAT == ma_format_s16) jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)AUDIO.System.pcmBuffer, framesToStream);
+                else if (AUDIO_DEVICE_FORMAT == ma_format_u8) jar_xm_generate_samples_8bit((jar_xm_context_t *)music.ctxData, (char *)AUDIO.System.pcmBuffer, framesToStream);
+
+                //jar_xm_reset((jar_xm_context_t *)music.ctxData);
 
             } break;
         #endif
@@ -1752,38 +1829,33 @@ void UpdateMusicStream(Music music)
             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, frameCountToStream, 0);
+                jar_mod_fillbuffer((jar_mod_context_t *)music.ctxData, (short *)AUDIO.System.pcmBuffer, framesToStream, 0);
+
+                //jar_mod_seek_start((jar_mod_context_t *)music.ctxData);
+
             } break;
         #endif
             default: break;
         }
 
-        UpdateAudioStream(music.stream, pcm, frameCountToStream);
+        UpdateAudioStream(music.stream, AUDIO.System.pcmBuffer, framesToStream);
 
-        framesLeft -= frameCountToStream;
+        music.stream.buffer->framesProcessed = music.stream.buffer->framesProcessed%music.frameCount;
 
-        if (framesLeft <= 0)
+        if (framesLeft <= subBufferSizeInFrames)
         {
-            streamEnding = true;
-            break;
+            if (!music.looping)
+            {
+                // Streaming is ending, we filled latest frames from input
+                StopMusicStream(music);
+                return;
+            }
         }
     }
 
-    // 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 (IsMusicStreamPlaying(music)) PlayMusicStream(music);
-    }
+    // NOTE: In case window is minimized, music stream is stopped,
+    // just make sure to play again on window restore
+    if (IsMusicStreamPlaying(music)) PlayMusicStream(music);
 }
 
 // Check if any music is playing
@@ -1804,6 +1876,12 @@ void SetMusicPitch(Music music, float pitch)
     SetAudioBufferPitch(music.stream.buffer, pitch);
 }
 
+// Set pan for a music
+void SetMusicPan(Music music, float pan)
+{
+    SetAudioBufferPan(music.stream.buffer, pan);
+}
+
 // Get music time length (in seconds)
 float GetMusicTimeLength(Music music)
 {
@@ -1832,7 +1910,13 @@ float GetMusicTimePlayed(Music music)
     #endif
         {
             //ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
-            unsigned int framesPlayed = music.stream.buffer->framesProcessed;
+            int framesProcessed = (int)music.stream.buffer->framesProcessed;
+            int subBufferSize = (int)music.stream.buffer->sizeInFrames/2;
+            int framesInFirstBuffer = music.stream.buffer->isSubBufferProcessed[0]? 0 : subBufferSize;
+            int framesInSecondBuffer = music.stream.buffer->isSubBufferProcessed[1]? 0 : subBufferSize;
+            int framesSentToMix = music.stream.buffer->frameCursorPos%subBufferSize;
+            int framesPlayed = (framesProcessed - framesInFirstBuffer - framesInSecondBuffer + framesSentToMix)%(int)music.frameCount;
+            if (framesPlayed < 0) framesPlayed += music.frameCount;
             secondsPlayed = (float)framesPlayed/music.stream.sampleRate;
         }
     }
@@ -1907,16 +1991,14 @@ void UpdateAudioStream(AudioStream stream, const void *data, int frameCount)
             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.
+            // Total frames processed in buffer is always the complete size, filled with 0 if required
             stream.buffer->framesProcessed += subBufferSizeInFrames;
 
             // Does this API expect a whole buffer to be updated in one go?
             // Assuming so, but if not will need to change this logic.
             if (subBufferSizeInFrames >= (ma_uint32)frameCount)
             {
-                ma_uint32 framesToWrite = subBufferSizeInFrames;
-
-                if (framesToWrite > (ma_uint32)frameCount) framesToWrite = (ma_uint32)frameCount;
+                ma_uint32 framesToWrite = (ma_uint32)frameCount;
 
                 ma_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8);
                 memcpy(subBuffer, data, bytesToWrite);
@@ -1984,28 +2066,98 @@ void SetAudioStreamPitch(AudioStream stream, float pitch)
     SetAudioBufferPitch(stream.buffer, pitch);
 }
 
+// Set pan for audio stream
+void SetAudioStreamPan(AudioStream stream, float pan)
+{
+    SetAudioBufferPan(stream.buffer, pan);
+}
+
 // Default size for new audio streams
 void SetAudioStreamBufferSizeDefault(int size)
 {
     AUDIO.Buffer.defaultSize = size;
 }
 
+// Audio thread callback to request new data
+void SetAudioStreamCallback(AudioStream stream, AudioCallback callback)
+{
+    if (stream.buffer != NULL) stream.buffer->callback = callback;
+}
+
+// Add processor to audio stream. Contrary to buffers, the order of processors is important.
+// The new processor must be added at the end. As there aren't supposed to be a lot of processors attached to
+// a given stream, we iterate through the list to find the end. That way we don't need a pointer to the last element.
+void AttachAudioStreamProcessor(AudioStream stream, AudioCallback process)
+{
+    ma_mutex_lock(&AUDIO.System.lock);
+
+    rAudioProcessor *processor = (rAudioProcessor *)RL_CALLOC(1, sizeof(rAudioProcessor));
+    processor->process = process;
+
+    rAudioProcessor *last = stream.buffer->processor;
+
+    while (last && last->next)
+    {
+        last = last->next;
+    }
+    if (last)
+    {
+        processor->prev = last;
+        last->next = processor;
+    }
+    else stream.buffer->processor = processor;
+
+    ma_mutex_unlock(&AUDIO.System.lock);
+}
+
+void DetachAudioStreamProcessor(AudioStream stream, AudioCallback process)
+{
+    ma_mutex_lock(&AUDIO.System.lock);
+
+    rAudioProcessor *processor = stream.buffer->processor;
+
+    while (processor)
+    {
+        rAudioProcessor *next = processor->next;
+        rAudioProcessor *prev = processor->prev;
+
+        if (processor->process == process)
+        {
+            if (stream.buffer->processor == processor) stream.buffer->processor = next;
+            if (prev) prev->next = next;
+            if (next) next->prev = prev;
+
+            RL_FREE(processor);
+        }
+
+        processor = next;
+    }
+
+    ma_mutex_unlock(&AUDIO.System.lock);
+}
+
 //----------------------------------------------------------------------------------
 // Module specific Functions Definition
 //----------------------------------------------------------------------------------
 
 // Log callback function
-static void OnLog(ma_context *pContext, ma_device *pDevice, ma_uint32 logLevel, const char *message)
+static void OnLog(void *pUserData, ma_uint32 level, const char *pMessage)
 {
-    (void)pContext;
-    (void)pDevice;
-
-    TRACELOG(LOG_WARNING, "miniaudio: %s", message);   // All log messages from miniaudio are errors
+    TRACELOG(LOG_WARNING, "miniaudio: %s", pMessage);   // All log messages from miniaudio are errors
 }
 
 // Reads audio data from an AudioBuffer object in internal format.
 static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer, void *framesOut, ma_uint32 frameCount)
 {
+    // Using audio buffer callback
+    if (audioBuffer->callback)
+    {
+        audioBuffer->callback(framesOut, frameCount);
+        audioBuffer->framesProcessed += frameCount;
+
+        return frameCount;
+    }
+
     ma_uint32 subBufferSizeInFrames = (audioBuffer->sizeInFrames > 1)? audioBuffer->sizeInFrames/2 : audioBuffer->sizeInFrames;
     ma_uint32 currentSubBufferIndex = audioBuffer->frameCursorPos/subBufferSizeInFrames;
 
@@ -2013,11 +2165,11 @@ static ma_uint32 ReadAudioBufferFramesInInternalFormat(AudioBuffer *audioBuffer,
 
     // 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];
+    bool isSubBufferProcessed[2] = { 0 };
     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);
+    ma_uint32 frameSizeInBytes = ma_get_bytes_per_frame(audioBuffer->converter.formatIn, audioBuffer->converter.channelsIn);
 
     // Fill out every frame until we find a buffer that's marked as processed. Then fill the remainder with 0
     ma_uint32 framesRead = 0;
@@ -2096,21 +2248,22 @@ static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, f
     // 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_uint8 inputBuffer[4096] = { 0 };
+    ma_uint32 inputBufferFrameCap = sizeof(inputBuffer)/ma_get_bytes_per_frame(audioBuffer->converter.formatIn, audioBuffer->converter.channelsIn);
 
     ma_uint32 totalOutputFramesProcessed = 0;
     while (totalOutputFramesProcessed < frameCount)
     {
         ma_uint64 outputFramesToProcessThisIteration = frameCount - totalOutputFramesProcessed;
+        ma_uint64 inputFramesToProcessThisIteration = 0;
 
-        ma_uint64 inputFramesToProcessThisIteration = ma_data_converter_get_required_input_frame_count(&audioBuffer->converter, outputFramesToProcessThisIteration);
+        (void)ma_data_converter_get_required_input_frame_count(&audioBuffer->converter, outputFramesToProcessThisIteration, &inputFramesToProcessThisIteration);
         if (inputFramesToProcessThisIteration > inputBufferFrameCap)
         {
             inputFramesToProcessThisIteration = inputBufferFrameCap;
         }
 
-        float *runningFramesOut = framesOut + (totalOutputFramesProcessed*audioBuffer->converter.config.channelsOut);
+        float *runningFramesOut = framesOut + (totalOutputFramesProcessed*audioBuffer->converter.channelsOut);
 
         /* At this point we can convert the data to our mixing format. */
         ma_uint64 inputFramesProcessedThisIteration = ReadAudioBufferFramesInInternalFormat(audioBuffer, inputBuffer, (ma_uint32)inputFramesToProcessThisIteration);    /* Safe cast. */
@@ -2165,7 +2318,7 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const
 
                 while (framesToRead > 0)
                 {
-                    float tempBuffer[1024]; // 512 frames for stereo
+                    float tempBuffer[1024] = { 0 }; // Frames for stereo
 
                     ma_uint32 framesToReadRightNow = framesToRead;
                     if (framesToReadRightNow > sizeof(tempBuffer)/sizeof(tempBuffer[0])/AUDIO_DEVICE_CHANNELS)
@@ -2179,7 +2332,15 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const
                         float *framesOut = (float *)pFramesOut + (framesRead*AUDIO.System.device.playback.channels);
                         float *framesIn = tempBuffer;
 
-                        MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer->volume);
+                        // Apply processors chain if defined
+                        rAudioProcessor *processor = audioBuffer->processor;
+                        while (processor)
+                        {
+                            processor->process(framesIn, framesJustRead);
+                            processor = processor->next;
+                        }
+
+                        MixAudioFrames(framesOut, framesIn, framesJustRead, audioBuffer);
 
                         framesToRead -= framesJustRead;
                         framesRead += framesJustRead;
@@ -2219,18 +2380,45 @@ static void OnSendAudioDataToDevice(ma_device *pDevice, void *pFramesOut, const
     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)
+// Main mixing function, pretty simple in this project, just an accumulation
+// NOTE: framesOut is both an input and an output, it is initially filled with zeros outside of this function
+static void MixAudioFrames(float *framesOut, const float *framesIn, ma_uint32 frameCount, AudioBuffer *buffer)
 {
-    for (ma_uint32 iFrame = 0; iFrame < frameCount; ++iFrame)
+    const float localVolume = buffer->volume;
+    const ma_uint32 channels = AUDIO.System.device.playback.channels;
+
+    if (channels == 2)  // We consider panning
     {
-        for (ma_uint32 iChannel = 0; iChannel < AUDIO.System.device.playback.channels; ++iChannel)
+        const float left = buffer->pan;
+        const float right = 1.0f - left;
+
+        // Fast sine approximation in [0..1] for pan law: y = 0.5f*x*(3 - x*x);
+        const float levels[2] = { localVolume*0.5f*left*(3.0f - left*left), localVolume*0.5f*right*(3.0f - right*right) };
+
+        float *frameOut = framesOut;
+        const float *frameIn = framesIn;
+
+        for (ma_uint32 frame = 0; frame < frameCount; frame++)
         {
-            float *frameOut = framesOut + (iFrame*AUDIO.System.device.playback.channels);
-            const float *frameIn = framesIn + (iFrame*AUDIO.System.device.playback.channels);
+            frameOut[0] += (frameIn[0]*levels[0]);
+            frameOut[1] += (frameIn[1]*levels[1]);
+
+            frameOut += 2;
+            frameIn += 2;
+        }
+    }
+    else  // We do not consider panning
+    {
+        for (ma_uint32 frame = 0; frame < frameCount; frame++)
+        {
+            for (ma_uint32 c = 0; c < channels; c++)
+            {
+                float *frameOut = framesOut + (frame*channels);
+                const float *frameIn = framesIn + (frame*channels);
 
-            frameOut[iChannel] += (frameIn[iChannel]*localVolume);
+                // Output accumulates input multiplied by volume to provided output (usually 0)
+                frameOut[c] += (frameIn[c]*localVolume);
+            }
         }
     }
 }
@@ -2261,38 +2449,6 @@ static const char *GetFileExtension(const char *fileName)
     return dot;
 }
 
-// Check if two text string are equal
-// REQUIRES: strcmp()
-static bool TextIsEqual(const char *text1, const char *text2)
-{
-    bool result = false;
-
-    if (strcmp(text1, text2) == 0) result = true;
-
-    return result;
-}
-
-// Get lower case version of provided string
-// REQUIRES: tolower()
-static const char *TextToLower(const char *text)
-{
-    #define MAX_TEXT_BUFFER_LENGTH      1024
-
-    static char buffer[MAX_TEXT_BUFFER_LENGTH] = { 0 };
-
-    for (int i = 0; i < MAX_TEXT_BUFFER_LENGTH; i++)
-    {
-        if (text[i] != '\0')
-        {
-            buffer[i] = (char)tolower(text[i]);
-            //if ((text[i] >= 'A') && (text[i] <= 'Z')) buffer[i] = text[i] + 32;
-        }
-        else { buffer[i] = '\0'; break; }
-    }
-
-    return buffer;
-}
-
 // Load data from file into a buffer
 static unsigned char *LoadFileData(const char *fileName, unsigned int *bytesRead)
 {
@@ -2378,3 +2534,5 @@ static bool SaveFileText(const char *fileName, char *text)
 #endif
 
 #undef AudioBuffer
+
+#endif      // SUPPORT_MODULE_RAUDIO